Added old abandoned KDE3 version of the RoseGarden MIDI tool

git-svn-id: svn://anonsvn.kde.org/home/kde/branches/trinity/applications/rosegarden@1097595 283d02a7-25f6-0310-bc7c-ecb5cbfe19da

89 files changed, 36577 insertions, 0 deletions

diff --git a/src/sound/AlsaDriver.cpp b/src/sound/AlsaDriver.cpp
new file mode 100644
index 0000000..9d512d9
--- /dev/null
+++ b/src/sound/AlsaDriver.cpp
@@ -0,0 +1,5476 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include <iostream>
+#include "misc/Debug.h"
+#include <cstdlib>
+#include <cstdio>
+#include <algorithm>
+
+#ifdef HAVE_ALSA
+
+// ALSA
+#include <alsa/asoundlib.h>
+#include <alsa/seq_event.h>
+#include <alsa/version.h>
+#include <alsa/seq.h>
+
+#include "AlsaDriver.h"
+#include "AlsaPort.h"
+#include "ExternalTransport.h"
+#include "MappedInstrument.h"
+#include "Midi.h"
+#include "MappedStudio.h"
+#include "misc/Strings.h"
+#include "MappedCommon.h"
+#include "MappedEvent.h"
+#include "Audit.h"
+#include "AudioPlayQueue.h"
+#include "ExternalTransport.h"
+
+#include <qregexp.h>
+#include <pthread.h>
+
+
+//#define DEBUG_ALSA 1
+//#define DEBUG_PROCESS_MIDI_OUT 1
+//#define DEBUG_PROCESS_SOFT_SYNTH_OUT 1
+//#define MTC_DEBUG 1
+
+// This driver implements MIDI in and out via the ALSA (www.alsa-project.org)
+// sequencer interface.
+
+using std::cerr;
+using std::endl;
+
+static size_t _debug_jack_frame_count = 0;
+
+#define AUTO_TIMER_NAME "(auto)"
+
+
+namespace Rosegarden
+{
+
+#define FAILURE_REPORT_COUNT 256
+static MappedEvent::FailureCode _failureReports[FAILURE_REPORT_COUNT];
+static int _failureReportWriteIndex = 0;
+static int _failureReportReadIndex = 0;
+
+AlsaDriver::AlsaDriver(MappedStudio *studio):
+        SoundDriver(studio,
+		    std::string("[ALSA library version ") +
+                    std::string(SND_LIB_VERSION_STR) + 
+		    std::string(", module version ") +
+		    getAlsaModuleVersionString() + 
+		    std::string(", kernel version ") +
+		    getKernelVersionString() +
+		    "]"),
+        m_client( -1),
+        m_inputPort( -1),
+        m_syncOutputPort( -1),
+        m_controllerPort( -1),
+        m_queue( -1),
+        m_maxClients( -1),
+        m_maxPorts( -1),
+        m_maxQueues( -1),
+        m_midiInputPortConnected(false),
+        m_midiSyncAutoConnect(false),
+        m_alsaPlayStartTime(0, 0),
+        m_alsaRecordStartTime(0, 0),
+        m_loopStartTime(0, 0),
+        m_loopEndTime(0, 0),
+        m_eat_mtc(0),
+        m_looping(false),
+        m_haveShutdown(false)
+#ifdef HAVE_LIBJACK
+        , m_jackDriver(0)
+#endif
+        , m_queueRunning(false)
+        , m_portCheckNeeded(false),
+        m_needJackStart(NeedNoJackStart),
+        m_doTimerChecks(false),
+        m_firstTimerCheck(true),
+        m_timerRatio(0),
+        m_timerRatioCalculated(false)
+
+{
+    Audit audit;
+    audit << "Rosegarden " << VERSION << " - AlsaDriver "
+    << m_name << std::endl;
+}
+
+AlsaDriver::~AlsaDriver()
+{
+    if (!m_haveShutdown) {
+        std::cerr << "WARNING: AlsaDriver::shutdown() was not called before destructor, calling now" << std::endl;
+        shutdown();
+    }
+}
+
+int
+AlsaDriver::checkAlsaError(int rc, const char *
+#ifdef DEBUG_ALSA
+                           message
+#endif
+                          )
+{
+#ifdef DEBUG_ALSA
+    if (rc < 0) {
+        std::cerr << "AlsaDriver::"
+        << message
+        << ": " << rc
+        << " (" << snd_strerror(rc) << ")"
+        << std::endl;
+    }
+#endif
+    return rc;
+}
+
+void
+AlsaDriver::shutdown()
+{
+#ifdef DEBUG_ALSA
+    std::cerr << "AlsaDriver::~AlsaDriver - shutting down" << std::endl;
+#endif
+
+    processNotesOff(getAlsaTime(), true, true);
+
+#ifdef HAVE_LIBJACK
+    delete m_jackDriver;
+    m_jackDriver = 0;
+#endif
+
+    if (m_midiHandle) {
+#ifdef DEBUG_ALSA
+        std::cerr << "AlsaDriver::shutdown - closing MIDI client" << std::endl;
+#endif
+
+        checkAlsaError(snd_seq_stop_queue(m_midiHandle, m_queue, 0), "shutdown(): stopping queue");
+        checkAlsaError(snd_seq_drain_output(m_midiHandle), "shutdown(): drain output");
+#ifdef DEBUG_ALSA
+
+        std::cerr << "AlsaDriver::shutdown - stopped queue" << std::endl;
+#endif
+
+        snd_seq_close(m_midiHandle);
+#ifdef DEBUG_ALSA
+
+        std::cerr << "AlsaDriver::shutdown - closed MIDI handle" << std::endl;
+#endif
+
+        m_midiHandle = 0;
+    }
+
+    DataBlockRepository::clear();
+
+    m_haveShutdown = true;
+}
+
+void
+AlsaDriver::setLoop(const RealTime &loopStart, const RealTime &loopEnd)
+{
+    m_loopStartTime = loopStart;
+    m_loopEndTime = loopEnd;
+
+    // currently we use this simple test for looping - it might need
+    // to get more sophisticated in the future.
+    //
+    if (m_loopStartTime != m_loopEndTime)
+        m_looping = true;
+    else
+        m_looping = false;
+}
+
+void
+AlsaDriver::getSystemInfo()
+{
+    int err;
+    snd_seq_system_info_t *sysinfo;
+
+    snd_seq_system_info_alloca(&sysinfo);
+
+    if ((err = snd_seq_system_info(m_midiHandle, sysinfo)) < 0) {
+        std::cerr << "System info error: " << snd_strerror(err)
+        << std::endl;
+        reportFailure(MappedEvent::FailureALSACallFailed);
+        m_maxQueues = 0;
+        m_maxClients = 0;
+        m_maxPorts = 0;
+        return ;
+    }
+
+    m_maxQueues = snd_seq_system_info_get_queues(sysinfo);
+    m_maxClients = snd_seq_system_info_get_clients(sysinfo);
+    m_maxPorts = snd_seq_system_info_get_ports(sysinfo);
+}
+
+void
+AlsaDriver::showQueueStatus(int queue)
+{
+    int err, idx, min, max;
+    snd_seq_queue_status_t *status;
+
+    snd_seq_queue_status_alloca(&status);
+    min = queue < 0 ? 0 : queue;
+    max = queue < 0 ? m_maxQueues : queue + 1;
+
+    for (idx = min; idx < max; ++idx) {
+        if ((err = snd_seq_get_queue_status(m_midiHandle, idx, status)) < 0) {
+
+            if (err == -ENOENT)
+                continue;
+
+            std::cerr << "Client " << idx << " info error: "
+            << snd_strerror(err) << std::endl;
+
+            reportFailure(MappedEvent::FailureALSACallFailed);
+            return ;
+        }
+
+#ifdef DEBUG_ALSA
+        std::cerr << "Queue " << snd_seq_queue_status_get_queue(status)
+        << std::endl;
+
+        std::cerr << "Tick       = "
+        << snd_seq_queue_status_get_tick_time(status)
+        << std::endl;
+
+        std::cerr << "Realtime   = "
+        << snd_seq_queue_status_get_real_time(status)->tv_sec
+        << "."
+        << snd_seq_queue_status_get_real_time(status)->tv_nsec
+        << std::endl;
+
+        std::cerr << "Flags      = 0x"
+        << snd_seq_queue_status_get_status(status)
+        << std::endl;
+#endif
+
+    }
+
+}
+
+
+void
+AlsaDriver::generateTimerList()
+{
+    // Enumerate the available timers
+
+    snd_timer_t *timerHandle;
+
+    snd_timer_id_t *timerId;
+    snd_timer_info_t *timerInfo;
+
+    snd_timer_id_alloca(&timerId);
+    snd_timer_info_alloca(&timerInfo);
+
+    snd_timer_query_t *timerQuery;
+    char timerName[64];
+
+    m_timers.clear();
+
+    if (snd_timer_query_open(&timerQuery, "hw", 0) >= 0) {
+
+        snd_timer_id_set_class(timerId, SND_TIMER_CLASS_NONE);
+
+        while (1) {
+
+            if (snd_timer_query_next_device(timerQuery, timerId) < 0)
+                break;
+            if (snd_timer_id_get_class(timerId) < 0)
+                break;
+
+            AlsaTimerInfo info = {
+                                     snd_timer_id_get_class(timerId),
+                                     snd_timer_id_get_sclass(timerId),
+                                     snd_timer_id_get_card(timerId),
+                                     snd_timer_id_get_device(timerId),
+                                     snd_timer_id_get_subdevice(timerId),
+                                     "",
+                                     0
+                                 };
+
+            if (info.card < 0)
+                info.card = 0;
+            if (info.device < 0)
+                info.device = 0;
+            if (info.subdevice < 0)
+                info.subdevice = 0;
+
+            //	    std::cerr << "got timer: class " << info.clas << std::endl;
+
+            sprintf(timerName, "hw:CLASS=%i,SCLASS=%i,CARD=%i,DEV=%i,SUBDEV=%i",
+                    info.clas, info.sclas, info.card, info.device, info.subdevice);
+
+            if (snd_timer_open(&timerHandle, timerName, SND_TIMER_OPEN_NONBLOCK) < 0) {
+                std::cerr << "Failed to open timer: " << timerName << std::endl;
+                continue;
+            }
+
+            if (snd_timer_info(timerHandle, timerInfo) < 0)
+                continue;
+
+            info.name = snd_timer_info_get_name(timerInfo);
+            info.resolution = snd_timer_info_get_resolution(timerInfo);
+            snd_timer_close(timerHandle);
+
+            //	    std::cerr << "adding timer: " << info.name << std::endl;
+
+            m_timers.push_back(info);
+        }
+
+        snd_timer_query_close(timerQuery);
+    }
+}
+
+
+std::string
+AlsaDriver::getAutoTimer(bool &wantTimerChecks)
+{
+    Audit audit;
+
+    // Look for the apparent best-choice timer.
+
+    if (m_timers.empty())
+        return "";
+
+    // The system RTC timer ought to be good, but it doesn't look like
+    // a very safe choice -- we've seen some system lockups apparently
+    // connected with use of this timer on 2.6 kernels.  So we avoid
+    // using that as an auto option.
+
+    // Looks like our most reliable options for timers are, in order:
+    //
+    // 1. System timer if at 1000Hz, with timer checks (i.e. automatic
+    //    drift correction against PCM frame count).  Only available
+    //    when JACK is running.
+    //
+    // 2. PCM playback timer currently in use by JACK (no drift, but
+    //    suffers from jitter).
+    //
+    // 3. System timer if at 1000Hz.
+    //
+    // 4. System RTC timer.
+    //
+    // 5. System timer.
+
+    // As of Linux kernel 2.6.13 (?) the default system timer
+    // resolution has been reduced from 1000Hz to 250Hz, giving us
+    // only 4ms accuracy instead of 1ms.  This may be better than the
+    // 10ms available from the stock 2.4 kernel, but it's not enough
+    // for really solid MIDI timing.  If JACK is running at 44.1 or
+    // 48KHz with a buffer size less than 256 frames, then the PCM
+    // timer will give us less jitter.  Even at 256 frames, it may be
+    // preferable in practice just because it's simpler.
+
+    // However, we can't safely choose the PCM timer over the system
+    // timer unless the latter has really awful resolution, because we
+    // don't know for certain which PCM JACK is using.  We guess at
+    // hw:0 for the moment, which gives us a stuck timer problem if
+    // it's actually using something else.  So if the system timer
+    // runs at 250Hz, we really have to choose it anyway and just give
+    // a warning.
+
+    bool pcmTimerAccepted = false;
+    wantTimerChecks = false; // for most options
+
+    bool rtcCouldBeOK = false;
+
+#ifdef HAVE_LIBJACK
+    if (m_jackDriver) {
+	wantTimerChecks = true;
+	pcmTimerAccepted = true;
+    }
+#endif
+
+    // look for a high frequency system timer
+
+    for (std::vector<AlsaTimerInfo>::iterator i = m_timers.begin();
+            i != m_timers.end(); ++i) {
+        if (i->sclas != SND_TIMER_SCLASS_NONE)
+            continue;
+        if (i->clas == SND_TIMER_CLASS_GLOBAL) {
+            if (i->device == SND_TIMER_GLOBAL_SYSTEM) {
+                long hz = 1000000000 / i->resolution;
+                if (hz >= 750) {
+                    return i->name;
+                }
+            }
+        }
+    }
+
+    // Look for the system RTC timer if available.  This has been
+    // known to hang some real-time kernels, but reports suggest that
+    // recent kernels are OK.  Avoid if the kernel is older than
+    // 2.6.20 or the ALSA driver is older than 1.0.14.
+
+    if (versionIsAtLeast(getAlsaModuleVersionString(),
+			 1, 0, 14) &&
+	versionIsAtLeast(getKernelVersionString(),
+			 2, 6, 20)) {
+
+	rtcCouldBeOK = true;
+
+        for (std::vector<AlsaTimerInfo>::iterator i = m_timers.begin();
+	     i != m_timers.end(); ++i) {
+	    if (i->sclas != SND_TIMER_SCLASS_NONE) continue;
+	    if (i->clas == SND_TIMER_CLASS_GLOBAL) {
+		if (i->device == SND_TIMER_GLOBAL_RTC) {
+		    return i->name;
+		}
+	    }
+        }
+    }
+
+    // look for the first PCM playback timer; that's all we know about
+    // for now (until JACK becomes able to tell us which PCM it's on)
+
+    if (pcmTimerAccepted) {
+
+	for (std::vector<AlsaTimerInfo>::iterator i = m_timers.begin();
+	     i != m_timers.end(); ++i) {
+	    if (i->sclas != SND_TIMER_SCLASS_NONE)
+		continue;
+	    if (i->clas == SND_TIMER_CLASS_PCM) {
+		if (i->resolution != 0) {
+		    long hz = 1000000000 / i->resolution;
+		    if (hz >= 750) {
+			wantTimerChecks = false; // pointless with PCM timer
+			return i->name;
+		    } else {
+			audit << "PCM timer: inadequate resolution " << i->resolution << std::endl;
+		    }
+                }
+	    }
+	}
+    }
+
+    // next look for slow, unpopular 100Hz (2.4) or 250Hz (2.6) system timer
+
+    for (std::vector<AlsaTimerInfo>::iterator i = m_timers.begin();
+            i != m_timers.end(); ++i) {
+        if (i->sclas != SND_TIMER_SCLASS_NONE)
+            continue;
+        if (i->clas == SND_TIMER_CLASS_GLOBAL) {
+            if (i->device == SND_TIMER_GLOBAL_SYSTEM) {
+                audit << "Using low-resolution system timer, sending a warning" << std::endl;
+		if (rtcCouldBeOK) {
+		    reportFailure(MappedEvent::WarningImpreciseTimerTryRTC);
+		} else {
+		    reportFailure(MappedEvent::WarningImpreciseTimer);
+		}
+                return i->name;
+            }
+        }
+    }
+
+    // falling back to something that almost certainly won't work,
+    // if for any reason all of the above failed
+
+    return m_timers.begin()->name;
+}
+
+
+
+void
+AlsaDriver::generatePortList(AlsaPortList *newPorts)
+{
+    Audit audit;
+    AlsaPortList alsaPorts;
+
+    snd_seq_client_info_t *cinfo;
+    snd_seq_port_info_t *pinfo;
+    int client;
+    unsigned int writeCap = SND_SEQ_PORT_CAP_SUBS_WRITE | SND_SEQ_PORT_CAP_WRITE;
+    unsigned int readCap = SND_SEQ_PORT_CAP_SUBS_READ | SND_SEQ_PORT_CAP_READ;
+
+    snd_seq_client_info_alloca(&cinfo);
+    snd_seq_client_info_set_client(cinfo, -1);
+
+    audit << std::endl << "  ALSA Client information:"
+    << std::endl << std::endl;
+
+    // Get only the client ports we're interested in and store them
+    // for sorting and then device creation.
+    //
+    while (snd_seq_query_next_client(m_midiHandle, cinfo) >= 0) {
+        client = snd_seq_client_info_get_client(cinfo);
+        snd_seq_port_info_alloca(&pinfo);
+        snd_seq_port_info_set_client(pinfo, client);
+        snd_seq_port_info_set_port(pinfo, -1);
+
+        // Ignore ourselves and the system client
+        //
+        if (client == m_client || client == 0)
+            continue;
+
+        while (snd_seq_query_next_port(m_midiHandle, pinfo) >= 0) {
+            int client = snd_seq_port_info_get_client(pinfo);
+            int port = snd_seq_port_info_get_port(pinfo);
+            unsigned int clientType = snd_seq_client_info_get_type(cinfo);
+            unsigned int portType = snd_seq_port_info_get_type(pinfo);
+            unsigned int capability = snd_seq_port_info_get_capability(pinfo);
+
+
+            if ((((capability & writeCap) == writeCap) ||
+                    ((capability & readCap) == readCap)) &&
+                    ((capability & SND_SEQ_PORT_CAP_NO_EXPORT) == 0)) {
+                audit << "    "
+                << client << ","
+                << port << " - ("
+                << snd_seq_client_info_get_name(cinfo) << ", "
+                << snd_seq_port_info_get_name(pinfo) << ")";
+
+                PortDirection direction;
+
+                if ((capability & SND_SEQ_PORT_CAP_DUPLEX) ||
+                        ((capability & SND_SEQ_PORT_CAP_WRITE) &&
+                         (capability & SND_SEQ_PORT_CAP_READ))) {
+                    direction = Duplex;
+                    audit << "\t\t\t(DUPLEX)";
+                } else if (capability & SND_SEQ_PORT_CAP_WRITE) {
+                    direction = WriteOnly;
+                    audit << "\t\t(WRITE ONLY)";
+                } else {
+                    direction = ReadOnly;
+                    audit << "\t\t(READ ONLY)";
+                }
+
+                audit << " [ctype " << clientType << ", ptype " << portType << ", cap " << capability << "]";
+
+                // Generate a unique name using the client id
+                //
+                char portId[40];
+                sprintf(portId, "%d:%d ", client, port);
+
+                std::string fullClientName =
+                    std::string(snd_seq_client_info_get_name(cinfo));
+
+                std::string fullPortName =
+                    std::string(snd_seq_port_info_get_name(pinfo));
+
+                std::string name;
+
+                // If the first part of the client name is the same as the
+                // start of the port name, just use the port name.  otherwise
+                // concatenate.
+                //
+                int firstSpace = fullClientName.find(" ");
+
+                // If no space is found then we try to match the whole string
+                //
+                if (firstSpace < 0)
+                    firstSpace = fullClientName.length();
+
+                if (firstSpace > 0 &&
+                        int(fullPortName.length()) >= firstSpace &&
+                        fullPortName.substr(0, firstSpace) ==
+                        fullClientName.substr(0, firstSpace)) {
+                    name = portId + fullPortName;
+                } else {
+                    name = portId + fullClientName + ": " + fullPortName;
+                }
+
+                // Sanity check for length
+                //
+                if (name.length() > 35)
+                    name = portId + fullPortName;
+
+                if (direction == WriteOnly) {
+                    name += " (write)";
+                } else if (direction == ReadOnly) {
+                    name += " (read)";
+                } else if (direction == Duplex) {
+                    name += " (duplex)";
+                }
+
+                AlsaPortDescription *portDescription =
+                    new AlsaPortDescription(
+                        Instrument::Midi,
+                        name,
+                        client,
+                        port,
+                        clientType,
+                        portType,
+                        capability,
+                        direction);
+
+                if (newPorts &&
+                        (getPortName(ClientPortPair(client, port)) == "")) {
+                    newPorts->push_back(portDescription);
+                }
+
+                alsaPorts.push_back(portDescription);
+
+                audit << std::endl;
+            }
+        }
+    }
+
+    audit << std::endl;
+
+    // Ok now sort by duplexicity
+    //
+    std::sort(alsaPorts.begin(), alsaPorts.end(), AlsaPortCmp());
+    m_alsaPorts = alsaPorts;
+}
+
+
+void
+AlsaDriver::generateInstruments()
+{
+    // Reset these before each Instrument hunt
+    //
+    int audioCount = 0;
+    getAudioInstrumentNumbers(m_audioRunningId, audioCount);
+    m_midiRunningId = MidiInstrumentBase;
+
+    // Clear these
+    //
+    m_instruments.clear();
+    m_devices.clear();
+    m_devicePortMap.clear();
+    m_suspendedPortMap.clear();
+
+    AlsaPortList::iterator it = m_alsaPorts.begin();
+    for (; it != m_alsaPorts.end(); it++) {
+        if ((*it)->m_client == m_client) {
+            std::cerr << "(Ignoring own port " << (*it)->m_client
+            << ":" << (*it)->m_port << ")" << std::endl;
+            continue;
+        } else if ((*it)->m_client == 0) {
+            std::cerr << "(Ignoring system port " << (*it)->m_client
+            << ":" << (*it)->m_port << ")" << std::endl;
+            continue;
+        }
+
+        if ((*it)->isWriteable()) {
+            MappedDevice *device = createMidiDevice(*it, MidiDevice::Play);
+            if (!device) {
+#ifdef DEBUG_ALSA
+                std::cerr << "WARNING: Failed to create play device" << std::endl;
+#else
+
+                ;
+#endif
+
+            } else {
+                addInstrumentsForDevice(device);
+                m_devices.push_back(device);
+            }
+        }
+        if ((*it)->isReadable()) {
+            MappedDevice *device = createMidiDevice(*it, MidiDevice::Record);
+            if (!device) {
+#ifdef DEBUG_ALSA
+                std::cerr << "WARNING: Failed to create record device" << std::endl;
+#else
+
+                ;
+#endif
+
+            } else {
+                m_devices.push_back(device);
+            }
+        }
+    }
+
+#ifdef HAVE_DSSI
+    // Create a number of soft synth Instruments
+    //
+    {
+        MappedInstrument *instr;
+        char number[100];
+        InstrumentId first;
+        int count;
+        getSoftSynthInstrumentNumbers(first, count);
+
+        DeviceId ssiDeviceId = getSpareDeviceId();
+
+        if (m_driverStatus & AUDIO_OK) {
+            for (int i = 0; i < count; ++i) {
+                sprintf(number, " #%d", i + 1);
+                std::string name = "Synth plugin" + std::string(number);
+                instr = new MappedInstrument(Instrument::SoftSynth,
+                                             i,
+                                             first + i,
+                                             name,
+                                             ssiDeviceId);
+                m_instruments.push_back(instr);
+
+                m_studio->createObject(MappedObject::AudioFader,
+                                       first + i);
+            }
+
+            MappedDevice *device =
+                new MappedDevice(ssiDeviceId,
+                                 Device::SoftSynth,
+                                 "Synth plugin",
+                                 "Soft synth connection");
+            m_devices.push_back(device);
+        }
+    }
+#endif
+
+#ifdef HAVE_LIBJACK
+
+    // Create a number of audio Instruments - these are just
+    // logical Instruments anyway and so we can create as
+    // many as we like and then use them as Tracks.
+    //
+    {
+        MappedInstrument *instr;
+        char number[100];
+        std::string audioName;
+
+        DeviceId audioDeviceId = getSpareDeviceId();
+
+        if (m_driverStatus & AUDIO_OK)
+        {
+            for (int channel = 0; channel < audioCount; ++channel) {
+                sprintf(number, " #%d", channel + 1);
+                audioName = "Audio" + std::string(number);
+                instr = new MappedInstrument(Instrument::Audio,
+                                             channel,
+                                             m_audioRunningId,
+                                             audioName,
+                                             audioDeviceId);
+                m_instruments.push_back(instr);
+
+                // Create a fader with a matching id - this is the starting
+                // point for all audio faders.
+                //
+                m_studio->createObject(MappedObject::AudioFader,
+                                       m_audioRunningId);
+
+                /*
+                std::cerr  << "AlsaDriver::generateInstruments - "
+                           << "added audio fader (id=" << m_audioRunningId
+                           << ")" << std::endl;
+                           */
+
+                m_audioRunningId++;
+            }
+
+            // Create audio device
+            //
+            MappedDevice *device =
+                new MappedDevice(audioDeviceId,
+                                 Device::Audio,
+                                 "Audio",
+                                 "Audio connection");
+            m_devices.push_back(device);
+        }
+    }
+#endif
+
+}
+
+MappedDevice *
+AlsaDriver::createMidiDevice(AlsaPortDescription *port,
+                             MidiDevice::DeviceDirection reqDirection)
+{
+    char deviceName[100];
+    std::string connectionName("");
+    Audit audit;
+
+    static int unknownCounter;
+
+    static int counters[3][2]; // [system/hardware/software][out/in]
+    const int UNKNOWN = -1, SYSTEM = 0, HARDWARE = 1, SOFTWARE = 2;
+    static const char *firstNames[4][2] = {
+                                              { "MIDI output system device", "MIDI input system device"
+                                              },
+                                              { "MIDI external device", "MIDI hardware input device" },
+                                              { "MIDI software device", "MIDI software input" }
+                                          };
+    static const char *countedNames[4][2] = {
+                                                { "MIDI output system device %d", "MIDI input system device %d"
+                                                },
+                                                { "MIDI external device %d", "MIDI hardware input device %d" },
+                                                { "MIDI software device %d", "MIDI software input %d" }
+                                            };
+
+    static int specificCounters[2];
+    static const char *specificNames[2] = {
+                                              "MIDI soundcard synth", "MIDI soft synth",
+                                          };
+    static const char *specificCountedNames[2] = {
+                "MIDI soundcard synth %d", "MIDI soft synth %d",
+            };
+
+    DeviceId deviceId = getSpareDeviceId();
+
+    if (port) {
+
+        if (reqDirection == MidiDevice::Record && !port->isReadable())
+            return 0;
+        if (reqDirection == MidiDevice::Play && !port->isWriteable())
+            return 0;
+
+        int category = UNKNOWN;
+        bool noConnect = false;
+	bool isSynth = false;
+	bool synthKnown = false;
+
+        if (port->m_client < 16) {
+
+            category = SYSTEM;
+            noConnect = true;
+	    isSynth = false;
+	    synthKnown = true;
+
+	} else {
+
+#ifdef SND_SEQ_PORT_TYPE_HARDWARE
+	    if (port->m_portType & SND_SEQ_PORT_TYPE_HARDWARE) {
+		category = HARDWARE;
+	    }
+#endif
+#ifdef SND_SEQ_PORT_TYPE_SOFTWARE
+	    if (port->m_portType & SND_SEQ_PORT_TYPE_SOFTWARE) {
+		category = SOFTWARE;
+	    }
+#endif	    
+#ifdef SND_SEQ_PORT_TYPE_SYNTHESIZER
+	    if (port->m_portType & SND_SEQ_PORT_TYPE_SYNTHESIZER) {
+		isSynth = true;
+		synthKnown = true;
+	    }
+#endif	    
+#ifdef SND_SEQ_PORT_TYPE_APPLICATION
+	    if (port->m_portType & SND_SEQ_PORT_TYPE_APPLICATION) {
+		category = SOFTWARE;
+		isSynth = false;
+		synthKnown = true;
+	    }
+#endif	    
+
+	    if (category == UNKNOWN) {
+
+		if (port->m_client < 64) {
+
+		    if (versionIsAtLeast(getAlsaModuleVersionString(),
+					 1, 0, 11)) {
+
+			category = HARDWARE;
+
+		    } else {
+
+			category = SYSTEM;
+			noConnect = true;
+		    }
+
+		} else if (port->m_client < 128) {
+
+		    category = HARDWARE;
+		    
+		} else {
+		    
+		    category = SOFTWARE;
+		}
+	    }
+	}
+
+        bool haveName = false;
+
+        if (!synthKnown) {
+
+	    if (category != SYSTEM && reqDirection == MidiDevice::Play) {
+
+		// We assume GM/GS/XG/MT32 devices are synths.
+
+		bool isSynth = (port->m_portType &
+				(SND_SEQ_PORT_TYPE_MIDI_GM |
+				 SND_SEQ_PORT_TYPE_MIDI_GS |
+				 SND_SEQ_PORT_TYPE_MIDI_XG |
+				 SND_SEQ_PORT_TYPE_MIDI_MT32));
+
+		if (!isSynth &&
+                    (port->m_name.find("ynth") < port->m_name.length()))
+		    isSynth = true;
+		if (!isSynth &&
+                    (port->m_name.find("nstrument") < port->m_name.length()))
+		    isSynth = true;
+		if (!isSynth &&
+                    (port->m_name.find("VSTi") < port->m_name.length()))
+		    isSynth = true;
+
+	    } else {
+		isSynth = false;
+	    }
+	}
+
+	if (isSynth) {
+	    int clientType = (category == SOFTWARE) ? 1 : 0;
+	    if (specificCounters[clientType] == 0) {
+		sprintf(deviceName, specificNames[clientType]);
+		++specificCounters[clientType];
+	    } else {
+		sprintf(deviceName,
+			specificCountedNames[clientType],
+			++specificCounters[clientType]);
+	    }
+	    haveName = true;
+        }
+
+        if (!haveName) {
+            if (counters[category][reqDirection] == 0) {
+                sprintf(deviceName, firstNames[category][reqDirection]);
+                ++counters[category][reqDirection];
+            } else {
+                sprintf(deviceName,
+                        countedNames[category][reqDirection],
+                        ++counters[category][reqDirection]);
+            }
+        }
+
+        if (!noConnect) {
+            m_devicePortMap[deviceId] = ClientPortPair(port->m_client,
+                                        port->m_port);
+            connectionName = port->m_name;
+        }
+
+        audit << "Creating device " << deviceId << " in "
+        << (reqDirection == MidiDevice::Play ? "Play" : "Record")
+        << " mode for connection " << port->m_name
+        << (noConnect ? " (not connecting)" : "")
+        << "\nDefault device name for this device is "
+        << deviceName << std::endl;
+
+    } else { // !port
+
+        sprintf(deviceName, "Anonymous MIDI device %d", ++unknownCounter);
+
+        audit << "Creating device " << deviceId << " in "
+        << (reqDirection == MidiDevice::Play ? "Play" : "Record")
+        << " mode -- no connection available "
+        << "\nDefault device name for this device is "
+        << deviceName << std::endl;
+    }
+
+    if (reqDirection == MidiDevice::Play) {
+
+        QString portName;
+
+        if (QString(deviceName).startsWith("Anonymous MIDI device ")) {
+            portName = QString("out %1")
+                       .arg(m_outputPorts.size() + 1);
+        } else {
+            portName = QString("out %1 - %2")
+                       .arg(m_outputPorts.size() + 1)
+                       .arg(deviceName);
+        }
+
+        int outputPort = checkAlsaError(snd_seq_create_simple_port
+                                        (m_midiHandle,
+                                         portName,
+                                         SND_SEQ_PORT_CAP_READ |
+                                         SND_SEQ_PORT_CAP_SUBS_READ,
+                                         SND_SEQ_PORT_TYPE_APPLICATION),
+                                        "createMidiDevice - can't create output port");
+
+        if (outputPort >= 0) {
+
+            std::cerr << "CREATED OUTPUT PORT " << outputPort << ":" << portName << " for device " << deviceId << std::endl;
+
+            m_outputPorts[deviceId] = outputPort;
+
+            if (port) {
+                if (connectionName != "") {
+                    std::cerr << "Connecting my port " << outputPort << " to " << port->m_client << ":" << port->m_port << " on initialisation" << std::endl;
+                    snd_seq_connect_to(m_midiHandle,
+                                       outputPort,
+                                       port->m_client,
+                                       port->m_port);
+                    if (m_midiSyncAutoConnect) {
+                        snd_seq_connect_to(m_midiHandle,
+                                           m_syncOutputPort,
+                                           port->m_client,
+                                           port->m_port);
+                    }
+                }
+                std::cerr << "done" << std::endl;
+            }
+        }
+    }
+
+    MappedDevice *device = new MappedDevice(deviceId,
+                                            Device::Midi,
+                                            deviceName,
+                                            connectionName);
+    device->setDirection(reqDirection);
+    return device;
+}
+
+DeviceId
+AlsaDriver::getSpareDeviceId()
+{
+    std::set
+        <DeviceId> ids;
+    for (unsigned int i = 0; i < m_devices.size(); ++i) {
+        ids.insert(m_devices[i]->getId());
+    }
+
+    DeviceId id = 0;
+    while (ids.find(id) != ids.end())
+        ++id;
+    return id;
+}
+
+void
+AlsaDriver::addInstrumentsForDevice(MappedDevice *device)
+{
+    std::string channelName;
+    char number[100];
+
+    for (int channel = 0; channel < 16; ++channel) {
+        // Create MappedInstrument for export to GUI
+        //
+        // name is just number, derive rest from device at gui
+        sprintf(number, "#%d", channel + 1);
+        channelName = std::string(number);
+
+        if (channel == 9)
+            channelName = std::string("#10[D]");
+        MappedInstrument *instr = new MappedInstrument(Instrument::Midi,
+                                  channel,
+                                  m_midiRunningId++,
+                                  channelName,
+                                  device->getId());
+        m_instruments.push_back(instr);
+    }
+}
+
+
+bool
+AlsaDriver::canReconnect(Device::DeviceType type)
+{
+    return (type == Device::Midi);
+}
+
+DeviceId
+AlsaDriver::addDevice(Device::DeviceType type,
+                      MidiDevice::DeviceDirection direction)
+{
+    if (type == Device::Midi) {
+
+        MappedDevice *device = createMidiDevice(0, direction);
+        if (!device) {
+#ifdef DEBUG_ALSA
+            std::cerr << "WARNING: Device creation failed" << std::endl;
+#else
+
+            ;
+#endif
+
+        } else {
+            addInstrumentsForDevice(device);
+            m_devices.push_back(device);
+
+            MappedEvent *mE =
+                new MappedEvent(0, MappedEvent::SystemUpdateInstruments,
+                                0, 0);
+            insertMappedEventForReturn(mE);
+
+            return device->getId();
+        }
+    }
+
+    return Device::NO_DEVICE;
+}
+
+void
+AlsaDriver::removeDevice(DeviceId id)
+{
+    DeviceIntMap::iterator i1 = m_outputPorts.find(id);
+    if (i1 == m_outputPorts.end()) {
+        std::cerr << "WARNING: AlsaDriver::removeDevice: Cannot find device "
+        << id << " in port map" << std::endl;
+        return ;
+    }
+    checkAlsaError( snd_seq_delete_port(m_midiHandle, i1->second),
+                    "removeDevice");
+    m_outputPorts.erase(i1);
+
+    for (MappedDeviceList::iterator i = m_devices.end();
+            i != m_devices.begin(); ) {
+
+        --i;
+
+        if ((*i)->getId() == id) {
+            delete *i;
+            m_devices.erase(i);
+        }
+    }
+
+    for (MappedInstrumentList::iterator i = m_instruments.end();
+            i != m_instruments.begin(); ) {
+
+        --i;
+
+        if ((*i)->getDevice() == id) {
+            delete *i;
+            m_instruments.erase(i);
+        }
+    }
+
+    MappedEvent *mE =
+        new MappedEvent(0, MappedEvent::SystemUpdateInstruments,
+                        0, 0);
+    insertMappedEventForReturn(mE);
+}
+
+void
+AlsaDriver::renameDevice(DeviceId id, QString name)
+{
+    DeviceIntMap::iterator i = m_outputPorts.find(id);
+    if (i == m_outputPorts.end()) {
+        std::cerr << "WARNING: AlsaDriver::renameDevice: Cannot find device "
+        << id << " in port map" << std::endl;
+        return ;
+    }
+
+    snd_seq_port_info_t *pinfo;
+    snd_seq_port_info_alloca(&pinfo);
+    snd_seq_get_port_info(m_midiHandle, i->second, pinfo);
+
+    QString oldName = snd_seq_port_info_get_name(pinfo);
+    int sep = oldName.find(" - ");
+
+    QString newName;
+
+    if (name.startsWith("Anonymous MIDI device ")) {
+        if (sep < 0)
+            sep = 0;
+        newName = oldName.left(sep);
+    } else if (sep < 0) {
+        newName = oldName + " - " + name;
+    } else {
+        newName = oldName.left(sep + 3) + name;
+    }
+
+    snd_seq_port_info_set_name(pinfo, newName.data());
+    checkAlsaError(snd_seq_set_port_info(m_midiHandle, i->second, pinfo),
+                   "renameDevice");
+
+    for (unsigned int i = 0; i < m_devices.size(); ++i) {
+        if (m_devices[i]->getId() == id) {
+            m_devices[i]->setName(newName.data());
+            break;
+        }
+    }
+
+    std::cerr << "Renamed " << m_client << ":" << i->second << " to " << name << std::endl;
+}
+
+ClientPortPair
+AlsaDriver::getPortByName(std::string name)
+{
+    for (unsigned int i = 0; i < m_alsaPorts.size(); ++i) {
+        if (m_alsaPorts[i]->m_name == name) {
+            return ClientPortPair(m_alsaPorts[i]->m_client,
+                                  m_alsaPorts[i]->m_port);
+        }
+    }
+    return ClientPortPair( -1, -1);
+}
+
+std::string
+AlsaDriver::getPortName(ClientPortPair port)
+{
+    for (unsigned int i = 0; i < m_alsaPorts.size(); ++i) {
+        if (m_alsaPorts[i]->m_client == port.first &&
+                m_alsaPorts[i]->m_port == port.second) {
+            return m_alsaPorts[i]->m_name;
+        }
+    }
+    return "";
+}
+
+
+unsigned int
+AlsaDriver::getConnections(Device::DeviceType type,
+                           MidiDevice::DeviceDirection direction)
+{
+    if (type != Device::Midi)
+        return 0;
+
+    int count = 0;
+    for (unsigned int j = 0; j < m_alsaPorts.size(); ++j) {
+        if ((direction == MidiDevice::Play && m_alsaPorts[j]->isWriteable()) ||
+                (direction == MidiDevice::Record && m_alsaPorts[j]->isReadable())) {
+            ++count;
+        }
+    }
+
+    return count;
+}
+
+QString
+AlsaDriver::getConnection(Device::DeviceType type,
+                          MidiDevice::DeviceDirection direction,
+                          unsigned int connectionNo)
+{
+    if (type != Device::Midi)
+        return "";
+
+    AlsaPortList tempList;
+    for (unsigned int j = 0; j < m_alsaPorts.size(); ++j) {
+        if ((direction == MidiDevice::Play && m_alsaPorts[j]->isWriteable()) ||
+                (direction == MidiDevice::Record && m_alsaPorts[j]->isReadable())) {
+            tempList.push_back(m_alsaPorts[j]);
+        }
+    }
+
+    if (connectionNo < tempList.size()) {
+        return tempList[connectionNo]->m_name.c_str();
+    }
+
+    return "";
+}
+
+void
+AlsaDriver::setConnectionToDevice(MappedDevice &device, QString connection)
+{
+    ClientPortPair pair( -1, -1);
+    if (connection && connection != "") {
+        pair = getPortByName(connection.data());
+    }
+    setConnectionToDevice(device, connection, pair);
+}
+
+void
+AlsaDriver::setConnectionToDevice(MappedDevice &device, QString connection,
+                                  const ClientPortPair &pair)
+{
+    QString prevConnection = device.getConnection().c_str();
+    device.setConnection(connection.data());
+
+    if (device.getDirection() == MidiDevice::Play) {
+
+        DeviceIntMap::iterator j = m_outputPorts.find(device.getId());
+
+        if (j != m_outputPorts.end()) {
+
+            if (prevConnection != "") {
+                ClientPortPair prevPair = getPortByName(prevConnection.data());
+                if (prevPair.first >= 0 && prevPair.second >= 0) {
+
+                    std::cerr << "Disconnecting my port " << j->second << " from " << prevPair.first << ":" << prevPair.second << " on reconnection" << std::endl;
+                    snd_seq_disconnect_to(m_midiHandle,
+                                          j->second,
+                                          prevPair.first,
+                                          prevPair.second);
+
+                    if (m_midiSyncAutoConnect) {
+                        bool foundElsewhere = false;
+                        for (MappedDeviceList::iterator k = m_devices.begin();
+                                k != m_devices.end(); ++k) {
+                            if ((*k)->getId() != device.getId()) {
+                                if ((*k)->getConnection() == prevConnection.data()) {
+                                    foundElsewhere = true;
+                                    break;
+                                }
+                            }
+                        }
+                        if (!foundElsewhere) {
+                            snd_seq_disconnect_to(m_midiHandle,
+                                                  m_syncOutputPort,
+                                                  pair.first,
+                                                  pair.second);
+                        }
+                    }
+                }
+            }
+
+            if (pair.first >= 0 && pair.second >= 0) {
+                std::cerr << "Connecting my port " << j->second << " to " << pair.first << ":" << pair.second << " on reconnection" << std::endl;
+                snd_seq_connect_to(m_midiHandle,
+                                   j->second,
+                                   pair.first,
+                                   pair.second);
+                if (m_midiSyncAutoConnect) {
+                    snd_seq_connect_to(m_midiHandle,
+                                       m_syncOutputPort,
+                                       pair.first,
+                                       pair.second);
+                }
+            }
+        }
+    }
+}
+
+void
+AlsaDriver::setConnection(DeviceId id, QString connection)
+{
+    Audit audit;
+    ClientPortPair port(getPortByName(connection.data()));
+
+    if (port.first != -1 && port.second != -1) {
+
+        m_devicePortMap[id] = port;
+
+        for (unsigned int i = 0; i < m_devices.size(); ++i) {
+
+            if (m_devices[i]->getId() == id) {
+                setConnectionToDevice(*m_devices[i], connection, port);
+
+                MappedEvent *mE =
+                    new MappedEvent(0, MappedEvent::SystemUpdateInstruments,
+                                    0, 0);
+                insertMappedEventForReturn(mE);
+
+                break;
+            }
+        }
+    }
+}
+
+void
+AlsaDriver::setPlausibleConnection(DeviceId id, QString idealConnection)
+{
+    Audit audit;
+    ClientPortPair port(getPortByName(idealConnection.data()));
+
+    audit << "AlsaDriver::setPlausibleConnection: connection like "
+    << idealConnection << " requested for device " << id << std::endl;
+
+    if (port.first != -1 && port.second != -1) {
+
+        m_devicePortMap[id] = port;
+
+        for (unsigned int i = 0; i < m_devices.size(); ++i) {
+
+            if (m_devices[i]->getId() == id) {
+                setConnectionToDevice(*m_devices[i], idealConnection, port);
+                break;
+            }
+        }
+
+        audit << "AlsaDriver::setPlausibleConnection: exact match available"
+        << std::endl;
+        return ;
+    }
+
+    // What we want is a connection that:
+    //
+    //  * is in the right "class" (the 0-63/64-127/128+ range of client id)
+    //  * has at least some text in common
+    //  * is not yet in use for any device.
+    //
+    // To do this, we exploit our privileged position as part of AlsaDriver
+    // and use our knowledge of how connection strings are made (see
+    // AlsaDriver::generatePortList above) to pick out the relevant parts
+    // of the requested string.
+
+    int client = -1;
+    int colon = idealConnection.find(":");
+    if (colon >= 0)
+        client = idealConnection.left(colon).toInt();
+
+    int portNo = -1;
+    if (client > 0) {
+        QString remainder = idealConnection.mid(colon + 1);
+        int space = remainder.find(" ");
+        if (space >= 0)
+            portNo = remainder.left(space).toInt();
+    }
+
+    int firstSpace = idealConnection.find(" ");
+    int endOfText = idealConnection.find(QRegExp("[^\\w ]"), firstSpace);
+
+    QString text;
+    if (endOfText < 2) {
+        text = idealConnection.mid(firstSpace + 1);
+    } else {
+        text = idealConnection.mid(firstSpace + 1, endOfText - firstSpace - 2);
+    }
+
+    for (int testUsed = 1; testUsed >= 0; --testUsed) {
+
+        for (int testNumbers = 1; testNumbers >= 0; --testNumbers) {
+
+            for (int testName = 1; testName >= 0; --testName) {
+
+                int fitness =
+                    (testName << 3) +
+                    (testNumbers << 2) +
+                    (testUsed << 1) + 1;
+
+                for (unsigned int i = 0; i < m_alsaPorts.size(); ++i) {
+
+                    AlsaPortDescription *port = m_alsaPorts[i];
+
+                    if (client > 0) {
+
+                        if (port->m_client / 64 != client / 64)
+                            continue;
+
+                        if (testNumbers) {
+                            // We always check the client class (above).
+                            // But we also prefer to have something in
+                            // common with client or port number, at least
+                            // for ports that aren't used elsewhere
+                            // already.  We don't check both because the
+                            // chances are the entire string would already
+                            // have matched if both figures did; instead
+                            // we check the port if it's > 0 (handy for
+                            // e.g. matching the MIDI synth port on a
+                            // multi-port soundcard) and the client
+                            // otherwise.
+                            if (portNo > 0) {
+                                if (port->m_port != portNo)
+                                    continue;
+                            } else {
+                                if (port->m_client != client)
+                                    continue;
+                            }
+                        }
+                    }
+
+                    if (testName && text != "" &&
+                            !QString(port->m_name.c_str()).contains(text))
+                        continue;
+
+                    if (testUsed) {
+                        bool used = false;
+                        for (DevicePortMap::iterator dpmi = m_devicePortMap.begin();
+                                dpmi != m_devicePortMap.end(); ++dpmi) {
+                            if (dpmi->second.first == port->m_client &&
+                                    dpmi->second.second == port->m_port) {
+                                used = true;
+                                break;
+                            }
+                        }
+                        if (used)
+                            continue;
+                    }
+
+                    // OK, this one will do
+
+                    audit << "AlsaDriver::setPlausibleConnection: fuzzy match "
+                    << port->m_name << " available with fitness "
+                    << fitness << std::endl;
+
+                    m_devicePortMap[id] = ClientPortPair(port->m_client, port->m_port);
+
+                    for (unsigned int i = 0; i < m_devices.size(); ++i) {
+
+                        if (m_devices[i]->getId() == id) {
+                            setConnectionToDevice(*m_devices[i],
+                                                  port->m_name.c_str(),
+                                                  m_devicePortMap[id]);
+
+                            // in this case we don't request a device resync,
+                            // because this is only invoked at times such as
+                            // file load when the GUI is well aware that the
+                            // whole situation is in upheaval anyway
+
+                            return ;
+                        }
+                    }
+                }
+            }
+        }
+    }
+
+    audit << "AlsaDriver::setPlausibleConnection: nothing suitable available"
+    << std::endl;
+}
+
+
+void
+AlsaDriver::checkTimerSync(size_t frames)
+{
+    if (!m_doTimerChecks)
+        return ;
+
+#ifdef HAVE_LIBJACK
+
+    if (!m_jackDriver || !m_queueRunning || frames == 0 ||
+            (getMTCStatus() == TRANSPORT_SLAVE)) {
+        m_firstTimerCheck = true;
+        return ;
+    }
+
+    static RealTime startAlsaTime;
+    static size_t startJackFrames = 0;
+    static size_t lastJackFrames = 0;
+
+    size_t nowJackFrames = m_jackDriver->getFramesProcessed();
+    RealTime nowAlsaTime = getAlsaTime();
+
+    if (m_firstTimerCheck ||
+            (nowJackFrames <= lastJackFrames) ||
+            (nowAlsaTime <= startAlsaTime)) {
+
+        startAlsaTime = nowAlsaTime;
+        startJackFrames = nowJackFrames;
+        lastJackFrames = nowJackFrames;
+
+        m_firstTimerCheck = false;
+        return ;
+    }
+
+    RealTime jackDiff = RealTime::frame2RealTime
+                        (nowJackFrames - startJackFrames,
+                         m_jackDriver->getSampleRate());
+
+    RealTime alsaDiff = nowAlsaTime - startAlsaTime;
+
+    if (alsaDiff > RealTime(10, 0)) {
+
+#ifdef DEBUG_ALSA
+        if (!m_playing) {
+            std::cout << "\nALSA:" << startAlsaTime << "\t->" << nowAlsaTime << "\nJACK: " << startJackFrames << "\t\t-> " << nowJackFrames << std::endl;
+            std::cout << "ALSA diff:  " << alsaDiff << "\nJACK diff:  " << jackDiff << std::endl;
+        }
+#endif
+
+        double ratio = (jackDiff - alsaDiff) / alsaDiff;
+
+        if (fabs(ratio) > 0.1) {
+#ifdef DEBUG_ALSA
+            if (!m_playing) {
+                std::cout << "Ignoring excessive ratio " << ratio
+                << ", hoping for a more likely result next time"
+                << std::endl;
+            }
+#endif
+
+        } else if (fabs(ratio) > 0.000001) {
+
+#ifdef DEBUG_ALSA
+            if (alsaDiff > RealTime::zeroTime && jackDiff > RealTime::zeroTime) {
+                if (!m_playing) {
+                    if (jackDiff < alsaDiff) {
+                        std::cout << "<<<< ALSA timer is faster by " << 100.0 * ((alsaDiff - jackDiff) / alsaDiff) << "% (1/" << int(1.0 / ratio) << ")" << std::endl;
+                    } else {
+                        std::cout << ">>>> JACK timer is faster by " << 100.0 * ((jackDiff - alsaDiff) / alsaDiff) << "% (1/" << int(1.0 / ratio) << ")" << std::endl;
+                    }
+                }
+            }
+#endif
+
+            m_timerRatio = ratio;
+            m_timerRatioCalculated = true;
+        }
+
+        m_firstTimerCheck = true;
+    }
+#endif
+}
+
+
+unsigned int
+AlsaDriver::getTimers()
+{
+    return m_timers.size() + 1; // one extra for auto
+}
+
+QString
+AlsaDriver::getTimer(unsigned int n)
+{
+    if (n == 0)
+        return AUTO_TIMER_NAME;
+    else
+        return m_timers[n -1].name.c_str();
+}
+
+QString
+AlsaDriver::getCurrentTimer()
+{
+    return m_currentTimer.c_str();
+}
+
+void
+AlsaDriver::setCurrentTimer(QString timer)
+{
+    Audit audit;
+
+    if (timer == getCurrentTimer())
+        return ;
+
+    std::cerr << "AlsaDriver::setCurrentTimer(" << timer << ")" << std::endl;
+
+    std::string name(timer.data());
+
+    if (name == AUTO_TIMER_NAME) {
+        name = getAutoTimer(m_doTimerChecks);
+    } else {
+        m_doTimerChecks = false;
+    }
+    m_timerRatioCalculated = false;
+
+    // Stop and restart the queue around the timer change.  We don't
+    // call stopClocks/startClocks here because they do the wrong
+    // thing if we're currently playing and on the JACK transport.
+
+    m_queueRunning = false;
+    checkAlsaError(snd_seq_stop_queue(m_midiHandle, m_queue, NULL), "setCurrentTimer(): stopping queue");
+    checkAlsaError(snd_seq_drain_output(m_midiHandle), "setCurrentTimer(): draining output to stop queue");
+
+    snd_seq_event_t event;
+    snd_seq_ev_clear(&event);
+    snd_seq_real_time_t z = { 0, 0 };
+    snd_seq_ev_set_queue_pos_real(&event, m_queue, &z);
+    snd_seq_ev_set_direct(&event);
+    checkAlsaError(snd_seq_control_queue(m_midiHandle, m_queue, SND_SEQ_EVENT_SETPOS_TIME,
+                                         0, &event), "setCurrentTimer(): control queue");
+    checkAlsaError(snd_seq_drain_output(m_midiHandle), "setCurrentTimer(): draining output to control queue");
+    m_alsaPlayStartTime = RealTime::zeroTime;
+
+    for (unsigned int i = 0; i < m_timers.size(); ++i) {
+        if (m_timers[i].name == name) {
+
+            snd_seq_queue_timer_t *timer;
+            snd_timer_id_t *timerid;
+
+            snd_seq_queue_timer_alloca(&timer);
+            snd_seq_get_queue_timer(m_midiHandle, m_queue, timer);
+
+            snd_timer_id_alloca(&timerid);
+            snd_timer_id_set_class(timerid, m_timers[i].clas);
+            snd_timer_id_set_sclass(timerid, m_timers[i].sclas);
+            snd_timer_id_set_card(timerid, m_timers[i].card);
+            snd_timer_id_set_device(timerid, m_timers[i].device);
+            snd_timer_id_set_subdevice(timerid, m_timers[i].subdevice);
+
+            snd_seq_queue_timer_set_id(timer, timerid);
+            snd_seq_set_queue_timer(m_midiHandle, m_queue, timer);
+
+            if (m_doTimerChecks) {
+                audit << "    Current timer set to \"" << name << "\" with timer checks"
+                << std::endl;
+            } else {
+                audit << "    Current timer set to \"" << name << "\""
+                << std::endl;
+            }
+
+            if (m_timers[i].clas == SND_TIMER_CLASS_GLOBAL &&
+                    m_timers[i].device == SND_TIMER_GLOBAL_SYSTEM) {
+                long hz = 1000000000 / m_timers[i].resolution;
+                if (hz < 900) {
+                    audit << "    WARNING: using system timer with only "
+                    << hz << "Hz resolution!" << std::endl;
+                }
+            }
+
+            break;
+        }
+    }
+
+#ifdef HAVE_LIBJACK
+    if (m_jackDriver)
+        m_jackDriver->prebufferAudio();
+#endif
+
+    checkAlsaError(snd_seq_continue_queue(m_midiHandle, m_queue, NULL), "checkAlsaError(): continue queue");
+    checkAlsaError(snd_seq_drain_output(m_midiHandle), "setCurrentTimer(): draining output to continue queue");
+    m_queueRunning = true;
+
+    m_firstTimerCheck = true;
+}
+
+bool
+AlsaDriver::initialise()
+{
+    bool result = true;
+
+    initialiseAudio();
+    result = initialiseMidi();
+
+    return result;
+}
+
+
+
+// Set up queue, client and port
+//
+bool
+AlsaDriver::initialiseMidi()
+{
+    Audit audit;
+
+    // Create a non-blocking handle.
+    //
+    if (snd_seq_open(&m_midiHandle,
+                     "default",
+                     SND_SEQ_OPEN_DUPLEX,
+                     SND_SEQ_NONBLOCK) < 0) {
+        audit << "AlsaDriver::initialiseMidi - "
+        << "couldn't open sequencer - " << snd_strerror(errno)
+        << " - perhaps you need to modprobe snd-seq-midi."
+        << std::endl;
+        reportFailure(MappedEvent::FailureALSACallFailed);
+        return false;
+    }
+
+    snd_seq_set_client_name(m_midiHandle, "rosegarden");
+
+    if ((m_client = snd_seq_client_id(m_midiHandle)) < 0) {
+#ifdef DEBUG_ALSA
+        std::cerr << "AlsaDriver::initialiseMidi - can't create client"
+        << std::endl;
+#endif
+
+        return false;
+    }
+
+    // Create a queue
+    //
+    if ((m_queue = snd_seq_alloc_named_queue(m_midiHandle,
+                   "Rosegarden queue")) < 0) {
+#ifdef DEBUG_ALSA
+        std::cerr << "AlsaDriver::initialiseMidi - can't allocate queue"
+        << std::endl;
+#endif
+
+        return false;
+    }
+
+    // Create the input port
+    //
+    snd_seq_port_info_t *pinfo;
+
+    snd_seq_port_info_alloca(&pinfo);
+    snd_seq_port_info_set_capability(pinfo,
+                                     SND_SEQ_PORT_CAP_WRITE |
+                                     SND_SEQ_PORT_CAP_SUBS_WRITE );
+    snd_seq_port_info_set_type(pinfo, SND_SEQ_PORT_TYPE_APPLICATION);
+    snd_seq_port_info_set_midi_channels(pinfo, 16);
+    /* we want to know when the events got delivered to us */
+    snd_seq_port_info_set_timestamping(pinfo, 1);
+    snd_seq_port_info_set_timestamp_real(pinfo, 1);
+    snd_seq_port_info_set_timestamp_queue(pinfo, m_queue);
+    snd_seq_port_info_set_name(pinfo, "record in");
+
+    if (checkAlsaError(snd_seq_create_port(m_midiHandle, pinfo),
+                       "initialiseMidi - can't create input port") < 0)
+        return false;
+    m_inputPort = snd_seq_port_info_get_port(pinfo);
+
+    // Subscribe the input port to the ALSA Announce port
+    // to receive notifications when clients, ports and subscriptions change
+    snd_seq_connect_from( m_midiHandle, m_inputPort,
+                          SND_SEQ_CLIENT_SYSTEM, SND_SEQ_PORT_SYSTEM_ANNOUNCE );
+
+    m_midiInputPortConnected = true;
+
+    // Set the input queue size
+    //
+    if (snd_seq_set_client_pool_output(m_midiHandle, 2000) < 0 ||
+            snd_seq_set_client_pool_input(m_midiHandle, 2000) < 0 ||
+            snd_seq_set_client_pool_output_room(m_midiHandle, 2000) < 0) {
+#ifdef DEBUG_ALSA
+        std::cerr << "AlsaDriver::initialiseMidi - "
+        << "can't modify pool parameters"
+        << std::endl;
+#endif
+
+        return false;
+    }
+
+    // Create sync output now as well
+    m_syncOutputPort = checkAlsaError(snd_seq_create_simple_port
+                                      (m_midiHandle,
+                                       "sync out",
+                                       SND_SEQ_PORT_CAP_READ |
+                                       SND_SEQ_PORT_CAP_SUBS_READ,
+                                       SND_SEQ_PORT_TYPE_APPLICATION),
+                                      "initialiseMidi - can't create sync output port");
+
+    // and port for hardware controller
+    m_controllerPort = checkAlsaError(snd_seq_create_simple_port
+                                      (m_midiHandle,
+                                       "external controller",
+                                       SND_SEQ_PORT_CAP_READ |
+                                       SND_SEQ_PORT_CAP_WRITE |
+                                       SND_SEQ_PORT_CAP_SUBS_READ |
+                                       SND_SEQ_PORT_CAP_SUBS_WRITE,
+                                       SND_SEQ_PORT_TYPE_APPLICATION),
+                                      "initialiseMidi - can't create controller port");
+
+    getSystemInfo();
+
+    generatePortList();
+    generateInstruments();
+
+    // Modify status with MIDI success
+    //
+    m_driverStatus |= MIDI_OK;
+
+    generateTimerList();
+    setCurrentTimer(AUTO_TIMER_NAME);
+
+    // Start the timer
+    if (checkAlsaError(snd_seq_start_queue(m_midiHandle, m_queue, NULL),
+                       "initialiseMidi(): couldn't start queue") < 0) {
+        reportFailure(MappedEvent::FailureALSACallFailed);
+        return false;
+    }
+
+    m_queueRunning = true;
+
+    // process anything pending
+    checkAlsaError(snd_seq_drain_output(m_midiHandle), "initialiseMidi(): couldn't drain output");
+
+    audit << "AlsaDriver::initialiseMidi -  initialised MIDI subsystem"
+    << std::endl << std::endl;
+
+    return true;
+}
+
+// We don't even attempt to use ALSA audio.  We just use JACK instead.
+// See comment at the top of this file and jackProcess() for further
+// information on how we use this.
+//
+void
+AlsaDriver::initialiseAudio()
+{
+#ifdef HAVE_LIBJACK
+    m_jackDriver = new JackDriver(this);
+
+    if (m_jackDriver->isOK()) {
+        m_driverStatus |= AUDIO_OK;
+    } else {
+        delete m_jackDriver;
+        m_jackDriver = 0;
+    }
+#endif
+}
+
+void
+AlsaDriver::initialisePlayback(const RealTime &position)
+{
+#ifdef DEBUG_ALSA
+    std::cerr << "\n\nAlsaDriver - initialisePlayback" << std::endl;
+#endif
+
+    // now that we restart the queue at each play, the origin is always zero
+    m_alsaPlayStartTime = RealTime::zeroTime;
+    m_playStartPosition = position;
+
+    m_startPlayback = true;
+
+    m_mtcFirstTime = -1;
+    m_mtcSigmaE = 0;
+    m_mtcSigmaC = 0;
+
+    if (getMMCStatus() == TRANSPORT_MASTER) {
+        sendMMC(127, MIDI_MMC_PLAY, true, "");
+        m_eat_mtc = 0;
+    }
+
+    if (getMTCStatus() == TRANSPORT_MASTER) {
+        insertMTCFullFrame(position);
+    }
+
+    // If MIDI Sync is enabled then adjust for the MIDI Clock to
+    // synchronise the sequencer with the clock.
+    //
+    if (getMIDISyncStatus() == TRANSPORT_MASTER) {
+        // Send the Song Position Pointer for MIDI CLOCK positioning
+        //
+        // Get time from current alsa time to start of alsa timing -
+        // add the initial starting point and divide by the MIDI Beat
+        // length.  The SPP is is the MIDI Beat upon which to start the song.
+        // Songs are always assumed to start on a MIDI Beat of 0. Each MIDI
+        // Beat spans 6 MIDI Clocks. In other words, each MIDI Beat is a 16th
+        // note (since there are 24 MIDI Clocks in a quarter note).
+        //
+        long spp =
+            long(((getAlsaTime() - m_alsaPlayStartTime + m_playStartPosition) /
+                  m_midiClockInterval) / 6.0 );
+
+        // Ok now we have the new SPP - stop the transport and restart with the
+        // new value.
+        //
+        sendSystemDirect(SND_SEQ_EVENT_STOP, NULL);
+
+        signed int args = spp;
+        sendSystemDirect(SND_SEQ_EVENT_SONGPOS, &args);
+
+        // Now send the START/CONTINUE
+        //
+        if (m_playStartPosition == RealTime::zeroTime)
+            sendSystemQueued(SND_SEQ_EVENT_START, "",
+                             m_alsaPlayStartTime);
+        else
+            sendSystemQueued(SND_SEQ_EVENT_CONTINUE, "",
+                             m_alsaPlayStartTime);
+    }
+
+#ifdef HAVE_LIBJACK
+    if (m_jackDriver) {
+        m_needJackStart = NeedJackStart;
+    }
+#endif
+}
+
+
+void
+AlsaDriver::stopPlayback()
+{
+#ifdef DEBUG_ALSA
+    std::cerr << "\n\nAlsaDriver - stopPlayback" << std::endl;
+#endif
+
+    if (getMIDISyncStatus() == TRANSPORT_MASTER) {
+        sendSystemDirect(SND_SEQ_EVENT_STOP, NULL);
+    }
+
+    if (getMMCStatus() == TRANSPORT_MASTER) {
+        sendMMC(127, MIDI_MMC_STOP, true, "");
+        //<VN> need to throw away the next MTC event
+        m_eat_mtc = 3;
+    }
+
+    allNotesOff();
+    m_playing = false;
+
+#ifdef HAVE_LIBJACK
+    if (m_jackDriver) {
+        m_jackDriver->stopTransport();
+        m_needJackStart = NeedNoJackStart;
+    }
+#endif
+
+    // Flush the output and input queues
+    //
+    snd_seq_remove_events_t *info;
+    snd_seq_remove_events_alloca(&info);
+    snd_seq_remove_events_set_condition(info, SND_SEQ_REMOVE_INPUT |
+                                        SND_SEQ_REMOVE_OUTPUT);
+    snd_seq_remove_events(m_midiHandle, info);
+
+    // send sounds-off to all play devices
+    //
+    for (MappedDeviceList::iterator i = m_devices.begin(); i != m_devices.end(); ++i) {
+        if ((*i)->getDirection() == MidiDevice::Play) {
+            sendDeviceController((*i)->getId(),
+                                 MIDI_CONTROLLER_SUSTAIN, 0);
+            sendDeviceController((*i)->getId(),
+                                 MIDI_CONTROLLER_ALL_NOTES_OFF, 0);
+        }
+    }
+
+    punchOut();
+
+    stopClocks(); // Resets ALSA timer to zero
+
+    clearAudioQueue();
+
+    startClocksApproved(); // restarts ALSA timer without starting JACK transport
+}
+
+void
+AlsaDriver::punchOut()
+{
+#ifdef DEBUG_ALSA
+    std::cerr << "AlsaDriver::punchOut" << std::endl;
+#endif
+
+#ifdef HAVE_LIBJACK
+    // Close any recording file
+    if (m_recordStatus == RECORD_ON) {
+        for (InstrumentSet::const_iterator i = m_recordingInstruments.begin();
+                i != m_recordingInstruments.end(); ++i) {
+
+            InstrumentId id = *i;
+
+            if (id >= AudioInstrumentBase &&
+                    id < MidiInstrumentBase) {
+
+                AudioFileId auid = 0;
+                if (m_jackDriver && m_jackDriver->closeRecordFile(id, auid)) {
+
+#ifdef DEBUG_ALSA
+                    std::cerr << "AlsaDriver::stopPlayback: sending back to GUI for instrument " << id << std::endl;
+#endif
+
+                    // Create event to return to gui to say that we've
+                    // completed an audio file and we can generate a
+                    // preview for it now.
+                    //
+                    // nasty hack -- don't have right audio id here, and
+                    // the sequencer will wipe out the instrument id and
+                    // replace it with currently-selected one in gui --
+                    // so use audio id slot to pass back instrument id
+                    // and handle accordingly in gui
+                    try {
+                        MappedEvent *mE =
+                            new MappedEvent(id,
+                                            MappedEvent::AudioGeneratePreview,
+                                            id % 256,
+                                            id / 256);
+
+                        // send completion event
+                        insertMappedEventForReturn(mE);
+                    } catch (...) {
+                        ;
+                    }
+                }
+            }
+        }
+    }
+#endif
+
+    // Change recorded state if any set
+    //
+    if (m_recordStatus == RECORD_ON)
+        m_recordStatus = RECORD_OFF;
+
+    m_recordingInstruments.clear();
+}
+
+void
+AlsaDriver::resetPlayback(const RealTime &oldPosition, const RealTime &position)
+{
+#ifdef DEBUG_ALSA
+    std::cerr << "\n\nAlsaDriver - resetPlayback(" << oldPosition << "," << position << ")" << std::endl;
+#endif
+
+    if (getMMCStatus() == TRANSPORT_MASTER) {
+        unsigned char t_sec = (unsigned char) position.sec % 60;
+        unsigned char t_min = (unsigned char) (position.sec / 60) % 60;
+        unsigned char t_hrs = (unsigned char) (position.sec / 3600);
+#define STUPID_BROKEN_EQUIPMENT
+#ifdef STUPID_BROKEN_EQUIPMENT
+        // Some recorders assume you are talking in 30fps...
+        unsigned char t_frm = (unsigned char) (position.nsec / 33333333U);
+        unsigned char t_sbf = (unsigned char) ((position.nsec / 333333U) % 100U);
+#else
+        // We always send at 25fps, it's the easiest to avoid rounding problems
+        unsigned char t_frm = (unsigned char) (position.nsec / 40000000U);
+        unsigned char t_sbf = (unsigned char) ((position.nsec / 400000U) % 100U);
+#endif
+
+        std::cerr << "\n Jump using MMC LOCATE to" << position << std::endl;
+        std::cerr << "\t which is " << int(t_hrs) << ":" << int(t_min) << ":" << int(t_sec) << "." << int(t_frm) << "." << int(t_sbf) << std::endl;
+        unsigned char locateDataArr[7] = {
+                                             0x06,
+                                             0x01,
+                                             0x60 + t_hrs,    // (30fps flag) + hh
+                                             t_min, 	    // mm
+                                             t_sec, 	    // ss
+                                             t_frm, 	    // frames
+                                             t_sbf	    // subframes
+                                         };
+
+        sendMMC(127, MIDI_MMC_LOCATE, true, std::string((const char *) locateDataArr, 7));
+    }
+
+    RealTime formerStartPosition = m_playStartPosition;
+
+    m_playStartPosition = position;
+    m_alsaPlayStartTime = getAlsaTime();
+
+    // Reset note offs to correct positions
+    //
+    RealTime jump = position - oldPosition;
+
+#ifdef DEBUG_PROCESS_MIDI_OUT
+    std::cerr << "Currently " << m_noteOffQueue.size() << " in note off queue" << std::endl;
+#endif
+
+    // modify the note offs that exist as they're relative to the
+    // playStartPosition terms.
+    //
+    for (NoteOffQueue::iterator i = m_noteOffQueue.begin();
+            i != m_noteOffQueue.end(); ++i) {
+
+        // if we're fast forwarding then we bring the note off closer
+        if (jump >= RealTime::zeroTime) {
+
+	    RealTime endTime = formerStartPosition + (*i)->getRealTime();
+
+#ifdef DEBUG_PROCESS_MIDI_OUT
+	    std::cerr << "Forward jump of " << jump << ": adjusting note off from "
+		      << (*i)->getRealTime() << " (absolute " << endTime
+		      << ") to ";
+#endif
+	    (*i)->setRealTime(endTime - position);
+#ifdef DEBUG_PROCESS_MIDI_OUT
+	    std::cerr << (*i)->getRealTime() << std::endl;
+#endif
+        } else // we're rewinding - kill the note immediately
+        {
+#ifdef DEBUG_PROCESS_MIDI_OUT
+	    std::cerr << "Rewind by " << jump << ": setting note off to zero" << std::endl;
+#endif
+	    (*i)->setRealTime(RealTime::zeroTime);
+        }
+    }
+
+    pushRecentNoteOffs();
+    processNotesOff(getAlsaTime(), true);
+    checkAlsaError(snd_seq_drain_output(m_midiHandle), "resetPlayback(): draining");
+
+    // Ensure we clear down output queue on reset - in the case of
+    // MIDI clock where we might have a long queue of events already
+    // posted.
+    //
+    snd_seq_remove_events_t *info;
+    snd_seq_remove_events_alloca(&info);
+    snd_seq_remove_events_set_condition(info, SND_SEQ_REMOVE_OUTPUT);
+    snd_seq_remove_events(m_midiHandle, info);
+
+    if (getMTCStatus() == TRANSPORT_MASTER) {
+        m_mtcFirstTime = -1;
+        m_mtcSigmaE = 0;
+        m_mtcSigmaC = 0;
+        insertMTCFullFrame(position);
+    }
+
+#ifdef HAVE_LIBJACK
+    if (m_jackDriver) {
+	m_jackDriver->clearSynthPluginEvents();
+        m_needJackStart = NeedJackReposition;
+    }
+#endif
+}
+
+void
+AlsaDriver::setMIDIClockInterval(RealTime interval)
+{
+#ifdef DEBUG_ALSA
+    std::cerr << "AlsaDriver::setMIDIClockInterval(" << interval << ")" << endl;
+#endif
+
+    // Reset the value
+    //
+    SoundDriver::setMIDIClockInterval(interval);
+
+    // Return if the clock isn't enabled
+    //
+    if (!m_midiClockEnabled)
+        return ;
+
+    if (false)  // don't remove any events quite yet
+    {
+
+        // Remove all queued events (although we should filter this
+        // down to just the clock events.
+        //
+        snd_seq_remove_events_t *info;
+        snd_seq_remove_events_alloca(&info);
+
+        //if (snd_seq_type_check(SND_SEQ_EVENT_CLOCK, SND_SEQ_EVFLG_CONTROL))
+        //snd_seq_remove_events_set_event_type(info,
+        snd_seq_remove_events_set_condition(info, SND_SEQ_REMOVE_OUTPUT);
+        snd_seq_remove_events_set_event_type(info, SND_SEQ_EVFLG_CONTROL);
+        std::cout << "AlsaDriver::setMIDIClockInterval - "
+        << "MIDI CLOCK TYPE IS CONTROL" << std::endl;
+        snd_seq_remove_events(m_midiHandle, info);
+    }
+
+}
+
+
+void
+AlsaDriver::pushRecentNoteOffs()
+{
+#ifdef DEBUG_PROCESS_MIDI_OUT
+    std::cerr << "AlsaDriver::pushRecentNoteOffs: have " << m_recentNoteOffs.size() << " in queue" << std::endl;
+#endif
+
+    for (NoteOffQueue::iterator i = m_recentNoteOffs.begin();
+	 i != m_recentNoteOffs.end(); ++i) {
+	(*i)->setRealTime(RealTime::zeroTime);
+	m_noteOffQueue.insert(*i);
+    }
+
+    m_recentNoteOffs.clear();
+}
+
+void
+AlsaDriver::cropRecentNoteOffs(const RealTime &t)
+{
+    while (!m_recentNoteOffs.empty()) {
+	NoteOffEvent *ev = *m_recentNoteOffs.begin();
+#ifdef DEBUG_PROCESS_MIDI_OUT
+	std::cerr << "AlsaDriver::cropRecentNoteOffs: " << ev->getRealTime() << " vs " << t << std::endl;
+#endif
+	if (ev->getRealTime() >= t) break;
+	delete ev;
+	m_recentNoteOffs.erase(m_recentNoteOffs.begin());
+    }
+}
+
+void
+AlsaDriver::weedRecentNoteOffs(unsigned int pitch, MidiByte channel,
+			       InstrumentId instrument)
+{
+    for (NoteOffQueue::iterator i = m_recentNoteOffs.begin();
+	 i != m_recentNoteOffs.end(); ++i) {
+	if ((*i)->getPitch() == pitch &&
+	    (*i)->getChannel() == channel && 
+	    (*i)->getInstrument() == instrument) {
+#ifdef DEBUG_PROCESS_MIDI_OUT
+	    std::cerr << "AlsaDriver::weedRecentNoteOffs: deleting one" << std::endl;
+#endif
+	    delete *i;
+	    m_recentNoteOffs.erase(i);
+	    break;
+	}
+    }
+}
+
+void
+AlsaDriver::allNotesOff()
+{
+    snd_seq_event_t event;
+    ClientPortPair outputDevice;
+    RealTime offTime;
+
+    // drop any pending notes
+    snd_seq_drop_output_buffer(m_midiHandle);
+    snd_seq_drop_output(m_midiHandle);
+
+    // prepare the event
+    snd_seq_ev_clear(&event);
+    offTime = getAlsaTime();
+
+    for (NoteOffQueue::iterator it = m_noteOffQueue.begin();
+            it != m_noteOffQueue.end(); ++it) {
+        // Set destination according to connection for instrument
+        //
+        outputDevice = getPairForMappedInstrument((*it)->getInstrument());
+        if (outputDevice.first < 0 || outputDevice.second < 0)
+            continue;
+
+        snd_seq_ev_set_subs(&event);
+
+        // Set source according to port for device
+        //
+        int src = getOutputPortForMappedInstrument((*it)->getInstrument());
+        if (src < 0)
+            continue;
+        snd_seq_ev_set_source(&event, src);
+
+        snd_seq_ev_set_noteoff(&event,
+                               (*it)->getChannel(),
+                               (*it)->getPitch(),
+                               127);
+
+        //snd_seq_event_output(m_midiHandle, &event);
+        int error = snd_seq_event_output_direct(m_midiHandle, &event);
+
+        if (error < 0) {
+#ifdef DEBUG_ALSA
+            std::cerr << "AlsaDriver::allNotesOff - "
+            << "can't send event" << std::endl;
+#endif
+
+        }
+
+        delete(*it);
+    }
+
+    m_noteOffQueue.erase(m_noteOffQueue.begin(), m_noteOffQueue.end());
+
+    /*
+    std::cerr << "AlsaDriver::allNotesOff - "
+              << " queue size = " << m_noteOffQueue.size() << std::endl;
+              */
+
+    // flush
+    checkAlsaError(snd_seq_drain_output(m_midiHandle), "allNotesOff(): draining");
+}
+
+void
+AlsaDriver::processNotesOff(const RealTime &time, bool now, bool everything)
+{
+    if (m_noteOffQueue.empty()) {
+        return;
+    }
+
+    snd_seq_event_t event;
+
+    ClientPortPair outputDevice;
+    RealTime offTime;
+
+    // prepare the event
+    snd_seq_ev_clear(&event);
+
+    RealTime alsaTime = getAlsaTime();
+
+#ifdef DEBUG_PROCESS_MIDI_OUT
+    std::cerr << "AlsaDriver::processNotesOff(" << time << "): alsaTime = " << alsaTime << ", now = " << now << std::endl;
+#endif
+
+    while (m_noteOffQueue.begin() != m_noteOffQueue.end()) {
+
+        NoteOffEvent *ev = *m_noteOffQueue.begin();
+
+	if (ev->getRealTime() > time) {
+#ifdef DEBUG_PROCESS_MIDI_OUT
+	    std::cerr << "Note off time " << ev->getRealTime() << " is beyond current time " << time << std::endl;
+#endif
+	    if (!everything) break;
+	}
+
+#ifdef DEBUG_PROCESS_MIDI_OUT
+        std::cerr << "AlsaDriver::processNotesOff(" << time << "): found event at " << ev->getRealTime() << ", instr " << ev->getInstrument() << ", channel " << int(ev->getChannel()) << ", pitch " << int(ev->getPitch()) << std::endl;
+#endif
+
+        bool isSoftSynth = (ev->getInstrument() >= SoftSynthInstrumentBase);
+
+        offTime = ev->getRealTime();
+	if (offTime < RealTime::zeroTime) offTime = RealTime::zeroTime;
+	bool scheduled = (offTime > alsaTime) && !now;
+	if (!scheduled) offTime = RealTime::zeroTime;
+
+        snd_seq_real_time_t alsaOffTime = { offTime.sec,
+                                            offTime.nsec };
+
+	snd_seq_ev_set_noteoff(&event,
+			       ev->getChannel(),
+			       ev->getPitch(),
+			       127);
+
+        if (!isSoftSynth) {
+
+            snd_seq_ev_set_subs(&event);
+
+            // Set source according to instrument
+            //
+            int src = getOutputPortForMappedInstrument(ev->getInstrument());
+            if (src < 0) {
+		std::cerr << "note off has no output port (instr = " << ev->getInstrument() << ")" << std::endl;
+                delete ev;
+                m_noteOffQueue.erase(m_noteOffQueue.begin());
+                continue;
+            }
+
+            snd_seq_ev_set_source(&event, src);
+
+            snd_seq_ev_set_subs(&event);
+
+	    snd_seq_ev_schedule_real(&event, m_queue, 0, &alsaOffTime);
+
+	    if (scheduled) {
+		snd_seq_event_output(m_midiHandle, &event);
+	    } else {
+		snd_seq_event_output_direct(m_midiHandle, &event);
+	    }
+
+        } else {
+
+            event.time.time = alsaOffTime;
+
+            processSoftSynthEventOut(ev->getInstrument(), &event, now);
+        }
+
+	if (!now) {
+	    m_recentNoteOffs.insert(ev);
+	} else {
+	    delete ev;
+	}
+        m_noteOffQueue.erase(m_noteOffQueue.begin());
+    }
+
+    // We don't flush the queue here, as this is called nested from
+    // processMidiOut, which does the flushing
+
+#ifdef DEBUG_PROCESS_MIDI_OUT
+    std::cerr << "AlsaDriver::processNotesOff - "
+    << " queue size now: " << m_noteOffQueue.size() << std::endl;
+#endif
+}
+
+// Get the queue time and convert it to RealTime for the gui
+// to use.
+//
+RealTime
+AlsaDriver::getSequencerTime()
+{
+    RealTime t(0, 0);
+
+    t = getAlsaTime() + m_playStartPosition - m_alsaPlayStartTime;
+
+    //    std::cerr << "AlsaDriver::getSequencerTime: alsa time is "
+    //	      << getAlsaTime() << ", start time is " << m_alsaPlayStartTime << ", play start position is " << m_playStartPosition << endl;
+
+    return t;
+}
+
+// Gets the time of the ALSA queue
+//
+RealTime
+AlsaDriver::getAlsaTime()
+{
+    RealTime sequencerTime(0, 0);
+
+    snd_seq_queue_status_t *status;
+    snd_seq_queue_status_alloca(&status);
+
+    if (snd_seq_get_queue_status(m_midiHandle, m_queue, status) < 0) {
+#ifdef DEBUG_ALSA
+        std::cerr << "AlsaDriver::getAlsaTime - can't get queue status"
+        << std::endl;
+#endif
+
+        return sequencerTime;
+    }
+
+    sequencerTime.sec = snd_seq_queue_status_get_real_time(status)->tv_sec;
+    sequencerTime.nsec = snd_seq_queue_status_get_real_time(status)->tv_nsec;
+
+    //    std::cerr << "AlsaDriver::getAlsaTime: alsa time is " << sequencerTime << std::endl;
+
+    return sequencerTime;
+}
+
+
+// Get all pending input events and turn them into a MappedComposition.
+//
+//
+MappedComposition*
+AlsaDriver::getMappedComposition()
+{
+    m_recordComposition.clear();
+
+    while (_failureReportReadIndex != _failureReportWriteIndex) {
+        MappedEvent::FailureCode code = _failureReports[_failureReportReadIndex];
+        //	std::cerr << "AlsaDriver::reportFailure(" << code << ")" << std::endl;
+        MappedEvent *mE = new MappedEvent
+                          (0, MappedEvent::SystemFailure, code, 0);
+        m_returnComposition.insert(mE);
+        _failureReportReadIndex =
+            (_failureReportReadIndex + 1) % FAILURE_REPORT_COUNT;
+    }
+
+    if (!m_returnComposition.empty()) {
+        for (MappedComposition::iterator i = m_returnComposition.begin();
+                i != m_returnComposition.end(); ++i) {
+            m_recordComposition.insert(new MappedEvent(**i));
+        }
+        m_returnComposition.clear();
+    }
+
+    // If the input port hasn't connected we shouldn't poll it
+    //
+    if (m_midiInputPortConnected == false) {
+        return &m_recordComposition;
+    }
+
+    RealTime eventTime(0, 0);
+
+    snd_seq_event_t *event;
+
+    while (snd_seq_event_input(m_midiHandle, &event) > 0) {
+
+        unsigned int channel = (unsigned int)event->data.note.channel;
+        unsigned int chanNoteKey = ( channel << 8 ) +
+                                   (unsigned int) event->data.note.note;
+
+        bool fromController = false;
+
+        if (event->dest.client == m_client &&
+	    event->dest.port == m_controllerPort) {
+#ifdef DEBUG_ALSA
+            std::cerr << "Received an external controller event" << std::endl;
+#endif
+
+            fromController = true;
+        }
+
+        unsigned int deviceId = Device::NO_DEVICE;
+
+        if (fromController) {
+            deviceId = Device::CONTROL_DEVICE;
+        } else {
+            for (MappedDeviceList::iterator i = m_devices.begin();
+                    i != m_devices.end(); ++i) {
+                ClientPortPair pair(m_devicePortMap[(*i)->getId()]);
+                if (((*i)->getDirection() == MidiDevice::Record) &&
+                        ( pair.first == event->source.client ) &&
+                        ( pair.second == event->source.port )) {
+                    deviceId = (*i)->getId();
+                    break;
+                }
+            }
+        }
+
+        eventTime.sec = event->time.time.tv_sec;
+        eventTime.nsec = event->time.time.tv_nsec;
+        eventTime = eventTime - m_alsaRecordStartTime + m_playStartPosition;
+
+#ifdef DEBUG_ALSA
+	if (!fromController) {
+            std::cerr << "Received normal event: type " << int(event->type) << ", chan " << channel << ", note " << int(event->data.note.note) << ", time " << eventTime << std::endl;
+	}
+#endif
+
+        switch (event->type) {
+        case SND_SEQ_EVENT_NOTE:
+        case SND_SEQ_EVENT_NOTEON:
+            if (fromController)
+                continue;
+            if (event->data.note.velocity > 0) {
+                MappedEvent *mE = new MappedEvent();
+                mE->setPitch(event->data.note.note);
+                mE->setVelocity(event->data.note.velocity);
+                mE->setEventTime(eventTime);
+                mE->setRecordedChannel(channel);
+                mE->setRecordedDevice(deviceId);
+
+                // Negative duration - we need to hear the NOTE ON
+                // so we must insert it now with a negative duration
+                // and pick and mix against the following NOTE OFF
+                // when we create the recorded segment.
+                //
+                mE->setDuration(RealTime( -1, 0));
+
+                // Create a copy of this when we insert the NOTE ON -
+                // keeping a copy alive on the m_noteOnMap.
+                //
+                // We shake out the two NOTE Ons after we've recorded
+                // them.
+                //
+                m_recordComposition.insert(new MappedEvent(mE));
+                m_noteOnMap[deviceId][chanNoteKey] = mE;
+
+                break;
+            }
+
+        case SND_SEQ_EVENT_NOTEOFF:
+            if (fromController)
+                continue;
+
+            if (m_noteOnMap[deviceId][chanNoteKey] != 0) {
+
+                // Set duration correctly on the NOTE OFF
+                //
+                MappedEvent *mE = m_noteOnMap[deviceId][chanNoteKey];
+                RealTime duration = eventTime - mE->getEventTime();
+
+#ifdef DEBUG_ALSA
+		std::cerr << "NOTE OFF: found NOTE ON at " << mE->getEventTime() << std::endl;
+#endif
+
+                if (duration < RealTime::zeroTime) {
+		    duration = RealTime::zeroTime;
+		    mE->setEventTime(eventTime);
+		}
+
+                // Velocity 0 - NOTE OFF.  Set duration correctly
+                // for recovery later.
+                //
+                mE->setVelocity(0);
+                mE->setDuration(duration);
+
+                // force shut off of note
+                m_recordComposition.insert(mE);
+
+                // reset the reference
+                //
+                m_noteOnMap[deviceId][chanNoteKey] = 0;
+
+            }
+            break;
+
+        case SND_SEQ_EVENT_KEYPRESS: {
+                if (fromController)
+                    continue;
+
+                // Fix for 632964 by Pedro Lopez-Cabanillas (20030523)
+                //
+                MappedEvent *mE = new MappedEvent();
+                mE->setType(MappedEvent::MidiKeyPressure);
+                mE->setEventTime(eventTime);
+                mE->setData1(event->data.note.note);
+                mE->setData2(event->data.note.velocity);
+                mE->setRecordedChannel(channel);
+                mE->setRecordedDevice(deviceId);
+                m_recordComposition.insert(mE);
+            }
+            break;
+
+        case SND_SEQ_EVENT_CONTROLLER: {
+                MappedEvent *mE = new MappedEvent();
+                mE->setType(MappedEvent::MidiController);
+                mE->setEventTime(eventTime);
+                mE->setData1(event->data.control.param);
+                mE->setData2(event->data.control.value);
+                mE->setRecordedChannel(channel);
+                mE->setRecordedDevice(deviceId);
+                m_recordComposition.insert(mE);
+            }
+            break;
+
+        case SND_SEQ_EVENT_PGMCHANGE: {
+                MappedEvent *mE = new MappedEvent();
+                mE->setType(MappedEvent::MidiProgramChange);
+                mE->setEventTime(eventTime);
+                mE->setData1(event->data.control.value);
+                mE->setRecordedChannel(channel);
+                mE->setRecordedDevice(deviceId);
+                m_recordComposition.insert(mE);
+
+            }
+            break;
+
+        case SND_SEQ_EVENT_PITCHBEND: {
+                if (fromController)
+                    continue;
+
+                // Fix for 711889 by Pedro Lopez-Cabanillas (20030523)
+                //
+                int s = event->data.control.value + 8192;
+                int d1 = (s >> 7) & 0x7f; // data1 = MSB
+                int d2 = s & 0x7f; // data2 = LSB
+                MappedEvent *mE = new MappedEvent();
+                mE->setType(MappedEvent::MidiPitchBend);
+                mE->setEventTime(eventTime);
+                mE->setData1(d1);
+                mE->setData2(d2);
+                mE->setRecordedChannel(channel);
+                mE->setRecordedDevice(deviceId);
+                m_recordComposition.insert(mE);
+            }
+            break;
+
+        case SND_SEQ_EVENT_CHANPRESS: {
+                if (fromController)
+                    continue;
+
+                // Fixed by Pedro Lopez-Cabanillas (20030523)
+                //
+                int s = event->data.control.value & 0x7f;
+                MappedEvent *mE = new MappedEvent();
+                mE->setType(MappedEvent::MidiChannelPressure);
+                mE->setEventTime(eventTime);
+                mE->setData1(s);
+                mE->setRecordedChannel(channel);
+                mE->setRecordedDevice(deviceId);
+                m_recordComposition.insert(mE);
+            }
+            break;
+
+        case SND_SEQ_EVENT_SYSEX:
+
+            if (fromController)
+                continue;
+
+            if (!testForMTCSysex(event) &&
+                    !testForMMCSysex(event)) {
+
+                // Bundle up the data into a block on the MappedEvent
+                //
+                std::string data;
+                char *ptr = (char*)(event->data.ext.ptr);
+                for (unsigned int i = 0; i < event->data.ext.len; ++i)
+                    data += *(ptr++);
+
+#ifdef DEBUG_ALSA
+
+                if ((MidiByte)(data[1]) == MIDI_SYSEX_RT) {
+                    std::cerr << "REALTIME SYSEX" << endl;
+                    for (unsigned int ii = 0; ii < event->data.ext.len; ++ii) {
+                        printf("B %d = %02x\n", ii, ((char*)(event->data.ext.ptr))[ii]);
+                    }
+                } else {
+                    std::cerr << "NON-REALTIME SYSEX" << endl;
+                    for (unsigned int ii = 0; ii < event->data.ext.len; ++ii) {
+                        printf("B %d = %02x\n", ii, ((char*)(event->data.ext.ptr))[ii]);
+                    }
+                }
+#endif
+
+                MappedEvent *mE = new MappedEvent();
+                mE->setType(MappedEvent::MidiSystemMessage);
+                mE->setData1(MIDI_SYSTEM_EXCLUSIVE);
+                mE->setRecordedDevice(deviceId);
+                // chop off SYX and EOX bytes from data block
+                // Fix for 674731 by Pedro Lopez-Cabanillas (20030601)
+                DataBlockRepository::setDataBlockForEvent(mE, data.substr(1, data.length() - 2));
+                mE->setEventTime(eventTime);
+                m_recordComposition.insert(mE);
+            }
+            break;
+
+
+        case SND_SEQ_EVENT_SENSING:  // MIDI device is still there
+            break;
+
+        case SND_SEQ_EVENT_QFRAME:
+            if (fromController)
+                continue;
+            if (getMTCStatus() == TRANSPORT_SLAVE) {
+                handleMTCQFrame(event->data.control.value, eventTime);
+            }
+            break;
+
+        case SND_SEQ_EVENT_CLOCK:
+#ifdef DEBUG_ALSA
+
+            std::cerr << "AlsaDriver::getMappedComposition - "
+            << "got realtime MIDI clock" << std::endl;
+#endif
+
+            break;
+
+        case SND_SEQ_EVENT_START:
+            if ((getMIDISyncStatus() == TRANSPORT_SLAVE) && !isPlaying()) {
+                ExternalTransport *transport = getExternalTransportControl();
+                if (transport) {
+                    transport->transportJump(ExternalTransport::TransportStopAtTime,
+                                             RealTime::zeroTime);
+                    transport->transportChange(ExternalTransport::TransportStart);
+                }
+            }
+#ifdef DEBUG_ALSA
+            std::cerr << "AlsaDriver::getMappedComposition - "
+            << "START" << std::endl;
+#endif
+
+            break;
+
+        case SND_SEQ_EVENT_CONTINUE:
+            if ((getMIDISyncStatus() == TRANSPORT_SLAVE) && !isPlaying()) {
+                ExternalTransport *transport = getExternalTransportControl();
+                if (transport) {
+                    transport->transportChange(ExternalTransport::TransportPlay);
+                }
+            }
+#ifdef DEBUG_ALSA
+            std::cerr << "AlsaDriver::getMappedComposition - "
+            << "CONTINUE" << std::endl;
+#endif
+
+            break;
+
+        case SND_SEQ_EVENT_STOP:
+            if ((getMIDISyncStatus() == TRANSPORT_SLAVE) && isPlaying()) {
+                ExternalTransport *transport = getExternalTransportControl();
+                if (transport) {
+                    transport->transportChange(ExternalTransport::TransportStop);
+                }
+            }
+#ifdef DEBUG_ALSA
+            std::cerr << "AlsaDriver::getMappedComposition - "
+            << "STOP" << std::endl;
+#endif
+
+            break;
+
+        case SND_SEQ_EVENT_SONGPOS:
+#ifdef DEBUG_ALSA
+
+            std::cerr << "AlsaDriver::getMappedComposition - "
+            << "SONG POSITION" << std::endl;
+#endif
+
+            break;
+
+            // these cases are handled by checkForNewClients
+            //
+        case SND_SEQ_EVENT_CLIENT_START:
+        case SND_SEQ_EVENT_CLIENT_EXIT:
+        case SND_SEQ_EVENT_CLIENT_CHANGE:
+        case SND_SEQ_EVENT_PORT_START:
+        case SND_SEQ_EVENT_PORT_EXIT:
+        case SND_SEQ_EVENT_PORT_CHANGE:
+        case SND_SEQ_EVENT_PORT_SUBSCRIBED:
+        case SND_SEQ_EVENT_PORT_UNSUBSCRIBED:
+            m_portCheckNeeded = true;
+#ifdef DEBUG_ALSA
+
+            std::cerr << "AlsaDriver::getMappedComposition - "
+            << "got announce event ("
+            << int(event->type) << ")" << std::endl;
+#endif
+
+            break;
+        case SND_SEQ_EVENT_TICK:
+        default:
+#ifdef DEBUG_ALSA
+
+            std::cerr << "AlsaDriver::getMappedComposition - "
+            << "got unhandled MIDI event type from ALSA sequencer"
+            << "(" << int(event->type) << ")" << std::endl;
+#endif
+
+            break;
+
+
+        }
+    }
+
+    if (getMTCStatus() == TRANSPORT_SLAVE && isPlaying()) {
+#ifdef MTC_DEBUG
+        std::cerr << "seq time is " << getSequencerTime() << ", last MTC receive "
+        << m_mtcLastReceive << ", first time " << m_mtcFirstTime << std::endl;
+#endif
+
+        if (m_mtcFirstTime == 0) { // have received _some_ MTC quarter-frame info
+            RealTime seqTime = getSequencerTime();
+            if (m_mtcLastReceive < seqTime &&
+                    seqTime - m_mtcLastReceive > RealTime(0, 500000000L)) {
+                ExternalTransport *transport = getExternalTransportControl();
+                if (transport) {
+                    transport->transportJump(ExternalTransport::TransportStopAtTime,
+                                             m_mtcLastEncoded);
+                }
+            }
+        }
+    }
+
+    return &m_recordComposition;
+}
+
+static int lock_count = 0;
+
+void
+AlsaDriver::handleMTCQFrame(unsigned int data_byte, RealTime the_time)
+{
+    if (getMTCStatus() != TRANSPORT_SLAVE)
+        return ;
+
+    switch (data_byte & 0xF0) {
+        /* Frame */
+    case 0x00:
+        /*
+         * Reset everything
+         */
+        m_mtcReceiveTime = the_time;
+        m_mtcFrames = data_byte & 0x0f;
+        m_mtcSeconds = 0;
+        m_mtcMinutes = 0;
+        m_mtcHours = 0;
+        m_mtcSMPTEType = 0;
+
+        break;
+
+    case 0x10:
+        m_mtcFrames |= (data_byte & 0x0f) << 4;
+        break;
+
+        /* Seconds */
+    case 0x20:
+        m_mtcSeconds = data_byte & 0x0f;
+        break;
+    case 0x30:
+        m_mtcSeconds |= (data_byte & 0x0f) << 4;
+        break;
+
+        /* Minutes */
+    case 0x40:
+        m_mtcMinutes = data_byte & 0x0f;
+        break;
+    case 0x50:
+        m_mtcMinutes |= (data_byte & 0x0f) << 4;
+        break;
+
+        /* Hours and SMPTE type */
+    case 0x60:
+        m_mtcHours = data_byte & 0x0f;
+        break;
+
+    case 0x70: {
+            m_mtcHours |= (data_byte & 0x01) << 4;
+            m_mtcSMPTEType = (data_byte & 0x06) >> 1;
+
+            int fps = 30;
+            if (m_mtcSMPTEType == 0)
+                fps = 24;
+            else if (m_mtcSMPTEType == 1)
+                fps = 25;
+
+            /*
+             * Ok, got all the bits now
+             * (Assuming time is rolling forward)
+             */
+
+            /* correct for 2-frame lag */
+            m_mtcFrames += 2;
+            if (m_mtcFrames >= fps) {
+                m_mtcFrames -= fps;
+                if (++m_mtcSeconds == 60) {
+                    m_mtcSeconds = 0;
+                    if (++m_mtcMinutes == 60) {
+                        m_mtcMinutes = 0;
+                        ++m_mtcHours;
+                    }
+                }
+            }
+
+#ifdef MTC_DEBUG
+            printf("RG MTC: Got a complete sequence: %02d:%02d:%02d.%02d (type %d)\n",
+                   m_mtcHours,
+                   m_mtcMinutes,
+                   m_mtcSeconds,
+                   m_mtcFrames,
+                   m_mtcSMPTEType);
+#endif
+
+            /* compute encoded time */
+            m_mtcEncodedTime.sec = m_mtcSeconds +
+                                   m_mtcMinutes * 60 +
+                                   m_mtcHours * 60 * 60;
+
+            switch (fps) {
+            case 24:
+                m_mtcEncodedTime.nsec = (int)
+                                        ((125000000UL * (unsigned)m_mtcFrames) / (unsigned) 3);
+                break;
+            case 25:
+                m_mtcEncodedTime.nsec = (int)
+                                        (40000000UL * (unsigned)m_mtcFrames);
+                break;
+            case 30:
+            default:
+                m_mtcEncodedTime.nsec = (int)
+                                        ((100000000UL * (unsigned)m_mtcFrames) / (unsigned) 3);
+                break;
+            }
+
+            /*
+             * We only mess with the clock if we are playing
+             */
+            if (m_playing) {
+#ifdef MTC_DEBUG
+                std::cerr << "RG MTC: Tstamp " << m_mtcEncodedTime;
+                std::cerr << " Received @ " << m_mtcReceiveTime << endl;
+#endif
+
+                calibrateMTC();
+
+                RealTime t_diff = m_mtcEncodedTime - m_mtcReceiveTime;
+#ifdef MTC_DEBUG
+
+                std::cerr << "Diff: " << t_diff << endl;
+#endif
+
+                /* -ve diff means ALSA time ahead of MTC time */
+
+                if (t_diff.sec > 0) {
+                    tweakSkewForMTC(60000);
+                } else if (t_diff.sec < 0) {
+                    tweakSkewForMTC( -60000);
+                } else {
+                    /* "small" diff - use adaptive technique */
+                    tweakSkewForMTC(t_diff.nsec / 1400);
+                    if ((t_diff.nsec / 1000000) == 0) {
+                        if (++lock_count == 3) {
+                            printf("Got a lock @ %02d:%02d:%02d.%02d (type %d)\n",
+                                   m_mtcHours,
+                                   m_mtcMinutes,
+                                   m_mtcSeconds,
+                                   m_mtcFrames,
+                                   m_mtcSMPTEType);
+                        }
+                    } else {
+                        lock_count = 0;
+                    }
+                }
+
+            } else if (m_eat_mtc > 0) {
+#ifdef MTC_DEBUG
+                std::cerr << "MTC: Received quarter frame just after issuing MMC stop - ignore it" << std::endl;
+#endif
+
+                --m_eat_mtc;
+            } else {
+                /* If we're not playing, we should be. */
+#ifdef MTC_DEBUG
+                std::cerr << "MTC: Received quarter frame while not playing - starting now" << std::endl;
+#endif
+
+                ExternalTransport *transport = getExternalTransportControl();
+                if (transport) {
+                    transport->transportJump
+                    (ExternalTransport::TransportStartAtTime,
+                     m_mtcEncodedTime);
+                }
+            }
+
+            break;
+        }
+
+        /* Oh dear, demented device! */
+    default:
+        break;
+    }
+}
+
+void
+AlsaDriver::insertMTCFullFrame(RealTime time)
+{
+    snd_seq_event_t event;
+
+    snd_seq_ev_clear(&event);
+    snd_seq_ev_set_source(&event, m_syncOutputPort);
+    snd_seq_ev_set_subs(&event);
+
+    m_mtcEncodedTime = time;
+    m_mtcSeconds = m_mtcEncodedTime.sec % 60;
+    m_mtcMinutes = (m_mtcEncodedTime.sec / 60) % 60;
+    m_mtcHours = (m_mtcEncodedTime.sec / 3600);
+
+    // We always send at 25fps, it's the easiest to avoid rounding problems
+    m_mtcFrames = (unsigned)m_mtcEncodedTime.nsec / 40000000U;
+
+    time = time + m_alsaPlayStartTime - m_playStartPosition;
+    snd_seq_real_time_t atime = { time.sec, time.nsec };
+
+    unsigned char data[10] =
+        { MIDI_SYSTEM_EXCLUSIVE,
+          MIDI_SYSEX_RT, 127, 1, 1,
+          0, 0, 0, 0,
+          MIDI_END_OF_EXCLUSIVE };
+
+    data[5] = ((unsigned char)m_mtcHours & 0x1f) + (1 << 5); // 1 indicates 25fps
+    data[6] = (unsigned char)m_mtcMinutes;
+    data[7] = (unsigned char)m_mtcSeconds;
+    data[8] = (unsigned char)m_mtcFrames;
+
+    snd_seq_ev_schedule_real(&event, m_queue, 0, &atime);
+    snd_seq_ev_set_sysex(&event, 10, data);
+
+    checkAlsaError(snd_seq_event_output(m_midiHandle, &event),
+                   "insertMTCFullFrame event send");
+
+    if (m_queueRunning) {
+        checkAlsaError(snd_seq_drain_output(m_midiHandle), "insertMTCFullFrame drain");
+    }
+}
+
+void
+AlsaDriver::insertMTCQFrames(RealTime sliceStart, RealTime sliceEnd)
+{
+    if (sliceStart == RealTime::zeroTime && sliceEnd == RealTime::zeroTime) {
+        // not a real slice
+        return ;
+    }
+
+    // We send at 25fps, it's the easiest to avoid rounding problems
+    RealTime twoFrames(0, 80000000U);
+    RealTime quarterFrame(0, 10000000U);
+    int fps = 25;
+
+#ifdef MTC_DEBUG
+
+    std::cout << "AlsaDriver::insertMTCQFrames(" << sliceStart << ","
+    << sliceEnd << "): first time " << m_mtcFirstTime << std::endl;
+#endif
+
+    RealTime t;
+
+    if (m_mtcFirstTime != 0) { // first time through, reset location
+        m_mtcEncodedTime = sliceStart;
+        t = sliceStart;
+        m_mtcFirstTime = 0;
+    } else {
+        t = m_mtcEncodedTime + quarterFrame;
+    }
+
+    m_mtcSeconds = m_mtcEncodedTime.sec % 60;
+    m_mtcMinutes = (m_mtcEncodedTime.sec / 60) % 60;
+    m_mtcHours = (m_mtcEncodedTime.sec / 3600);
+    m_mtcFrames = (unsigned)m_mtcEncodedTime.nsec / 40000000U; // 25fps
+
+    std::string bytes = " ";
+
+    int type = 0;
+
+    while (m_mtcEncodedTime < sliceEnd) {
+
+        snd_seq_event_t event;
+        snd_seq_ev_clear(&event);
+        snd_seq_ev_set_source(&event, m_syncOutputPort);
+        snd_seq_ev_set_subs(&event);
+
+#ifdef MTC_DEBUG
+
+        std::cout << "Sending MTC quarter frame at " << t << std::endl;
+#endif
+
+        unsigned char c = (type << 4);
+
+        switch (type) {
+        case 0:
+            c += ((unsigned char)m_mtcFrames & 0x0f);
+            break;
+        case 1:
+            c += (((unsigned char)m_mtcFrames & 0xf0) >> 4);
+            break;
+        case 2:
+            c += ((unsigned char)m_mtcSeconds & 0x0f);
+            break;
+        case 3:
+            c += (((unsigned char)m_mtcSeconds & 0xf0) >> 4);
+            break;
+        case 4:
+            c += ((unsigned char)m_mtcMinutes & 0x0f);
+            break;
+        case 5:
+            c += (((unsigned char)m_mtcMinutes & 0xf0) >> 4);
+            break;
+        case 6:
+            c += ((unsigned char)m_mtcHours & 0x0f);
+            break;
+        case 7:  // hours high nibble + smpte type
+            c += (m_mtcHours >> 4) & 0x01;
+            c += (1 << 1); // type 1 indicates 25fps
+            break;
+        }
+
+        RealTime scheduleTime = t + m_alsaPlayStartTime - m_playStartPosition;
+        snd_seq_real_time_t atime = { scheduleTime.sec, scheduleTime.nsec };
+
+        event.type = SND_SEQ_EVENT_QFRAME;
+        event.data.control.value = c;
+
+        snd_seq_ev_schedule_real(&event, m_queue, 0, &atime);
+
+        checkAlsaError(snd_seq_event_output(m_midiHandle, &event),
+                       "insertMTCQFrames sending qframe event");
+
+        if (++type == 8) {
+            m_mtcFrames += 2;
+            if (m_mtcFrames >= fps) {
+                m_mtcFrames -= fps;
+                if (++m_mtcSeconds == 60) {
+                    m_mtcSeconds = 0;
+                    if (++m_mtcMinutes == 60) {
+                        m_mtcMinutes = 0;
+                        ++m_mtcHours;
+                    }
+                }
+            }
+            m_mtcEncodedTime = t;
+            type = 0;
+        }
+
+        t = t + quarterFrame;
+    }
+}
+
+bool
+AlsaDriver::testForMTCSysex(const snd_seq_event_t *event)
+{
+    if (getMTCStatus() != TRANSPORT_SLAVE)
+        return false;
+
+    // At this point, and possibly for the foreseeable future, the only
+    // sysex we're interested in is full-frame transport location
+
+#ifdef MTC_DEBUG
+
+    std::cerr << "MTC: testing sysex of length " << event->data.ext.len << ":" << std::endl;
+    for (int i = 0; i < event->data.ext.len; ++i) {
+        std::cerr << (int)*((unsigned char *)event->data.ext.ptr + i) << " ";
+    }
+    std::cerr << endl;
+#endif
+
+    if (event->data.ext.len != 10)
+        return false;
+
+    unsigned char *ptr = (unsigned char *)(event->data.ext.ptr);
+
+    if (*ptr++ != MIDI_SYSTEM_EXCLUSIVE)
+        return false;
+    if (*ptr++ != MIDI_SYSEX_RT)
+        return false;
+    if (*ptr++ > 127)
+        return false;
+
+    // 01 01 for MTC full frame
+
+    if (*ptr++ != 1)
+        return false;
+    if (*ptr++ != 1)
+        return false;
+
+    int htype = *ptr++;
+    int min = *ptr++;
+    int sec = *ptr++;
+    int frame = *ptr++;
+
+    if (*ptr != MIDI_END_OF_EXCLUSIVE)
+        return false;
+
+    int hour = (htype & 0x1f);
+    int type = (htype & 0xe0) >> 5;
+
+    m_mtcFrames = frame;
+    m_mtcSeconds = sec;
+    m_mtcMinutes = min;
+    m_mtcHours = hour;
+    m_mtcSMPTEType = type;
+
+    int fps = 30;
+    if (m_mtcSMPTEType == 0)
+        fps = 24;
+    else if (m_mtcSMPTEType == 1)
+        fps = 25;
+
+    m_mtcEncodedTime.sec = sec + min * 60 + hour * 60 * 60;
+
+    switch (fps) {
+    case 24:
+        m_mtcEncodedTime.nsec = (int)
+                                ((125000000UL * (unsigned)m_mtcFrames) / (unsigned) 3);
+        break;
+    case 25:
+        m_mtcEncodedTime.nsec = (int)
+                                (40000000UL * (unsigned)m_mtcFrames);
+        break;
+    case 30:
+    default:
+        m_mtcEncodedTime.nsec = (int)
+                                ((100000000UL * (unsigned)m_mtcFrames) / (unsigned) 3);
+        break;
+    }
+
+#ifdef MTC_DEBUG
+    std::cerr << "MTC: MTC sysex found (frame type " << type
+    << "), jumping to " << m_mtcEncodedTime << std::endl;
+#endif
+
+    ExternalTransport *transport = getExternalTransportControl();
+    if (transport) {
+        transport->transportJump
+        (ExternalTransport::TransportJumpToTime,
+         m_mtcEncodedTime);
+    }
+
+    return true;
+}
+
+static int last_factor = 0;
+static int bias_factor = 0;
+
+void
+AlsaDriver::calibrateMTC()
+{
+    if (m_mtcFirstTime < 0)
+        return ;
+    else if (m_mtcFirstTime > 0) {
+        --m_mtcFirstTime;
+        m_mtcSigmaC = 0;
+        m_mtcSigmaE = 0;
+    } else {
+        RealTime diff_e = m_mtcEncodedTime - m_mtcLastEncoded;
+        RealTime diff_c = m_mtcReceiveTime - m_mtcLastReceive;
+
+#ifdef MTC_DEBUG
+
+        printf("RG MTC: diffs %d %d %d\n", diff_c.nsec, diff_e.nsec, m_mtcSkew);
+#endif
+
+        m_mtcSigmaE += ((long long int) diff_e.nsec) * m_mtcSkew;
+        m_mtcSigmaC += diff_c.nsec;
+
+
+        int t_bias = (m_mtcSigmaE / m_mtcSigmaC) - 0x10000;
+
+#ifdef MTC_DEBUG
+
+        printf("RG MTC: sigmas %lld %lld %d\n", m_mtcSigmaE, m_mtcSigmaC, t_bias);
+#endif
+
+        bias_factor = t_bias;
+    }
+
+    m_mtcLastReceive = m_mtcReceiveTime;
+    m_mtcLastEncoded = m_mtcEncodedTime;
+
+}
+
+void
+AlsaDriver::tweakSkewForMTC(int factor)
+{
+    if (factor > 50000) {
+        factor = 50000;
+    } else if (factor < -50000) {
+        factor = -50000;
+    } else if (factor == last_factor) {
+        return ;
+    } else {
+        if (m_mtcFirstTime == -1)
+            m_mtcFirstTime = 5;
+    }
+    last_factor = factor;
+
+    snd_seq_queue_tempo_t *q_ptr;
+    snd_seq_queue_tempo_alloca(&q_ptr);
+
+    snd_seq_get_queue_tempo( m_midiHandle, m_queue, q_ptr);
+
+    unsigned int t_skew = snd_seq_queue_tempo_get_skew(q_ptr);
+#ifdef MTC_DEBUG
+
+    std::cerr << "RG MTC: skew: " << t_skew;
+#endif
+
+    t_skew = 0x10000 + factor + bias_factor;
+
+#ifdef MTC_DEBUG
+
+    std::cerr << " changed to " << factor << "+" << bias_factor << endl;
+#endif
+
+    snd_seq_queue_tempo_set_skew(q_ptr, t_skew);
+    snd_seq_set_queue_tempo( m_midiHandle, m_queue, q_ptr);
+
+    m_mtcSkew = t_skew;
+}
+
+bool
+AlsaDriver::testForMMCSysex(const snd_seq_event_t *event)
+{
+    if (getMMCStatus() != TRANSPORT_SLAVE)
+        return false;
+
+    if (event->data.ext.len != 6)
+        return false;
+
+    unsigned char *ptr = (unsigned char *)(event->data.ext.ptr);
+
+    if (*ptr++ != MIDI_SYSTEM_EXCLUSIVE)
+        return false;
+    if (*ptr++ != MIDI_SYSEX_RT)
+        return false;
+    if (*ptr++ > 127)
+        return false;
+    if (*ptr++ != MIDI_SYSEX_RT_COMMAND)
+        return false;
+
+    int instruction = *ptr++;
+
+    if (*ptr != MIDI_END_OF_EXCLUSIVE)
+        return false;
+
+    if (instruction == MIDI_MMC_PLAY ||
+            instruction == MIDI_MMC_DEFERRED_PLAY) {
+        ExternalTransport *transport = getExternalTransportControl();
+        if (transport) {
+            transport->transportChange(ExternalTransport::TransportPlay);
+        }
+    } else if (instruction == MIDI_MMC_STOP) {
+        ExternalTransport *transport = getExternalTransportControl();
+        if (transport) {
+            transport->transportChange(ExternalTransport::TransportStop);
+        }
+    }
+
+    return true;
+}
+
+void
+AlsaDriver::processMidiOut(const MappedComposition &mC,
+                           const RealTime &sliceStart,
+                           const RealTime &sliceEnd)
+{
+    RealTime outputTime;
+    RealTime outputStopTime;
+    MappedInstrument *instrument;
+    ClientPortPair outputDevice;
+    MidiByte channel;
+    snd_seq_event_t event;
+
+    // special case for unqueued events
+    bool now = (sliceStart == RealTime::zeroTime && sliceEnd == RealTime::zeroTime);
+    
+    if (!now) {
+	// This 0.5 sec is arbitrary, but it must be larger than the
+	// sequencer's read-ahead
+	RealTime diff = RealTime::fromSeconds(0.5);
+	RealTime cutoff = sliceStart - diff;
+	cropRecentNoteOffs(cutoff - m_playStartPosition + m_alsaPlayStartTime);
+    }
+
+    // These won't change in this slice
+    //
+    snd_seq_ev_clear(&event);
+
+    if ((mC.begin() != mC.end()) && getSequencerDataBlock()) {
+        getSequencerDataBlock()->setVisual(*mC.begin());
+    }
+
+#ifdef DEBUG_PROCESS_MIDI_OUT
+    std::cerr << "AlsaDriver::processMidiOut(" << sliceStart << "," << sliceEnd
+    << "), " << mC.size() << " events, now is " << now << std::endl;
+#endif
+
+    // NB the MappedComposition is implicitly ordered by time (std::multiset)
+
+    for (MappedComposition::const_iterator i = mC.begin(); i != mC.end(); ++i) {
+        if ((*i)->getType() >= MappedEvent::Audio)
+            continue;
+
+        bool isControllerOut = ((*i)->getRecordedDevice() ==
+                                Device::CONTROL_DEVICE);
+
+        bool isSoftSynth = (!isControllerOut &&
+                            ((*i)->getInstrument() >= SoftSynthInstrumentBase));
+
+        outputTime = (*i)->getEventTime() - m_playStartPosition +
+                     m_alsaPlayStartTime;
+
+        if (now && !m_playing && m_queueRunning) {
+            // stop queue to ensure exact timing and make sure the
+            // event gets through right now
+#ifdef DEBUG_PROCESS_MIDI_OUT
+            std::cerr << "processMidiOut: stopping queue for now-event" << std::endl;
+#endif
+
+            checkAlsaError(snd_seq_stop_queue(m_midiHandle, m_queue, NULL), "processMidiOut(): stop queue");
+            checkAlsaError(snd_seq_drain_output(m_midiHandle), "processMidiOut(): draining");
+        }
+
+        RealTime alsaTimeNow = getAlsaTime();
+
+        if (now) {
+            if (!m_playing) {
+                outputTime = alsaTimeNow;
+            } else if (outputTime < alsaTimeNow) {
+                outputTime = alsaTimeNow + RealTime(0, 10000000);
+            }
+        }
+
+#ifdef DEBUG_PROCESS_MIDI_OUT
+        std::cerr << "processMidiOut[" << now << "]: event is at " << outputTime << " (" << outputTime - alsaTimeNow << " ahead of queue time), type " << int((*i)->getType()) << ", duration " << (*i)->getDuration() << std::endl;
+#endif
+
+	if (!m_queueRunning && outputTime < alsaTimeNow) {
+            RealTime adjust = alsaTimeNow - outputTime;
+            if ((*i)->getDuration() > RealTime::zeroTime) {
+                if ((*i)->getDuration() <= adjust) {
+#ifdef DEBUG_PROCESS_MIDI_OUT
+                    std::cerr << "processMidiOut[" << now << "]: too late for this event, abandoning it" << std::endl;
+#endif
+
+                    continue;
+                } else {
+#ifdef DEBUG_PROCESS_MIDI_OUT
+                    std::cerr << "processMidiOut[" << now << "]: pushing event forward and reducing duration by " << adjust << std::endl;
+#endif
+
+                    (*i)->setDuration((*i)->getDuration() - adjust);
+                }
+            } else {
+#ifdef DEBUG_PROCESS_MIDI_OUT
+                std::cerr << "processMidiOut[" << now << "]: pushing zero-duration event forward by " << adjust << std::endl;
+#endif
+
+            }
+            outputTime = alsaTimeNow;
+        }
+
+        processNotesOff(outputTime, now);
+
+#ifdef HAVE_LIBJACK
+
+        if (m_jackDriver) {
+            size_t frameCount = m_jackDriver->getFramesProcessed();
+            size_t elapsed = frameCount - _debug_jack_frame_count;
+            RealTime rt = RealTime::frame2RealTime(elapsed, m_jackDriver->getSampleRate());
+            rt = rt - getAlsaTime();
+#ifdef DEBUG_PROCESS_MIDI_OUT
+
+            std::cerr << "processMidiOut[" << now << "]: JACK time is " << rt << " ahead of ALSA time" << std::endl;
+#endif
+
+        }
+#endif
+
+        // Second and nanoseconds for ALSA
+        //
+        snd_seq_real_time_t time = { outputTime.sec, outputTime.nsec };
+
+        if (!isSoftSynth) {
+
+#ifdef DEBUG_PROCESS_MIDI_OUT
+            std::cout << "processMidiOut[" << now << "]: instrument " << (*i)->getInstrument() << std::endl;
+            std::cout << "pitch: " << (int)(*i)->getPitch() << ", velocity " << (int)(*i)->getVelocity() << ", duration " << (*i)->getDuration() << std::endl;
+#endif
+
+            snd_seq_ev_set_subs(&event);
+
+            // Set source according to port for device
+            //
+            int src;
+
+            if (isControllerOut) {
+                src = m_controllerPort;
+            } else {
+                src = getOutputPortForMappedInstrument((*i)->getInstrument());
+            }
+
+            if (src < 0) continue;
+            snd_seq_ev_set_source(&event, src);
+
+            snd_seq_ev_schedule_real(&event, m_queue, 0, &time);
+
+        } else {
+            event.time.time = time;
+        }
+
+        instrument = getMappedInstrument((*i)->getInstrument());
+
+        // set the stop time for Note Off
+        //
+        outputStopTime = outputTime + (*i)->getDuration()
+                         - RealTime(0, 1); // notch it back 1nsec just to ensure
+        // correct ordering against any other
+        // note-ons at the same nominal time
+        bool needNoteOff = false;
+
+        if (isControllerOut) {
+            channel = (*i)->getRecordedChannel();
+#ifdef DEBUG_ALSA
+
+            std::cerr << "processMidiOut() - Event of type " << (int)((*i)->getType()) << " (data1 " << (int)(*i)->getData1() << ", data2 " << (int)(*i)->getData2() << ") for external controller channel " << (int)channel << std::endl;
+#endif
+
+        } else if (instrument != 0) {
+            channel = instrument->getChannel();
+        } else {
+#ifdef DEBUG_ALSA
+            std::cerr << "processMidiOut() - No instrument for event of type "
+            << (int)(*i)->getType() << " at " << (*i)->getEventTime()
+            << std::endl;
+#endif
+
+            channel = 0;
+        }
+
+        switch ((*i)->getType()) {
+
+        case MappedEvent::MidiNoteOneShot:
+	{
+	    snd_seq_ev_set_noteon(&event,
+				  channel,
+				  (*i)->getPitch(),
+				  (*i)->getVelocity());
+	    needNoteOff = true;
+	    
+	    if (!isSoftSynth && getSequencerDataBlock()) {
+		LevelInfo info;
+		info.level = (*i)->getVelocity();
+		info.levelRight = 0;
+		getSequencerDataBlock()->setInstrumentLevel
+                    ((*i)->getInstrument(), info);
+	    }
+
+	    weedRecentNoteOffs((*i)->getPitch(), channel, (*i)->getInstrument());
+	}
+	break;
+
+        case MappedEvent::MidiNote:
+            // We always use plain NOTE ON here, not ALSA
+            // time+duration notes, because we have our own NOTE
+            // OFF stack (which will be augmented at the bottom of
+            // this function) and we want to ensure it gets used
+            // for the purposes of e.g. soft synths
+            //
+            if ((*i)->getVelocity() > 0) {
+                snd_seq_ev_set_noteon(&event,
+                                      channel,
+                                      (*i)->getPitch(),
+                                      (*i)->getVelocity());
+
+                if (!isSoftSynth && getSequencerDataBlock()) {
+                    LevelInfo info;
+                    info.level = (*i)->getVelocity();
+                    info.levelRight = 0;
+                    getSequencerDataBlock()->setInstrumentLevel
+                    ((*i)->getInstrument(), info);
+                }
+
+		weedRecentNoteOffs((*i)->getPitch(), channel, (*i)->getInstrument());
+            } else {
+                snd_seq_ev_set_noteoff(&event,
+                                       channel,
+                                       (*i)->getPitch(),
+                                       (*i)->getVelocity());
+            }
+
+            break;
+
+        case MappedEvent::MidiProgramChange:
+            snd_seq_ev_set_pgmchange(&event,
+                                     channel,
+                                     (*i)->getData1());
+            break;
+
+        case MappedEvent::MidiKeyPressure:
+            snd_seq_ev_set_keypress(&event,
+                                    channel,
+                                    (*i)->getData1(),
+                                    (*i)->getData2());
+            break;
+
+        case MappedEvent::MidiChannelPressure:
+            snd_seq_ev_set_chanpress(&event,
+                                     channel,
+                                     (*i)->getData1());
+            break;
+
+        case MappedEvent::MidiPitchBend: {
+                int d1 = (int)((*i)->getData1());
+                int d2 = (int)((*i)->getData2());
+                int value = ((d1 << 7) | d2) - 8192;
+
+                // keep within -8192 to +8192
+                //
+                // if (value & 0x4000)
+                //    value -= 0x8000;
+
+                snd_seq_ev_set_pitchbend(&event,
+                                         channel,
+                                         value);
+            }
+            break;
+
+        case MappedEvent::MidiSystemMessage: {
+                switch ((*i)->getData1()) {
+                case MIDI_SYSTEM_EXCLUSIVE: {
+                        char out[2];
+                        sprintf(out, "%c", MIDI_SYSTEM_EXCLUSIVE);
+                        std::string data = out;
+
+                        data += DataBlockRepository::getDataBlockForEvent((*i));
+
+                        sprintf(out, "%c", MIDI_END_OF_EXCLUSIVE);
+                        data += out;
+
+                        snd_seq_ev_set_sysex(&event,
+                                             data.length(),
+                                             (char*)(data.c_str()));
+                    }
+                    break;
+
+                case MIDI_TIMING_CLOCK: {
+                        RealTime rt =
+                            RealTime(time.tv_sec, time.tv_nsec);
+
+                        /*
+                        std::cerr << "AlsaDriver::processMidiOut - "
+                                  << "send clock @ " << rt << std::endl;
+                                  */
+
+                        sendSystemQueued(SND_SEQ_EVENT_CLOCK, "", rt);
+
+                        continue;
+
+                    }
+                    break;
+
+                default:
+                    std::cerr << "AlsaDriver::processMidiOut - "
+                    << "unrecognised system message"
+                    << std::endl;
+                    break;
+                }
+            }
+            break;
+
+        case MappedEvent::MidiController:
+            snd_seq_ev_set_controller(&event,
+                                      channel,
+                                      (*i)->getData1(),
+                                      (*i)->getData2());
+            break;
+
+        case MappedEvent::Audio:
+        case MappedEvent::AudioCancel:
+        case MappedEvent::AudioLevel:
+        case MappedEvent::AudioStopped:
+        case MappedEvent::SystemUpdateInstruments:
+        case MappedEvent::SystemJackTransport:  //???
+        case MappedEvent::SystemMMCTransport:
+        case MappedEvent::SystemMIDIClock:
+        case MappedEvent::SystemMIDISyncAuto:
+            break;
+
+        default:
+        case MappedEvent::InvalidMappedEvent:
+#ifdef DEBUG_ALSA
+
+            std::cerr << "AlsaDriver::processMidiOut - "
+            << "skipping unrecognised or invalid MappedEvent type"
+            << std::endl;
+#endif
+
+            continue;
+        }
+
+        if (isSoftSynth) {
+
+            processSoftSynthEventOut((*i)->getInstrument(), &event, now);
+
+        } else {
+            checkAlsaError(snd_seq_event_output(m_midiHandle, &event),
+                           "processMidiOut(): output queued");
+
+            if (now) {
+                if (m_queueRunning && !m_playing) {
+                    // restart queue
+#ifdef DEBUG_PROCESS_MIDI_OUT
+                    std::cerr << "processMidiOut: restarting queue after now-event" << std::endl;
+#endif
+
+                    checkAlsaError(snd_seq_continue_queue(m_midiHandle, m_queue, NULL), "processMidiOut(): continue queue");
+                }
+                checkAlsaError(snd_seq_drain_output(m_midiHandle), "processMidiOut(): draining");
+            }
+        }
+
+        // Add note to note off stack
+        //
+        if (needNoteOff) {
+            NoteOffEvent *noteOffEvent =
+                new NoteOffEvent(outputStopTime,  // already calculated
+                                 (*i)->getPitch(),
+                                 channel,
+                                 (*i)->getInstrument());
+
+#ifdef DEBUG_ALSA
+
+            std::cerr << "Adding NOTE OFF at " << outputStopTime
+            << std::endl;
+#endif
+
+            m_noteOffQueue.insert(noteOffEvent);
+        }
+    }
+
+    processNotesOff(sliceEnd - m_playStartPosition + m_alsaPlayStartTime, now);
+
+    if (getMTCStatus() == TRANSPORT_MASTER) {
+        insertMTCQFrames(sliceStart, sliceEnd);
+    }
+
+    if (m_queueRunning) {
+
+        if (now && !m_playing) {
+            // just to be sure
+#ifdef DEBUG_PROCESS_MIDI_OUT
+            std::cerr << "processMidiOut: restarting queue after all now-events" << std::endl;
+#endif
+
+            checkAlsaError(snd_seq_continue_queue(m_midiHandle, m_queue, NULL), "processMidiOut(): continue queue");
+        }
+
+#ifdef DEBUG_PROCESS_MIDI_OUT 
+        //	std::cerr << "processMidiOut: m_queueRunning " << m_queueRunning
+        //		  << ", now " << now << std::endl;
+#endif
+        checkAlsaError(snd_seq_drain_output(m_midiHandle), "processMidiOut(): draining");
+    }
+}
+
+void
+AlsaDriver::processSoftSynthEventOut(InstrumentId id, const snd_seq_event_t *ev, bool now)
+{
+#ifdef DEBUG_PROCESS_SOFT_SYNTH_OUT
+    std::cerr << "AlsaDriver::processSoftSynthEventOut: instrument " << id << ", now " << now << std::endl;
+#endif
+
+#ifdef HAVE_LIBJACK
+
+    if (!m_jackDriver)
+        return ;
+    RunnablePluginInstance *synthPlugin = m_jackDriver->getSynthPlugin(id);
+
+    if (synthPlugin) {
+
+        RealTime t(ev->time.time.tv_sec, ev->time.time.tv_nsec);
+
+        if (now)
+            t = RealTime::zeroTime;
+        else
+            t = t + m_playStartPosition - m_alsaPlayStartTime;
+
+#ifdef DEBUG_PROCESS_SOFT_SYNTH_OUT
+
+        std::cerr << "AlsaDriver::processSoftSynthEventOut: event time " << t << std::endl;
+#endif
+
+        synthPlugin->sendEvent(t, ev);
+
+        if (now) {
+#ifdef DEBUG_PROCESS_SOFT_SYNTH_OUT
+            std::cerr << "AlsaDriver::processSoftSynthEventOut: setting haveAsyncAudioEvent" << std::endl;
+#endif
+
+            m_jackDriver->setHaveAsyncAudioEvent();
+        }
+    }
+#endif
+}
+
+void
+AlsaDriver::startClocks()
+{
+    int result;
+
+#ifdef DEBUG_ALSA
+
+    std::cerr << "AlsaDriver::startClocks" << std::endl;
+#endif
+
+    if (m_needJackStart) {
+#ifdef DEBUG_ALSA
+        std::cerr << "AlsaDriver::startClocks: Need JACK start (m_playing = " << m_playing << ")" << std::endl;
+#endif
+
+    }
+
+#ifdef HAVE_LIBJACK
+
+    // New JACK transport scheme: The initialisePlayback,
+    // resetPlayback and stopPlayback methods set m_needJackStart, and
+    // then this method checks it and calls the appropriate JACK
+    // transport start or relocate method, which calls back on
+    // startClocksApproved when ready.  (Previously this method always
+    // called the JACK transport start method, so we couldn't handle
+    // moving the pointer when not playing, and we had to stop the
+    // transport explicitly from resetPlayback when repositioning
+    // during playback.)
+
+    if (m_jackDriver) {
+
+        // Don't need any locks on this, except for those that the
+        // driver methods take and hold for themselves
+
+        if (m_needJackStart != NeedNoJackStart) {
+            if (m_needJackStart == NeedJackStart ||
+                    m_playing) {
+#ifdef DEBUG_ALSA
+                std::cerr << "AlsaDriver::startClocks: playing, prebuffer audio" << std::endl;
+#endif
+
+                m_jackDriver->prebufferAudio();
+            } else {
+#ifdef DEBUG_ALSA
+                std::cerr << "AlsaDriver::startClocks: prepare audio only" << std::endl;
+#endif
+
+                m_jackDriver->prepareAudio();
+            }
+            bool rv;
+            if (m_needJackStart == NeedJackReposition) {
+                rv = m_jackDriver->relocateTransport();
+            } else {
+                rv = m_jackDriver->startTransport();
+                if (!rv) {
+#ifdef DEBUG_ALSA
+                    std::cerr << "AlsaDriver::startClocks: Waiting for startClocksApproved" << std::endl;
+#endif
+                    // need to wait for transport sync
+                    _debug_jack_frame_count = m_jackDriver->getFramesProcessed();
+                    return ;
+                }
+            }
+        }
+    }
+#endif
+
+    // Restart the timer
+    if ((result = snd_seq_continue_queue(m_midiHandle, m_queue, NULL)) < 0) {
+        std::cerr << "AlsaDriver::startClocks - couldn't start queue - "
+        << snd_strerror(result)
+        << std::endl;
+        reportFailure(MappedEvent::FailureALSACallFailed);
+    }
+
+#ifdef DEBUG_ALSA
+    std::cerr << "AlsaDriver::startClocks: started clocks" << std::endl;
+#endif
+
+    m_queueRunning = true;
+
+#ifdef HAVE_LIBJACK
+
+    if (m_jackDriver) {
+        _debug_jack_frame_count = m_jackDriver->getFramesProcessed();
+    }
+#endif
+
+    // process pending MIDI events
+    checkAlsaError(snd_seq_drain_output(m_midiHandle), "startClocks(): draining");
+}
+
+void
+AlsaDriver::startClocksApproved()
+{
+#ifdef DEBUG_ALSA
+    std::cerr << "AlsaDriver::startClocks: startClocksApproved" << std::endl;
+#endif
+
+    //!!!
+    m_needJackStart = NeedNoJackStart;
+    startClocks();
+    return ;
+
+    int result;
+
+    // Restart the timer
+    if ((result = snd_seq_continue_queue(m_midiHandle, m_queue, NULL)) < 0) {
+        std::cerr << "AlsaDriver::startClocks - couldn't start queue - "
+        << snd_strerror(result)
+        << std::endl;
+        reportFailure(MappedEvent::FailureALSACallFailed);
+    }
+
+    m_queueRunning = true;
+
+    // process pending MIDI events
+    checkAlsaError(snd_seq_drain_output(m_midiHandle), "startClocksApproved(): draining");
+}
+
+void
+AlsaDriver::stopClocks()
+{
+#ifdef DEBUG_ALSA
+    std::cerr << "AlsaDriver::stopClocks" << std::endl;
+#endif
+
+    if (checkAlsaError(snd_seq_stop_queue(m_midiHandle, m_queue, NULL), "stopClocks(): stopping queue") < 0) {
+        reportFailure(MappedEvent::FailureALSACallFailed);
+    }
+    checkAlsaError(snd_seq_drain_output(m_midiHandle), "stopClocks(): draining output to stop queue");
+
+    m_queueRunning = false;
+
+    // We used to call m_jackDriver->stop() from here, but we no
+    // longer do -- it's now called from stopPlayback() so as to
+    // handle repositioning during playback (when stopClocks is
+    // necessary but stopPlayback and m_jackDriver->stop() are not).
+
+    snd_seq_event_t event;
+    snd_seq_ev_clear(&event);
+    snd_seq_real_time_t z = { 0, 0 };
+    snd_seq_ev_set_queue_pos_real(&event, m_queue, &z);
+    snd_seq_ev_set_direct(&event);
+    checkAlsaError(snd_seq_control_queue(m_midiHandle, m_queue, SND_SEQ_EVENT_SETPOS_TIME,
+                                         0, &event), "stopClocks(): setting zpos to queue");
+    // process that
+    checkAlsaError(snd_seq_drain_output(m_midiHandle), "stopClocks(): draining output to zpos queue");
+
+#ifdef DEBUG_ALSA
+
+    std::cerr << "AlsaDriver::stopClocks: ALSA time now is " << getAlsaTime() << std::endl;
+#endif
+
+    m_alsaPlayStartTime = RealTime::zeroTime;
+}
+
+
+void
+AlsaDriver::processEventsOut(const MappedComposition &mC)
+{
+    processEventsOut(mC, RealTime::zeroTime, RealTime::zeroTime);
+}
+
+void
+AlsaDriver::processEventsOut(const MappedComposition &mC,
+                             const RealTime &sliceStart,
+                             const RealTime &sliceEnd)
+{
+    // special case for unqueued events
+    bool now = (sliceStart == RealTime::zeroTime && sliceEnd == RealTime::zeroTime);
+
+    if (m_startPlayback) {
+        m_startPlayback = false;
+        // This only records whether we're playing in principle,
+        // not whether the clocks are actually ticking.  Contrariwise,
+        // areClocksRunning tells us whether the clocks are ticking
+        // but not whether we're actually playing (the clocks go even
+        // when we're not).  Check both if you want to know whether
+        // we're really rolling.
+        m_playing = true;
+
+        if (getMTCStatus() == TRANSPORT_SLAVE) {
+            tweakSkewForMTC(0);
+        }
+    }
+
+    AudioFile *audioFile = 0;
+    bool haveNewAudio = false;
+
+    // insert audio events if we find them
+    for (MappedComposition::const_iterator i = mC.begin(); i != mC.end(); ++i) {
+#ifdef HAVE_LIBJACK
+
+        // Play an audio file
+        //
+        if ((*i)->getType() == MappedEvent::Audio) {
+            if (!m_jackDriver)
+                continue;
+
+            // This is used for handling asynchronous
+            // (i.e. unexpected) audio events only
+
+            if ((*i)->getEventTime() > RealTime( -120, 0)) {
+                // Not an asynchronous event
+                continue;
+            }
+
+            // Check for existence of file - if the sequencer has died
+            // and been restarted then we're not always loaded up with
+            // the audio file references we should have.  In the future
+            // we could make this just get the gui to reload our files
+            // when (or before) this fails.
+            //
+            audioFile = getAudioFile((*i)->getAudioID());
+
+            if (audioFile) {
+                MappedAudioFader *fader =
+                    dynamic_cast<MappedAudioFader*>
+                    (getMappedStudio()->getAudioFader((*i)->getInstrument()));
+
+                if (!fader) {
+                    std::cerr << "WARNING: AlsaDriver::processEventsOut: no fader for audio instrument " << (*i)->getInstrument() << std::endl;
+                    continue;
+                }
+
+                unsigned int channels = fader->getPropertyList(
+                                            MappedAudioFader::Channels)[0].toInt();
+
+                RealTime bufferLength = getAudioReadBufferLength();
+                int bufferFrames = RealTime::realTime2Frame
+                                   (bufferLength, m_jackDriver->getSampleRate());
+                if (bufferFrames % m_jackDriver->getBufferSize()) {
+                    bufferFrames /= m_jackDriver->getBufferSize();
+                    bufferFrames ++;
+                    bufferFrames *= m_jackDriver->getBufferSize();
+                }
+
+                //#define DEBUG_PLAYING_AUDIO
+#ifdef DEBUG_PLAYING_AUDIO
+                std::cout << "Creating playable audio file: id " << audioFile->getId() << ", event time " << (*i)->getEventTime() << ", time now " << getAlsaTime() << ", start marker " << (*i)->getAudioStartMarker() << ", duration " << (*i)->getDuration() << ", instrument " << (*i)->getInstrument() << " channels " << channels << std::endl;
+
+                std::cout << "Read buffer length is " << bufferLength << " (" << bufferFrames << " frames)" << std::endl;
+#endif
+
+                PlayableAudioFile *paf = 0;
+
+                try {
+                    paf = new PlayableAudioFile((*i)->getInstrument(),
+                                                audioFile,
+                                                getSequencerTime() +
+                                                (RealTime(1, 0) / 4),
+                                                (*i)->getAudioStartMarker(),
+                                                (*i)->getDuration(),
+                                                bufferFrames,
+                                                getSmallFileSize() * 1024,
+                                                channels,
+                                                m_jackDriver->getSampleRate());
+                } catch (...) {
+                    continue;
+                }
+
+                if ((*i)->isAutoFading()) {
+                    paf->setAutoFade(true);
+                    paf->setFadeInTime((*i)->getFadeInTime());
+                    paf->setFadeOutTime((*i)->getFadeInTime());
+
+                    //#define DEBUG_AUTOFADING
+#ifdef DEBUG_AUTOFADING
+
+                    std::cout << "PlayableAudioFile is AUTOFADING - "
+                    << "in = " << (*i)->getFadeInTime()
+                    << ", out = " << (*i)->getFadeOutTime()
+                    << std::endl;
+#endif
+
+                }
+#ifdef DEBUG_AUTOFADING
+                else {
+                    std::cout << "PlayableAudioFile has no AUTOFADE"
+                    << std::endl;
+                }
+#endif
+
+
+                // segment runtime id
+                paf->setRuntimeSegmentId((*i)->getRuntimeSegmentId());
+
+                m_audioQueue->addUnscheduled(paf);
+
+                haveNewAudio = true;
+            } else {
+#ifdef DEBUG_ALSA
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "can't find audio file reference"
+                << std::endl;
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "try reloading the current Rosegarden file"
+                << std::endl;
+#else
+
+                ;
+#endif
+
+            }
+        }
+
+        // Cancel a playing audio file preview (this is predicated on
+        // runtime segment ID and optionally start time)
+        //
+        if ((*i)->getType() == MappedEvent::AudioCancel) {
+            cancelAudioFile(*i);
+        }
+
+#endif // HAVE_LIBJACK
+
+        if ((*i)->getType() == MappedEvent::SystemMIDIClock) {
+            switch ((int)(*i)->getData1()) {
+            case 0:
+                m_midiClockEnabled = false;
+#ifdef DEBUG_ALSA
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "Rosegarden MIDI CLOCK, START and STOP DISABLED"
+                << std::endl;
+#endif
+
+                setMIDISyncStatus(TRANSPORT_OFF);
+                break;
+
+            case 1:
+                m_midiClockEnabled = true;
+#ifdef DEBUG_ALSA
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "Rosegarden send MIDI CLOCK, START and STOP ENABLED"
+                << std::endl;
+#endif
+
+                setMIDISyncStatus(TRANSPORT_MASTER);
+                break;
+
+            case 2:
+                m_midiClockEnabled = false;
+#ifdef DEBUG_ALSA
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "Rosegarden accept START and STOP ENABLED"
+                << std::endl;
+#endif
+
+                setMIDISyncStatus(TRANSPORT_SLAVE);
+                break;
+            }
+        }
+
+        if ((*i)->getType() == MappedEvent::SystemMIDISyncAuto) {
+            if ((*i)->getData1()) {
+                m_midiSyncAutoConnect = true;
+#ifdef DEBUG_ALSA
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "Rosegarden MIDI SYNC AUTO ENABLED"
+                << std::endl;
+#endif
+
+                for (DevicePortMap::iterator dpmi = m_devicePortMap.begin();
+                        dpmi != m_devicePortMap.end(); ++dpmi) {
+                    snd_seq_connect_to(m_midiHandle,
+                                       m_syncOutputPort,
+                                       dpmi->second.first,
+                                       dpmi->second.second);
+                }
+            } else {
+                m_midiSyncAutoConnect = false;
+#ifdef DEBUG_ALSA
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "Rosegarden MIDI SYNC AUTO DISABLED"
+                << std::endl;
+#endif
+
+            }
+        }
+
+#ifdef HAVE_LIBJACK
+
+        // Set the JACK transport
+        if ((*i)->getType() == MappedEvent::SystemJackTransport) {
+            bool enabled = false;
+            bool master = false;
+
+            switch ((int)(*i)->getData1()) {
+            case 2:
+                master = true;
+                enabled = true;
+#ifdef DEBUG_ALSA
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "Rosegarden to follow JACK transport and request JACK timebase master role (not yet implemented)"
+                << std::endl;
+#endif
+
+                break;
+
+            case 1:
+                enabled = true;
+#ifdef DEBUG_ALSA
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "Rosegarden to follow JACK transport"
+                << std::endl;
+#endif
+
+                break;
+
+            case 0:
+            default:
+#ifdef DEBUG_ALSA
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "Rosegarden to ignore JACK transport"
+                << std::endl;
+#endif
+
+                break;
+            }
+
+            if (m_jackDriver) {
+                m_jackDriver->setTransportEnabled(enabled);
+                m_jackDriver->setTransportMaster(master);
+            }
+        }
+#endif // HAVE_LIBJACK
+
+
+        if ((*i)->getType() == MappedEvent::SystemMMCTransport) {
+            switch ((int)(*i)->getData1()) {
+            case 1:
+#ifdef DEBUG_ALSA
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "Rosegarden is MMC MASTER"
+                << std::endl;
+#endif
+
+                setMMCStatus(TRANSPORT_MASTER);
+                break;
+
+            case 2:
+#ifdef DEBUG_ALSA
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "Rosegarden is MMC SLAVE"
+                << std::endl;
+#endif
+
+                setMMCStatus(TRANSPORT_SLAVE);
+                break;
+
+            case 0:
+            default:
+#ifdef DEBUG_ALSA
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "Rosegarden MMC Transport DISABLED"
+                << std::endl;
+#endif
+
+                setMMCStatus(TRANSPORT_OFF);
+                break;
+            }
+        }
+
+        if ((*i)->getType() == MappedEvent::SystemMTCTransport) {
+            switch ((int)(*i)->getData1()) {
+            case 1:
+#ifdef DEBUG_ALSA
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "Rosegarden is MTC MASTER"
+                << std::endl;
+#endif
+
+                setMTCStatus(TRANSPORT_MASTER);
+                tweakSkewForMTC(0);
+                m_mtcFirstTime = -1;
+                break;
+
+            case 2:
+#ifdef DEBUG_ALSA
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "Rosegarden is MTC SLAVE"
+                << std::endl;
+#endif
+
+                setMTCStatus(TRANSPORT_SLAVE);
+                m_mtcFirstTime = -1;
+                break;
+
+            case 0:
+            default:
+#ifdef DEBUG_ALSA
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "Rosegarden MTC Transport DISABLED"
+                << std::endl;
+#endif
+
+                setMTCStatus(TRANSPORT_OFF);
+                m_mtcFirstTime = -1;
+                break;
+            }
+        }
+
+        if ((*i)->getType() == MappedEvent::SystemRecordDevice) {
+            DeviceId recordDevice =
+                (DeviceId)((*i)->getData1());
+            bool conn = (bool) ((*i)->getData2());
+
+            // Unset connections
+            //
+            // unsetRecordDevices();
+
+            // Special case to set for all record ports
+            //
+            if (recordDevice == Device::ALL_DEVICES) {
+                /* set all record devices */
+#ifdef DEBUG_ALSA
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "set all record devices - not implemented"
+                << std::endl;
+#endif
+
+                /*
+                MappedDeviceList::iterator it = m_devices.begin();
+                std::vector<int> ports;
+                std::vector<int>::iterator pIt;
+
+                for (; it != m_devices.end(); ++it)
+                {
+                    std::cout << "DEVICE = " << (*it)->getName() << " - DIR = "
+                         << (*it)->getDirection() << endl;
+                    // ignore ports we can't connect to
+                    if ((*it)->getDirection() == MidiDevice::WriteOnly) continue;
+
+                    std::cout << "PORTS = " << ports.size() << endl;
+                    ports = (*it)->getPorts();
+                    for (pIt = ports.begin(); pIt != ports.end(); ++pIt)
+                    {
+                        setRecordDevice((*it)->getClient(), *pIt);
+                    }
+                }
+                */
+            } else {
+                // Otherwise just for the one device and port
+                //
+                setRecordDevice(recordDevice, conn);
+            }
+        }
+
+        if ((*i)->getType() == MappedEvent::SystemAudioPortCounts) {
+            // never actually used, I think?
+        }
+
+        if ((*i)->getType() == MappedEvent::SystemAudioPorts) {
+#ifdef HAVE_LIBJACK
+            if (m_jackDriver) {
+                int data = (*i)->getData1();
+                m_jackDriver->setAudioPorts(data & MappedEvent::FaderOuts,
+                                            data & MappedEvent::SubmasterOuts);
+            }
+#else
+#ifdef DEBUG_ALSA
+            std::cerr << "AlsaDriver::processEventsOut - "
+            << "MappedEvent::SystemAudioPorts - no audio subsystem"
+            << std::endl;
+#endif
+#endif
+
+        }
+
+        if ((*i)->getType() == MappedEvent::SystemAudioFileFormat) {
+#ifdef HAVE_LIBJACK
+            int format = (*i)->getData1();
+            switch (format) {
+            case 0:
+                m_audioRecFileFormat = RIFFAudioFile::PCM;
+                break;
+            case 1:
+                m_audioRecFileFormat = RIFFAudioFile::FLOAT;
+                break;
+            default:
+#ifdef DEBUG_ALSA
+
+                std::cerr << "AlsaDriver::processEventsOut - "
+                << "MappedEvent::SystemAudioFileFormat - unexpected format number " << format
+                << std::endl;
+#endif
+
+                break;
+            }
+#else
+#ifdef DEBUG_ALSA
+            std::cerr << "AlsaDriver::processEventsOut - "
+            << "MappedEvent::SystemAudioFileFormat - no audio subsystem"
+            << std::endl;
+#endif
+#endif
+
+        }
+
+        if ((*i)->getType() == MappedEvent::Panic) {
+            for (MappedDeviceList::iterator i = m_devices.begin();
+                    i != m_devices.end(); ++i) {
+                if ((*i)->getDirection() == MidiDevice::Play) {
+                    sendDeviceController((*i)->getId(),
+                                         MIDI_CONTROLLER_SUSTAIN, 0);
+                    sendDeviceController((*i)->getId(),
+                                         MIDI_CONTROLLER_ALL_NOTES_OFF, 0);
+                    sendDeviceController((*i)->getId(),
+                                         MIDI_CONTROLLER_RESET, 0);
+                }
+            }
+        }
+    }
+
+    // Process Midi and Audio
+    //
+    processMidiOut(mC, sliceStart, sliceEnd);
+
+#ifdef HAVE_LIBJACK
+
+    if (m_jackDriver) {
+        if (haveNewAudio) {
+            if (now) {
+                m_jackDriver->prebufferAudio();
+                m_jackDriver->setHaveAsyncAudioEvent();
+            }
+            if (m_queueRunning) {
+                m_jackDriver->kickAudio();
+            }
+        }
+    }
+#endif
+}
+
+bool
+AlsaDriver::record(RecordStatus recordStatus,
+                   const std::vector<InstrumentId> *armedInstruments,
+                   const std::vector<QString> *audioFileNames)
+{
+    m_recordingInstruments.clear();
+
+    if (recordStatus == RECORD_ON) {
+        // start recording
+        m_recordStatus = RECORD_ON;
+        m_alsaRecordStartTime = RealTime::zeroTime;
+
+        unsigned int audioCount = 0;
+
+        if (armedInstruments) {
+
+            for (unsigned int i = 0; i < armedInstruments->size(); ++i) {
+
+                InstrumentId id = (*armedInstruments)[i];
+
+                m_recordingInstruments.insert(id);
+                if (!audioFileNames || (audioCount >= audioFileNames->size())) {
+                    continue;
+                }
+
+                QString fileName = (*audioFileNames)[audioCount];
+
+                if (id >= AudioInstrumentBase &&
+                        id < MidiInstrumentBase) {
+
+                    bool good = false;
+
+#ifdef DEBUG_ALSA
+
+                    std::cerr << "AlsaDriver::record: Requesting new record file \"" << fileName << "\" for instrument " << id << std::endl;
+#endif
+
+#ifdef HAVE_LIBJACK
+
+                    if (m_jackDriver &&
+                            m_jackDriver->openRecordFile(id, fileName.data())) {
+                        good = true;
+                    }
+#endif
+
+                    if (!good) {
+                        m_recordStatus = RECORD_OFF;
+                        std::cerr << "AlsaDriver::record: No JACK driver, or JACK driver failed to prepare for recording audio" << std::endl;
+                        return false;
+                    }
+
+                    ++audioCount;
+                }
+            }
+        }
+    } else
+        if (recordStatus == RECORD_OFF) {
+            m_recordStatus = RECORD_OFF;
+        }
+#ifdef DEBUG_ALSA
+        else {
+            std::cerr << "AlsaDriver::record - unsupported recording mode"
+            << std::endl;
+        }
+#endif
+
+    return true;
+}
+
+ClientPortPair
+AlsaDriver::getFirstDestination(bool duplex)
+{
+    ClientPortPair destPair( -1, -1);
+    AlsaPortList::iterator it;
+
+    for (it = m_alsaPorts.begin(); it != m_alsaPorts.end(); ++it) {
+        destPair.first = (*it)->m_client;
+        destPair.second = (*it)->m_port;
+
+        // If duplex port is required then choose first one
+        //
+        if (duplex) {
+            if ((*it)->m_direction == Duplex)
+                return destPair;
+        } else {
+            // If duplex port isn't required then choose first
+            // specifically non-duplex port (should be a synth)
+            //
+            if ((*it)->m_direction != Duplex)
+                return destPair;
+        }
+    }
+
+    return destPair;
+}
+
+
+// Sort through the ALSA client/port pairs for the range that
+// matches the one we're querying.  If none matches then send
+// back -1 for each.
+//
+ClientPortPair
+AlsaDriver::getPairForMappedInstrument(InstrumentId id)
+{
+    MappedInstrument *instrument = getMappedInstrument(id);
+    if (instrument) {
+        DeviceId device = instrument->getDevice();
+        DevicePortMap::iterator i = m_devicePortMap.find(device);
+        if (i != m_devicePortMap.end()) {
+            return i->second;
+        }
+    }
+#ifdef DEBUG_ALSA
+    /*
+    else
+    {
+    cerr << "WARNING: AlsaDriver::getPairForMappedInstrument: couldn't find instrument for id " << id << ", falling through" << endl;
+    }
+    */
+#endif
+
+    return ClientPortPair( -1, -1);
+}
+
+int
+AlsaDriver::getOutputPortForMappedInstrument(InstrumentId id)
+{
+    MappedInstrument *instrument = getMappedInstrument(id);
+    if (instrument) {
+        DeviceId device = instrument->getDevice();
+        DeviceIntMap::iterator i = m_outputPorts.find(device);
+        if (i != m_outputPorts.end()) {
+            return i->second;
+        }
+#ifdef DEBUG_ALSA
+        else {
+            cerr << "WARNING: AlsaDriver::getOutputPortForMappedInstrument: couldn't find output port for device for instrument " << id << ", falling through" << endl;
+        }
+#endif
+
+    }
+
+    return -1;
+}
+
+// Send a direct controller to the specified port/client
+//
+void
+AlsaDriver::sendDeviceController(DeviceId device,
+                                 MidiByte controller,
+                                 MidiByte value)
+{
+    snd_seq_event_t event;
+
+    snd_seq_ev_clear(&event);
+
+    snd_seq_ev_set_subs(&event);
+
+    DeviceIntMap::iterator dimi = m_outputPorts.find(device);
+    if (dimi == m_outputPorts.end())
+        return ;
+
+    snd_seq_ev_set_source(&event, dimi->second);
+    snd_seq_ev_set_direct(&event);
+
+    for (int i = 0; i < 16; i++) {
+        snd_seq_ev_set_controller(&event,
+                                  i,
+                                  controller,
+                                  value);
+        snd_seq_event_output_direct(m_midiHandle, &event);
+    }
+
+    // we probably don't need this:
+    checkAlsaError(snd_seq_drain_output(m_midiHandle), "sendDeviceController(): draining");
+}
+
+void
+AlsaDriver::processPending()
+{
+    if (!m_playing) {
+        processNotesOff(getAlsaTime(), true);
+        checkAlsaError(snd_seq_drain_output(m_midiHandle), "processPending(): draining");
+    }
+
+#ifdef HAVE_LIBJACK
+    if (m_jackDriver) {
+        m_jackDriver->updateAudioData();
+    }
+#endif
+
+    scavengePlugins();
+    m_audioQueueScavenger.scavenge();
+}
+
+void
+AlsaDriver::insertMappedEventForReturn(MappedEvent *mE)
+{
+    // Insert the event ready for return at the next opportunity.
+    //
+    m_returnComposition.insert(mE);
+}
+
+// check for recording status on any ALSA Port
+//
+bool
+AlsaDriver::isRecording(AlsaPortDescription *port)
+{
+    if (port->isReadable()) {
+
+        snd_seq_query_subscribe_t *qSubs;
+        snd_seq_addr_t rg_addr, sender_addr;
+        snd_seq_query_subscribe_alloca(&qSubs);
+
+        rg_addr.client = m_client;
+        rg_addr.port = m_inputPort;
+
+        snd_seq_query_subscribe_set_type(qSubs, SND_SEQ_QUERY_SUBS_WRITE);
+        snd_seq_query_subscribe_set_index(qSubs, 0);
+        snd_seq_query_subscribe_set_root(qSubs, &rg_addr);
+
+        while (snd_seq_query_port_subscribers(m_midiHandle, qSubs) >= 0) {
+            sender_addr = *snd_seq_query_subscribe_get_addr(qSubs);
+            if (sender_addr.client == port->m_client &&
+                    sender_addr.port == port->m_port)
+                return true;
+
+            snd_seq_query_subscribe_set_index(qSubs,
+                                              snd_seq_query_subscribe_get_index(qSubs) + 1);
+        }
+    }
+    return false;
+}
+
+bool
+AlsaDriver::checkForNewClients()
+{
+    Audit audit;
+    bool madeChange = false;
+
+    if (!m_portCheckNeeded)
+        return false;
+
+    AlsaPortList newPorts;
+    generatePortList(&newPorts); // updates m_alsaPorts, returns new ports as well
+
+    // If any devices have connections that no longer exist,
+    // clear those connections and stick them in the suspended
+    // port map in case they come back online later.
+
+    for (MappedDeviceList::iterator i = m_devices.begin();
+            i != m_devices.end(); ++i) {
+
+        ClientPortPair pair(m_devicePortMap[(*i)->getId()]);
+
+        bool found = false;
+        for (AlsaPortList::iterator j = m_alsaPorts.begin();
+                j != m_alsaPorts.end(); ++j) {
+            if ((*j)->m_client == pair.first &&
+                    (*j)->m_port == pair.second) {
+                if ((*i)->getDirection() == MidiDevice::Record) {
+                    bool recState = isRecording(*j);
+                    if (recState != (*i)->isRecording()) {
+                        madeChange = true;
+                        (*i)->setRecording(recState);
+                    }
+                } else {
+                    (*i)->setRecording(false);
+                }
+                found = true;
+                break;
+            }
+        }
+
+        if (!found) {
+            m_suspendedPortMap[pair] = (*i)->getId();
+            m_devicePortMap[(*i)->getId()] = ClientPortPair( -1, -1);
+            setConnectionToDevice(**i, "");
+            (*i)->setRecording(false);
+            madeChange = true;
+        }
+    }
+
+    // If we've increased the number of connections, we need
+    // to assign the new connections to existing devices that
+    // have none, where possible, and create new devices for
+    // any left over.
+
+    if (newPorts.size() > 0) {
+
+        audit << "New ports:" << std::endl;
+
+        for (AlsaPortList::iterator i = newPorts.begin();
+                i != newPorts.end(); ++i) {
+
+            if ((*i)->m_client == m_client) {
+                audit << "(Ignoring own port " << (*i)->m_client << ":" << (*i)->m_port << ")" << std::endl;
+                continue;
+            } else if ((*i)->m_client == 0) {
+                audit << "(Ignoring system port " << (*i)->m_client << ":" << (*i)->m_port << ")" << std::endl;
+                continue;
+            }
+
+            audit << (*i)->m_name << std::endl;
+
+            QString portName = (*i)->m_name.c_str();
+            ClientPortPair portPair = ClientPortPair((*i)->m_client,
+                                      (*i)->m_port);
+
+            if (m_suspendedPortMap.find(portPair) != m_suspendedPortMap.end()) {
+
+                DeviceId id = m_suspendedPortMap[portPair];
+
+                audit << "(Reusing suspended device " << id << ")" << std::endl;
+
+                for (MappedDeviceList::iterator j = m_devices.begin();
+                        j != m_devices.end(); ++j) {
+                    if ((*j)->getId() == id) {
+                        setConnectionToDevice(**j, portName);
+                    }
+                }
+
+                m_suspendedPortMap.erase(m_suspendedPortMap.find(portPair));
+                m_devicePortMap[id] = portPair;
+                madeChange = true;
+                continue;
+            }
+
+            bool needPlayDevice = true, needRecordDevice = true;
+
+            if ((*i)->isReadable()) {
+                for (MappedDeviceList::iterator j = m_devices.begin();
+                        j != m_devices.end(); ++j) {
+                    if ((*j)->getType() == Device::Midi &&
+                            (*j)->getConnection() == "" &&
+                            (*j)->getDirection() == MidiDevice::Record) {
+                        audit << "(Reusing record device " << (*j)->getId()
+                        << ")" << std::endl;
+                        m_devicePortMap[(*j)->getId()] = portPair;
+                        setConnectionToDevice(**j, portName);
+                        needRecordDevice = false;
+                        madeChange = true;
+                        break;
+                    }
+                }
+            } else {
+                needRecordDevice = false;
+            }
+
+            if ((*i)->isWriteable()) {
+                for (MappedDeviceList::iterator j = m_devices.begin();
+                        j != m_devices.end(); ++j) {
+                    if ((*j)->getType() == Device::Midi &&
+                            (*j)->getConnection() == "" &&
+                            (*j)->getDirection() == MidiDevice::Play) {
+                        audit << "(Reusing play device " << (*j)->getId()
+                        << ")" << std::endl;
+                        m_devicePortMap[(*j)->getId()] = portPair;
+                        setConnectionToDevice(**j, portName);
+                        needPlayDevice = false;
+                        madeChange = true;
+                        break;
+                    }
+                }
+            } else {
+                needPlayDevice = false;
+            }
+
+            if (needRecordDevice) {
+                MappedDevice *device = createMidiDevice(*i, MidiDevice::Record);
+                if (!device) {
+#ifdef DEBUG_ALSA
+                    std::cerr << "WARNING: Failed to create record device" << std::endl;
+#else
+
+                    ;
+#endif
+
+                } else {
+                    audit << "(Created new record device " << device->getId() << ")" << std::endl;
+                    addInstrumentsForDevice(device);
+                    m_devices.push_back(device);
+                    madeChange = true;
+                }
+            }
+
+            if (needPlayDevice) {
+                MappedDevice *device = createMidiDevice(*i, MidiDevice::Play);
+                if (!device) {
+#ifdef DEBUG_ALSA
+                    std::cerr << "WARNING: Failed to create play device" << std::endl;
+#else
+
+                    ;
+#endif
+
+                } else {
+                    audit << "(Created new play device " << device->getId() << ")" << std::endl;
+                    addInstrumentsForDevice(device);
+                    m_devices.push_back(device);
+                    madeChange = true;
+                }
+            }
+        }
+    }
+
+    // If one of our ports is connected to a single other port and
+    // it isn't the one we thought, we should update our connection
+
+    for (MappedDeviceList::iterator i = m_devices.begin();
+            i != m_devices.end(); ++i) {
+
+        DevicePortMap::iterator j = m_devicePortMap.find((*i)->getId());
+
+        snd_seq_addr_t addr;
+        addr.client = m_client;
+
+        DeviceIntMap::iterator ii = m_outputPorts.find((*i)->getId());
+        if (ii == m_outputPorts.end())
+            continue;
+        addr.port = ii->second;
+
+        snd_seq_query_subscribe_t *subs;
+        snd_seq_query_subscribe_alloca(&subs);
+        snd_seq_query_subscribe_set_root(subs, &addr);
+        snd_seq_query_subscribe_set_index(subs, 0);
+
+        bool haveOurs = false;
+        int others = 0;
+        ClientPortPair firstOther;
+
+        while (!snd_seq_query_port_subscribers(m_midiHandle, subs)) {
+
+            const snd_seq_addr_t *otherEnd =
+                snd_seq_query_subscribe_get_addr(subs);
+
+            if (!otherEnd)
+                continue;
+
+            if (j != m_devicePortMap.end() &&
+                    otherEnd->client == j->second.first &&
+                    otherEnd->port == j->second.second) {
+                haveOurs = true;
+            } else {
+                ++others;
+                firstOther = ClientPortPair(otherEnd->client, otherEnd->port);
+            }
+
+            snd_seq_query_subscribe_set_index
+            (subs, snd_seq_query_subscribe_get_index(subs) + 1);
+        }
+
+        if (haveOurs) { // leave our own connection alone, and stop worrying
+            continue;
+
+        } else {
+            if (others == 0) {
+                if (j != m_devicePortMap.end()) {
+                    j->second = ClientPortPair( -1, -1);
+                    setConnectionToDevice(**i, "");
+                    madeChange = true;
+                }
+            } else {
+                for (AlsaPortList::iterator k = m_alsaPorts.begin();
+                        k != m_alsaPorts.end(); ++k) {
+                    if ((*k)->m_client == firstOther.first &&
+                            (*k)->m_port == firstOther.second) {
+                        m_devicePortMap[(*i)->getId()] = firstOther;
+                        setConnectionToDevice(**i, (*k)->m_name.c_str(),
+                                              firstOther);
+                        madeChange = true;
+                        break;
+                    }
+                }
+            }
+        }
+    }
+
+    if (madeChange) {
+        MappedEvent *mE =
+            new MappedEvent(0, MappedEvent::SystemUpdateInstruments,
+                            0, 0);
+        // send completion event
+        insertMappedEventForReturn(mE);
+    }
+
+    m_portCheckNeeded = false;
+
+    return true;
+}
+
+
+// From a DeviceId get a client/port pair for connecting as the
+// MIDI record device.
+//
+void
+AlsaDriver::setRecordDevice(DeviceId id, bool connectAction)
+{
+    Audit audit;
+
+    // Locate a suitable port
+    //
+    if (m_devicePortMap.find(id) == m_devicePortMap.end()) {
+#ifdef DEBUG_ALSA
+        audit << "AlsaDriver::setRecordDevice - "
+        << "couldn't match device id (" << id << ") to ALSA port"
+        << std::endl;
+#endif
+
+        return ;
+    }
+
+    ClientPortPair pair = m_devicePortMap[id];
+
+    snd_seq_addr_t sender, dest;
+    sender.client = pair.first;
+    sender.port = pair.second;
+
+    for (MappedDeviceList::iterator i = m_devices.begin();
+            i != m_devices.end(); ++i) {
+        if ((*i)->getId() == id) {
+            if ((*i)->getDirection() == MidiDevice::Record) {
+                if ((*i)->isRecording() && connectAction) {
+#ifdef DEBUG_ALSA
+                    audit << "AlsaDriver::setRecordDevice - "
+                    << "attempting to subscribe (" << id
+                    << ") already subscribed" << std::endl;
+#endif
+
+                    return ;
+                }
+                if (!(*i)->isRecording() && !connectAction) {
+#ifdef DEBUG_ALSA
+                    audit << "AlsaDriver::setRecordDevice - "
+                    << "attempting to unsubscribe (" << id
+                    << ") already unsubscribed" << std::endl;
+#endif
+
+                    return ;
+                }
+            } else {
+#ifdef DEBUG_ALSA
+                audit << "AlsaDriver::setRecordDevice - "
+                << "attempting to set play device (" << id
+                << ") to record device" << std::endl;
+#endif
+
+                return ;
+            }
+            break;
+        }
+    }
+
+    snd_seq_port_subscribe_t *subs;
+    snd_seq_port_subscribe_alloca(&subs);
+
+    dest.client = m_client;
+    dest.port = m_inputPort;
+
+    // Set destinations and senders
+    //
+    snd_seq_port_subscribe_set_sender(subs, &sender);
+    snd_seq_port_subscribe_set_dest(subs, &dest);
+
+    // subscribe or unsubscribe the port
+    //
+    if (connectAction) {
+        if (checkAlsaError(snd_seq_subscribe_port(m_midiHandle, subs),
+                           "setRecordDevice - failed subscription of input port") < 0) {
+            // Not the end of the world if this fails but we
+            // have to flag it internally.
+            //
+            audit << "AlsaDriver::setRecordDevice - "
+            << int(sender.client) << ":" << int(sender.port)
+            << " failed to subscribe device "
+            << id << " as record port" << std::endl;
+        } else {
+            m_midiInputPortConnected = true;
+            audit << "AlsaDriver::setRecordDevice - "
+            << "successfully subscribed device "
+            << id << " as record port" << std::endl;
+        }
+    } else {
+        if (checkAlsaError(snd_seq_unsubscribe_port(m_midiHandle, subs),
+                           "setRecordDevice - failed to unsubscribe a device") == 0)
+            audit << "AlsaDriver::setRecordDevice - "
+            << "successfully unsubscribed device "
+            << id << " as record port" << std::endl;
+
+    }
+}
+
+// Clear any record device connections
+//
+void
+AlsaDriver::unsetRecordDevices()
+{
+    snd_seq_addr_t dest;
+    dest.client = m_client;
+    dest.port = m_inputPort;
+
+    snd_seq_query_subscribe_t *qSubs;
+    snd_seq_addr_t tmp_addr;
+    snd_seq_query_subscribe_alloca(&qSubs);
+
+    tmp_addr.client = m_client;
+    tmp_addr.port = m_inputPort;
+
+    // Unsubsribe any existing connections
+    //
+    snd_seq_query_subscribe_set_type(qSubs, SND_SEQ_QUERY_SUBS_WRITE);
+    snd_seq_query_subscribe_set_index(qSubs, 0);
+    snd_seq_query_subscribe_set_root(qSubs, &tmp_addr);
+
+    while (snd_seq_query_port_subscribers(m_midiHandle, qSubs) >= 0) {
+        tmp_addr = *snd_seq_query_subscribe_get_addr(qSubs);
+
+        snd_seq_port_subscribe_t *dSubs;
+        snd_seq_port_subscribe_alloca(&dSubs);
+
+        snd_seq_addr_t dSender;
+        dSender.client = tmp_addr.client;
+        dSender.port = tmp_addr.port;
+
+        snd_seq_port_subscribe_set_sender(dSubs, &dSender);
+        snd_seq_port_subscribe_set_dest(dSubs, &dest);
+
+        int error = snd_seq_unsubscribe_port(m_midiHandle, dSubs);
+
+        if (error < 0) {
+#ifdef DEBUG_ALSA
+            std::cerr << "AlsaDriver::unsetRecordDevices - "
+            << "can't unsubscribe record port" << std::endl;
+#endif
+
+        }
+
+        snd_seq_query_subscribe_set_index(qSubs,
+                                          snd_seq_query_subscribe_get_index(qSubs) + 1);
+    }
+}
+
+void
+AlsaDriver::sendMMC(MidiByte deviceArg,
+                    MidiByte instruction,
+                    bool isCommand,
+                    const std::string &data)
+{
+    snd_seq_event_t event;
+
+    snd_seq_ev_clear(&event);
+    snd_seq_ev_set_source(&event, m_syncOutputPort);
+    snd_seq_ev_set_subs(&event);
+
+    unsigned char dataArr[10] =
+        { MIDI_SYSTEM_EXCLUSIVE,
+          MIDI_SYSEX_RT, deviceArg,
+          (isCommand ? MIDI_SYSEX_RT_COMMAND : MIDI_SYSEX_RT_RESPONSE),
+          instruction };
+
+    std::string dataString = std::string((const char *)dataArr) +
+                             data + (char)MIDI_END_OF_EXCLUSIVE;
+
+    snd_seq_ev_set_sysex(&event, dataString.length(),
+                         (char *)dataString.c_str());
+
+    event.queue = SND_SEQ_QUEUE_DIRECT;
+
+    checkAlsaError(snd_seq_event_output_direct(m_midiHandle, &event),
+                   "sendMMC event send");
+
+    if (m_queueRunning) {
+        checkAlsaError(snd_seq_drain_output(m_midiHandle), "sendMMC drain");
+    }
+}
+
+// Send a system real-time message from the sync output port
+//
+void
+AlsaDriver::sendSystemDirect(MidiByte command, int *args)
+{
+    snd_seq_event_t event;
+
+    snd_seq_ev_clear(&event);
+    snd_seq_ev_set_source(&event, m_syncOutputPort);
+    snd_seq_ev_set_subs(&event);
+
+    event.queue = SND_SEQ_QUEUE_DIRECT;
+
+    // set the command
+    event.type = command;
+
+    // set args if we have them
+    if (args) {
+        event.data.control.value = *args;
+    }
+
+    int error = snd_seq_event_output_direct(m_midiHandle, &event);
+
+    if (error < 0) {
+#ifdef DEBUG_ALSA
+        std::cerr << "AlsaDriver::sendSystemDirect - "
+        << "can't send event (" << int(command) << ")"
+        << std::endl;
+#endif
+
+    }
+
+    //    checkAlsaError(snd_seq_drain_output(m_midiHandle),
+    //		   "sendSystemDirect(): draining");
+}
+
+
+void
+AlsaDriver::sendSystemQueued(MidiByte command,
+                             const std::string &args,
+                             const RealTime &time)
+{
+    snd_seq_event_t event;
+
+    snd_seq_ev_clear(&event);
+    snd_seq_ev_set_source(&event, m_syncOutputPort);
+    snd_seq_ev_set_subs(&event);
+
+    snd_seq_real_time_t sendTime = { time.sec, time.nsec };
+
+    // Schedule the command
+    //
+    event.type = command;
+
+    snd_seq_ev_schedule_real(&event, m_queue, 0, &sendTime);
+
+    // set args if we have them
+    switch (args.length()) {
+    case 1:
+        event.data.control.value = args[0];
+        break;
+
+    case 2:
+        event.data.control.value = int(args[0]) | (int(args[1]) << 7);
+        break;
+
+    default:  // do nothing
+        break;
+    }
+
+    int error = snd_seq_event_output(m_midiHandle, &event);
+
+    if (error < 0) {
+#ifdef DEBUG_ALSA
+        std::cerr << "AlsaDriver::sendSystemQueued - "
+        << "can't send event (" << int(command) << ")"
+        << " - error = (" << error << ")"
+        << std::endl;
+#endif
+
+    }
+
+    //    if (m_queueRunning) {
+    //	checkAlsaError(snd_seq_drain_output(m_midiHandle), "sendSystemQueued(): draining");
+    //    }
+}
+
+
+void
+AlsaDriver::claimUnwantedPlugin(void *plugin)
+{
+    m_pluginScavenger.claim((RunnablePluginInstance *)plugin);
+}
+
+
+void
+AlsaDriver::scavengePlugins()
+{
+    m_pluginScavenger.scavenge();
+}
+
+
+QString
+AlsaDriver::getStatusLog()
+{
+    return QString::fromUtf8(Audit::getAudit().c_str());
+}
+
+
+void
+AlsaDriver::sleep(const RealTime &rt)
+{
+    int npfd = snd_seq_poll_descriptors_count(m_midiHandle, POLLIN);
+    struct pollfd *pfd = (struct pollfd *)alloca(npfd * sizeof(struct pollfd));
+    snd_seq_poll_descriptors(m_midiHandle, pfd, npfd, POLLIN);
+    poll(pfd, npfd, rt.sec * 1000 + rt.msec());
+}
+
+void
+AlsaDriver::runTasks()
+{
+#ifdef HAVE_LIBJACK
+    if (m_jackDriver) {
+        if (!m_jackDriver->isOK()) {
+            m_jackDriver->restoreIfRestorable();
+        }
+    }
+
+    if (m_doTimerChecks && m_timerRatioCalculated) {
+
+        double ratio = m_timerRatio;
+        m_timerRatioCalculated = false;
+
+        snd_seq_queue_tempo_t *q_ptr;
+        snd_seq_queue_tempo_alloca(&q_ptr);
+
+        snd_seq_get_queue_tempo(m_midiHandle, m_queue, q_ptr);
+
+        unsigned int t_skew = snd_seq_queue_tempo_get_skew(q_ptr);
+#ifdef DEBUG_ALSA
+
+        unsigned int t_base = snd_seq_queue_tempo_get_skew_base(q_ptr);
+        if (!m_playing) {
+            std::cerr << "Skew: " << t_skew << "/" << t_base;
+        }
+#endif
+
+        unsigned int newSkew = t_skew + (unsigned int)(t_skew * ratio);
+
+        if (newSkew != t_skew) {
+#ifdef DEBUG_ALSA
+            if (!m_playing) {
+                std::cerr << " changed to " << newSkew << endl;
+            }
+#endif
+            snd_seq_queue_tempo_set_skew(q_ptr, newSkew);
+            snd_seq_set_queue_tempo( m_midiHandle, m_queue, q_ptr);
+        } else {
+#ifdef DEBUG_ALSA
+            if (!m_playing) {
+                std::cerr << endl;
+            }
+#endif
+
+        }
+
+        m_firstTimerCheck = true;
+    }
+
+#endif
+}
+
+void
+AlsaDriver::reportFailure(MappedEvent::FailureCode code)
+{
+    //#define REPORT_XRUNS 1
+#ifndef REPORT_XRUNS
+    if (code == MappedEvent::FailureXRuns ||
+            code == MappedEvent::FailureDiscUnderrun ||
+            code == MappedEvent::FailureBussMixUnderrun ||
+            code == MappedEvent::FailureMixUnderrun) {
+        return ;
+    }
+#endif
+
+    // Ignore consecutive duplicates
+    if (_failureReportWriteIndex > 0 &&
+            _failureReportWriteIndex != _failureReportReadIndex) {
+        if (code == _failureReports[_failureReportWriteIndex - 1])
+            return ;
+    }
+
+    _failureReports[_failureReportWriteIndex] = code;
+    _failureReportWriteIndex =
+        (_failureReportWriteIndex + 1) % FAILURE_REPORT_COUNT;
+}
+
+std::string
+AlsaDriver::getAlsaModuleVersionString()
+{
+    FILE *v = fopen("/proc/asound/version", "r");
+    
+    // Examples:
+    // Advanced Linux Sound Architecture Driver Version 1.0.14rc3.
+    // Advanced Linux Sound Architecture Driver Version 1.0.14 (Thu May 31 09:03:25 2008 UTC).
+
+    if (v) {
+	char buf[256];
+	fgets(buf, 256, v);
+	fclose(v);
+
+	std::string vs(buf);
+	std::string::size_type sp = vs.find_first_of('.');
+	if (sp > 0 && sp != std::string::npos) {
+	    while (sp > 0 && isdigit(vs[sp-1])) --sp;
+	    vs = vs.substr(sp);
+	    if (vs.length() > 0 && vs[vs.length()-1] == '\n') {
+		vs = vs.substr(0, vs.length()-1);
+	    }
+	    if (vs.length() > 0 && vs[vs.length()-1] == '.') {
+		vs = vs.substr(0, vs.length()-1);
+	    }
+	    return vs;
+	}
+    }
+
+    return "(unknown)";
+}
+
+std::string
+AlsaDriver::getKernelVersionString()
+{
+    FILE *v = fopen("/proc/version", "r");
+
+    if (v) {
+	char buf[256];
+	fgets(buf, 256, v);
+	fclose(v);
+
+	std::string vs(buf);
+	std::string key(" version ");
+	std::string::size_type sp = vs.find(key);
+	if (sp != std::string::npos) {
+	    vs = vs.substr(sp + key.length());
+	    sp = vs.find(' ');
+	    if (sp != std::string::npos) {
+		vs = vs.substr(0, sp);
+	    }
+	    if (vs.length() > 0 && vs[vs.length()-1] == '\n') {
+		vs = vs.substr(0, vs.length()-1);
+	    }
+	    return vs;
+	}
+    }
+
+    return "(unknown)";
+}
+
+void
+AlsaDriver::extractVersion(std::string v, int &major, int &minor, int &subminor, std::string &suffix)
+{
+    major = minor = subminor = 0;
+    suffix = "";
+    if (v == "(unknown)") return;
+
+    std::string::size_type sp, pp;
+
+    sp = v.find('.');
+    if (sp == std::string::npos) goto done;
+    major = atoi(v.substr(0, sp).c_str());
+    pp = sp + 1;
+
+    sp = v.find('.', pp);
+    if (sp == std::string::npos) goto done;
+    minor = atoi(v.substr(pp, sp - pp).c_str());
+    pp = sp + 1;
+
+    while (++sp < v.length() && (::isdigit(v[sp]) || v[sp] == '-'));
+    subminor = atoi(v.substr(pp, sp - pp).c_str());
+
+    if (sp >= v.length()) goto done;
+    suffix = v.substr(sp);
+
+done:
+    std::cerr << "extractVersion: major = " << major << ", minor = " << minor << ", subminor = " << subminor << ", suffix = \"" << suffix << "\"" << std::endl;
+}
+
+bool
+AlsaDriver::versionIsAtLeast(std::string v, int major, int minor, int subminor)
+{
+    int actualMajor, actualMinor, actualSubminor;
+    std::string actualSuffix;
+
+    extractVersion(v, actualMajor, actualMinor, actualSubminor, actualSuffix);
+
+    bool ok = false;
+
+    if (actualMajor > major) {
+	ok = true;
+    } else if (actualMajor == major) {
+	if (actualMinor > minor) {
+	    ok = true;
+	} else if (actualMinor == minor) {
+	    if (actualSubminor > subminor) {
+		ok = true;
+	    } else if (actualSubminor == subminor) {
+		if (strncmp(actualSuffix.c_str(), "rc", 2) &&
+		    strncmp(actualSuffix.c_str(), "pre", 3)) {
+		    ok = true;
+		}
+	    }
+	}
+    }
+
+    std::cerr << "AlsaDriver::versionIsAtLeast: is version " << v << " at least " << major << "." << minor << "." << subminor << "? " << (ok ? "yes" : "no") << std::endl;
+    return ok;
+}    
+
+}
+
+
+#endif // HAVE_ALSA
diff --git a/src/sound/AlsaDriver.h b/src/sound/AlsaDriver.h
new file mode 100644
index 0000000..e80e30f
--- /dev/null
+++ b/src/sound/AlsaDriver.h
@@ -0,0 +1,561 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+// Specialisation of SoundDriver to support ALSA (http://www.alsa-project.org)
+//
+//
+#ifndef _ALSADRIVER_H_
+#define _ALSADRIVER_H_
+
+#include <vector>
+#include <set>
+#include <map>
+
+#ifdef HAVE_ALSA
+
+#include <alsa/asoundlib.h> // ALSA
+
+#include "SoundDriver.h"
+#include "Instrument.h"
+#include "Device.h"
+#include "AlsaPort.h"
+#include "Scavenger.h"
+#include "RunnablePluginInstance.h"
+
+#ifdef HAVE_LIBJACK
+#include "JackDriver.h"
+#endif
+
+namespace Rosegarden
+{
+
+class AlsaDriver : public SoundDriver
+{
+public:
+    AlsaDriver(MappedStudio *studio);
+    virtual ~AlsaDriver();
+
+    // shutdown everything that's currently open
+    void shutdown();
+
+    virtual bool initialise();
+    virtual void initialisePlayback(const RealTime &position);
+    virtual void stopPlayback();
+    virtual void punchOut();
+    virtual void resetPlayback(const RealTime &oldPosition, const RealTime &position);
+    virtual void allNotesOff();
+    virtual void processNotesOff(const RealTime &time, bool now, bool everything = false);
+
+    virtual RealTime getSequencerTime();
+
+    virtual MappedComposition *getMappedComposition();
+    
+    virtual bool record(RecordStatus recordStatus,
+                        const std::vector<InstrumentId> *armedInstruments = 0,
+                        const std::vector<QString> *audioFileNames = 0);
+
+    virtual void startClocks();
+    virtual void startClocksApproved(); // called by JACK driver in sync mode
+    virtual void stopClocks();
+    virtual bool areClocksRunning() const { return m_queueRunning; }
+
+    virtual void processEventsOut(const MappedComposition &mC);
+    virtual void processEventsOut(const MappedComposition &mC,
+                                  const RealTime &sliceStart,
+                                  const RealTime &sliceEnd);
+
+    // Return the sample rate
+    //
+    virtual unsigned int getSampleRate() const {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) return m_jackDriver->getSampleRate();
+        else return 0;
+#else
+        return 0;
+#endif
+    }
+
+    // Define here to catch this being reset
+    //
+    virtual void setMIDIClockInterval(RealTime interval);
+
+    // initialise subsystems
+    //
+    bool initialiseMidi();
+    void initialiseAudio();
+
+    // Some stuff to help us debug this
+    //
+    void getSystemInfo();
+    void showQueueStatus(int queue);
+
+    // Process pending
+    //
+    virtual void processPending();
+
+    // We can return audio control signals to the gui using MappedEvents.
+    // Meter levels or audio file completions can go in here.
+    //
+    void insertMappedEventForReturn(MappedEvent *mE);
+
+    
+    virtual RealTime getAudioPlayLatency() {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) return m_jackDriver->getAudioPlayLatency();
+#endif
+        return RealTime::zeroTime;
+    }
+
+    virtual RealTime getAudioRecordLatency() {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) return m_jackDriver->getAudioRecordLatency();
+#endif
+        return RealTime::zeroTime;
+    }
+
+    virtual RealTime getInstrumentPlayLatency(InstrumentId id) {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) return m_jackDriver->getInstrumentPlayLatency(id);
+#endif
+        return RealTime::zeroTime;
+    }
+
+    virtual RealTime getMaximumPlayLatency() {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) return m_jackDriver->getMaximumPlayLatency();
+#endif
+        return RealTime::zeroTime;
+    }
+        
+
+    // Plugin instance management
+    //
+    virtual void setPluginInstance(InstrumentId id,
+                                   QString identifier,
+                                   int position) {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) m_jackDriver->setPluginInstance(id, identifier, position);
+#endif
+    }
+
+    virtual void removePluginInstance(InstrumentId id, int position) {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) m_jackDriver->removePluginInstance(id, position);
+#endif
+    }
+
+    // Remove all plugin instances
+    //
+    virtual void removePluginInstances() {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) m_jackDriver->removePluginInstances();
+#endif
+    }
+
+    virtual void setPluginInstancePortValue(InstrumentId id,
+                                            int position,
+                                            unsigned long portNumber,
+                                            float value) {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) m_jackDriver->setPluginInstancePortValue(id, position, portNumber, value);
+#endif
+    }
+
+    virtual float getPluginInstancePortValue(InstrumentId id,
+                                             int position,
+                                             unsigned long portNumber) {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) return m_jackDriver->getPluginInstancePortValue(id, position, portNumber);
+#endif
+        return 0;
+    }
+
+    virtual void setPluginInstanceBypass(InstrumentId id,
+                                         int position,
+                                         bool value) {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) m_jackDriver->setPluginInstanceBypass(id, position, value);
+#endif
+    }
+
+    virtual QStringList getPluginInstancePrograms(InstrumentId id,
+                                                  int position) {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) return m_jackDriver->getPluginInstancePrograms(id, position);
+#endif
+        return QStringList();
+    }
+
+    virtual QString getPluginInstanceProgram(InstrumentId id,
+                                             int position) {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) return m_jackDriver->getPluginInstanceProgram(id, position);
+#endif
+        return QString();
+    }
+
+    virtual QString getPluginInstanceProgram(InstrumentId id,
+                                             int position,
+                                             int bank,
+                                             int program) {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) return m_jackDriver->getPluginInstanceProgram(id, position, bank, program);
+#endif
+        return QString();
+    }
+
+    virtual unsigned long getPluginInstanceProgram(InstrumentId id,
+                                                   int position,
+                                                   QString name) {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) return m_jackDriver->getPluginInstanceProgram(id, position, name);
+#endif
+        return 0;
+    }
+    
+    virtual void setPluginInstanceProgram(InstrumentId id,
+                                          int position,
+                                          QString program) {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) m_jackDriver->setPluginInstanceProgram(id, position, program);
+#endif
+    }
+
+    virtual QString configurePlugin(InstrumentId id,
+                                    int position,
+                                    QString key,
+                                    QString value) {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) return m_jackDriver->configurePlugin(id, position, key, value);
+#endif
+        return QString();
+    }
+
+    virtual void setAudioBussLevels(int bussId,
+                                    float dB,
+                                    float pan) {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) m_jackDriver->setAudioBussLevels(bussId, dB, pan);
+#endif
+    }
+
+    virtual void setAudioInstrumentLevels(InstrumentId instrument,
+                                          float dB,
+                                          float pan) {
+#ifdef HAVE_LIBJACK
+        if (m_jackDriver) m_jackDriver->setAudioInstrumentLevels(instrument, dB, pan);
+#endif
+    }
+
+    virtual void claimUnwantedPlugin(void *plugin);
+    virtual void scavengePlugins();
+
+    virtual bool checkForNewClients();
+
+    virtual void setLoop(const RealTime &loopStart, const RealTime &loopEnd);
+
+    virtual void sleep(const RealTime &);
+
+    // ----------------------- End of Virtuals ----------------------
+
+    // Create and send an MMC command
+    //
+    void sendMMC(MidiByte deviceId,
+                 MidiByte instruction,
+                 bool isCommand,
+                 const std::string &data);
+
+    // Check whether the given event is an MMC command we need to act on
+    // (and if so act on it)
+    //
+    bool testForMMCSysex(const snd_seq_event_t *event);
+
+    // Create and enqueue a batch of MTC quarter-frame events
+    //
+    void insertMTCQFrames(RealTime sliceStart, RealTime sliceEnd);
+
+    // Create and enqueue an MTC full-frame system exclusive event
+    //
+    void insertMTCFullFrame(RealTime time);
+
+    // Parse and accept an incoming MTC quarter-frame event
+    //
+    void handleMTCQFrame(unsigned int data_byte, RealTime the_time);
+
+    // Check whether the given event is an MTC sysex we need to act on
+    // (and if so act on it)
+    //
+    bool testForMTCSysex(const snd_seq_event_t *event);
+
+    // Adjust the ALSA clock skew for MTC lock
+    //
+    void tweakSkewForMTC(int factor);
+
+    // Recalibrate internal MTC factors
+    //
+    void calibrateMTC();
+
+    // Send a System message straight away
+    //
+    void sendSystemDirect(MidiByte command, int *arg);
+
+    // Scheduled system message with arguments
+    //
+    void sendSystemQueued(MidiByte command,
+                          const std::string &args,
+                          const RealTime &time);
+
+    // Set the record device
+    //
+    void setRecordDevice(DeviceId id, bool connectAction);
+    void unsetRecordDevices();
+
+    virtual bool canReconnect(Device::DeviceType type);
+
+    virtual DeviceId addDevice(Device::DeviceType type,
+                               MidiDevice::DeviceDirection direction);
+    virtual void removeDevice(DeviceId id);
+    virtual void renameDevice(DeviceId id, QString name);
+
+    // Get available connections per device
+    // 
+    virtual unsigned int getConnections(Device::DeviceType type,
+                                        MidiDevice::DeviceDirection direction);
+    virtual QString getConnection(Device::DeviceType type,
+                                  MidiDevice::DeviceDirection direction,
+                                  unsigned int connectionNo);
+    virtual void setConnection(DeviceId deviceId, QString connection);
+    virtual void setPlausibleConnection(DeviceId deviceId, QString connection);
+
+    virtual unsigned int getTimers();
+    virtual QString getTimer(unsigned int);
+    virtual QString getCurrentTimer();
+    virtual void setCurrentTimer(QString);
+ 
+    virtual void getAudioInstrumentNumbers(InstrumentId &audioInstrumentBase,
+                                           int &audioInstrumentCount) {
+        audioInstrumentBase = AudioInstrumentBase;
+#ifdef HAVE_LIBJACK
+        audioInstrumentCount = AudioInstrumentCount;
+#else
+        audioInstrumentCount = 0;
+#endif
+    }
+ 
+    virtual void getSoftSynthInstrumentNumbers(InstrumentId &ssInstrumentBase,
+                                               int &ssInstrumentCount) {
+        ssInstrumentBase = SoftSynthInstrumentBase;
+#ifdef HAVE_DSSI
+        ssInstrumentCount = SoftSynthInstrumentCount;
+#else
+        ssInstrumentCount = 0;
+#endif
+    }
+
+    virtual QString getStatusLog();
+
+    // To be called regularly from JACK driver when idle
+    void checkTimerSync(size_t frames);
+
+    virtual void runTasks();
+
+    // Report a failure back to the GUI
+    //
+    virtual void reportFailure(MappedEvent::FailureCode code);
+
+protected:
+    typedef std::vector<AlsaPortDescription *> AlsaPortList;
+
+    ClientPortPair getFirstDestination(bool duplex);
+    ClientPortPair getPairForMappedInstrument(InstrumentId id);
+    int getOutputPortForMappedInstrument(InstrumentId id);
+    std::map<unsigned int, std::map<unsigned int, MappedEvent*> >  m_noteOnMap;
+
+    /**
+     * Bring m_alsaPorts up-to-date; if newPorts is non-null, also
+     * return the new ports (not previously in m_alsaPorts) through it
+     */
+    virtual void generatePortList(AlsaPortList *newPorts = 0);
+    virtual void generateInstruments();
+
+    virtual void generateTimerList();
+    virtual std::string getAutoTimer(bool &wantTimerChecks);
+
+    void addInstrumentsForDevice(MappedDevice *device);
+    MappedDevice *createMidiDevice(AlsaPortDescription *,
+                                   MidiDevice::DeviceDirection);
+
+    virtual void processMidiOut(const MappedComposition &mC,
+                                const RealTime &sliceStart,
+                                const RealTime &sliceEnd);
+
+    virtual void processSoftSynthEventOut(InstrumentId id,
+                                          const snd_seq_event_t *event,
+                                          bool now);
+
+    virtual bool isRecording(AlsaPortDescription *port);
+
+    virtual void processAudioQueue(bool /* now */) { }
+
+    virtual void setConnectionToDevice(MappedDevice &device, QString connection);
+    virtual void setConnectionToDevice(MappedDevice &device, QString connection,
+                                       const ClientPortPair &pair);
+
+private:
+    RealTime getAlsaTime();
+
+    // Locally convenient to control our devices
+    //
+    void sendDeviceController(DeviceId device,
+                              MidiByte byte1,
+                              MidiByte byte2);
+                              
+    int checkAlsaError(int rc, const char *message);
+
+    AlsaPortList m_alsaPorts;
+
+    // ALSA MIDI/Sequencer stuff
+    //
+    snd_seq_t                   *m_midiHandle;
+    int                          m_client;
+
+    int                          m_inputPort;
+    
+    typedef std::map<DeviceId, int> DeviceIntMap;
+    DeviceIntMap                 m_outputPorts;
+
+    int                          m_syncOutputPort;
+    int                          m_controllerPort;
+
+    int                          m_queue;
+    int                          m_maxClients;
+    int                          m_maxPorts;
+    int                          m_maxQueues;
+
+    // Because this can fail even if the driver's up (if
+    // another service is using the port say)
+    //
+    bool                         m_midiInputPortConnected;
+
+    bool                         m_midiSyncAutoConnect;
+
+    RealTime                     m_alsaPlayStartTime;
+    RealTime                     m_alsaRecordStartTime;
+
+    RealTime                     m_loopStartTime;
+    RealTime                     m_loopEndTime;
+
+    // MIDI Time Code handling:
+
+    unsigned int                 m_eat_mtc;
+    // Received/emitted MTC data breakdown:
+    RealTime                     m_mtcReceiveTime;
+    RealTime                     m_mtcEncodedTime;
+    int                          m_mtcFrames;
+    int                          m_mtcSeconds;
+    int                          m_mtcMinutes;
+    int                          m_mtcHours;
+    int                          m_mtcSMPTEType;
+
+    // Calculated MTC factors:
+    int                          m_mtcFirstTime;
+    RealTime                     m_mtcLastEncoded;
+    RealTime                     m_mtcLastReceive;
+    long long int                m_mtcSigmaE;
+    long long int                m_mtcSigmaC;
+    unsigned int                 m_mtcSkew;
+
+    bool                         m_looping;
+
+    bool                         m_haveShutdown;
+
+#ifdef HAVE_LIBJACK
+    JackDriver *m_jackDriver;
+#endif
+
+    Scavenger<RunnablePluginInstance> m_pluginScavenger;
+
+    //!!! -- hoist to SoundDriver w/setter?
+    typedef std::set<InstrumentId> InstrumentSet;
+    InstrumentSet m_recordingInstruments;
+
+    typedef std::map<DeviceId, ClientPortPair> DevicePortMap;
+    DevicePortMap m_devicePortMap;
+
+    typedef std::map<ClientPortPair, DeviceId> PortDeviceMap;
+    PortDeviceMap m_suspendedPortMap;
+
+    std::string getPortName(ClientPortPair port);
+    ClientPortPair getPortByName(std::string name);
+
+    DeviceId getSpareDeviceId();
+
+    struct AlsaTimerInfo {
+        int clas;
+        int sclas;
+        int card;
+        int device;
+        int subdevice;
+        std::string name;
+        long resolution;
+    };
+    std::vector<AlsaTimerInfo> m_timers;
+    std::string m_currentTimer;
+
+    // This auxiliary queue is here as a hack, to avoid stuck notes if
+    // resetting playback while a note-off is currently in the ALSA
+    // queue.  When playback is reset by ffwd or rewind etc, we drop
+    // all the queued events (which is generally what is desired,
+    // except for note offs) and reset the queue timer (so the note
+    // offs would have the wrong time stamps even if we hadn't dropped
+    // them).  Thus, we need to re-send any recent note offs before
+    // continuing.  This queue records which note offs have been
+    // added to the ALSA queue recently.
+    //
+    NoteOffQueue m_recentNoteOffs;
+    void pushRecentNoteOffs(); // move from recent to normal queue after reset
+    void cropRecentNoteOffs(const RealTime &t); // remove old note offs
+    void weedRecentNoteOffs(unsigned int pitch, MidiByte channel,
+			    InstrumentId instrument); // on subsequent note on
+
+    bool m_queueRunning;
+    
+    bool m_portCheckNeeded;
+
+    enum { NeedNoJackStart, NeedJackReposition, NeedJackStart } m_needJackStart;
+    
+    bool m_doTimerChecks;
+    bool m_firstTimerCheck;
+    double m_timerRatio;
+    bool m_timerRatioCalculated;
+
+    std::string getAlsaModuleVersionString();
+    std::string getKernelVersionString();
+    void extractVersion(std::string vstr, int &major, int &minor, int &subminor, std::string &suffix);
+    bool versionIsAtLeast(std::string vstr, int major, int minor, int subminor);
+};
+
+}
+
+#endif // HAVE_ALSA
+
+#endif // _ALSADRIVER_H_
+
diff --git a/src/sound/AlsaPort.cpp b/src/sound/AlsaPort.cpp
new file mode 100644
index 0000000..dc12610
--- /dev/null
+++ b/src/sound/AlsaPort.cpp
@@ -0,0 +1,192 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+ 
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+ 
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+ 
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include "AlsaPort.h"
+
+#include <iostream>
+#include <cstdlib>
+#include <cstdio>
+
+#ifdef HAVE_ALSA
+
+// ALSA
+#include <alsa/asoundlib.h>
+#include <alsa/seq_event.h>
+#include <alsa/version.h>
+
+#include "MappedInstrument.h"
+#include "Midi.h"
+#include "WAVAudioFile.h"
+#include "MappedStudio.h"
+#include "misc/Strings.h"
+
+#ifdef HAVE_LIBJACK
+#include <jack/types.h>
+#include <unistd.h> // for usleep
+#include <cmath>
+#endif
+
+namespace Rosegarden
+{
+
+AlsaPortDescription::AlsaPortDescription(Instrument::InstrumentType type,
+        const std::string &name,
+        int client,
+        int port,
+        unsigned int clientType,
+        unsigned int portType,
+        unsigned int capability,
+        PortDirection direction):
+        m_type(type),
+        m_name(name),
+        m_client(client),
+        m_port(port),
+        m_clientType(clientType),
+        m_portType(portType),
+        m_capability(capability),
+        m_direction(direction)
+{}
+
+
+bool
+AlsaPortCmp::operator()(AlsaPortDescription *a1, AlsaPortDescription *a2)
+{
+    // Ordering for ALSA ports in the list:
+    //
+    // * Hardware ports (client id 64-127) sorted by direction
+    //   (write, duplex, read) then client id then port id
+    //
+    // * Software ports (client id 128+) sorted by client id
+    //   then port id
+    //
+    // * System ports (client id 0-63) sorted by client id then
+    //   port id
+
+
+    // See comment in AlsaDriver::createMidiDevice -- the client
+    // numbering scheme changed in ALSA driver 1.0.11rc1.
+    // We now order:
+    //
+    // * Write-only software ports (client id 128+) sorted by client
+    //   id then port id
+    //
+    // * Probable hardware ports (client id 16-127) sorted by
+    //   direction (write, duplex, read) then client id (64+
+    //   preferred) then port id
+    //
+    // * Read-write or read-only software ports (client id 128+)
+    //   sorted by client id then port id
+    //
+    // * System ports (client id 0-15) sorted by client id then
+    //   port id
+    //
+    // It's necessary to handle software ports ahead of
+    // hardware/system ports, because we want to keep all the hardware
+    // ports together (we don't want to change the priority of a
+    // hardware port relative to a software port based on its client
+    // ID) and we can't know for sure whether the 16-63 range are
+    // hardware or system ports.
+
+    enum Category {
+        WRITE_ONLY_SOFTWARE,
+        HARDWARE_PROBABLY,
+        MIXED_SOFTWARE,
+        SYSTEM
+    };
+
+    bool oldScheme = (SND_LIB_MAJOR == 0 ||
+                      (SND_LIB_MAJOR == 1 &&
+                       SND_LIB_MINOR == 0 &&
+                       SND_LIB_SUBMINOR < 11));
+
+    Category a1cat;
+    if (a1->m_client < 16)
+        a1cat = SYSTEM;
+    else if (oldScheme && (a1->m_client < 64))
+        a1cat = SYSTEM;
+    else if (a1->m_client < 128)
+        a1cat = HARDWARE_PROBABLY;
+    else
+        a1cat = MIXED_SOFTWARE;
+
+    if (a1cat == MIXED_SOFTWARE) {
+        if (a1->m_direction == WriteOnly)
+            a1cat = WRITE_ONLY_SOFTWARE;
+    }
+
+    Category a2cat;
+    if (a2->m_client < 16)
+        a2cat = SYSTEM;
+    else if (oldScheme && (a2->m_client < 64))
+        a2cat = SYSTEM;
+    else if (a2->m_client < 128)
+        a2cat = HARDWARE_PROBABLY;
+    else
+        a2cat = MIXED_SOFTWARE;
+
+    if (a2cat == MIXED_SOFTWARE) {
+        if (a2->m_direction == WriteOnly)
+            a2cat = WRITE_ONLY_SOFTWARE;
+    }
+
+    if (a1cat != a2cat)
+        return int(a1cat) < int(a2cat);
+
+    if (a1cat == HARDWARE_PROBABLY) {
+
+        if (a1->m_direction == WriteOnly) {
+            if (a2->m_direction != WriteOnly)
+                return true;
+        } else if (a1->m_direction == Duplex) {
+            if (a2->m_direction == ReadOnly)
+                return true;
+        }
+
+        int c1 = a1->m_client;
+        int c2 = a2->m_client;
+        if (c1 < 64)
+            c1 += 1000;
+        if (c2 < 64)
+            c2 += 1000;
+        if (c1 != c2)
+            return c1 < c2;
+
+    } else if (a1cat == SYSTEM) {
+
+        int c1 = a1->m_client;
+        int c2 = a2->m_client;
+        if (c1 < 16)
+            c1 += 1000;
+        if (c2 < 16)
+            c2 += 1000;
+        if (c1 != c2)
+            return c1 < c2;
+    }
+
+    if (a1->m_client != a2->m_client) {
+        return a1->m_client < a2->m_client;
+    } else {
+        return a1->m_port < a2->m_port;
+    }
+}
+
+}
+
+#endif // HAVE_ALSA
diff --git a/src/sound/AlsaPort.h b/src/sound/AlsaPort.h
new file mode 100644
index 0000000..39cf009
--- /dev/null
+++ b/src/sound/AlsaPort.h
@@ -0,0 +1,86 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include <vector>
+#include <set>
+#include "Instrument.h"
+#include "MappedCommon.h"
+
+#ifndef _ALSAPORT_H_
+#define _ALSAPORT_H_
+
+#ifdef HAVE_ALSA
+#include <alsa/asoundlib.h> // ALSA
+
+namespace Rosegarden
+{
+
+typedef std::pair<int, int> ClientPortPair;
+
+// Use this to hold all client information so that we can sort it
+// before generating devices - we want to put non-duplex devices
+// at the front of any device list (makes thing much easier at the
+// GUI and we already have some backwards compatability issues with
+// this).
+//
+class AlsaPortDescription
+{
+public:
+    AlsaPortDescription(Instrument::InstrumentType type,
+                        const std::string &name,
+                        int client,
+                        int port,
+                        unsigned int clientType,
+                        unsigned int portType,
+                        unsigned int capability,
+                        PortDirection direction);
+
+    Instrument::InstrumentType m_type;
+    std::string                m_name;
+    int                        m_client;
+    int                        m_port;
+    unsigned int               m_clientType;
+    unsigned int               m_portType;
+    unsigned int               m_capability;
+    PortDirection              m_direction; // or can deduce from capability
+
+    bool isReadable()  { return m_direction == ReadOnly ||
+                                m_direction == Duplex; }
+
+    bool isWriteable() { return m_direction == WriteOnly ||
+                                m_direction == Duplex; }
+
+};
+
+// Sort by checking direction
+//
+struct AlsaPortCmp
+{
+    bool operator()(AlsaPortDescription *a1,
+                    AlsaPortDescription *a2);
+};
+
+
+}
+
+#endif // HAVE_ALSA
+
+#endif // _RG_ALSA_PORT_H_
+
diff --git a/src/sound/AudioCache.cpp b/src/sound/AudioCache.cpp
new file mode 100644
index 0000000..e1e9000
--- /dev/null
+++ b/src/sound/AudioCache.cpp
@@ -0,0 +1,139 @@
+
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "AudioCache.h"
+#include <iostream>
+
+//#define DEBUG_AUDIO_CACHE 1
+
+namespace Rosegarden
+{
+
+AudioCache::~AudioCache()
+{
+    clear();
+}
+
+bool
+AudioCache::has(void *index)
+{
+    return m_cache.find(index) != m_cache.end();
+}
+
+float **
+AudioCache::getData(void *index, size_t &channels, size_t &frames)
+{
+    if (m_cache.find(index) == m_cache.end())
+        return 0;
+    CacheRec *rec = m_cache[index];
+    channels = rec->channels;
+    frames = rec->nframes;
+    return rec->data;
+}
+
+void
+AudioCache::addData(void *index, size_t channels, size_t nframes, float **data)
+{
+#ifdef DEBUG_AUDIO_CACHE
+    std::cerr << "AudioCache::addData(" << index << ")" << std::endl;
+#endif
+
+    if (m_cache.find(index) != m_cache.end()) {
+        std::cerr << "WARNING: AudioCache::addData(" << index << ", "
+        << channels << ", " << nframes
+        << ": already cached" << std::endl;
+        return ;
+    }
+
+    m_cache[index] = new CacheRec(data, channels, nframes);
+}
+
+void
+AudioCache::incrementReference(void *index)
+{
+    if (m_cache.find(index) == m_cache.end()) {
+        std::cerr << "WARNING: AudioCache::incrementReference(" << index
+        << "): not found" << std::endl;
+        return ;
+    }
+    ++m_cache[index]->refCount;
+
+#ifdef DEBUG_AUDIO_CACHE
+
+    std::cerr << "AudioCache::incrementReference(" << index << ") [to " << (m_cache[index]->refCount) << "]" << std::endl;
+#endif
+}
+
+void
+AudioCache::decrementReference(void *index)
+{
+    std::map<void *, CacheRec *>::iterator i = m_cache.find(index);
+
+    if (i == m_cache.end()) {
+        std::cerr << "WARNING: AudioCache::decrementReference(" << index
+        << "): not found" << std::endl;
+        return ;
+    }
+    if (i->second->refCount <= 1) {
+        delete i->second;
+        m_cache.erase(i);
+#ifdef DEBUG_AUDIO_CACHE
+
+        std::cerr << "AudioCache::decrementReference(" << index << ") [deleting]" << std::endl;
+#endif
+
+    } else {
+        --i->second->refCount;
+#ifdef DEBUG_AUDIO_CACHE
+
+        std::cerr << "AudioCache::decrementReference(" << index << ") [to " << (m_cache[index]->refCount) << "]" << std::endl;
+#endif
+
+    }
+}
+
+void
+AudioCache::clear()
+{
+#ifdef DEBUG_AUDIO_CACHE
+    std::cerr << "AudioCache::clear()" << std::endl;
+#endif
+
+    for (std::map<void *, CacheRec *>::iterator i = m_cache.begin();
+            i != m_cache.end(); ++i) {
+        if (i->second->refCount > 0) {
+            std::cerr << "WARNING: AudioCache::clear: deleting cached data with refCount " << i->second->refCount << std::endl;
+        }
+    }
+    m_cache.clear();
+}
+
+AudioCache::CacheRec::~CacheRec()
+{
+    for (size_t j = 0; j < channels; ++j)
+        delete[] data[j];
+    delete[] data;
+}
+
+}
+
+
diff --git a/src/sound/AudioCache.h b/src/sound/AudioCache.h
new file mode 100644
index 0000000..6dd55ff
--- /dev/null
+++ b/src/sound/AudioCache.h
@@ -0,0 +1,98 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _AUDIO_CACHE_H_
+#define _AUDIO_CACHE_H_
+
+#include <map>
+
+namespace Rosegarden
+{
+
+/**
+ * A simple cache for smallish bits of audio data, indexed by some
+ * opaque pointer type.  (The PlayableAudioFile uses this with an
+ * AudioFile* index type, for example.)  With reference counting.
+ */
+
+class AudioCache
+{
+public:
+    AudioCache() { }
+    virtual ~AudioCache();
+
+    /**
+     * Look some audio data up in the cache and report whether it
+     * exists.
+     */
+    bool has(void *index);
+
+    /**
+     * Look some audio data up in the cache and return it if it
+     * exists, returning the number of channels in channels, the frame
+     * count in frames, and one pointer per channel to samples in the
+     * return value.  Return 0 if the data is not in cache.  Does not
+     * affect the reference count.  Ownership of the returned data
+     * remains with the cache object.  You should not call this unless
+     * you have already called incrementReference (or addData) to
+     * register your interest.
+     */
+    float **getData(void *index, size_t &channels, size_t &frames);
+
+    /**
+     * Add a piece of data to the cache, and increment the reference
+     * count for that data (to 1).  Ownership of the data is passed
+     * to the cache, which will delete it with delete[] when done.
+     */
+    void addData(void *index, size_t channels, size_t nframes, float **data);
+
+    /**
+     * Increment the reference count for a given piece of data.
+     */
+    void incrementReference(void *index);
+
+    /**
+     * Decrement the reference count for a given piece of data,
+     * and delete the data from the cache if the reference count has
+     * reached zero.
+     */
+    void decrementReference(void *index);
+
+protected:
+    void clear();
+
+    struct CacheRec {
+        CacheRec() : data(0), channels(0), nframes(0), refCount(0) { }
+        CacheRec(float **d, size_t c, size_t n) :
+            data(d), channels(c), nframes(n), refCount(1) { }
+        ~CacheRec();
+        float **data;
+        size_t channels;
+        size_t nframes;
+        int refCount;
+    };
+
+    std::map<void *, CacheRec *> m_cache;
+};
+
+}
+
+#endif
diff --git a/src/sound/AudioFile.cpp b/src/sound/AudioFile.cpp
new file mode 100644
index 0000000..6eba15a
--- /dev/null
+++ b/src/sound/AudioFile.cpp
@@ -0,0 +1,75 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+ 
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+ 
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+ 
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+
+#include "AudioFile.h"
+
+namespace Rosegarden
+
+{
+
+AudioFile::AudioFile(unsigned int id,
+                     const std::string &name,
+                     const std::string &fileName):
+        SoundFile(fileName),
+        m_type(UNKNOWN),
+        m_id(id),
+        m_name(name),
+        m_bitsPerSample(0),
+        m_sampleRate(0),
+        m_channels(0),
+        m_dataChunkIndex( -1)
+{
+    m_fileInfo = new QFileInfo(QString(fileName.c_str()));
+}
+
+AudioFile::AudioFile(const std::string &fileName,
+                     unsigned int channels = 1,
+                     unsigned int sampleRate = 48000,
+                     unsigned int bitsPerSample = 16):
+        SoundFile(fileName),
+        m_type(UNKNOWN),
+        m_id(0),
+        m_name(""),
+        m_bitsPerSample(bitsPerSample),
+        m_sampleRate(sampleRate),
+        m_channels(channels),
+        m_dataChunkIndex( -1)
+{
+    m_fileInfo = new QFileInfo(QString(fileName.c_str()));
+}
+
+AudioFile::~AudioFile()
+{
+    delete m_fileInfo;
+}
+
+QDateTime
+AudioFile::getModificationDateTime()
+{
+    if (m_fileInfo)
+        return m_fileInfo->lastModified();
+    else
+        return QDateTime();
+}
+
+
+}
+
diff --git a/src/sound/AudioFile.h b/src/sound/AudioFile.h
new file mode 100644
index 0000000..1acbe61
--- /dev/null
+++ b/src/sound/AudioFile.h
@@ -0,0 +1,216 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+
+#ifndef _AUDIOFILE_H_
+#define _AUDIOFILE_H_
+
+#include <string>
+#include <vector>
+#include <cmath>
+
+#include <qfileinfo.h>
+
+#include "SoundFile.h"
+#include "RealTime.h"
+
+// An AudioFile maintains information pertaining to an audio sample.
+// This is an abstract base class from which we derive our actual
+// AudioFile types - WAV, BWF, AIFF etc.
+//
+//
+
+namespace Rosegarden
+{
+
+typedef unsigned int AudioFileId;
+
+// The different types of audio file we support.
+//
+typedef enum
+{
+
+    UNKNOWN, // not yet known
+    WAV,     // RIFF (Resource Interchange File Format) wav file 
+    BWF,     // RIFF Broadcast Wave File
+    AIFF,    // Audio Interchange File Format
+    MP3
+
+} AudioFileType;
+
+class AudioFile : public SoundFile
+{
+public:
+    // The "read" constructor - open a file
+    // an assign a name and id to it.
+    //
+    AudioFile(AudioFileId id,
+              const std::string &name,
+              const std::string &fileName);
+
+    // The "write" constructor - we only need to
+    // specify a filename and some parameters and
+    // then write it out.
+    //
+    AudioFile(const std::string &fileName,
+              unsigned int channels,
+              unsigned int sampleRate,
+              unsigned int bitsPerSample);
+
+    ~AudioFile();
+
+    // Id of this audio file (used by AudioFileManager)
+    //
+    void setId(AudioFileId id) { m_id = id; }
+    AudioFileId getId() const { return m_id; }
+
+    // Name of this sample - in addition to a filename
+    //
+    void setName(const std::string &name) { m_name = name; }
+    std::string getName() const { return m_name; }
+
+    // Used for waveform interpolation at a point
+    //
+    float sinc(float value) { return sin(M_PI * value)/ M_PI * value; }
+
+    // Audio file identifier - set in constructor of file type
+    //
+    AudioFileType getType() const { return m_type; }
+
+    unsigned int getBitsPerSample() const { return m_bitsPerSample; }
+    unsigned int getSampleRate() const { return m_sampleRate; }
+    unsigned int getChannels() const { return m_channels; }
+    
+    // We must continue our main control abstract methods from SoundFile
+    // and add our own close() method that will add any relevant header
+    // information to an audio file that has been written and is now
+    // being closed.
+    //
+    virtual bool open() = 0;
+    virtual bool write() = 0;
+    virtual void close() = 0;
+
+    // Show the information we have on this file
+    //
+    virtual void printStats() = 0;
+
+    // Move file pointer to relative time in data chunk - shouldn't be
+    // less than zero.  Returns true if the scan time was valid and
+    // successful.  Need two interfaces because when playing we use an
+    // external file handle (one per playback instance - PlayableAudioFile)
+    // 
+    virtual bool scanTo(const RealTime &time) = 0;
+    virtual bool scanTo(std::ifstream *file, const RealTime &time) = 0;
+
+    // Scan forward in a file by a certain amount of time - same
+    // double interface (simple one for peak file generation, other
+    // for playback).
+    //
+    virtual bool scanForward(const RealTime &time) = 0;
+    virtual bool scanForward(std::ifstream *file, const RealTime &time) = 0;
+
+    // Return a number of samples - caller will have to
+    // de-interleave n-channel samples themselves.
+    //
+    virtual std::string getSampleFrames(std::ifstream *file,
+                                        unsigned int frames) = 0;
+
+    // Return a number of samples - caller will have to
+    // de-interleave n-channel samples themselves.  Place
+    // results in buf, return actual number of frames read.
+    //
+    virtual unsigned int getSampleFrames(std::ifstream *file,
+                                         char *buf,
+                                         unsigned int frames) = 0;
+
+    // Return a number of (possibly) interleaved samples
+    // over a time slice from current file pointer position.
+    //
+    virtual std::string getSampleFrameSlice(std::ifstream *file,
+                                            const RealTime &time) = 0;
+
+    // Append a string of samples to an already open (for writing)
+    // audio file.  Caller must have interleaved samples etc.
+    //
+    virtual bool appendSamples(const std::string &buffer) = 0;
+
+    // Append a string of samples to an already open (for writing)
+    // audio file.  Caller must have interleaved samples etc.
+    //
+    virtual bool appendSamples(const char *buffer, unsigned int frames) = 0;
+
+    // Get the length of the sample file in RealTime
+    //
+    virtual RealTime getLength() = 0;
+
+    // Offset to start of sample data
+    //
+    virtual std::streampos getDataOffset() = 0;
+
+    // Return the peak file name
+    //
+    virtual std::string getPeakFilename() = 0;
+
+    // Return the modification timestamp
+    //
+    QDateTime getModificationDateTime();
+
+    // Implement in actual file type
+    //
+    virtual unsigned int getBytesPerFrame() = 0;
+
+    // Decode and de-interleave the given samples that were retrieved
+    // from this file or another with the same format as it.  Place
+    // the results in the given float buffer.  Return true for
+    // success.  This function does crappy resampling if necessary.
+    // 
+    virtual bool decode(const unsigned char *sourceData,
+                        size_t sourceBytes,
+                        size_t targetSampleRate,
+                        size_t targetChannels,
+                        size_t targetFrames,
+                        std::vector<float *> &targetData,
+                        bool addToResultBuffers = false) = 0;
+
+protected:
+
+    AudioFileType  m_type;   // AudioFile type
+    AudioFileId    m_id;     // AudioFile ID
+    std::string    m_name;   // AudioFile name (not filename)
+
+    unsigned int   m_bitsPerSample;
+    unsigned int   m_sampleRate;
+    unsigned int   m_channels;
+
+    // How many bytes do we read before we get to the data?
+    // Could be huge so we make it a long long. -1 means it
+    // hasn't been set yet.
+    //
+    long long      m_dataChunkIndex;
+
+    QFileInfo     *m_fileInfo;
+
+};
+
+}
+
+
+#endif // _AUDIOFILE_H_
diff --git a/src/sound/AudioFileManager.cpp b/src/sound/AudioFileManager.cpp
new file mode 100644
index 0000000..93be26c
--- /dev/null
+++ b/src/sound/AudioFileManager.cpp
@@ -0,0 +1,1257 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+ 
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+ 
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+ 
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+
+#include <iostream>
+#include <kapplication.h>
+#include <fstream>
+#include <string>
+#include <dirent.h> // for new recording file
+#include <cstdio>   // sprintf
+#include <cstdlib>
+#include <pthread.h>
+
+#if (__GNUC__ < 3)
+#include <strstream>
+#define stringstream strstream
+#else
+#include <sstream>
+#endif
+
+#include <kapp.h>
+#include <klocale.h>
+#include <kprocess.h>
+#include <kio/netaccess.h>
+#include <kmessagebox.h>
+
+#include <qpixmap.h>
+#include <qpainter.h>
+#include <qdatetime.h>
+#include <qfile.h>
+
+#include "AudioFile.h"
+#include "AudioFileManager.h"
+#include "WAVAudioFile.h"
+#include "BWFAudioFile.h"
+#include "MP3AudioFile.h"
+#include "misc/Strings.h"
+
+namespace Rosegarden
+{
+
+static pthread_mutex_t _audioFileManagerLock;
+
+class MutexLock
+{
+public:
+    MutexLock(pthread_mutex_t *mutex) : m_mutex(mutex)
+    {
+        pthread_mutex_lock(m_mutex);
+    }
+    ~MutexLock()
+    {
+        pthread_mutex_unlock(m_mutex);
+    }
+private:
+    pthread_mutex_t *m_mutex;
+};
+
+AudioFileManager::AudioFileManager() :
+    m_importProcess(0),
+    m_expectedSampleRate(0)
+{
+    pthread_mutexattr_t attr;
+    pthread_mutexattr_init(&attr);
+#ifdef HAVE_PTHREAD_MUTEX_RECURSIVE
+
+    pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE);
+#else
+#ifdef PTHREAD_MUTEX_RECURSIVE
+
+    pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE);
+#else
+
+    pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE_NP);
+#endif
+#endif
+
+    pthread_mutex_init(&_audioFileManagerLock, &attr);
+
+    // Set this through the set method so that the tilde gets
+    // shaken out.
+    //
+    setAudioPath("~/rosegarden");
+
+    // Retransmit progress
+    //
+    connect(&m_peakManager, SIGNAL(setProgress(int)),
+            this, SIGNAL(setProgress(int)));
+}
+
+AudioFileManager::~AudioFileManager()
+{
+    clear();
+}
+
+// Add a file from an absolute path
+//
+AudioFileId
+AudioFileManager::addFile(const std::string &filePath)
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    QString ext;
+
+    if (filePath.length() > 3) {
+	ext = QString(filePath.substr(filePath.length() - 3, 3).c_str()).lower();
+    }
+
+    // Check for file existing already in manager by path
+    //
+    int check = fileExists(filePath);
+    if (check != -1) {
+        return AudioFileId(check);
+    }
+
+    // prepare for audio file
+    AudioFile *aF = 0;
+    AudioFileId id = getFirstUnusedID();
+
+    if (ext == "wav") {
+        // identify file type
+        AudioFileType subType = RIFFAudioFile::identifySubType(filePath);
+
+        if (subType == BWF) {
+#ifdef DEBUG_AUDIOFILEMANAGER
+            std::cout << "FOUND BWF" << std::endl;
+#endif
+
+            try {
+                aF = new BWFAudioFile(id, getShortFilename(filePath), filePath);
+            } catch (SoundFile::BadSoundFileException e) {
+                delete aF;
+                throw BadAudioPathException(e);
+            }
+        } else if (subType == WAV) {
+            try {
+                aF = new WAVAudioFile(id, getShortFilename(filePath), filePath);
+            } catch (SoundFile::BadSoundFileException e) {
+                delete aF;
+                throw BadAudioPathException(e);
+            }
+        }
+
+        // Ensure we have a valid file handle
+        //
+        if (aF == 0) {
+            std::cerr << "AudioFileManager: Unknown WAV audio file subtype in " << filePath << std::endl;
+            throw BadAudioPathException(filePath, __FILE__, __LINE__);
+        }
+
+        // Add file type on extension
+        try {
+            if (aF->open() == false) {
+                delete aF;
+                std::cerr << "AudioFileManager: Malformed audio file in " << filePath << std::endl;
+                throw BadAudioPathException(filePath, __FILE__, __LINE__);
+            }
+        } catch (SoundFile::BadSoundFileException e) {
+            delete aF;
+            throw BadAudioPathException(e);
+        }
+    }
+#ifdef HAVE_LIBMAD
+    else if (ext == "mp3") {
+        try {
+            aF = new MP3AudioFile(id, getShortFilename(filePath), filePath);
+
+            if (aF->open() == false) {
+                delete aF;
+                std::cerr << "AudioFileManager: Malformed mp3 audio file in " << filePath << std::endl;
+                throw BadAudioPathException(filePath, __FILE__, __LINE__);
+            }
+        } catch (SoundFile::BadSoundFileException e) {
+            delete aF;
+            throw BadAudioPathException(e);
+        }
+    }
+#endif // HAVE_LIBMAD
+    else {
+        std::cerr << "AudioFileManager: Unsupported audio file extension in " << filePath << std::endl;
+        throw BadAudioPathException(filePath, __FILE__, __LINE__);
+    }
+
+    if (aF) {
+        m_audioFiles.push_back(aF);
+        return id;
+    }
+
+    return 0;
+}
+
+// Convert long filename to shorter version
+std::string
+AudioFileManager::getShortFilename(const std::string &fileName)
+{
+    std::string rS = fileName;
+    unsigned int pos = rS.find_last_of("/");
+
+    if (pos > 0 && ( pos + 1 ) < rS.length())
+        rS = rS.substr(pos + 1, rS.length());
+
+    return rS;
+}
+
+// Turn a long path into a directory ending with a slash
+//
+std::string
+AudioFileManager::getDirectory(const std::string &path)
+{
+    std::string rS = path;
+    unsigned int pos = rS.find_last_of("/");
+
+    if (pos > 0 && ( pos + 1 ) < rS.length())
+        rS = rS.substr(0, pos + 1);
+
+    return rS;
+}
+
+
+// Create a new AudioFile with unique ID and label - insert from
+// our RG4 file
+//
+AudioFileId
+AudioFileManager::insertFile(const std::string &name,
+                             const std::string &fileName)
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    // first try to expand any beginning tilde
+    //
+    std::string foundFileName = substituteTildeForHome(fileName);
+
+    // If we've expanded and we can't find the file
+    // then try to find it in audio file directory.
+    //
+    QFileInfo info(foundFileName.c_str());
+    if (!info.exists())
+        foundFileName = getFileInPath(foundFileName);
+
+#ifdef DEBUG_AUDIOFILEMANAGER_INSERT_FILE
+
+    std::cout << "AudioFileManager::insertFile - "
+    << "expanded fileName = \""
+    << foundFileName << "\"" << std::endl;
+#endif
+
+    // bail if we haven't found any reasonable filename
+    if (foundFileName == "")
+        return false;
+
+    AudioFileId id = getFirstUnusedID();
+
+    WAVAudioFile *aF = 0;
+
+    try {
+
+        aF = new WAVAudioFile(id, name, foundFileName);
+
+        // if we don't recognise the file then don't insert it
+        //
+        if (aF->open() == false) {
+            delete aF;
+            std::cerr << "AudioFileManager::insertFile - don't recognise file type in " << foundFileName << std::endl;
+            throw BadAudioPathException(foundFileName, __FILE__, __LINE__);
+        }
+        m_audioFiles.push_back(aF);
+
+    } catch (SoundFile::BadSoundFileException e) {
+        delete aF;
+        throw BadAudioPathException(e);
+    }
+
+    return id;
+}
+
+
+bool
+AudioFileManager::removeFile(AudioFileId id)
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    std::vector<AudioFile*>::iterator it;
+
+    for (it = m_audioFiles.begin();
+            it != m_audioFiles.end();
+            ++it) {
+        if ((*it)->getId() == id) {
+            m_peakManager.removeAudioFile(*it);
+            m_recordedAudioFiles.erase(*it);
+	    m_derivedAudioFiles.erase(*it);
+            delete(*it);
+            m_audioFiles.erase(it);
+            return true;
+        }
+    }
+
+    return false;
+}
+
+AudioFileId
+AudioFileManager::getFirstUnusedID()
+{
+    AudioFileId rI = 0;
+
+    if (m_audioFiles.size() == 0)
+        return rI;
+
+    std::vector<AudioFile*>::iterator it;
+
+    for (it = m_audioFiles.begin();
+            it != m_audioFiles.end();
+            ++it) {
+        if (rI < (*it)->getId())
+            rI = (*it)->getId();
+    }
+
+    rI++;
+
+    return rI;
+}
+
+bool
+AudioFileManager::insertFile(const std::string &name,
+                             const std::string &fileName,
+                             AudioFileId id)
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    // first try to expany any beginning tilde
+    std::string foundFileName = substituteTildeForHome(fileName);
+
+    // If we've expanded and we can't find the file
+    // then try to find it in audio file directory.
+    //
+    QFileInfo info(foundFileName.c_str());
+    if (!info.exists())
+        foundFileName = getFileInPath(foundFileName);
+
+#ifdef DEBUG_AUDIOFILEMANAGER_INSERT_FILE
+
+    std::cout << "AudioFileManager::insertFile - "
+    << "expanded fileName = \""
+    << foundFileName << "\"" << std::endl;
+#endif
+
+    // If no joy here then we can't find this file
+    if (foundFileName == "")
+        return false;
+
+    // make sure we don't have a file of this ID hanging around already
+    removeFile(id);
+
+    // and insert
+    WAVAudioFile *aF = 0;
+
+    try {
+
+        aF = new WAVAudioFile(id, name, foundFileName);
+
+        // Test the file
+        if (aF->open() == false) {
+            delete aF;
+            return false;
+        }
+
+        m_audioFiles.push_back(aF);
+
+    } catch (SoundFile::BadSoundFileException e) {
+        delete aF;
+        throw BadAudioPathException(e);
+    }
+
+    return true;
+}
+
+// Add a given path to our sample search path
+//
+void
+AudioFileManager::setAudioPath(const std::string &path)
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    std::string hPath = path;
+
+    // add a trailing / if we don't have one
+    //
+    if (hPath[hPath.size() - 1] != '/')
+        hPath += std::string("/");
+
+    // get the home directory
+    if (hPath[0] == '~') {
+        hPath.erase(0, 1);
+        hPath = std::string(getenv("HOME")) + hPath;
+    }
+
+    m_audioPath = hPath;
+
+}
+
+void
+AudioFileManager::testAudioPath() throw (BadAudioPathException)
+{
+    QFileInfo info(m_audioPath.c_str());
+    if (!(info.exists() && info.isDir() && !info.isRelative() &&
+            info.isWritable() && info.isReadable()))
+        throw BadAudioPathException(m_audioPath.data());
+}
+
+
+// See if we can find a given file in our search path
+// return the first occurence of a match or the empty
+// std::string if no match.
+//
+std::string
+AudioFileManager::getFileInPath(const std::string &file)
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    QFileInfo info(file.c_str());
+
+    if (info.exists())
+        return file;
+
+    // Build the search filename from the audio path and
+    // the file basename.
+    //
+    QString searchFile = QString(m_audioPath.c_str()) + info.fileName();
+    QFileInfo searchInfo(searchFile);
+
+    if (searchInfo.exists())
+        return searchFile.latin1();
+
+    std::cout << "AudioFileManager::getFileInPath - "
+    << "searchInfo = " << searchFile << std::endl;
+
+    return "";
+}
+
+
+// Check for file path existence
+//
+int
+AudioFileManager::fileExists(const std::string &path)
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    std::vector<AudioFile*>::iterator it;
+
+    for (it = m_audioFiles.begin();
+            it != m_audioFiles.end();
+            ++it) {
+        if ((*it)->getFilename() == path)
+            return (*it)->getId();
+    }
+
+    return -1;
+
+}
+
+// Does a specific file id exist on the manager?
+//
+bool
+AudioFileManager::fileExists(AudioFileId id)
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    std::vector<AudioFile*>::iterator it;
+
+    for (it = m_audioFiles.begin();
+            it != m_audioFiles.end();
+            ++it) {
+        if ((*it)->getId() == id)
+            return true;
+    }
+
+    return false;
+
+}
+
+void
+AudioFileManager::clear()
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    std::vector<AudioFile*>::iterator it;
+
+    for (it = m_audioFiles.begin();
+            it != m_audioFiles.end();
+            ++it) {
+        m_recordedAudioFiles.erase(*it);
+	m_derivedAudioFiles.erase(*it);
+        delete(*it);
+    }
+
+    m_audioFiles.erase(m_audioFiles.begin(), m_audioFiles.end());
+
+    // Clear the PeakFileManager too
+    //
+    m_peakManager.clear();
+}
+
+AudioFile *
+AudioFileManager::createRecordingAudioFile()
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    AudioFileId newId = getFirstUnusedID();
+    QString fileName = "";
+
+    while (fileName == "") {
+
+        fileName = QString("rg-%1-%2.wav")
+                   .arg(QDateTime::currentDateTime().toString("yyyyMMdd-hhmmss"))
+                   .arg(newId + 1);
+
+        if (QFile(m_audioPath.c_str() + fileName).exists()) {
+            fileName = "";
+            ++newId;
+        }
+    }
+
+    // insert file into vector
+    WAVAudioFile *aF = 0;
+
+    try {
+        aF = new WAVAudioFile(newId, fileName.data(), m_audioPath + fileName.data());
+        m_audioFiles.push_back(aF);
+        m_recordedAudioFiles.insert(aF);
+    } catch (SoundFile::BadSoundFileException e) {
+        delete aF;
+        throw BadAudioPathException(e);
+    }
+
+    return aF;
+}
+
+std::vector<std::string>
+AudioFileManager::createRecordingAudioFiles(unsigned int n)
+{
+    std::vector<std::string> v;
+    for (unsigned int i = 0; i < n; ++i) {
+        AudioFile *af = createRecordingAudioFile();
+        if (af)
+            v.push_back(m_audioPath + af->getFilename().data());
+        // !af should not happen, and we have no good recovery if it does
+    }
+    return v;
+}
+
+bool
+AudioFileManager::wasAudioFileRecentlyRecorded(AudioFileId id)
+{
+    AudioFile *file = getAudioFile(id);
+    if (file)
+        return (m_recordedAudioFiles.find(file) !=
+                m_recordedAudioFiles.end());
+    return false;
+}
+
+bool
+AudioFileManager::wasAudioFileRecentlyDerived(AudioFileId id)
+{
+    AudioFile *file = getAudioFile(id);
+    if (file)
+        return (m_derivedAudioFiles.find(file) !=
+                m_derivedAudioFiles.end());
+    return false;
+}
+
+void
+AudioFileManager::resetRecentlyCreatedFiles()
+{
+    m_recordedAudioFiles.clear();
+    m_derivedAudioFiles.clear();
+}
+
+AudioFile *
+AudioFileManager::createDerivedAudioFile(AudioFileId source,
+					 const char *prefix)
+{
+    MutexLock lock (&_audioFileManagerLock);
+
+    AudioFile *sourceFile = getAudioFile(source);
+    if (!sourceFile) return 0;
+
+    AudioFileId newId = getFirstUnusedID();
+    QString fileName = "";
+
+    std::string sourceBase = sourceFile->getShortFilename();
+    if (sourceBase.length() > 4 && sourceBase.substr(0, 3) == "rg-") {
+	sourceBase = sourceBase.substr(3);
+    }
+    if (sourceBase.length() > 15) sourceBase = sourceBase.substr(0, 15);
+
+    while (fileName == "") {
+
+        fileName = QString("%1-%2-%3-%4.wav")
+	    .arg(prefix)
+	    .arg(sourceBase)
+	    .arg(QDateTime::currentDateTime().toString("yyyyMMdd-hhmmss"))
+	    .arg(newId + 1);
+
+        if (QFile(m_audioPath.c_str() + fileName).exists()) {
+            fileName = "";
+            ++newId;
+        }
+    }
+
+    // insert file into vector
+    WAVAudioFile *aF = 0;
+
+    try {
+        aF = new WAVAudioFile(newId,
+			      fileName.data(),
+			      m_audioPath + fileName.data());
+        m_audioFiles.push_back(aF);
+	m_derivedAudioFiles.insert(aF);
+    } catch (SoundFile::BadSoundFileException e) {
+        delete aF;
+        throw BadAudioPathException(e);
+    }
+
+    return aF;
+}
+
+AudioFileId
+AudioFileManager::importURL(const KURL &url, int sampleRate)
+{
+    if (url.isLocalFile()) return importFile(url.path(), sampleRate);
+
+    std::cerr << "AudioFileManager::importURL("<< url.prettyURL() << ", " << sampleRate << ")" << std::endl;
+
+    emit setOperationName(i18n("Downloading file %1").arg(url.prettyURL()));
+
+    QString localPath = "";
+    if (!KIO::NetAccess::download(url, localPath)) {
+	KMessageBox::error(0, i18n("Cannot download file %1").arg(url.prettyURL()));
+	throw SoundFile::BadSoundFileException(url.prettyURL());
+    }
+    
+    AudioFileId id = 0;
+
+    try {
+	id = importFile(localPath.data(), sampleRate);
+    } catch (BadAudioPathException ape) {
+	KIO::NetAccess::removeTempFile(localPath);
+	throw ape;
+    } catch (SoundFile::BadSoundFileException bse) {
+	KIO::NetAccess::removeTempFile(localPath);
+	throw bse;
+    }
+    
+    return id;
+}
+
+bool
+AudioFileManager::fileNeedsConversion(const std::string &fileName,
+                                      int sampleRate)
+{
+    KProcess *proc = new KProcess();
+    *proc << "rosegarden-audiofile-importer";
+    if (sampleRate > 0) {
+        *proc << "-r";
+        *proc << QString("%1").arg(sampleRate);
+    }
+    *proc << "-w";
+    *proc << fileName.c_str();
+
+    proc->start(KProcess::Block, KProcess::NoCommunication);
+
+    int es = proc->exitStatus();
+    delete proc;
+
+    if (es == 0 || es == 1) { // 1 == "other error" -- wouldn't be able to convert
+        return false;
+    }
+    return true;
+}
+
+AudioFileId
+AudioFileManager::importFile(const std::string &fileName, int sampleRate)
+{
+    MutexLock lock (&_audioFileManagerLock);
+
+    std::cerr << "AudioFileManager::importFile("<< fileName << ", " << sampleRate << ")" << std::endl;
+
+    KProcess *proc = new KProcess();
+    *proc << "rosegarden-audiofile-importer";
+    if (sampleRate > 0) {
+	*proc << "-r";
+	*proc << QString("%1").arg(sampleRate);
+    }
+    *proc << "-w";
+    *proc << fileName.c_str();
+
+    proc->start(KProcess::Block, KProcess::NoCommunication);
+
+    int es = proc->exitStatus();
+    delete proc;
+
+    if (es == 0) {
+	AudioFileId id = addFile(fileName);
+	m_expectedSampleRate = sampleRate;
+	return id;
+    }
+
+    if (es == 2) {
+	emit setOperationName(i18n("Converting audio file..."));
+    } else if (es == 3) {
+	emit setOperationName(i18n("Resampling audio file..."));
+    } else if (es == 4) {
+	emit setOperationName(i18n("Converting and resampling audio file..."));
+    } else {
+	emit setOperationName(i18n("Importing audio file..."));
+    }
+
+    AudioFileId newId = getFirstUnusedID();
+    QString targetName = "";
+
+    QString sourceBase = QFileInfo(fileName.c_str()).baseName();
+    if (sourceBase.length() > 3 && sourceBase.startsWith("rg-")) {
+	sourceBase = sourceBase.right(sourceBase.length() - 3);
+    }
+    if (sourceBase.length() > 15) sourceBase = sourceBase.left(15);
+
+    while (targetName == "") {
+
+        targetName = QString("conv-%2-%3-%4.wav")
+	    .arg(sourceBase)
+	    .arg(QDateTime::currentDateTime().toString("yyyyMMdd-hhmmss"))
+	    .arg(newId + 1);
+
+        if (QFile(m_audioPath.c_str() + targetName).exists()) {
+            targetName = "";
+            ++newId;
+        }
+    }
+
+    m_importProcess = new KProcess;
+
+    *m_importProcess << "rosegarden-audiofile-importer";
+    if (sampleRate > 0) {
+	*m_importProcess << "-r";
+	*m_importProcess << QString("%1").arg(sampleRate);
+    }
+    *m_importProcess << "-c";
+    *m_importProcess << fileName.c_str();
+    *m_importProcess << (m_audioPath.c_str() + targetName);
+    
+    m_importProcess->start(KProcess::NotifyOnExit, KProcess::NoCommunication);
+
+    while (m_importProcess->isRunning()) {
+	kapp->processEvents(100);
+    }
+
+    if (!m_importProcess->normalExit()) {
+	// interrupted
+	throw SoundFile::BadSoundFileException(fileName, "Import cancelled");
+    }
+
+    es = m_importProcess->exitStatus();
+    delete m_importProcess;
+    m_importProcess = 0;
+
+    if (es) {
+	std::cerr << "audio file importer failed" << std::endl;
+	throw SoundFile::BadSoundFileException(fileName, i18n("Failed to convert or resample audio file on import"));
+    } else {
+	std::cerr << "audio file importer succeeded" << std::endl;
+    }
+
+    // insert file into vector
+    WAVAudioFile *aF = 0;
+
+    aF = new WAVAudioFile(newId,
+			  targetName.data(),
+			  m_audioPath + targetName.data());
+    m_audioFiles.push_back(aF);
+    m_derivedAudioFiles.insert(aF);
+    // Don't catch SoundFile::BadSoundFileException
+
+    m_expectedSampleRate = sampleRate;
+
+    return aF->getId();
+}
+
+void
+AudioFileManager::slotStopImport()
+{
+    if (m_importProcess) {
+	m_importProcess->kill(SIGTERM);
+	sleep(1);
+	m_importProcess->kill(SIGKILL);
+    }
+}
+
+AudioFile*
+AudioFileManager::getLastAudioFile()
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    std::vector<AudioFile*>::iterator it = m_audioFiles.begin();
+    AudioFile* audioFile = 0;
+
+    while (it != m_audioFiles.end()) {
+        audioFile = (*it);
+        it++;
+    }
+
+    return audioFile;
+}
+
+std::string
+AudioFileManager::substituteHomeForTilde(const std::string &path)
+{
+    std::string rS = path;
+    std::string homePath = std::string(getenv("HOME"));
+
+    // if path length is less than homePath then just return unchanged
+    if (rS.length() < homePath.length())
+        return rS;
+
+    // if the first section matches the path then substitute
+    if (rS.substr(0, homePath.length()) == homePath) {
+        rS.erase(0, homePath.length());
+        rS = "~" + rS;
+    }
+
+    return rS;
+}
+
+std::string
+AudioFileManager::substituteTildeForHome(const std::string &path)
+{
+    std::string rS = path;
+    std::string homePath = std::string(getenv("HOME"));
+
+    if (rS.substr(0, 2) == std::string("~/")) {
+        rS.erase(0, 1); // erase tilde and prepend HOME env
+        rS = homePath + rS;
+    }
+
+    return rS;
+}
+
+
+
+// Export audio files and assorted bits and bobs - make sure
+// that we store the files in a format that's user independent
+// so that people can pack up and swap their songs (including
+// audio files) and shift them about easily.
+//
+std::string
+AudioFileManager::toXmlString()
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    std::stringstream audioFiles;
+    std::string audioPath = substituteHomeForTilde(m_audioPath);
+
+    audioFiles << "<audiofiles";
+    if (m_expectedSampleRate != 0) {
+	audioFiles << " expectedRate=\"" << m_expectedSampleRate << "\"";
+    }
+    audioFiles << ">" << std::endl;
+    audioFiles << "    <audioPath value=\""
+    << audioPath << "\"/>" << std::endl;
+
+    std::string fileName;
+    std::vector<AudioFile*>::iterator it;
+
+    for (it = m_audioFiles.begin(); it != m_audioFiles.end(); ++it) {
+        fileName = (*it)->getFilename();
+
+        // attempt two substitutions - If the prefix to the filename
+        // is the same as the audio path then we can dock the prefix
+        // as it'll be added again next time.  If the path doesn't
+        // have the audio path in it but has our home directory in it
+        // then swap this out for a tilde '~'
+        //
+#ifdef DEBUG_AUDIOFILEMANAGER
+
+        std::cout << "DIR = " << getDirectory(fileName) << " : "
+        " PATH = " << m_audioPath << std::endl;
+#endif
+
+        if (getDirectory(fileName) == m_audioPath)
+            fileName = getShortFilename(fileName);
+        else
+            fileName = substituteHomeForTilde(fileName);
+
+        audioFiles << "    <audio id=\""
+        << (*it)->getId()
+        << "\" file=\""
+        << fileName
+        << "\" label=\""
+        << encode((*it)->getName())
+        << "\"/>" << std::endl;
+    }
+
+    audioFiles << "</audiofiles>" << std::endl;
+
+#if (__GNUC__ < 3)
+
+    audioFiles << std::ends;
+#else
+
+    audioFiles << std::endl;
+#endif
+
+    return audioFiles.str();
+}
+
+// Generate preview peak files or peak chunks according
+// to file type.
+//
+void
+AudioFileManager::generatePreviews()
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+#ifdef DEBUG_AUDIOFILEMANAGER
+
+    std::cout << "AudioFileManager::generatePreviews - "
+    << "for " << m_audioFiles.size() << " files"
+    << std::endl;
+#endif
+
+
+    // Generate peaks if we need to
+    //
+    std::vector<AudioFile*>::iterator it;
+    for (it = m_audioFiles.begin(); it != m_audioFiles.end(); ++it) {
+        if (!m_peakManager.hasValidPeaks(*it))
+            m_peakManager.generatePeaks(*it, 1);
+    }
+}
+
+// Attempt to stop a preview
+//
+void
+AudioFileManager::slotStopPreview()
+{
+    MutexLock lock (&_audioFileManagerLock);
+    m_peakManager.stopPreview();
+}
+
+
+// Generate a preview for a specific audio file - say if
+// one has just been added to the AudioFileManager.
+// Also used for generating previews if the file has been
+// modified.
+//
+bool
+AudioFileManager::generatePreview(AudioFileId id)
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    AudioFile *audioFile = getAudioFile(id);
+
+    if (audioFile == 0)
+        return false;
+
+    if (!m_peakManager.hasValidPeaks(audioFile))
+        m_peakManager.generatePeaks(audioFile, 1);
+
+    return true;
+}
+
+AudioFile*
+AudioFileManager::getAudioFile(AudioFileId id)
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    std::vector<AudioFile*>::iterator it;
+
+    for (it = m_audioFiles.begin();
+            it != m_audioFiles.end();
+            it++) {
+        if ((*it)->getId() == id)
+            return (*it);
+    }
+    return 0;
+}
+
+std::vector<float>
+AudioFileManager::getPreview(AudioFileId id,
+                             const RealTime &startTime,
+                             const RealTime &endTime,
+                             int width,
+                             bool withMinima)
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    AudioFile *audioFile = getAudioFile(id);
+
+    if (audioFile == 0) {
+        return std::vector<float>();
+    }
+
+    if (!m_peakManager.hasValidPeaks(audioFile)) {
+        std::cerr << "AudioFileManager::getPreview: No peaks for audio file " << audioFile->getFilename() << std::endl;
+        throw PeakFileManager::BadPeakFileException
+        (audioFile->getFilename(), __FILE__, __LINE__);
+    }
+
+    return m_peakManager.getPreview(audioFile,
+                                    startTime,
+                                    endTime,
+                                    width,
+                                    withMinima);
+}
+
+void
+AudioFileManager::drawPreview(AudioFileId id,
+                              const RealTime &startTime,
+                              const RealTime &endTime,
+                              QPixmap *pixmap)
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    AudioFile *audioFile = getAudioFile(id);
+    if (!audioFile)
+        return ;
+
+    if (!m_peakManager.hasValidPeaks(audioFile)) {
+        std::cerr << "AudioFileManager::getPreview: No peaks for audio file " << audioFile->getFilename() << std::endl;
+        throw PeakFileManager::BadPeakFileException
+        (audioFile->getFilename(), __FILE__, __LINE__);
+    }
+
+    std::vector<float> values = m_peakManager.getPreview
+                                (audioFile,
+                                 startTime,
+                                 endTime,
+                                 pixmap->width(),
+                                 false);
+
+    QPainter painter(pixmap);
+    pixmap->fill(kapp->palette().color(QPalette::Active,
+                                       QColorGroup::Base));
+    painter.setPen(kapp->palette().color(QPalette::Active,
+                                         QColorGroup::Dark));
+
+    if (values.size() == 0) {
+#ifdef DEBUG_AUDIOFILEMANAGER
+        std::cerr << "AudioFileManager::drawPreview - "
+        << "no preview values returned!" << std::endl;
+#endif
+
+        return ;
+    }
+
+    float yStep = pixmap->height() / 2;
+    int channels = audioFile->getChannels();
+    float ch1Value, ch2Value;
+
+    if (channels == 0) {
+#ifdef DEBUG_AUDIOFILEMANAGER
+        std::cerr << "AudioFileManager::drawPreview - "
+        << "no channels in audio file!" << std::endl;
+#endif
+
+        return ;
+    }
+
+
+    // Render pixmap
+    //
+    for (int i = 0; i < pixmap->width(); i++) {
+        // Always get two values for our pixmap no matter how many
+        // channels in AudioFile as that's all we can display.
+        //
+        if (channels == 1) {
+            ch1Value = values[i];
+            ch2Value = values[i];
+        } else {
+            ch1Value = values[i * channels];
+            ch2Value = values[i * channels + 1];
+        }
+
+        painter.drawLine(i, static_cast<int>(yStep - ch1Value * yStep),
+                         i, static_cast<int>(yStep + ch2Value * yStep));
+    }
+}
+
+void
+AudioFileManager::drawHighlightedPreview(AudioFileId id,
+        const RealTime &startTime,
+        const RealTime &endTime,
+        const RealTime &highlightStart,
+        const RealTime &highlightEnd,
+        QPixmap *pixmap)
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    AudioFile *audioFile = getAudioFile(id);
+    if (!audioFile)
+        return ;
+
+    if (!m_peakManager.hasValidPeaks(audioFile)) {
+        std::cerr << "AudioFileManager::getPreview: No peaks for audio file " << audioFile->getFilename() << std::endl;
+        throw PeakFileManager::BadPeakFileException
+        (audioFile->getFilename(), __FILE__, __LINE__);
+    }
+
+    std::vector<float> values = m_peakManager.getPreview
+                                (audioFile,
+                                 startTime,
+                                 endTime,
+                                 pixmap->width(),
+                                 false);
+
+    int startWidth = (int)(double(pixmap->width()) * (highlightStart /
+                           (endTime - startTime)));
+    int endWidth = (int)(double(pixmap->width()) * (highlightEnd /
+                         (endTime - startTime)));
+
+    QPainter painter(pixmap);
+    pixmap->fill(kapp->palette().color(QPalette::Active,
+                                       QColorGroup::Base));
+
+    float yStep = pixmap->height() / 2;
+    int channels = audioFile->getChannels();
+    float ch1Value, ch2Value;
+
+    // Render pixmap
+    //
+    for (int i = 0; i < pixmap->width(); ++i) {
+        if ((i * channels + (channels - 1)) >= int(values.size()))
+            break;
+
+        // Always get two values for our pixmap no matter how many
+        // channels in AudioFile as that's all we can display.
+        //
+        if (channels == 1) {
+            ch1Value = values[i];
+            ch2Value = values[i];
+        } else {
+            ch1Value = values[i * channels];
+            ch2Value = values[i * channels + 1];
+        }
+
+        if (i < startWidth || i > endWidth)
+            painter.setPen(kapp->palette().color(QPalette::Active,
+                                                 QColorGroup::Mid));
+        else
+            painter.setPen(kapp->palette().color(QPalette::Active,
+                                                 QColorGroup::Dark));
+
+        painter.drawLine(i, static_cast<int>(yStep - ch1Value * yStep),
+                         i, static_cast<int>(yStep + ch2Value * yStep));
+    }
+}
+
+
+void
+AudioFileManager::print()
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+#ifdef DEBUG_AUDIOFILEMANAGER
+
+    std::cout << "AudioFileManager - " << m_audioFiles.size() << " entr";
+
+    if (m_audioFiles.size() == 1)
+        std::cout << "y";
+    else
+        std::cout << "ies";
+
+    std::cout << std::endl << std::endl;
+
+    std::vector<AudioFile*>::iterator it;
+    for (it = m_audioFiles.begin(); it != m_audioFiles.end(); ++it) {
+        std::cout << (*it)->getId() << " : " << (*it)->getName()
+        << " : \"" << (*it)->getFilename() << "\"" << std::endl;
+    }
+#endif
+}
+
+std::vector<SplitPointPair>
+AudioFileManager::getSplitPoints(AudioFileId id,
+                                 const RealTime &startTime,
+                                 const RealTime &endTime,
+                                 int threshold,
+                                 const RealTime &minTime)
+{
+    MutexLock lock (&_audioFileManagerLock)
+        ;
+
+    AudioFile *audioFile = getAudioFile(id);
+
+    if (audioFile == 0)
+        return std::vector<SplitPointPair>();
+
+    return m_peakManager.getSplitPoints(audioFile,
+                                        startTime,
+                                        endTime,
+                                        threshold,
+                                        minTime);
+}
+
+std::set<int>
+AudioFileManager::getActualSampleRates() const
+{
+    std::set<int> rates;
+
+    for (std::vector<AudioFile *>::const_iterator i = m_audioFiles.begin();
+	 i != m_audioFiles.end(); ++i) {
+
+	unsigned int sr = (*i)->getSampleRate();
+	if (sr != 0) rates.insert(int(sr));
+    }
+
+    return rates;
+}
+
+}
+
+
+#include "AudioFileManager.moc"
diff --git a/src/sound/AudioFileManager.h b/src/sound/AudioFileManager.h
new file mode 100644
index 0000000..9721669
--- /dev/null
+++ b/src/sound/AudioFileManager.h
@@ -0,0 +1,327 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#ifndef _AUDIOFILEMANAGER_H_
+#define _AUDIOFILEMANAGER_H_
+
+#include <string>
+#include <vector>
+#include <set>
+#include <map>
+
+#include <qpixmap.h>
+#include <qobject.h>
+
+#include "AudioFile.h"
+#include "XmlExportable.h"
+#include "PeakFileManager.h"
+#include "PeakFile.h"
+#include "Exception.h"
+
+#include <kurl.h>
+
+// AudioFileManager loads and maps audio files to their
+// internal references (ids).  A point of contact for
+// AudioFile information - loading a Composition should
+// use this class to pick up the AudioFile references,
+// editing the AudioFiles in a Composition will be
+// made through this manager.
+
+// This is in the sound library because it's so closely
+// connected to other sound classes like the AudioFile
+// ones.  However, the audio file manager itself within
+// Rosegarden is stored in the GUI process.  This class
+// is not (and should not be) used elsewhere within the
+// sound or sequencer libraries.
+
+class KProcess;
+
+namespace Rosegarden
+{
+
+typedef std::vector<AudioFile*>::const_iterator AudioFileManagerIterator;
+
+class AudioFileManager : public QObject, public XmlExportable
+{
+    Q_OBJECT
+public:
+    AudioFileManager();
+    virtual ~AudioFileManager();
+    
+    class BadAudioPathException : public Exception
+    {
+    public:
+        BadAudioPathException(std::string path) :
+            Exception("Bad audio file path " + path), m_path(path) { }
+        BadAudioPathException(std::string path, std::string file, int line) :
+            Exception("Bad audio file path " + path, file, line), m_path(path) { }
+        BadAudioPathException(const SoundFile::BadSoundFileException &e) :
+            Exception("Bad audio file path (malformed file?) " + e.getPath()), m_path(e.getPath()) { }
+
+        ~BadAudioPathException() throw() { }
+
+        std::string getPath() const { return m_path; }
+
+    private:
+        std::string m_path;
+    };
+
+private:
+    AudioFileManager(const AudioFileManager &aFM);
+    AudioFileManager& operator=(const AudioFileManager &);
+
+public:
+
+    // Create an audio file from an absolute path - we use this interface
+    // to add an actual file.
+    //
+    AudioFileId addFile(const std::string &filePath);
+    // throw BadAudioPathException
+
+    // Return true if a file would require importFile to import it, rather
+    // than a simple addFile.  You can use importFile even when a file
+    // doesn't need conversion, but this tells you whether it's necessary
+    //
+    bool fileNeedsConversion(const std::string &filePath,
+                             int targetSampleRate = 0);
+
+    // Create an audio file by importing (i.e. converting and/or
+    // resampling) an existing file using the external conversion
+    // utility
+    //
+    AudioFileId importFile(const std::string &filePath,
+			   int targetSampleRate = 0);
+    // throw BadAudioPathException, BadSoundFileException
+
+    // Create an audio file by importing from a URL
+    //
+    AudioFileId importURL(const KURL &filePath,
+			  int targetSampleRate = 0);
+    // throw BadAudioPathException, BadSoundFileException
+
+    // Insert an audio file into the AudioFileManager and get the
+    // first allocated id for it.  Used from the RG file as we already
+    // have both name and filename/path.
+    //
+    AudioFileId insertFile(const std::string &name,
+                           const std::string &fileName);
+    // throw BadAudioPathException
+
+    // And insert an AudioFile and specify an id
+    //
+    bool insertFile(const std::string &name, const std::string &fileName,
+                    AudioFileId id);
+    // throw BadAudioPathException
+
+    // Remove a file from the AudioManager by id
+    //
+    bool removeFile(AudioFileId id);
+
+    // Does a specific file id exist?
+    //
+    bool fileExists(AudioFileId id);
+
+    // Does a specific file path exist?  Return ID or -1.
+    //
+    int fileExists(const std::string &path);
+
+    // get audio file by id
+    //
+    AudioFile* getAudioFile(AudioFileId id);
+
+    // Get the list of files
+    //
+    std::vector<AudioFile*>::const_iterator begin() const
+        { return m_audioFiles.begin(); }
+
+    std::vector<AudioFile*>::const_iterator end() const
+        { return m_audioFiles.end(); }
+
+    // Clear down all audio file references
+    //
+    void clear();
+
+    // Get and set the record path
+    //
+    std::string getAudioPath() const { return m_audioPath; }
+    void setAudioPath(const std::string &path);
+
+    // Throw if the current audio path does not exist or is not writable
+    //
+    void testAudioPath() throw(BadAudioPathException);
+
+    // Get a new audio filename at the audio record path
+    //
+    AudioFile *createRecordingAudioFile();
+    // throw BadAudioPathException
+
+    // Get a set of new audio filenames at the audio record path
+    //
+    std::vector<std::string> createRecordingAudioFiles(unsigned int number);
+    // throw BadAudioPathException
+
+    // Return whether a file was created by recording within this "session"
+    //
+    bool wasAudioFileRecentlyRecorded(AudioFileId id);
+
+    // Return whether a file was created by derivation within this "session"
+    //
+    bool wasAudioFileRecentlyDerived(AudioFileId id);
+
+    // Indicate that a new "session" has started from the point of
+    // view of recorded and derived audio files (e.g. that the
+    // document has been saved)
+    //
+    void resetRecentlyCreatedFiles();
+    
+    // Create an empty file "derived from" the source (used by e.g. stretcher)
+    // 
+    AudioFile *createDerivedAudioFile(AudioFileId source,
+				      const char *prefix);
+
+    // return the last file in the vector - the last created
+    //
+    AudioFile* getLastAudioFile();
+
+    // Export to XML
+    //
+    virtual std::string toXmlString();
+
+    // Convenience function generate all previews on the audio file.
+    //
+    void generatePreviews();
+    // throw BadSoundFileException, BadPeakFileException
+
+    // Generate for a single audio file
+    //
+    bool generatePreview(AudioFileId id);
+    // throw BadSoundFileException, BadPeakFileException
+
+    // Get a preview for an AudioFile adjusted to Segment start and
+    // end parameters (assuming they fall within boundaries).
+    // 
+    // We can get back a set of values (floats) or a Pixmap if we 
+    // supply the details.
+    //
+    std::vector<float> getPreview(AudioFileId id,
+                                  const RealTime &startTime, 
+                                  const RealTime &endTime,
+                                  int width,
+                                  bool withMinima);
+    // throw BadPeakFileException, BadAudioPathException
+
+    // Draw a fixed size (fixed by QPixmap) preview of an audio file
+    //
+    void drawPreview(AudioFileId id,
+                     const RealTime &startTime, 
+                     const RealTime &endTime,
+                     QPixmap *pixmap);
+    // throw BadPeakFileException, BadAudioPathException
+
+    // Usually used to show how an audio Segment makes up part of
+    // an audio file.
+    //
+    void drawHighlightedPreview(AudioFileId it,
+                                const RealTime &startTime,
+                                const RealTime &endTime,
+                                const RealTime &highlightStart,
+                                const RealTime &highlightEnd,
+                                QPixmap *pixmap);
+    // throw BadPeakFileException, BadAudioPathException
+
+    // Get a short file name from a long one (with '/'s)
+    //
+    std::string getShortFilename(const std::string &fileName);
+
+    // Get a directory from a full file path
+    //
+    std::string getDirectory(const std::string &path);
+
+    // Attempt to subsititute a tilde '~' for a home directory
+    // to make paths a little more generic when saving.  Also
+    // provide the inverse function as convenience here.
+    //
+    std::string substituteHomeForTilde(const std::string &path);
+    std::string substituteTildeForHome(const std::string &path);
+
+    // Show entries for debug purposes
+    //
+    void print(); 
+
+    // Get a split point vector from a peak file
+    //
+    std::vector<SplitPointPair> 
+        getSplitPoints(AudioFileId id,
+                       const RealTime &startTime,
+                       const RealTime &endTime,
+                       int threshold,
+                       const RealTime &minTime = RealTime(0, 100000000));
+    // throw BadPeakFileException, BadAudioPathException
+
+    // Get the peak file manager
+    //
+    const PeakFileManager& getPeakFileManager() const { return m_peakManager; }
+
+    // Get the peak file manager
+    //
+    PeakFileManager& getPeakFileManager() { return m_peakManager; }
+
+    int getExpectedSampleRate() const { return m_expectedSampleRate; }
+    void setExpectedSampleRate(int rate) { m_expectedSampleRate = rate; }
+
+    std::set<int> getActualSampleRates() const;
+
+signals:
+    void setProgress(int);
+    void setOperationName(QString);
+
+public slots:
+    // Cancel a running preview
+    //
+    void slotStopPreview();
+
+    void slotStopImport();
+
+private:
+    std::string getFileInPath(const std::string &file);
+
+    AudioFileId getFirstUnusedID();
+
+    std::vector<AudioFile*> m_audioFiles;
+    std::string m_audioPath;
+
+    PeakFileManager m_peakManager;
+
+    // All audio files are stored in m_audioFiles.  These additional
+    // sets of pointers just refer to those that have been created by
+    // recording or derivations within the current session, and thus
+    // that the user may wish to remove at the end of the session if
+    // the document is not saved.
+    std::set<AudioFile *> m_recordedAudioFiles;
+    std::set<AudioFile *> m_derivedAudioFiles;
+
+    KProcess *m_importProcess;
+
+    int m_expectedSampleRate;
+};
+
+}
+
+#endif // _AUDIOFILEMANAGER_H_
diff --git a/src/sound/AudioFileTimeStretcher.cpp b/src/sound/AudioFileTimeStretcher.cpp
new file mode 100644
index 0000000..d5b2321
--- /dev/null
+++ b/src/sound/AudioFileTimeStretcher.cpp
@@ -0,0 +1,268 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rosegarden
+    A MIDI and audio sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <richard.bown@ferventsoftware.com>
+
+    The moral rights of Guillaume Laurent, Chris Cannam, and Richard
+    Bown to claim authorship of this work have been asserted.
+
+    Other copyrights also apply to some parts of this work.  Please
+    see the AUTHORS file and individual file headers for details.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "AudioFileTimeStretcher.h"
+#include "AudioTimeStretcher.h"
+#include "AudioFileManager.h"
+#include "WAVAudioFile.h"
+#include "base/RealTime.h"
+
+#include <kapplication.h>
+
+#include <iostream>
+#include <fstream>
+
+namespace Rosegarden {
+
+
+AudioFileTimeStretcher::AudioFileTimeStretcher(AudioFileManager *manager) :
+    m_manager(manager),
+    m_timestretchCancelled(false)
+{
+}
+
+AudioFileTimeStretcher::~AudioFileTimeStretcher()
+{
+}
+
+AudioFileId
+AudioFileTimeStretcher::getStretchedAudioFile(AudioFileId source,
+                                              float ratio)
+{
+    AudioFile *sourceFile = m_manager->getAudioFile(source);
+    if (!sourceFile) {
+	throw SoundFile::BadSoundFileException
+	    ("<unknown source>",
+	     "Source file not found in AudioFileTimeStretcher::getStretchedAudioFile");
+    }
+
+    std::cerr << "AudioFileTimeStretcher: got source file id " << source
+              << ", name " << sourceFile->getFilename() << std::endl;
+
+    AudioFile *file = m_manager->createDerivedAudioFile(source, "stretch");
+    if (!file) {
+	throw AudioFileManager::BadAudioPathException(m_manager->getAudioPath());
+    }
+
+    std::cerr << "AudioFileTimeStretcher: got derived file id " << file->getId()
+              << ", name " << file->getFilename() << std::endl;
+
+    std::ifstream streamIn(sourceFile->getFilename().c_str(),
+			   std::ios::in | std::ios::binary);
+    if (!streamIn) {
+	throw SoundFile::BadSoundFileException
+	    (file->getFilename().c_str(),
+	     "Failed to open source stream for time stretcher");
+    }
+    
+    //!!!
+    //...
+    // Need to make SoundDriver::getAudioRecFileFormat available?
+    // -- the sound file classes should just have a float interface
+    // (like libsndfile, or hey!, we could use libsndfile...)
+
+    WAVAudioFile writeFile
+        (file->getFilename(),
+         sourceFile->getChannels(),
+         sourceFile->getSampleRate(),
+         sourceFile->getSampleRate() * 4 * sourceFile->getChannels(),
+         4 * sourceFile->getChannels(),
+         32);
+
+    if (!writeFile.write()) {
+        throw AudioFileManager::BadAudioPathException
+            (file->getFilename());
+    }
+    
+    int obs = 1024;
+    int ibs = obs / ratio;
+    int ch = sourceFile->getChannels();
+    int sr = sourceFile->getSampleRate();
+	    
+    AudioTimeStretcher stretcher(sr, ch, ratio, true, obs);
+
+    // We'll first prime the timestretcher with half its window size
+    // of silence, an amount which we then discard at the start of the
+    // output (as well as its own processing latency).  Really the
+    // timestretcher should handle this itself and report it in its
+    // own latency calculation
+
+    size_t padding = stretcher.getWindowSize()/2;
+
+    char *ebf = (char *)alloca
+        (ch * ibs * sourceFile->getBytesPerFrame());
+    
+    std::vector<float *> dbfs;
+    for (int c = 0; c < ch; ++c) {
+        dbfs.push_back((float *)alloca((ibs > padding ? ibs : padding)
+                                       * sizeof(float)));
+    }
+    
+    float **ibfs = (float **)alloca(ch * sizeof(float *));
+    float **obfs = (float **)alloca(ch * sizeof(float *));
+            
+    for (int c = 0; c < ch; ++c) {
+        ibfs[c] = dbfs[c];
+    }
+        
+    for (int c = 0; c < ch; ++c) {
+        obfs[c] = (float *)alloca(obs * sizeof(float));
+    }
+        
+    char *oebf = (char *)alloca(ch * obs * sizeof(float));
+        
+    int totalIn = 0, totalOut = 0;
+        
+    for (int c = 0; c < ch; ++c) {
+        for (size_t i = 0; i < padding; ++i) {
+            ibfs[c][i] = 0.f;
+        }
+    }
+    stretcher.putInput(ibfs, padding);
+
+    RealTime totalTime = sourceFile->getLength();
+    long fileTotalIn = RealTime::realTime2Frame
+        (totalTime, sourceFile->getSampleRate());
+    int progressCount = 0;
+	
+    long expectedOut = ceil(fileTotalIn * ratio);
+
+    m_timestretchCancelled = false;
+    bool inputExhausted = false;
+
+    sourceFile->scanTo(&streamIn, RealTime::zeroTime);
+
+    while (1) {
+            
+        if (m_timestretchCancelled) {
+            std::cerr << "AudioFileTimeStretcher::getStretchedAudioFile: cancelled" << std::endl;
+            throw CancelledException();
+        }
+            
+        unsigned int thisRead = 0;
+
+        if (!inputExhausted) {
+            thisRead = sourceFile->getSampleFrames(&streamIn, ebf, ibs);
+            if (thisRead < ibs) inputExhausted = true;
+        }
+            
+        if (thisRead == 0) {
+            if (totalOut >= expectedOut) break;
+            else {
+                // run out of input data, continue feeding zeroes until
+                // we have enough output data
+                for (int c = 0; c < ch; ++c) {
+                    for (int i = 0; i < ibs; ++i) {
+                        ibfs[c][i] = 0.f;
+                    }
+                }
+                thisRead = ibs;
+            }
+        }
+            
+        if (!sourceFile->decode((unsigned char *)ebf,
+                                thisRead * sourceFile->getBytesPerFrame(),
+                                sr, ch,
+                                thisRead, dbfs, false)) {
+            std::cerr << "ERROR: Stupid audio file class failed to decode its own output" << std::endl;
+            break;
+        }
+            
+        stretcher.putInput(ibfs, thisRead);
+        totalIn += thisRead;
+            
+        unsigned int available = stretcher.getAvailableOutputSamples();
+            
+        while (available > 0) {
+                
+            unsigned int count = available;
+            if (count > obs) count = obs;
+
+            if (padding > 0) {
+                if (count <= padding) {
+                    stretcher.getOutput(obfs, count);
+                    padding -= count;
+                    available -= count;
+                    continue;
+                } else {
+                    stretcher.getOutput(obfs, padding);
+                    count -= padding;
+                    available -= padding;
+                    padding = 0;
+                }
+            }
+                
+            stretcher.getOutput(obfs, count);
+                
+            char *encodePointer = oebf;
+            for (int i = 0; i < count; ++i) {
+                for (int c = 0; c < ch; ++c) {
+                    float sample = obfs[c][i];
+                    *(float *)encodePointer = sample;
+                    encodePointer += sizeof(float);
+                }
+            }
+                
+            if (totalOut < expectedOut &&
+                totalOut + count > expectedOut) {
+                count = expectedOut - totalOut;
+            }
+
+            writeFile.appendSamples(oebf, count);
+            totalOut += count;
+            available -= count;
+
+            if (totalOut >= expectedOut) break;
+        }
+            
+        if (++progressCount == 100) {
+            int progress = int
+                ((100.f * float(totalIn)) / float(fileTotalIn));
+            emit setProgress(progress);
+            kapp->processEvents();
+            progressCount = 0;
+        }
+    }		
+        
+    emit setProgress(100);
+    kapp->processEvents();
+    writeFile.close();
+    
+    std::cerr << "AudioFileTimeStretcher::getStretchedAudioFile: success, id is "
+                 << file->getId() << std::endl;
+
+    return file->getId();
+}
+
+void
+AudioFileTimeStretcher::slotStopTimestretch()
+{
+    m_timestretchCancelled = true;
+}
+
+
+}
+
+#include "AudioFileTimeStretcher.moc"
+
diff --git a/src/sound/AudioFileTimeStretcher.h b/src/sound/AudioFileTimeStretcher.h
new file mode 100644
index 0000000..c02e286
--- /dev/null
+++ b/src/sound/AudioFileTimeStretcher.h
@@ -0,0 +1,76 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rosegarden
+    A MIDI and audio sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <richard.bown@ferventsoftware.com>
+
+    The moral rights of Guillaume Laurent, Chris Cannam, and Richard
+    Bown to claim authorship of this work have been asserted.
+
+    Other copyrights also apply to some parts of this work.  Please
+    see the AUTHORS file and individual file headers for details.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _AUDIO_FILE_TIME_STRETCHER_H_
+#define _AUDIO_FILE_TIME_STRETCHER_H_
+
+#include <qobject.h>
+#include "AudioFile.h"
+#include "base/Exception.h"
+
+namespace Rosegarden {
+
+class AudioFileManager;
+
+class AudioFileTimeStretcher : public QObject
+{
+    Q_OBJECT
+    
+public:
+    AudioFileTimeStretcher(AudioFileManager *mgr);
+    virtual ~AudioFileTimeStretcher();
+
+    /**
+     * Stretch an audio file and return the ID of the stretched
+     * version.  May throw SoundFile::BadSoundFileException,
+     * AudioFileManager::BadAudioPathException, CancelledException
+     */
+    AudioFileId getStretchedAudioFile(AudioFileId source,
+                                      float ratio);
+
+    class CancelledException : public Exception
+    {
+    public:
+	CancelledException() : Exception("Cancelled") { }
+	~CancelledException() throw() { }
+    };
+
+signals:
+    void setProgress(int);
+
+public slots:
+    /**
+     * Cancel an ongoing getStretchedAudioFile
+     */
+    void slotStopTimestretch();
+    
+protected:
+    AudioFileManager *m_manager;
+
+    bool m_timestretchCancelled;
+};
+
+}
+
+#endif
diff --git a/src/sound/AudioPlayQueue.cpp b/src/sound/AudioPlayQueue.cpp
new file mode 100644
index 0000000..2bd07c3
--- /dev/null
+++ b/src/sound/AudioPlayQueue.cpp
@@ -0,0 +1,501 @@
+
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "AudioPlayQueue.h"
+#include "misc/Debug.h"
+#include "PlayableAudioFile.h"
+#include "Profiler.h"
+
+//#define DEBUG_AUDIO_PLAY_QUEUE 1
+//#define FINE_DEBUG_AUDIO_PLAY_QUEUE 1
+
+namespace Rosegarden
+{
+
+
+static inline unsigned int instrumentId2Index(InstrumentId id)
+{
+    if (id < AudioInstrumentBase)
+        return 0;
+    else
+        return (id - AudioInstrumentBase);
+}
+
+bool
+AudioPlayQueue::FileTimeCmp::operator()(const PlayableAudioFile &f1,
+                                        const PlayableAudioFile &f2) const
+{
+    return operator()(&f1, &f2);
+}
+
+bool
+AudioPlayQueue::FileTimeCmp::operator()(const PlayableAudioFile *f1,
+                                        const PlayableAudioFile *f2) const
+{
+    RealTime t1 = f1->getStartTime(), t2 = f2->getStartTime();
+    if (t1 < t2)
+        return true;
+    else if (t2 < t1)
+        return false;
+    else
+        return f1 < f2;
+}
+
+
+AudioPlayQueue::AudioPlayQueue() :
+        m_maxBuffers(0)
+{
+    // nothing to do
+}
+
+AudioPlayQueue::~AudioPlayQueue()
+{
+    std::cerr << "AudioPlayQueue::~AudioPlayQueue()" << std::endl;
+    clear();
+}
+
+void
+AudioPlayQueue::addScheduled(PlayableAudioFile *file)
+{
+    if (m_files.find(file) != m_files.end()) {
+        std::cerr << "WARNING: AudioPlayQueue::addScheduled("
+        << file << "): already in queue" << std::endl;
+        return ;
+    }
+
+    m_files.insert(file);
+
+    RealTime startTime = file->getStartTime();
+    RealTime endTime = file->getStartTime() + file->getDuration();
+
+    InstrumentId instrument = file->getInstrument();
+    unsigned int index = instrumentId2Index(instrument);
+
+    while (m_instrumentIndex.size() <= index) {
+        m_instrumentIndex.push_back(ReverseFileMap());
+    }
+
+#ifdef DEBUG_AUDIO_PLAY_QUEUE
+    std::cerr << "AudioPlayQueue[" << this << "]::addScheduled(" << file << "): start " << file->getStartTime() << ", end " << file->getEndTime() << ", slots: " << std::endl;
+#endif
+
+    for (int i = startTime.sec; i <= endTime.sec; ++i) {
+        m_index[i].push_back(file);
+        m_instrumentIndex[index][i].push_back(file);
+        if (!file->isSmallFile()) {
+            m_counts[i] += file->getTargetChannels();
+            if (m_counts[i] > m_maxBuffers) {
+                m_maxBuffers = m_counts[i];
+            }
+        }
+#ifdef DEBUG_AUDIO_PLAY_QUEUE
+        std::cerr << i << " ";
+#endif
+
+    }
+
+#ifdef DEBUG_AUDIO_PLAY_QUEUE
+    std::cerr << std::endl << "(max buffers now "
+    << m_maxBuffers << ")" << std::endl;
+#endif
+}
+
+void
+AudioPlayQueue::addUnscheduled(PlayableAudioFile *file)
+{
+#ifdef DEBUG_AUDIO_PLAY_QUEUE
+    std::cerr << "AudioPlayQueue[" << this << "]::addUnscheduled(" << file << "): start " << file->getStartTime() << ", end " << file->getEndTime() << ", instrument " << file->getInstrument() << std::endl;
+#endif
+
+    m_unscheduled.push_back(file);
+
+#ifdef DEBUG_AUDIO_PLAY_QUEUE
+
+    std::cerr << "AudioPlayQueue[" << this << "]::addUnscheduled: now " << m_unscheduled.size() << " unscheduled files" << std::endl;
+#endif
+
+}
+
+void
+AudioPlayQueue::erase(PlayableAudioFile *file)
+{
+#ifdef DEBUG_AUDIO_PLAY_QUEUE
+    std::cerr << "AudioPlayQueue::erase(" << file << "): start " << file->getStartTime() << ", end " << file->getEndTime() << std::endl;
+#endif
+
+    FileSet::iterator fi = m_files.find(file);
+    if (fi == m_files.end()) {
+        for (FileList::iterator fli = m_unscheduled.begin();
+                fli != m_unscheduled.end(); ++fli) {
+            if (*fli == file) {
+                m_unscheduled.erase(fli);
+                delete file;
+                return ;
+            }
+        }
+        return ;
+    }
+    m_files.erase(fi);
+
+    InstrumentId instrument = file->getInstrument();
+    unsigned int index = instrumentId2Index(instrument);
+
+    for (ReverseFileMap::iterator mi = m_instrumentIndex[index].begin();
+            mi != m_instrumentIndex[index].end(); ++mi) {
+
+        for (FileVector::iterator fi = mi->second.begin();
+                fi != mi->second.end(); ++fi) {
+
+            if (*fi == file) {
+                mi->second.erase(fi);
+                if (m_counts[mi->first] > 0)
+                    --m_counts[mi->first];
+                break;
+            }
+        }
+    }
+
+    for (ReverseFileMap::iterator mi = m_index.begin();
+            mi != m_index.end(); ++mi) {
+
+        for (FileVector::iterator fi = mi->second.begin();
+                fi != mi->second.end(); ++fi) {
+
+            if (*fi == file) {
+                mi->second.erase(fi);
+                if (m_counts[mi->first] > 0)
+                    --m_counts[mi->first];
+                break;
+            }
+        }
+    }
+
+    delete file;
+}
+
+void
+AudioPlayQueue::clear()
+{
+#ifdef DEBUG_AUDIO_PLAY_QUEUE
+    std::cerr << "AudioPlayQueue::clear()" << std::endl;
+#endif
+
+    while (m_files.begin() != m_files.end()) {
+        delete *m_files.begin();
+        m_files.erase(m_files.begin());
+    }
+
+    while (m_unscheduled.begin() != m_unscheduled.end()) {
+        delete *m_unscheduled.begin();
+        m_unscheduled.erase(m_unscheduled.begin());
+    }
+
+    m_instrumentIndex.clear();
+    m_index.clear();
+    m_counts.clear();
+    m_maxBuffers = 0;
+}
+
+bool
+AudioPlayQueue::empty() const
+{
+    return m_unscheduled.empty() && m_files.empty();
+}
+
+size_t
+AudioPlayQueue::size() const
+{
+    return m_unscheduled.size() + m_files.size();
+}
+
+void
+AudioPlayQueue::getPlayingFiles(const RealTime &sliceStart,
+                                const RealTime &sliceDuration,
+                                FileSet &playing) const
+{
+    //    Profiler profiler("AudioPlayQueue::getPlayingFiles");
+
+    // This one needs to be quick.
+
+    playing.clear();
+
+    RealTime sliceEnd = sliceStart + sliceDuration;
+
+    for (int i = sliceStart.sec; i <= sliceEnd.sec; ++i) {
+
+        ReverseFileMap::const_iterator mi(m_index.find(i));
+        if (mi == m_index.end())
+            continue;
+
+        for (FileVector::const_iterator fi = mi->second.begin();
+                fi != mi->second.end(); ++fi) {
+
+            PlayableAudioFile *f = *fi;
+
+            if (f->getStartTime() > sliceEnd ||
+                    f->getStartTime() + f->getDuration() <= sliceStart)
+                continue;
+
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+
+            std::cerr << "... found " << f << " in slot " << i << std::endl;
+#endif
+
+            playing.insert(f);
+        }
+    }
+
+    for (FileList::const_iterator fli = m_unscheduled.begin();
+            fli != m_unscheduled.end(); ++fli) {
+        PlayableAudioFile *file = *fli;
+        if (file->getStartTime() <= sliceEnd &&
+                file->getStartTime() + file->getDuration() > sliceStart) {
+            playing.insert(file);
+        }
+    }
+
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+    if (playing.size() > 0) {
+        std::cerr << "AudioPlayQueue::getPlayingFiles(" << sliceStart << ","
+        << sliceDuration << "): total "
+        << playing.size() << " files" << std::endl;
+    }
+#endif
+}
+
+void
+AudioPlayQueue::getPlayingFilesForInstrument(const RealTime &sliceStart,
+        const RealTime &sliceDuration,
+        InstrumentId instrumentId,
+        PlayableAudioFile **playing,
+        size_t &size) const
+{
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+    bool printed = false;
+    Profiler profiler("AudioPlayQueue::getPlayingFilesForInstrument", true);
+#endif
+
+    // This one needs to be quick.
+
+    size_t written = 0;
+
+    RealTime sliceEnd = sliceStart + sliceDuration;
+
+    unsigned int index = instrumentId2Index(instrumentId);
+    if (index >= m_instrumentIndex.size()) {
+        goto unscheduled; // nothing scheduled here
+    }
+
+    for (int i = sliceStart.sec; i <= sliceEnd.sec; ++i) {
+
+        ReverseFileMap::const_iterator mi
+        (m_instrumentIndex[index].find(i));
+
+        if (mi == m_instrumentIndex[index].end())
+            continue;
+
+        for (FileVector::const_iterator fi = mi->second.begin();
+                fi != mi->second.end(); ++fi) {
+
+            PlayableAudioFile *f = *fi;
+
+            if (f->getInstrument() != instrumentId)
+                continue;
+
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+
+            if (!printed) {
+                std::cerr << "AudioPlayQueue::getPlayingFilesForInstrument(" << sliceStart
+                << ", " << sliceDuration << ", " << instrumentId << ")"
+                << std::endl;
+                printed = true;
+            }
+#endif
+
+            if (f->getStartTime() > sliceEnd ||
+                    f->getEndTime() <= sliceStart) {
+
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+                std::cerr << "... rejected " << f << " in slot " << i << std::endl;
+                if (f->getStartTime() > sliceEnd) {
+                    std::cerr << "(" << f->getStartTime() << " > " << sliceEnd
+                    << ")" << std::endl;
+                } else {
+                    std::cerr << "(" << f->getEndTime() << " <= " << sliceStart
+                    << ")" << std::endl;
+                }
+#endif
+
+                continue;
+            }
+
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+            std::cerr << "... found " << f << " in slot " << i << " ("
+            << f->getStartTime() << " -> " << f->getEndTime()
+            << ")" << std::endl;
+#endif
+
+            size_t j = 0;
+            for (j = 0; j < written; ++j) {
+                if (playing[j] == f)
+                    break;
+            }
+            if (j < written)
+                break; // already have it
+
+            if (written >= size) {
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+                std::cerr << "No room to write it!" << std::endl;
+#endif
+
+                break;
+            }
+
+            playing[written++] = f;
+        }
+    }
+
+unscheduled:
+
+    for (FileList::const_iterator fli = m_unscheduled.begin();
+            fli != m_unscheduled.end(); ++fli) {
+
+        PlayableAudioFile *f = *fli;
+
+        if (f->getInstrument() != instrumentId) {
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+            std::cerr << "rejecting unscheduled " << f << " as wrong instrument ("
+            << f->getInstrument() << " != " << instrumentId << ")" << std::endl;
+#endif
+
+            continue;
+        }
+
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+        if (!printed) {
+            std::cerr << "AudioPlayQueue::getPlayingFilesForInstrument(" << sliceStart
+            << ", " << sliceDuration << ", " << instrumentId << ")"
+            << std::endl;
+            printed = true;
+        }
+#endif
+
+        if (f->getStartTime() <= sliceEnd &&
+                f->getStartTime() + f->getDuration() > sliceStart) {
+
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+            std::cerr << "... found " << f << " in unscheduled list ("
+            << f->getStartTime() << " -> " << f->getEndTime()
+            << ")" << std::endl;
+#endif
+
+            if (written >= size)
+                break;
+            playing[written++] = f;
+
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+
+        } else {
+
+            std::cerr << "... rejected " << f << " in unscheduled list" << std::endl;
+            if (f->getStartTime() > sliceEnd) {
+                std::cerr << "(" << f->getStartTime() << " > " << sliceEnd
+                << ")" << std::endl;
+            } else {
+                std::cerr << "(" << f->getEndTime() << " <= " << sliceStart
+                << ")" << std::endl;
+            }
+#endif
+
+        }
+    }
+
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+    if (written > 0) {
+        std::cerr << "AudioPlayQueue::getPlayingFilesForInstrument: total "
+        << written << " files" << std::endl;
+    }
+#endif
+
+    size = written;
+}
+
+bool
+AudioPlayQueue::haveFilesForInstrument(InstrumentId instrumentId) const
+{
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+    std::cerr << "AudioPlayQueue::haveFilesForInstrument(" << instrumentId << ")...";
+#endif
+
+    unsigned int index = instrumentId2Index(instrumentId);
+
+    if (index < m_instrumentIndex.size() &&
+            !m_instrumentIndex[index].empty()) {
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+        std::cerr << " yes (scheduled)" << std::endl;
+#endif
+
+        return true;
+    }
+
+    for (FileList::const_iterator fli = m_unscheduled.begin();
+            fli != m_unscheduled.end(); ++fli) {
+        PlayableAudioFile *file = *fli;
+        if (file->getInstrument() == instrumentId) {
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+            std::cerr << " yes (unscheduled)" << std::endl;
+#endif
+
+            return true;
+        }
+    }
+
+#ifdef FINE_DEBUG_AUDIO_PLAY_QUEUE
+    std::cerr << " no" << std::endl;
+#endif
+
+    return false;
+}
+
+const AudioPlayQueue::FileSet &
+AudioPlayQueue::getAllScheduledFiles() const
+{
+#ifdef DEBUG_AUDIO_PLAY_QUEUE
+    std::cerr << "AudioPlayQueue[" << this << "]::getAllScheduledFiles: have " << m_files.size() << " files" << std::endl;
+#endif
+
+    return m_files;
+}
+
+const AudioPlayQueue::FileList &
+AudioPlayQueue::getAllUnscheduledFiles() const
+{
+#ifdef DEBUG_AUDIO_PLAY_QUEUE
+    std::cerr << "AudioPlayQueue[" << this << "]::getAllUnscheduledFiles: have " << m_unscheduled.size() << " files" << std::endl;
+#endif
+
+    return m_unscheduled;
+}
+
+
+}
+
diff --git a/src/sound/AudioPlayQueue.h b/src/sound/AudioPlayQueue.h
new file mode 100644
index 0000000..2a7067c
--- /dev/null
+++ b/src/sound/AudioPlayQueue.h
@@ -0,0 +1,168 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _AUDIO_PLAY_QUEUE_H_
+#define _AUDIO_PLAY_QUEUE_H_
+
+#include "RealTime.h"
+#include "Instrument.h"
+
+#include <set>
+#include <vector>
+#include <map>
+#include <list>
+
+namespace Rosegarden
+{
+
+class PlayableAudioFile;
+
+/**
+ * An ordered list of PlayableAudioFiles that does not aim to be quick
+ * to add to or remove files from, but that aims to quickly answer the
+ * question of which files are playing within a given time slice.
+ *
+ * Note that there is no locking between audio file add/remove and
+ * lookup.  Add/remove should only be carried out when it is known
+ * that no threads will be performing lookup.
+ */
+
+class AudioPlayQueue
+{
+public:
+    AudioPlayQueue();
+    virtual ~AudioPlayQueue();
+
+    struct FileTimeCmp {
+        bool operator()(const PlayableAudioFile &, const PlayableAudioFile &) const;
+        bool operator()(const PlayableAudioFile *, const PlayableAudioFile *) const;
+    };
+    typedef std::set<PlayableAudioFile *, FileTimeCmp> FileSet;
+    typedef std::list<PlayableAudioFile *> FileList;
+
+    /**
+     * Add a file to the queue.  AudioPlayQueue takes ownership of the
+     * file and will delete it when removed.
+     */
+    void addScheduled(PlayableAudioFile *file);
+
+    /**
+     * Add a file to the unscheduled list.  AudioPlayQueue takes
+     * ownership of the file and will delete it when removed.
+     * Unscheduled files will be returned along with schuled ones for
+     * normal lookups, but everything will be less efficient when
+     * there are unscheduled files on the queue.  This is intended
+     * for asynchronous (preview) playback.
+     */
+    void addUnscheduled(PlayableAudioFile *file);
+
+    /**
+     * Remove a scheduled or unscheduled file from the queue and
+     * delete it.
+     */
+    void erase(PlayableAudioFile *file);
+
+    /**
+     * Remove all files and delete them.
+     */
+    void clear();
+
+    /**
+     * Return true if the queue is empty.
+     */
+    bool empty() const;
+
+    /**
+     * Return the total number of files in the queue.  (May be slow.)
+     */
+    size_t size() const;
+
+    /**
+     * Look up the files playing during a given slice and return them
+     * in the passed FileSet.  The pointers returned are still owned
+     * by me and the caller should not delete them.
+     */ 
+    void getPlayingFiles(const RealTime &sliceStart,
+                         const RealTime &sliceDuration,
+                         FileSet &) const;
+
+    /**
+     * Look up the files playing during a given slice on a given
+     * instrument and return them in the passed array.  The size arg
+     * gives the available size of the array and is used to return the
+     * number of file pointers written.  The pointers returned are
+     * still owned by me and the caller should not delete them.
+     */
+    void getPlayingFilesForInstrument(const RealTime &sliceStart,
+                                      const RealTime &sliceDuration,
+                                      InstrumentId instrumentId,
+                                      PlayableAudioFile **files,
+                                      size_t &size) const;
+
+    /**
+     * Return true if at least one scheduled or unscheduled file is
+     * associated with the given instrument somewhere in the queue.
+     */
+    bool haveFilesForInstrument(InstrumentId instrumentId) const;
+
+    /**
+     * Return a (shared reference to an) ordered set of all files on
+     * the scheduled queue.
+     */
+    const FileSet &getAllScheduledFiles() const;
+
+    /**
+     * Return a (shared reference to an) ordered set of all files on
+     * the unscheduled queue.
+     */
+    const FileList &getAllUnscheduledFiles() const;
+
+    /**
+     * Get an approximate (but always pessimistic) estimate of the
+     * number of ring buffers required for the current queue -- that
+     * is, the maximum possible number of audio channels playing at
+     * once from non-small-file-cached-files.
+     */
+    size_t getMaxBuffersRequired() const { return m_maxBuffers; }
+
+private:
+    FileSet m_files;
+
+    typedef std::vector<PlayableAudioFile *> FileVector;
+    typedef std::map<int, FileVector> ReverseFileMap;
+    ReverseFileMap m_index;
+
+    typedef std::vector<ReverseFileMap> InstrumentReverseFileMap;
+    InstrumentReverseFileMap m_instrumentIndex;
+
+    FileList m_unscheduled;
+
+    typedef std::map<int, size_t> FileCountMap;
+    FileCountMap m_counts;
+
+    size_t m_maxBuffers;
+};
+
+
+}
+
+#endif
+
diff --git a/src/sound/AudioProcess.cpp b/src/sound/AudioProcess.cpp
new file mode 100644
index 0000000..9b44e13
--- /dev/null
+++ b/src/sound/AudioProcess.cpp
@@ -0,0 +1,2463 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "AudioProcess.h"
+
+#include "RunnablePluginInstance.h"
+#include "PlayableAudioFile.h"
+#include "RecordableAudioFile.h"
+#include "WAVAudioFile.h"
+#include "MappedStudio.h"
+#include "Profiler.h"
+#include "AudioLevel.h"
+#include "AudioPlayQueue.h"
+#include "PluginFactory.h"
+
+#include <sys/time.h>
+#include <pthread.h>
+
+#include <cmath>
+
+//#define DEBUG_THREAD_CREATE_DESTROY 1
+//#define DEBUG_BUSS_MIXER 1
+//#define DEBUG_MIXER 1
+//#define DEBUG_MIXER_LIGHTWEIGHT 1
+//#define DEBUG_LOCKS 1
+//#define DEBUG_READER 1
+//#define DEBUG_WRITER 1
+
+namespace Rosegarden
+{
+
+/* Branch-free optimizer-resistant denormal killer courtesy of Simon
+   Jenkins on LAD: */
+
+static inline float flushToZero(volatile float f)
+{
+    f += 9.8607615E-32f;
+    return f - 9.8607615E-32f;
+}
+
+static inline void denormalKill(float *buffer, int size)
+{
+    for (int i = 0; i < size; ++i) {
+        buffer[i] = flushToZero(buffer[i]);
+    }
+}
+
+AudioThread::AudioThread(std::string name,
+                         SoundDriver *driver,
+                         unsigned int sampleRate) :
+        m_name(name),
+        m_driver(driver),
+        m_sampleRate(sampleRate),
+        m_thread(0),
+        m_running(false),
+        m_exiting(false)
+{
+#ifdef DEBUG_THREAD_CREATE_DESTROY
+    std::cerr << "AudioThread::AudioThread() [" << m_name << "]" << std::endl;
+#endif
+
+    pthread_mutex_t initialisingMutex = PTHREAD_MUTEX_INITIALIZER;
+    memcpy(&m_lock, &initialisingMutex, sizeof(pthread_mutex_t));
+
+    pthread_cond_t initialisingCondition = PTHREAD_COND_INITIALIZER;
+    memcpy(&m_condition, &initialisingCondition, sizeof(pthread_cond_t));
+}
+
+AudioThread::~AudioThread()
+{
+#ifdef DEBUG_THREAD_CREATE_DESTROY
+    std::cerr << "AudioThread::~AudioThread() [" << m_name << "]" << std::endl;
+#endif
+
+    if (m_thread) {
+        pthread_mutex_destroy(&m_lock);
+        m_thread = 0;
+    }
+
+#ifdef DEBUG_THREAD_CREATE_DESTROY
+    std::cerr << "AudioThread::~AudioThread() exiting" << std::endl;
+#endif
+}
+
+void
+AudioThread::run()
+{
+#ifdef DEBUG_THREAD_CREATE_DESTROY
+    std::cerr << m_name << "::run()" << std::endl;
+#endif
+
+    pthread_attr_t attr;
+    pthread_attr_init(&attr);
+
+    int priority = getPriority();
+
+    if (priority > 0) {
+
+        if (pthread_attr_setschedpolicy(&attr, SCHED_FIFO)) {
+
+            std::cerr << m_name << "::run: WARNING: couldn't set FIFO scheduling "
+            << "on new thread" << std::endl;
+            pthread_attr_init(&attr); // reset to safety
+
+        } else {
+
+            struct sched_param param;
+            memset(&param, 0, sizeof(struct sched_param));
+            param.sched_priority = priority;
+
+            if (pthread_attr_setschedparam(&attr, &param)) {
+                std::cerr << m_name << "::run: WARNING: couldn't set priority "
+                << priority << " on new thread" << std::endl;
+                pthread_attr_init(&attr); // reset to safety
+            }
+        }
+    }
+
+    pthread_attr_setstacksize(&attr, 1048576);
+    int rv = pthread_create(&m_thread, &attr, staticThreadRun, this);
+
+    if (rv != 0 && priority > 0) {
+#ifdef DEBUG_THREAD_CREATE_DESTROY
+        std::cerr << m_name << "::run: WARNING: unable to start RT thread;"
+        << "\ntrying again with normal scheduling" << std::endl;
+#endif
+
+        pthread_attr_init(&attr);
+        pthread_attr_setstacksize(&attr, 1048576);
+        rv = pthread_create(&m_thread, &attr, staticThreadRun, this);
+    }
+
+    if (rv != 0) {
+        // This is quite fatal.
+        std::cerr << m_name << "::run: ERROR: failed to start thread!" << std::endl;
+        ::exit(1);
+    }
+
+    m_running = true;
+
+#ifdef DEBUG_THREAD_CREATE_DESTROY
+
+    std::cerr << m_name << "::run() done" << std::endl;
+#endif
+}
+
+void
+AudioThread::terminate()
+{
+#ifdef DEBUG_THREAD_CREATE_DESTROY
+    std::string name = m_name;
+    std::cerr << name << "::terminate()" << std::endl;
+#endif
+
+    m_running = false;
+
+    if (m_thread) {
+
+        pthread_cancel(m_thread);
+
+#ifdef DEBUG_THREAD_CREATE_DESTROY
+
+        std::cerr << name << "::terminate(): cancel requested" << std::endl;
+#endif
+
+        int rv = pthread_join(m_thread, 0);
+
+#ifdef DEBUG_THREAD_CREATE_DESTROY
+
+        std::cerr << name << "::terminate(): thread exited with return value " << rv << std::endl;
+#endif
+
+    }
+
+#ifdef DEBUG_THREAD_CREATE_DESTROY
+    std::cerr << name << "::terminate(): done" << std::endl;
+#endif
+}
+
+void *
+AudioThread::staticThreadRun(void *arg)
+{
+    AudioThread *inst = static_cast<AudioThread *>(arg);
+    if (!inst)
+        return 0;
+
+    pthread_cleanup_push(staticThreadCleanup, arg);
+
+    inst->getLock();
+    inst->m_exiting = false;
+    inst->threadRun();
+
+#ifdef DEBUG_THREAD_CREATE_DESTROY
+
+    std::cerr << inst->m_name << "::staticThreadRun(): threadRun exited" << std::endl;
+#endif
+
+    inst->releaseLock();
+    pthread_cleanup_pop(0);
+
+    return 0;
+}
+
+void
+AudioThread::staticThreadCleanup(void *arg)
+{
+    AudioThread *inst = static_cast<AudioThread *>(arg);
+    if (!inst || inst->m_exiting)
+        return ;
+
+#ifdef DEBUG_THREAD_CREATE_DESTROY
+
+    std::string name = inst->m_name;
+    std::cerr << name << "::staticThreadCleanup()" << std::endl;
+#endif
+
+    inst->m_exiting = true;
+    inst->releaseLock();
+
+#ifdef DEBUG_THREAD_CREATE_DESTROY
+
+    std::cerr << name << "::staticThreadCleanup() done" << std::endl;
+#endif
+}
+
+int
+AudioThread::getLock()
+{
+    int rv;
+#ifdef DEBUG_LOCKS
+
+    std::cerr << m_name << "::getLock()" << std::endl;
+#endif
+
+    rv = pthread_mutex_lock(&m_lock);
+#ifdef DEBUG_LOCKS
+
+    std::cerr << "OK" << std::endl;
+#endif
+
+    return rv;
+}
+
+int
+AudioThread::tryLock()
+{
+    int rv;
+#ifdef DEBUG_LOCKS
+
+    std::cerr << m_name << "::tryLock()" << std::endl;
+#endif
+
+    rv = pthread_mutex_trylock(&m_lock);
+#ifdef DEBUG_LOCKS
+
+    std::cerr << "OK (rv is " << rv << ")" << std::endl;
+#endif
+
+    return rv;
+}
+
+int
+AudioThread::releaseLock()
+{
+    int rv;
+#ifdef DEBUG_LOCKS
+
+    std::cerr << m_name << "::releaseLock()" << std::endl;
+#endif
+
+    rv = pthread_mutex_unlock(&m_lock);
+#ifdef DEBUG_LOCKS
+
+    std::cerr << "OK" << std::endl;
+#endif
+
+    return rv;
+}
+
+void
+AudioThread::signal()
+{
+#ifdef DEBUG_LOCKS
+    std::cerr << m_name << "::signal()" << std::endl;
+#endif
+
+    pthread_cond_signal(&m_condition);
+}
+
+
+AudioBussMixer::AudioBussMixer(SoundDriver *driver,
+                               AudioInstrumentMixer *instrumentMixer,
+                               unsigned int sampleRate,
+                               unsigned int blockSize) :
+        AudioThread("AudioBussMixer", driver, sampleRate),
+        m_instrumentMixer(instrumentMixer),
+        m_blockSize(blockSize),
+        m_bussCount(0)
+{
+    // nothing else here
+}
+
+AudioBussMixer::~AudioBussMixer()
+{
+    for (unsigned int i = 0; i < m_processBuffers.size(); ++i) {
+        delete[] m_processBuffers[i];
+    }
+}
+
+AudioBussMixer::BufferRec::~BufferRec()
+{
+    for (size_t i = 0; i < buffers.size(); ++i)
+        delete buffers[i];
+}
+
+void
+AudioBussMixer::generateBuffers()
+{
+    // Not RT safe
+
+#ifdef DEBUG_BUSS_MIXER
+    std::cerr << "AudioBussMixer::generateBuffers" << std::endl;
+#endif
+
+    // This returns one too many, as the master is counted as buss 0
+    m_bussCount =
+        m_driver->getMappedStudio()->getObjectCount(MappedStudio::AudioBuss) - 1;
+
+#ifdef DEBUG_BUSS_MIXER
+
+    std::cerr << "AudioBussMixer::generateBuffers: have " << m_bussCount << " busses" << std::endl;
+#endif
+
+    int bufferSamples = m_blockSize;
+
+    if (!m_driver->getLowLatencyMode()) {
+        RealTime bufferLength = m_driver->getAudioMixBufferLength();
+        int bufferSamples = RealTime::realTime2Frame(bufferLength, m_sampleRate);
+        bufferSamples = ((bufferSamples / m_blockSize) + 1) * m_blockSize;
+    }
+
+    for (int i = 0; i < m_bussCount; ++i) {
+
+        BufferRec &rec = m_bufferMap[i];
+
+        if (rec.buffers.size() == 2)
+            continue;
+
+        for (unsigned int ch = 0; ch < 2; ++ch) {
+            RingBuffer<sample_t> *rb = new RingBuffer<sample_t>(bufferSamples);
+            if (!rb->mlock()) {
+                //		std::cerr << "WARNING: AudioBussMixer::generateBuffers: couldn't lock ring buffer into real memory, performance may be impaired" << std::endl;
+            }
+            rec.buffers.push_back(rb);
+        }
+
+        MappedAudioBuss *mbuss =
+            m_driver->getMappedStudio()->getAudioBuss(i + 1); // master is 0
+
+        if (mbuss) {
+
+            float level = 0.0;
+            (void)mbuss->getProperty(MappedAudioBuss::Level, level);
+
+            float pan = 0.0;
+            (void)mbuss->getProperty(MappedAudioBuss::Pan, pan);
+
+            setBussLevels(i + 1, level, pan);
+        }
+    }
+
+    if (m_processBuffers.size() == 0) {
+        m_processBuffers.push_back(new sample_t[m_blockSize]);
+        m_processBuffers.push_back(new sample_t[m_blockSize]);
+    }
+}
+
+void
+AudioBussMixer::fillBuffers(const RealTime &currentTime)
+{
+    // Not RT safe
+
+#ifdef DEBUG_BUSS_MIXER
+    std::cerr << "AudioBussMixer::fillBuffers" << std::endl;
+#endif
+
+    emptyBuffers();
+    m_instrumentMixer->fillBuffers(currentTime);
+    kick();
+}
+
+void
+AudioBussMixer::emptyBuffers()
+{
+    // Not RT safe
+
+    getLock();
+
+#ifdef DEBUG_BUSS_MIXER
+
+    std::cerr << "AudioBussMixer::emptyBuffers" << std::endl;
+#endif
+
+    // We can't generate buffers before this, because we don't know how
+    // many busses there are
+    generateBuffers();
+
+    for (int i = 0; i < m_bussCount; ++i) {
+        m_bufferMap[i].dormant = true;
+        for (int ch = 0; ch < 2; ++ch) {
+            if (int(m_bufferMap[i].buffers.size()) > ch) {
+                m_bufferMap[i].buffers[ch]->reset();
+            }
+        }
+    }
+
+    releaseLock();
+}
+
+void
+AudioBussMixer::kick(bool wantLock, bool signalInstrumentMixer)
+{
+    // Needs to be RT safe if wantLock is not specified
+
+    if (wantLock)
+        getLock();
+
+#ifdef DEBUG_BUSS_MIXER
+
+    std::cerr << "AudioBussMixer::kick" << std::endl;
+#endif
+
+    processBlocks();
+
+#ifdef DEBUG_BUSS_MIXER
+
+    std::cerr << "AudioBussMixer::kick: processed" << std::endl;
+#endif
+
+    if (wantLock)
+        releaseLock();
+
+    if (signalInstrumentMixer) {
+        m_instrumentMixer->signal();
+    }
+}
+
+void
+AudioBussMixer::setBussLevels(int bussId, float dB, float pan)
+{
+    // No requirement to be RT safe
+
+    if (bussId == 0)
+        return ; // master
+    int buss = bussId - 1;
+
+    BufferRec &rec = m_bufferMap[buss];
+
+    float volume = AudioLevel::dB_to_multiplier(dB);
+
+    rec.gainLeft = volume * ((pan > 0.0) ? (1.0 - (pan / 100.0)) : 1.0);
+    rec.gainRight = volume * ((pan < 0.0) ? ((pan + 100.0) / 100.0) : 1.0);
+}
+
+void
+AudioBussMixer::updateInstrumentConnections()
+{
+    // Not RT safe
+
+    if (m_bussCount <= 0)
+        generateBuffers();
+
+    InstrumentId audioInstrumentBase;
+    int audioInstruments;
+    m_driver->getAudioInstrumentNumbers(audioInstrumentBase, audioInstruments);
+
+    InstrumentId synthInstrumentBase;
+    int synthInstruments;
+    m_driver->getSoftSynthInstrumentNumbers(synthInstrumentBase, synthInstruments);
+
+    for (int buss = 0; buss < m_bussCount; ++buss) {
+
+        MappedAudioBuss *mbuss =
+            m_driver->getMappedStudio()->getAudioBuss(buss + 1); // master is 0
+
+        if (!mbuss) {
+#ifdef DEBUG_BUSS_MIXER
+            std::cerr << "AudioBussMixer::updateInstrumentConnections: buss " << buss << " not found" << std::endl;
+#endif
+
+            continue;
+        }
+
+        BufferRec &rec = m_bufferMap[buss];
+
+        while (int(rec.instruments.size()) < audioInstruments + synthInstruments) {
+            rec.instruments.push_back(false);
+        }
+
+        std::vector<InstrumentId> instruments = mbuss->getInstruments();
+
+        for (int i = 0; i < audioInstruments + synthInstruments; ++i) {
+
+            InstrumentId id;
+            if (i < audioInstruments)
+                id = audioInstrumentBase + i;
+            else
+                id = synthInstrumentBase + (i - audioInstruments);
+
+            size_t j = 0;
+            for (j = 0; j < instruments.size(); ++j) {
+                if (instruments[j] == id) {
+                    rec.instruments[i] = true;
+                    break;
+                }
+            }
+            if (j == instruments.size())
+                rec.instruments[i] = false;
+        }
+    }
+}
+
+void
+AudioBussMixer::processBlocks()
+{
+    // Needs to be RT safe
+
+    if (m_bussCount == 0)
+        return ;
+
+#ifdef DEBUG_BUSS_MIXER
+
+    if (m_driver->isPlaying())
+        std::cerr << "AudioBussMixer::processBlocks" << std::endl;
+#endif
+
+    InstrumentId audioInstrumentBase;
+    int audioInstruments;
+    m_driver->getAudioInstrumentNumbers(audioInstrumentBase, audioInstruments);
+
+    InstrumentId synthInstrumentBase;
+    int synthInstruments;
+    m_driver->getSoftSynthInstrumentNumbers(synthInstrumentBase, synthInstruments);
+
+    bool *processedInstruments = (bool *)alloca
+                                 ((audioInstruments + synthInstruments) * sizeof(bool));
+
+    for (int i = 0; i < audioInstruments + synthInstruments; ++i) {
+        processedInstruments[i] = false;
+    }
+
+    int minBlocks = 0;
+    bool haveMinBlocks = false;
+
+    for (int buss = 0; buss < m_bussCount; ++buss) {
+
+        BufferRec &rec = m_bufferMap[buss];
+
+        float gain[2];
+        gain[0] = rec.gainLeft;
+        gain[1] = rec.gainRight;
+
+        // The dormant calculation here depends on the buffer length
+        // for this mixer being the same as that for the instrument mixer
+
+        size_t minSpace = 0;
+
+        for (int ch = 0; ch < 2; ++ch) {
+
+            size_t w = rec.buffers[ch]->getWriteSpace();
+            if (ch == 0 || w < minSpace)
+                minSpace = w;
+
+#ifdef DEBUG_BUSS_MIXER
+
+            std::cerr << "AudioBussMixer::processBlocks: buss " << buss << ": write space " << w << " on channel " << ch << std::endl;
+#endif
+
+            if (minSpace == 0)
+                break;
+
+            for (int i = 0; i < audioInstruments + synthInstruments; ++i) {
+
+                // is this instrument on this buss?
+                if (int(rec.instruments.size()) <= i ||
+                        !rec.instruments[i])
+                    continue;
+
+                InstrumentId id;
+                if (i < audioInstruments)
+                    id = audioInstrumentBase + i;
+                else
+                    id = synthInstrumentBase + (i - audioInstruments);
+
+                if (m_instrumentMixer->isInstrumentEmpty(id))
+                    continue;
+
+                RingBuffer<sample_t, 2> *rb =
+                    m_instrumentMixer->getRingBuffer(id, ch);
+                if (rb) {
+                    size_t r = rb->getReadSpace(1);
+                    if (r < minSpace)
+                        minSpace = r;
+
+#ifdef DEBUG_BUSS_MIXER
+
+                    if (id == 1000) {
+                        std::cerr << "AudioBussMixer::processBlocks: buss " << buss << ": read space " << r << " on instrument " << id << ", channel " << ch << std::endl;
+                    }
+#endif
+
+                    if (minSpace == 0)
+                        break;
+                }
+            }
+
+            if (minSpace == 0)
+                break;
+        }
+
+        int blocks = minSpace / m_blockSize;
+        if (!haveMinBlocks || (blocks < minBlocks)) {
+            minBlocks = blocks;
+            haveMinBlocks = true;
+        }
+
+#ifdef DEBUG_BUSS_MIXER
+        if (m_driver->isPlaying())
+            std::cerr << "AudioBussMixer::processBlocks: doing " << blocks << " blocks at block size " << m_blockSize << std::endl;
+#endif
+
+        for (int block = 0; block < blocks; ++block) {
+
+            memset(m_processBuffers[0], 0, m_blockSize * sizeof(sample_t));
+            memset(m_processBuffers[1], 0, m_blockSize * sizeof(sample_t));
+
+            bool dormant = true;
+
+            for (int i = 0; i < audioInstruments + synthInstruments; ++i) {
+
+                // is this instrument on this buss?
+                if (int(rec.instruments.size()) <= i ||
+                        !rec.instruments[i])
+                    continue;
+
+                if (processedInstruments[i]) {
+                    // we aren't set up to process any instrument to
+                    // more than one buss
+                    continue;
+                } else {
+                    processedInstruments[i] = true;
+                }
+
+                InstrumentId id;
+                if (i < audioInstruments)
+                    id = audioInstrumentBase + i;
+                else
+                    id = synthInstrumentBase + (i - audioInstruments);
+
+                if (m_instrumentMixer->isInstrumentEmpty(id))
+                    continue;
+
+                if (m_instrumentMixer->isInstrumentDormant(id)) {
+
+                    for (int ch = 0; ch < 2; ++ch) {
+                        RingBuffer<sample_t, 2> *rb =
+                            m_instrumentMixer->getRingBuffer(id, ch);
+
+                        if (rb)
+                            rb->skip(m_blockSize,
+                                     1);
+                    }
+                } else {
+                    dormant = false;
+
+                    for (int ch = 0; ch < 2; ++ch) {
+                        RingBuffer<sample_t, 2> *rb =
+                            m_instrumentMixer->getRingBuffer(id, ch);
+
+                        if (rb)
+                            rb->readAdding(m_processBuffers[ch],
+                                           m_blockSize,
+                                           1);
+                    }
+                }
+            }
+
+            if (m_instrumentMixer) {
+                AudioInstrumentMixer::PluginList &plugins =
+                    m_instrumentMixer->getBussPlugins(buss + 1);
+
+                // This will have to do for now!
+                if (!plugins.empty())
+                    dormant = false;
+
+                for (AudioInstrumentMixer::PluginList::iterator pli =
+                            plugins.begin(); pli != plugins.end(); ++pli) {
+
+                    RunnablePluginInstance *plugin = *pli;
+                    if (!plugin || plugin->isBypassed())
+                        continue;
+
+                    unsigned int ch = 0;
+
+                    while (ch < plugin->getAudioInputCount()) {
+                        if (ch < 2) {
+                            memcpy(plugin->getAudioInputBuffers()[ch],
+                                   m_processBuffers[ch],
+                                   m_blockSize * sizeof(sample_t));
+                        } else {
+                            memset(plugin->getAudioInputBuffers()[ch], 0,
+                                   m_blockSize * sizeof(sample_t));
+                        }
+                        ++ch;
+                    }
+
+#ifdef DEBUG_BUSS_MIXER
+                    std::cerr << "Running buss plugin with " << plugin->getAudioInputCount()
+                    << " inputs, " << plugin->getAudioOutputCount() << " outputs" << std::endl;
+#endif
+
+                    // We don't currently maintain a record of our
+                    // frame time in the buss mixer.  This will screw
+                    // up any plugin that requires a good frame count:
+                    // at the moment that only means DSSI effects
+                    // plugins using run_multiple_synths, which would
+                    // be an unusual although plausible combination
+                    plugin->run(RealTime::zeroTime);
+
+                    ch = 0;
+
+                    while (ch < 2 && ch < plugin->getAudioOutputCount()) {
+
+                        denormalKill(plugin->getAudioOutputBuffers()[ch],
+                                     m_blockSize);
+
+                        memcpy(m_processBuffers[ch],
+                               plugin->getAudioOutputBuffers()[ch],
+                               m_blockSize * sizeof(sample_t));
+
+                        ++ch;
+                    }
+                }
+            }
+
+            for (int ch = 0; ch < 2; ++ch) {
+                if (dormant) {
+                    rec.buffers[ch]->zero(m_blockSize);
+                } else {
+                    for (size_t j = 0; j < m_blockSize; ++j) {
+                        m_processBuffers[ch][j] *= gain[ch];
+                    }
+                    rec.buffers[ch]->write(m_processBuffers[ch], m_blockSize);
+                }
+            }
+
+            rec.dormant = dormant;
+
+#ifdef DEBUG_BUSS_MIXER
+
+            if (m_driver->isPlaying())
+                std::cerr << "AudioBussMixer::processBlocks: buss " << buss << (dormant ? " dormant" : " not dormant") << std::endl;
+#endif
+
+        }
+    }
+
+    // any unprocessed instruments need to be skipped, or they'll block
+
+    for (int i = 0; i < audioInstruments + synthInstruments; ++i) {
+
+        if (processedInstruments[i])
+            continue;
+
+        InstrumentId id;
+        if (i < audioInstruments)
+            id = audioInstrumentBase + i;
+        else
+            id = synthInstrumentBase + (i - audioInstruments);
+
+        if (m_instrumentMixer->isInstrumentEmpty(id))
+            continue;
+
+        for (int ch = 0; ch < 2; ++ch) {
+            RingBuffer<sample_t, 2> *rb =
+                m_instrumentMixer->getRingBuffer(id, ch);
+
+            if (rb)
+                rb->skip(m_blockSize * minBlocks,
+                         1);
+        }
+    }
+
+
+#ifdef DEBUG_BUSS_MIXER
+    std::cerr << "AudioBussMixer::processBlocks: done" << std::endl;
+#endif
+}
+
+void
+AudioBussMixer::threadRun()
+{
+    while (!m_exiting) {
+
+        if (m_driver->areClocksRunning()) {
+            kick(false);
+        }
+
+        RealTime t = m_driver->getAudioMixBufferLength();
+        t = t / 2;
+        if (t < RealTime(0, 10000000))
+            t = RealTime(0, 10000000); // 10ms minimum
+
+        struct timeval now;
+        gettimeofday(&now, 0);
+        t = t + RealTime(now.tv_sec, now.tv_usec * 1000);
+
+        struct timespec timeout;
+        timeout.tv_sec = t.sec;
+        timeout.tv_nsec = t.nsec;
+
+        pthread_cond_timedwait(&m_condition, &m_lock, &timeout);
+        pthread_testcancel();
+    }
+}
+
+
+AudioInstrumentMixer::AudioInstrumentMixer(SoundDriver *driver,
+        AudioFileReader *fileReader,
+        unsigned int sampleRate,
+        unsigned int blockSize) :
+        AudioThread("AudioInstrumentMixer", driver, sampleRate),
+        m_fileReader(fileReader),
+        m_bussMixer(0),
+        m_blockSize(blockSize)
+{
+    // Pregenerate empty plugin slots
+
+    InstrumentId audioInstrumentBase;
+    int audioInstruments;
+    m_driver->getAudioInstrumentNumbers(audioInstrumentBase, audioInstruments);
+
+    InstrumentId synthInstrumentBase;
+    int synthInstruments;
+    m_driver->getSoftSynthInstrumentNumbers(synthInstrumentBase, synthInstruments);
+
+    for (int i = 0; i < audioInstruments + synthInstruments; ++i) {
+
+        InstrumentId id;
+        if (i < audioInstruments)
+            id = audioInstrumentBase + i;
+        else
+            id = synthInstrumentBase + (i - audioInstruments);
+
+        PluginList &list = m_plugins[id];
+        for (int j = 0; j < int(Instrument::PLUGIN_COUNT); ++j) {
+            list.push_back(0);
+        }
+
+        if (i >= audioInstruments) {
+            m_synths[id] = 0;
+        }
+    }
+
+    // Leave the buffer map and process buffer list empty for now.
+    // The buffer length can change between plays, so we always
+    // examine the buffers in fillBuffers and are prepared to
+    // regenerate from scratch if necessary.  Don't like it though.
+}
+
+AudioInstrumentMixer::~AudioInstrumentMixer()
+{
+    std::cerr << "AudioInstrumentMixer::~AudioInstrumentMixer" << std::endl;
+    // BufferRec dtor will handle the BufferMap
+
+    removeAllPlugins();
+
+    for (std::vector<sample_t *>::iterator i = m_processBuffers.begin();
+            i != m_processBuffers.end(); ++i) {
+        delete[] *i;
+    }
+
+    std::cerr << "AudioInstrumentMixer::~AudioInstrumentMixer exiting" << std::endl;
+}
+
+AudioInstrumentMixer::BufferRec::~BufferRec()
+{
+    for (size_t i = 0; i < buffers.size(); ++i)
+        delete buffers[i];
+}
+
+
+void
+AudioInstrumentMixer::setPlugin(InstrumentId id, int position, QString identifier)
+{
+    // Not RT safe
+
+    std::cerr << "AudioInstrumentMixer::setPlugin(" << id << ", " << position << ", " << identifier << ")" << std::endl;
+
+    int channels = 2;
+    if (m_bufferMap.find(id) != m_bufferMap.end()) {
+        channels = m_bufferMap[id].channels;
+    }
+
+    RunnablePluginInstance *instance = 0;
+
+    PluginFactory *factory = PluginFactory::instanceFor(identifier);
+    if (factory) {
+        instance = factory->instantiatePlugin(identifier,
+                                              id,
+                                              position,
+                                              m_sampleRate,
+                                              m_blockSize,
+                                              channels);
+        if (instance && !instance->isOK()) {
+            std::cerr << "AudioInstrumentMixer::setPlugin(" << id << ", " << position
+            << ": instance is not OK" << std::endl;
+            delete instance;
+            instance = 0;
+        }
+    } else {
+        std::cerr << "AudioInstrumentMixer::setPlugin: No factory for identifier "
+        << identifier << std::endl;
+    }
+
+    RunnablePluginInstance *oldInstance = 0;
+
+    if (position == int(Instrument::SYNTH_PLUGIN_POSITION)) {
+
+        oldInstance = m_synths[id];
+        m_synths[id] = instance;
+
+    } else {
+
+        PluginList &list = m_plugins[id];
+
+        if (position < Instrument::PLUGIN_COUNT) {
+            while (position >= (int)list.size()) {
+                list.push_back(0);
+            }
+            oldInstance = list[position];
+            list[position] = instance;
+        } else {
+            std::cerr << "AudioInstrumentMixer::setPlugin: No position "
+            << position << " for instrument " << id << std::endl;
+            delete instance;
+        }
+    }
+
+    if (oldInstance) {
+        m_driver->claimUnwantedPlugin(oldInstance);
+    }
+}
+
+void
+AudioInstrumentMixer::removePlugin(InstrumentId id, int position)
+{
+    // Not RT safe
+
+    std::cerr << "AudioInstrumentMixer::removePlugin(" << id << ", " << position << ")" << std::endl;
+
+    RunnablePluginInstance *oldInstance = 0;
+
+    if (position == int(Instrument::SYNTH_PLUGIN_POSITION)) {
+
+        if (m_synths[id]) {
+            oldInstance = m_synths[id];
+            m_synths[id] = 0;
+        }
+
+    } else {
+
+        PluginList &list = m_plugins[id];
+        if (position < (int)list.size()) {
+            oldInstance = list[position];
+            list[position] = 0;
+        }
+    }
+
+    if (oldInstance) {
+        m_driver->claimUnwantedPlugin(oldInstance);
+    }
+}
+
+void
+AudioInstrumentMixer::removeAllPlugins()
+{
+    // Not RT safe
+
+    std::cerr << "AudioInstrumentMixer::removeAllPlugins" << std::endl;
+
+    for (SynthPluginMap::iterator i = m_synths.begin();
+            i != m_synths.end(); ++i) {
+        if (i->second) {
+            RunnablePluginInstance *instance = i->second;
+            i->second = 0;
+            m_driver->claimUnwantedPlugin(instance);
+        }
+    }
+
+    for (PluginMap::iterator j = m_plugins.begin();
+            j != m_plugins.end(); ++j) {
+
+        PluginList &list = j->second;
+
+        for (PluginList::iterator i = list.begin(); i != list.end(); ++i) {
+            RunnablePluginInstance *instance = *i;
+            *i = 0;
+            m_driver->claimUnwantedPlugin(instance);
+        }
+    }
+}
+
+
+RunnablePluginInstance *
+AudioInstrumentMixer::getPluginInstance(InstrumentId id, int position)
+{
+    // Not RT safe
+
+    if (position == int(Instrument::SYNTH_PLUGIN_POSITION)) {
+        return m_synths[id];
+    } else {
+        PluginList &list = m_plugins[id];
+        if (position < int(list.size()))
+            return list[position];
+    }
+    return 0;
+}
+
+
+void
+AudioInstrumentMixer::setPluginPortValue(InstrumentId id, int position,
+        unsigned int port, float value)
+{
+    // Not RT safe
+
+    RunnablePluginInstance *instance = getPluginInstance(id, position);
+
+    if (instance) {
+        instance->setPortValue(port, value);
+    }
+}
+
+float
+AudioInstrumentMixer::getPluginPortValue(InstrumentId id, int position,
+        unsigned int port)
+{
+    // Not RT safe
+
+    RunnablePluginInstance *instance = getPluginInstance(id, position);
+
+    if (instance) {
+        return instance->getPortValue(port);
+    }
+
+    return 0;
+}
+
+void
+AudioInstrumentMixer::setPluginBypass(InstrumentId id, int position, bool bypass)
+{
+    // Not RT safe
+
+    RunnablePluginInstance *instance = getPluginInstance(id, position);
+    if (instance)
+        instance->setBypassed(bypass);
+}
+
+QStringList
+AudioInstrumentMixer::getPluginPrograms(InstrumentId id, int position)
+{
+    // Not RT safe
+
+    QStringList programs;
+    RunnablePluginInstance *instance = getPluginInstance(id, position);
+    if (instance)
+        programs = instance->getPrograms();
+    return programs;
+}
+
+QString
+AudioInstrumentMixer::getPluginProgram(InstrumentId id, int position)
+{
+    // Not RT safe
+
+    QString program;
+    RunnablePluginInstance *instance = getPluginInstance(id, position);
+    if (instance)
+        program = instance->getCurrentProgram();
+    return program;
+}
+
+QString
+AudioInstrumentMixer::getPluginProgram(InstrumentId id, int position, int bank,
+                                       int program)
+{
+    // Not RT safe
+
+    QString programName;
+    RunnablePluginInstance *instance = getPluginInstance(id, position);
+    if (instance)
+        programName = instance->getProgram(bank, program);
+    return programName;
+}
+
+unsigned long
+AudioInstrumentMixer::getPluginProgram(InstrumentId id, int position, QString name)
+{
+    // Not RT safe
+
+    unsigned long program = 0;
+    RunnablePluginInstance *instance = getPluginInstance(id, position);
+    if (instance)
+        program = instance->getProgram(name);
+    return program;
+}
+
+void
+AudioInstrumentMixer::setPluginProgram(InstrumentId id, int position, QString program)
+{
+    // Not RT safe
+
+    RunnablePluginInstance *instance = getPluginInstance(id, position);
+    if (instance)
+        instance->selectProgram(program);
+}
+
+QString
+AudioInstrumentMixer::configurePlugin(InstrumentId id, int position, QString key, QString value)
+{
+    // Not RT safe
+
+    RunnablePluginInstance *instance = getPluginInstance(id, position);
+    if (instance)
+        return instance->configure(key, value);
+    return QString();
+}
+
+void
+AudioInstrumentMixer::discardPluginEvents()
+{
+    getLock();
+    if (m_bussMixer) m_bussMixer->getLock();
+
+    for (SynthPluginMap::iterator j = m_synths.begin();
+            j != m_synths.end(); ++j) {
+
+        RunnablePluginInstance *instance = j->second;
+        if (instance) instance->discardEvents();
+    }
+
+    for (PluginMap::iterator j = m_plugins.begin();
+            j != m_plugins.end(); ++j) {
+
+        InstrumentId id = j->first;
+
+        for (PluginList::iterator i = m_plugins[id].begin();
+	     i != m_plugins[id].end(); ++i) {
+
+            RunnablePluginInstance *instance = *i;
+	    if (instance) instance->discardEvents();
+        }
+    }
+
+    if (m_bussMixer) m_bussMixer->releaseLock();
+    releaseLock();
+}
+
+void
+AudioInstrumentMixer::resetAllPlugins(bool discardEvents)
+{
+    // Not RT safe
+
+    // lock required here to protect against calling
+    // activate/deactivate at the same time as run()
+
+#ifdef DEBUG_MIXER
+    std::cerr << "AudioInstrumentMixer::resetAllPlugins!" << std::endl;
+    if (discardEvents) std::cerr << "(discardEvents true)" << std::endl;
+#endif
+
+    getLock();
+    if (m_bussMixer)
+        m_bussMixer->getLock();
+
+    for (SynthPluginMap::iterator j = m_synths.begin();
+            j != m_synths.end(); ++j) {
+
+        InstrumentId id = j->first;
+
+        int channels = 2;
+        if (m_bufferMap.find(id) != m_bufferMap.end()) {
+            channels = m_bufferMap[id].channels;
+        }
+
+        RunnablePluginInstance *instance = j->second;
+
+        if (instance) {
+#ifdef DEBUG_MIXER
+            std::cerr << "AudioInstrumentMixer::resetAllPlugins: (re)setting " << channels << " channels on synth for instrument " << id << std::endl;
+#endif
+
+            if (discardEvents)
+                instance->discardEvents();
+            instance->setIdealChannelCount(channels);
+        }
+    }
+
+    for (PluginMap::iterator j = m_plugins.begin();
+            j != m_plugins.end(); ++j) {
+
+        InstrumentId id = j->first;
+
+        int channels = 2;
+        if (m_bufferMap.find(id) != m_bufferMap.end()) {
+            channels = m_bufferMap[id].channels;
+        }
+
+        for (PluginList::iterator i = m_plugins[id].begin();
+                i != m_plugins[id].end(); ++i) {
+
+            RunnablePluginInstance *instance = *i;
+
+            if (instance) {
+#ifdef DEBUG_MIXER
+                std::cerr << "AudioInstrumentMixer::resetAllPlugins: (re)setting " << channels << " channels on plugin for instrument " << id << std::endl;
+#endif
+
+                if (discardEvents)
+                    instance->discardEvents();
+                instance->setIdealChannelCount(channels);
+            }
+        }
+    }
+
+    if (m_bussMixer)
+        m_bussMixer->releaseLock();
+    releaseLock();
+}
+
+void
+AudioInstrumentMixer::destroyAllPlugins()
+{
+    // Not RT safe
+
+    getLock();
+    if (m_bussMixer)
+        m_bussMixer->getLock();
+
+    // Delete immediately, as we're probably exiting here -- don't use
+    // the scavenger.
+
+    std::cerr << "AudioInstrumentMixer::destroyAllPlugins" << std::endl;
+
+    for (SynthPluginMap::iterator j = m_synths.begin();
+            j != m_synths.end(); ++j) {
+        RunnablePluginInstance *instance = j->second;
+        j->second = 0;
+        delete instance;
+    }
+
+    for (PluginMap::iterator j = m_plugins.begin();
+            j != m_plugins.end(); ++j) {
+
+        InstrumentId id = j->first;
+
+        for (PluginList::iterator i = m_plugins[id].begin();
+                i != m_plugins[id].end(); ++i) {
+
+            RunnablePluginInstance *instance = *i;
+            *i = 0;
+            delete instance;
+        }
+    }
+
+    // and tell the driver to get rid of anything already scavenged.
+    m_driver->scavengePlugins();
+
+    if (m_bussMixer)
+        m_bussMixer->releaseLock();
+    releaseLock();
+}
+
+size_t
+AudioInstrumentMixer::getPluginLatency(unsigned int id)
+{
+    // Not RT safe
+
+    size_t latency = 0;
+
+    RunnablePluginInstance *synth = m_synths[id];
+    if (synth)
+        latency += m_synths[id]->getLatency();
+
+    for (PluginList::iterator i = m_plugins[id].begin();
+            i != m_plugins[id].end(); ++i) {
+        RunnablePluginInstance *plugin = *i;
+        if (plugin)
+            latency += plugin->getLatency();
+    }
+
+    return latency;
+}
+
+void
+AudioInstrumentMixer::generateBuffers()
+{
+    // Not RT safe
+
+    InstrumentId audioInstrumentBase;
+    int audioInstruments;
+    m_driver->getAudioInstrumentNumbers(audioInstrumentBase, audioInstruments);
+
+    InstrumentId synthInstrumentBase;
+    int synthInstruments;
+    m_driver->getSoftSynthInstrumentNumbers(synthInstrumentBase, synthInstruments);
+
+    unsigned int maxChannels = 0;
+
+    int bufferSamples = m_blockSize;
+
+    if (!m_driver->getLowLatencyMode()) {
+        RealTime bufferLength = m_driver->getAudioMixBufferLength();
+        int bufferSamples = RealTime::realTime2Frame(bufferLength, m_sampleRate);
+        bufferSamples = ((bufferSamples / m_blockSize) + 1) * m_blockSize;
+#ifdef DEBUG_MIXER
+
+        std::cerr << "AudioInstrumentMixer::generateBuffers: Buffer length is " << bufferLength << "; buffer samples " << bufferSamples << " (sample rate " << m_sampleRate << ")" << std::endl;
+#endif
+
+    }
+
+    for (int i = 0; i < audioInstruments + synthInstruments; ++i) {
+
+        InstrumentId id;
+        if (i < audioInstruments)
+            id = audioInstrumentBase + i;
+        else
+            id = synthInstrumentBase + (i - audioInstruments);
+
+        // Get a fader for this instrument - if we can't then this
+        // isn't a valid audio track.
+        MappedAudioFader *fader = m_driver->getMappedStudio()->getAudioFader(id);
+
+        if (!fader) {
+#ifdef DEBUG_MIXER
+            std::cerr << "AudioInstrumentMixer::generateBuffers: no fader for audio instrument " << id << std::endl;
+#endif
+
+            continue;
+        }
+
+        float fch = 2;
+        (void)fader->getProperty(MappedAudioFader::Channels, fch);
+        unsigned int channels = (unsigned int)fch;
+
+        BufferRec &rec = m_bufferMap[id];
+
+        rec.channels = channels;
+
+        // We always have stereo buffers (for output of pan)
+        // even on a mono instrument.
+        if (channels < 2)
+            channels = 2;
+        if (channels > maxChannels)
+            maxChannels = channels;
+
+        bool replaceBuffers = (rec.buffers.size() > channels);
+
+        if (!replaceBuffers) {
+            for (size_t i = 0; i < rec.buffers.size(); ++i) {
+                if (rec.buffers[i]->getSize() != bufferSamples) {
+                    replaceBuffers = true;
+                    break;
+                }
+            }
+        }
+
+        if (replaceBuffers) {
+            for (size_t i = 0; i < rec.buffers.size(); ++i) {
+                delete rec.buffers[i];
+            }
+            rec.buffers.clear();
+        }
+
+        while (rec.buffers.size() < channels) {
+
+            // All our ringbuffers are set up for two readers: the
+            // buss mix thread and the main process thread for
+            // e.g. JACK.  The main process thread gets the zero-id
+            // reader, so it gets the same API as if this was a
+            // single-reader buffer; the buss mixer has to remember to
+            // explicitly request reader 1.
+
+            RingBuffer<sample_t, 2> *rb =
+                new RingBuffer<sample_t, 2>(bufferSamples);
+
+            if (!rb->mlock()) {
+                //		std::cerr << "WARNING: AudioInstrumentMixer::generateBuffers: couldn't lock ring buffer into real memory, performance may be impaired" << std::endl;
+            }
+            rec.buffers.push_back(rb);
+        }
+
+        float level = 0.0;
+        (void)fader->getProperty(MappedAudioFader::FaderLevel, level);
+
+        float pan = 0.0;
+        (void)fader->getProperty(MappedAudioFader::Pan, pan);
+
+        setInstrumentLevels(id, level, pan);
+    }
+
+    // Make room for up to 16 busses here, to avoid reshuffling later
+    int busses = 16;
+    if (m_bussMixer)
+        busses = std::max(busses, m_bussMixer->getBussCount());
+    for (int i = 0; i < busses; ++i) {
+        PluginList &list = m_plugins[i + 1];
+        while (list.size() < Instrument::PLUGIN_COUNT) {
+            list.push_back(0);
+        }
+    }
+
+    while (m_processBuffers.size() > maxChannels) {
+        std::vector<sample_t *>::iterator bi = m_processBuffers.end();
+        --bi;
+        delete[] *bi;
+        m_processBuffers.erase(bi);
+    }
+    while (m_processBuffers.size() < maxChannels) {
+        m_processBuffers.push_back(new sample_t[m_blockSize]);
+    }
+}
+
+void
+AudioInstrumentMixer::fillBuffers(const RealTime &currentTime)
+{
+    // Not RT safe
+
+    emptyBuffers(currentTime);
+
+    getLock();
+
+#ifdef DEBUG_MIXER
+
+    std::cerr << "AudioInstrumentMixer::fillBuffers(" << currentTime << ")" << std::endl;
+#endif
+
+    bool discard;
+    processBlocks(discard);
+
+    releaseLock();
+}
+
+void
+AudioInstrumentMixer::allocateBuffers()
+{
+    // Not RT safe
+
+    getLock();
+
+#ifdef DEBUG_MIXER
+
+    std::cerr << "AudioInstrumentMixer::allocateBuffers()" << std::endl;
+#endif
+
+    generateBuffers();
+
+    releaseLock();
+}
+
+void
+AudioInstrumentMixer::emptyBuffers(RealTime currentTime)
+{
+    // Not RT safe
+
+    getLock();
+
+#ifdef DEBUG_MIXER
+
+    std::cerr << "AudioInstrumentMixer::emptyBuffers(" << currentTime << ")" << std::endl;
+#endif
+
+    generateBuffers();
+
+    InstrumentId audioInstrumentBase;
+    int audioInstruments;
+    m_driver->getAudioInstrumentNumbers(audioInstrumentBase, audioInstruments);
+
+    InstrumentId synthInstrumentBase;
+    int synthInstruments;
+    m_driver->getSoftSynthInstrumentNumbers(synthInstrumentBase, synthInstruments);
+
+    for (int i = 0; i < audioInstruments + synthInstruments; ++i) {
+
+        InstrumentId id;
+        if (i < audioInstruments)
+            id = audioInstrumentBase + i;
+        else
+            id = synthInstrumentBase + (i - audioInstruments);
+
+        m_bufferMap[id].dormant = true;
+        m_bufferMap[id].muted = false;
+        m_bufferMap[id].zeroFrames = 0;
+        m_bufferMap[id].filledTo = currentTime;
+
+        for (size_t i = 0; i < m_bufferMap[id].buffers.size(); ++i) {
+            m_bufferMap[id].buffers[i]->reset();
+        }
+    }
+
+    releaseLock();
+}
+
+void
+AudioInstrumentMixer::setInstrumentLevels(InstrumentId id, float dB, float pan)
+{
+    // No requirement to be RT safe
+
+    BufferRec &rec = m_bufferMap[id];
+
+    float volume = AudioLevel::dB_to_multiplier(dB);
+
+    rec.gainLeft = volume * ((pan > 0.0) ? (1.0 - (pan / 100.0)) : 1.0);
+    rec.gainRight = volume * ((pan < 0.0) ? ((pan + 100.0) / 100.0) : 1.0);
+    rec.volume = volume;
+}
+
+void
+AudioInstrumentMixer::updateInstrumentMuteStates()
+{
+    SequencerDataBlock *sdb = m_driver->getSequencerDataBlock();
+    if (sdb) {
+        ControlBlock *cb = sdb->getControlBlock();
+        if (cb) {
+
+            for (BufferMap::iterator i = m_bufferMap.begin();
+                    i != m_bufferMap.end(); ++i) {
+
+                InstrumentId id = i->first;
+                BufferRec &rec = i->second;
+
+                if (id >= SoftSynthInstrumentBase) {
+                    rec.muted = cb->isInstrumentMuted(id);
+                } else {
+                    rec.muted = cb->isInstrumentUnused(id);
+                }
+            }
+        }
+    }
+}
+
+void
+AudioInstrumentMixer::processBlocks(bool &readSomething)
+{
+    // Needs to be RT safe
+
+#ifdef DEBUG_MIXER
+    if (m_driver->isPlaying())
+        std::cerr << "AudioInstrumentMixer::processBlocks" << std::endl;
+#endif
+
+    //    Profiler profiler("processBlocks", true);
+
+    const AudioPlayQueue *queue = m_driver->getAudioQueue();
+
+    for (BufferMap::iterator i = m_bufferMap.begin();
+            i != m_bufferMap.end(); ++i) {
+
+        InstrumentId id = i->first;
+        BufferRec &rec = i->second;
+
+        // This "muted" flag actually only strictly means muted when
+        // applied to synth instruments.  For audio instruments it's
+        // only true if the instrument is not in use at all (see
+        // updateInstrumentMuteStates above).  It's not safe to base
+        // the empty calculation on muted state for audio tracks,
+        // because that causes buffering problems when the mute is
+        // toggled for an audio track while it's playing a file.
+
+        bool empty = false;
+
+        if (rec.muted) {
+            empty = true;
+        } else {
+            if (id >= SoftSynthInstrumentBase) {
+                empty = (!m_synths[id] || m_synths[id]->isBypassed());
+            } else {
+                empty = !queue->haveFilesForInstrument(id);
+            }
+
+            if (empty) {
+                for (PluginList::iterator j = m_plugins[id].begin();
+                        j != m_plugins[id].end(); ++j) {
+                    if (*j != 0) {
+                        empty = false;
+                        break;
+                    }
+                }
+            }
+        }
+
+        if (!empty && rec.empty) {
+
+            // This instrument is becoming freshly non-empty.  We need
+            // to set its filledTo field to match that of an existing
+            // non-empty instrument, if we can find one.
+
+            for (BufferMap::iterator j = m_bufferMap.begin();
+                    j != m_bufferMap.end(); ++j) {
+
+                if (j->first == i->first)
+                    continue;
+                if (j->second.empty)
+                    continue;
+
+                rec.filledTo = j->second.filledTo;
+                break;
+            }
+        }
+
+        rec.empty = empty;
+
+        // For a while we were setting empty to true if the volume on
+        // the track was zero, but that breaks continuity if there is
+        // actually a file on the track -- processEmptyBlocks won't
+        // read it, so it'll fall behind if we put the volume up again.
+    }
+
+    bool more = true;
+
+    static const int MAX_FILES_PER_INSTRUMENT = 500;
+    static PlayableAudioFile *playing[MAX_FILES_PER_INSTRUMENT];
+
+    RealTime blockDuration = RealTime::frame2RealTime(m_blockSize, m_sampleRate);
+
+    while (more) {
+
+        more = false;
+
+        for (BufferMap::iterator i = m_bufferMap.begin();
+                i != m_bufferMap.end(); ++i) {
+
+            InstrumentId id = i->first;
+            BufferRec &rec = i->second;
+
+            if (rec.empty) {
+                rec.dormant = true;
+                continue;
+            }
+
+            size_t playCount = MAX_FILES_PER_INSTRUMENT;
+
+            if (id >= SoftSynthInstrumentBase)
+                playCount = 0;
+            else {
+                queue->getPlayingFilesForInstrument(rec.filledTo,
+                                                    blockDuration, id,
+                                                    playing, playCount);
+            }
+
+            if (processBlock(id, playing, playCount, readSomething)) {
+                more = true;
+            }
+        }
+    }
+}
+
+
+bool
+AudioInstrumentMixer::processBlock(InstrumentId id,
+                                   PlayableAudioFile **playing,
+                                   size_t playCount,
+                                   bool &readSomething)
+{
+    // Needs to be RT safe
+
+    //    Profiler profiler("processBlock", true);
+
+    BufferRec &rec = m_bufferMap[id];
+    RealTime bufferTime = rec.filledTo;
+
+#ifdef DEBUG_MIXER 
+    //    if (m_driver->isPlaying()) {
+    if ((id % 100) == 0)
+        std::cerr << "AudioInstrumentMixer::processBlock(" << id << "): buffer time is " << bufferTime << std::endl;
+    //    }
+#endif
+
+    unsigned int channels = rec.channels;
+    if (channels > rec.buffers.size())
+        channels = rec.buffers.size();
+    if (channels > m_processBuffers.size())
+        channels = m_processBuffers.size();
+    if (channels == 0) {
+#ifdef DEBUG_MIXER
+        if ((id % 100) == 0)
+            std::cerr << "AudioInstrumentMixer::processBlock(" << id << "): nominal channels " << rec.channels << ", ring buffers " << rec.buffers.size() << ", process buffers " << m_processBuffers.size() << std::endl;
+#endif
+
+        return false; // buffers just haven't been set up yet
+    }
+
+    unsigned int targetChannels = channels;
+    if (targetChannels < 2)
+        targetChannels = 2; // fill at least two buffers
+
+    size_t minWriteSpace = 0;
+    for (unsigned int ch = 0; ch < targetChannels; ++ch) {
+        size_t thisWriteSpace = rec.buffers[ch]->getWriteSpace();
+        if (ch == 0 || thisWriteSpace < minWriteSpace) {
+            minWriteSpace = thisWriteSpace;
+            if (minWriteSpace < m_blockSize) {
+#ifdef DEBUG_MIXER 
+                //		if (m_driver->isPlaying()) {
+                if ((id % 100) == 0)
+                    std::cerr << "AudioInstrumentMixer::processBlock(" << id << "): only " << minWriteSpace << " write space on channel " << ch << " for block size " << m_blockSize << std::endl;
+                //		}
+#endif
+
+                return false;
+            }
+        }
+    }
+
+    PluginList &plugins = m_plugins[id];
+
+#ifdef DEBUG_MIXER
+
+    if ((id % 100) == 0 && m_driver->isPlaying())
+        std::cerr << "AudioInstrumentMixer::processBlock(" << id << "): minWriteSpace is " << minWriteSpace << std::endl;
+#else
+#ifdef DEBUG_MIXER_LIGHTWEIGHT
+
+    if ((id % 100) == 0 && m_driver->isPlaying())
+        std::cout << minWriteSpace << "/" << rec.buffers[0]->getSize() << std::endl;
+#endif
+#endif
+
+#ifdef DEBUG_MIXER
+
+    if ((id % 100) == 0 && playCount > 0)
+        std::cerr << "AudioInstrumentMixer::processBlock(" << id << "): " << playCount << " audio file(s) to consider" << std::endl;
+#endif
+
+    bool haveBlock = true;
+    bool haveMore = false;
+
+    for (size_t fileNo = 0; fileNo < playCount; ++fileNo) {
+
+        bool acceptable = false;
+        PlayableAudioFile *file = playing[fileNo];
+
+        size_t frames = file->getSampleFramesAvailable();
+        acceptable = ((frames >= m_blockSize) || file->isFullyBuffered());
+
+        if (acceptable &&
+                (minWriteSpace >= m_blockSize * 2) &&
+                (frames >= m_blockSize * 2)) {
+
+#ifdef DEBUG_MIXER
+            if ((id % 100) == 0)
+                std::cerr << "AudioInstrumentMixer::processBlock(" << id << "): will be asking for more" << std::endl;
+#endif
+
+            haveMore = true;
+        }
+
+#ifdef DEBUG_MIXER
+        if ((id % 100) == 0)
+            std::cerr << "AudioInstrumentMixer::processBlock(" << id << "): file has " << frames << " frames available" << std::endl;
+#endif
+
+        if (!acceptable) {
+
+            std::cerr << "AudioInstrumentMixer::processBlock(" << id << "): file " << file->getAudioFile()->getFilename() << " has " << frames << " frames available, says isBuffered " << file->isBuffered() << std::endl;
+
+            if (!m_driver->getLowLatencyMode()) {
+
+                // Not a serious problem, just block on this
+                // instrument and return to it a little later.
+                haveBlock = false;
+
+            } else {
+                // In low latency mode, this is a serious problem if
+                // the file has been buffered and simply isn't filling
+                // fast enough.  Otherwise we have to assume that the
+                // problem is something like a new file being dropped
+                // in by unmute during playback, in which case we have
+                // to accept that it won't be available for a while
+                // and just read silence from it instead.
+                if (file->isBuffered()) {
+                    m_driver->reportFailure(MappedEvent::FailureDiscUnderrun);
+                    haveBlock = false;
+                } else {
+                    // ignore happily.
+                }
+            }
+        }
+    }
+
+    if (!haveBlock) {
+        return false; // blocked;
+    }
+
+#ifdef DEBUG_MIXER
+    if (!haveMore) {
+        if ((id % 100) == 0)
+            std::cerr << "AudioInstrumentMixer::processBlock(" << id << "): won't be asking for more" << std::endl;
+    }
+#endif
+
+    for (unsigned int ch = 0; ch < targetChannels; ++ch) {
+        memset(m_processBuffers[ch], 0, sizeof(sample_t) * m_blockSize);
+    }
+
+    RunnablePluginInstance *synth = m_synths[id];
+
+    if (synth && !synth->isBypassed()) {
+
+        synth->run(bufferTime);
+
+        unsigned int ch = 0;
+
+        while (ch < synth->getAudioOutputCount() && ch < channels) {
+            denormalKill(synth->getAudioOutputBuffers()[ch],
+                         m_blockSize);
+            memcpy(m_processBuffers[ch],
+                   synth->getAudioOutputBuffers()[ch],
+                   m_blockSize * sizeof(sample_t));
+            ++ch;
+        }
+    }
+
+    if (haveBlock) {
+
+        // Mix in a block from each playing file on this instrument.
+
+        for (size_t fileNo = 0; fileNo < playCount; ++fileNo) {
+
+            PlayableAudioFile *file = playing[fileNo];
+
+            size_t offset = 0;
+            size_t blockSize = m_blockSize;
+
+            if (file->getStartTime() > bufferTime) {
+                offset = RealTime::realTime2Frame
+                         (file->getStartTime() - bufferTime, m_sampleRate);
+                if (offset < blockSize)
+                    blockSize -= offset;
+                else
+                    blockSize = 0;
+#ifdef DEBUG_MIXER
+
+                std::cerr << "AudioInstrumentMixer::processBlock: file starts at offset " << offset << ", block size now " << blockSize << std::endl;
+#endif
+
+            }
+
+            //!!! This addSamples call is what is supposed to signal
+            // to a playable audio file when the end of the file has
+            // been reached.  But for some playables it appears the
+            // file overruns, possibly due to rounding errors in
+            // sample rate conversion, and so we stop reading from it
+            // before it's actually done.  I don't particularly mind
+            // that from a sound quality POV (after all it's badly
+            // resampled already) but unfortunately it means we leak
+            // pooled buffers.
+
+            if (blockSize > 0) {
+                file->addSamples(m_processBuffers, channels, blockSize, offset);
+                readSomething = true;
+            }
+        }
+    }
+
+    // Apply plugins.  There are various copy-reducing
+    // optimisations available here, but we're not even going to
+    // think about them yet.  Note that we force plugins to mono
+    // on a mono track, even though we have stereo output buffers
+    // -- stereo only comes into effect at the pan stage, and
+    // these are pre-fader plugins.
+
+    for (PluginList::iterator pli = plugins.begin();
+            pli != plugins.end(); ++pli) {
+
+        RunnablePluginInstance *plugin = *pli;
+        if (!plugin || plugin->isBypassed())
+            continue;
+
+        unsigned int ch = 0;
+
+        // If a plugin has more input channels than we have
+        // available, we duplicate up to stereo and leave any
+        // remaining channels empty.
+
+        while (ch < plugin->getAudioInputCount()) {
+
+            if (ch < channels || ch < 2) {
+                memcpy(plugin->getAudioInputBuffers()[ch],
+                       m_processBuffers[ch % channels],
+                       m_blockSize * sizeof(sample_t));
+            } else {
+                memset(plugin->getAudioInputBuffers()[ch], 0,
+                       m_blockSize * sizeof(sample_t));
+            }
+            ++ch;
+        }
+
+#ifdef DEBUG_MIXER
+        std::cerr << "Running plugin with " << plugin->getAudioInputCount()
+        << " inputs, " << plugin->getAudioOutputCount() << " outputs" << std::endl;
+#endif
+
+        plugin->run(bufferTime);
+
+        ch = 0;
+
+        while (ch < plugin->getAudioOutputCount()) {
+
+            denormalKill(plugin->getAudioOutputBuffers()[ch],
+                         m_blockSize);
+
+            if (ch < channels) {
+                memcpy(m_processBuffers[ch],
+                       plugin->getAudioOutputBuffers()[ch],
+                       m_blockSize * sizeof(sample_t));
+            } else if (ch == 1) {
+                // stereo output from plugin on a mono track
+                for (size_t i = 0; i < m_blockSize; ++i) {
+                    m_processBuffers[0][i] +=
+                        plugin->getAudioOutputBuffers()[ch][i];
+                    m_processBuffers[0][i] /= 2;
+                }
+            } else {
+                break;
+            }
+
+            ++ch;
+        }
+    }
+
+    // special handling for pan on mono tracks
+
+    bool allZeros = true;
+
+    if (targetChannels == 2 && channels == 1) {
+
+        for (size_t i = 0; i < m_blockSize; ++i) {
+
+            sample_t sample = m_processBuffers[0][i];
+
+            m_processBuffers[0][i] = sample * rec.gainLeft;
+            m_processBuffers[1][i] = sample * rec.gainRight;
+
+            if (allZeros && sample != 0.0)
+                allZeros = false;
+        }
+
+        rec.buffers[0]->write(m_processBuffers[0], m_blockSize);
+        rec.buffers[1]->write(m_processBuffers[1], m_blockSize);
+
+    } else {
+
+        for (unsigned int ch = 0; ch < targetChannels; ++ch) {
+
+            float gain = ((ch == 0) ? rec.gainLeft :
+                          (ch == 1) ? rec.gainRight : rec.volume);
+
+            for (size_t i = 0; i < m_blockSize; ++i) {
+
+                // handle volume and pan
+                m_processBuffers[ch][i] *= gain;
+
+                if (allZeros && m_processBuffers[ch][i] != 0.0)
+                    allZeros = false;
+            }
+
+            rec.buffers[ch]->write(m_processBuffers[ch], m_blockSize);
+        }
+    }
+
+    bool dormant = true;
+
+    if (allZeros) {
+        rec.zeroFrames += m_blockSize;
+        for (unsigned int ch = 0; ch < targetChannels; ++ch) {
+            if (rec.buffers[ch]->getReadSpace() > rec.zeroFrames) {
+                dormant = false;
+            }
+        }
+    } else {
+        rec.zeroFrames = 0;
+        dormant = false;
+    }
+
+#ifdef DEBUG_MIXER
+    if ((id % 100) == 0 && m_driver->isPlaying())
+        std::cerr << "AudioInstrumentMixer::processBlock(" << id << "): setting dormant to " << dormant << std::endl;
+#endif
+
+    rec.dormant = dormant;
+    bufferTime = bufferTime + RealTime::frame2RealTime(m_blockSize,
+                 m_sampleRate);
+
+    rec.filledTo = bufferTime;
+
+#ifdef DEBUG_MIXER
+
+    if ((id % 100) == 0)
+        std::cerr << "AudioInstrumentMixer::processBlock(" << id << "): done, returning " << haveMore << std::endl;
+#endif
+
+    return haveMore;
+}
+
+void
+AudioInstrumentMixer::kick(bool wantLock)
+{
+    // Needs to be RT safe if wantLock is not specified
+
+    if (wantLock)
+        getLock();
+
+    bool readSomething = false;
+    processBlocks(readSomething);
+    if (readSomething)
+        m_fileReader->signal();
+
+    if (wantLock)
+        releaseLock();
+}
+
+
+void
+AudioInstrumentMixer::threadRun()
+{
+    while (!m_exiting) {
+
+        if (m_driver->areClocksRunning()) {
+            kick(false);
+        }
+
+        RealTime t = m_driver->getAudioMixBufferLength();
+        t = t / 2;
+        if (t < RealTime(0, 10000000))
+            t = RealTime(0, 10000000); // 10ms minimum
+
+        struct timeval now;
+        gettimeofday(&now, 0);
+        t = t + RealTime(now.tv_sec, now.tv_usec * 1000);
+
+        struct timespec timeout;
+        timeout.tv_sec = t.sec;
+        timeout.tv_nsec = t.nsec;
+
+        pthread_cond_timedwait(&m_condition, &m_lock, &timeout);
+        pthread_testcancel();
+    }
+}
+
+
+
+AudioFileReader::AudioFileReader(SoundDriver *driver,
+                                 unsigned int sampleRate) :
+        AudioThread("AudioFileReader", driver, sampleRate)
+{
+    // nothing else here
+}
+
+AudioFileReader::~AudioFileReader()
+{}
+
+void
+AudioFileReader::fillBuffers(const RealTime &currentTime)
+{
+    getLock();
+
+    // Tell every audio file the play start time.
+
+    const AudioPlayQueue *queue = m_driver->getAudioQueue();
+
+    RealTime bufferLength = m_driver->getAudioReadBufferLength();
+    int bufferFrames = RealTime::realTime2Frame(bufferLength, m_sampleRate);
+
+    int poolSize = queue->getMaxBuffersRequired() * 2 + 4;
+    PlayableAudioFile::setRingBufferPoolSizes(poolSize, bufferFrames);
+
+    const AudioPlayQueue::FileSet &files = queue->getAllScheduledFiles();
+
+#ifdef DEBUG_READER
+
+    std::cerr << "AudioFileReader::fillBuffers: have " << files.size() << " audio files total" << std::endl;
+#endif
+
+    for (AudioPlayQueue::FileSet::const_iterator fi = files.begin();
+            fi != files.end(); ++fi) {
+        (*fi)->clearBuffers();
+    }
+
+    int allocated = 0;
+    for (AudioPlayQueue::FileSet::const_iterator fi = files.begin();
+            fi != files.end(); ++fi) {
+        (*fi)->fillBuffers(currentTime);
+        if ((*fi)->getEndTime() >= currentTime) {
+            if (++allocated == poolSize)
+                break;
+        } // else the file's ring buffers will have been returned
+    }
+
+    releaseLock();
+}
+
+bool
+AudioFileReader::kick(bool wantLock)
+{
+    if (wantLock)
+        getLock();
+
+    RealTime now = m_driver->getSequencerTime();
+    const AudioPlayQueue *queue = m_driver->getAudioQueue();
+
+    bool someFilled = false;
+
+    // Tell files that are playing or will be playing in the next few
+    // seconds to update.
+
+    AudioPlayQueue::FileSet playing;
+
+    queue->getPlayingFiles
+    (now, RealTime(3, 0) + m_driver->getAudioReadBufferLength(), playing);
+
+    for (AudioPlayQueue::FileSet::iterator fi = playing.begin();
+            fi != playing.end(); ++fi) {
+
+        if (!(*fi)->isBuffered()) {
+            // fillBuffers has not been called on this file.  This
+            // happens when a file is unmuted during playback.  The
+            // results are unpredictable because we can no longer
+            // synchronise with the correct JACK callback slice at
+            // this point, but this is better than allowing the file
+            // to update from its start as would otherwise happen.
+            (*fi)->fillBuffers(now);
+            someFilled = true;
+        } else {
+            if ((*fi)->updateBuffers())
+                someFilled = true;
+        }
+    }
+
+    if (wantLock)
+        releaseLock();
+
+    return someFilled;
+}
+
+void
+AudioFileReader::threadRun()
+{
+    while (!m_exiting) {
+
+        //	struct timeval now;
+        //	gettimeofday(&now, 0);
+        //	RealTime t = RealTime(now.tv_sec, now.tv_usec * 1000);
+
+        bool someFilled = false;
+
+        if (m_driver->areClocksRunning()) {
+            someFilled = kick(false);
+        }
+
+        if (someFilled) {
+
+            releaseLock();
+            getLock();
+
+        } else {
+
+            RealTime bt = m_driver->getAudioReadBufferLength();
+            bt = bt / 2;
+            if (bt < RealTime(0, 10000000))
+                bt = RealTime(0, 10000000); // 10ms minimum
+
+            struct timeval now;
+            gettimeofday(&now, 0);
+            RealTime t = bt + RealTime(now.tv_sec, now.tv_usec * 1000);
+
+            struct timespec timeout;
+            timeout.tv_sec = t.sec;
+            timeout.tv_nsec = t.nsec;
+
+            pthread_cond_timedwait(&m_condition, &m_lock, &timeout);
+            pthread_testcancel();
+        }
+    }
+}
+
+
+
+AudioFileWriter::AudioFileWriter(SoundDriver *driver,
+                                 unsigned int sampleRate) :
+        AudioThread("AudioFileWriter", driver, sampleRate)
+{
+    InstrumentId instrumentBase;
+    int instrumentCount;
+    m_driver->getAudioInstrumentNumbers(instrumentBase, instrumentCount);
+
+    for (InstrumentId id = instrumentBase;
+            id < instrumentBase + instrumentCount; ++id) {
+
+        // prefill with zero files in all slots, so that we can
+        // refer to the map without a lock (as the number of
+        // instruments won't change)
+
+        m_files[id] = FilePair(0, 0);
+    }
+}
+
+AudioFileWriter::~AudioFileWriter()
+{}
+
+
+bool
+AudioFileWriter::openRecordFile(InstrumentId id,
+                                const std::string &fileName)
+{
+    getLock();
+
+    if (m_files[id].first) {
+        releaseLock();
+        std::cerr << "AudioFileWriter::openRecordFile: already have record file for instrument " << id << "!" << std::endl;
+        return false; // already have one
+    }
+
+#ifdef DEBUG_WRITER
+    std::cerr << "AudioFileWriter::openRecordFile: instrument id is " << id << std::endl;
+#endif
+
+    MappedAudioFader *fader = m_driver->getMappedStudio()->getAudioFader(id);
+
+    RealTime bufferLength = m_driver->getAudioWriteBufferLength();
+    int bufferSamples = RealTime::realTime2Frame(bufferLength, m_sampleRate);
+    bufferSamples = ((bufferSamples / 1024) + 1) * 1024;
+
+    if (fader) {
+        float fch = 2;
+        (void)fader->getProperty(MappedAudioFader::Channels, fch);
+        int channels = (int)fch;
+
+        RIFFAudioFile::SubFormat format = m_driver->getAudioRecFileFormat();
+
+        int bytesPerSample = (format == RIFFAudioFile::PCM ? 2 : 4) * channels;
+        int bitsPerSample = (format == RIFFAudioFile::PCM ? 16 : 32);
+
+        AudioFile *recordFile = 0;
+
+        try {
+            recordFile =
+                new WAVAudioFile(fileName,
+                                 channels,             // channels
+                                 m_sampleRate,         // samples per second
+                                 m_sampleRate *
+                                 bytesPerSample,       // bytes per second
+                                 bytesPerSample,       // bytes per frame
+                                 bitsPerSample);       // bits per sample
+
+            // open the file for writing
+            //
+            if (!recordFile->write()) {
+                std::cerr << "AudioFileWriter::openRecordFile: failed to open " << fileName << " for writing" << std::endl;
+                delete recordFile;
+                releaseLock();
+                return false;
+            }
+        } catch (SoundFile::BadSoundFileException e) {
+            std::cerr << "AudioFileWriter::openRecordFile: failed to open " << fileName << " for writing: " << e.getMessage() << std::endl;
+            delete recordFile;
+            releaseLock();
+            return false;
+        }
+
+        RecordableAudioFile *raf = new RecordableAudioFile(recordFile,
+                                   bufferSamples);
+        m_files[id].second = raf;
+        m_files[id].first = recordFile;
+
+#ifdef DEBUG_WRITER
+
+        std::cerr << "AudioFileWriter::openRecordFile: created " << channels << "-channel file at " << fileName << " (id is " << recordFile->getId() << ")" << std::endl;
+#endif
+
+        releaseLock();
+        return true;
+    }
+
+    std::cerr << "AudioFileWriter::openRecordFile: no audio fader for record instrument " << id << "!" << std::endl;
+    releaseLock();
+    return false;
+}
+
+
+void
+AudioFileWriter::write(InstrumentId id,
+                       const sample_t *samples,
+                       int channel,
+                       size_t sampleCount)
+{
+    if (!m_files[id].first)
+        return ; // no file
+    if (m_files[id].second->buffer(samples, channel, sampleCount) < sampleCount) {
+        m_driver->reportFailure(MappedEvent::FailureDiscOverrun);
+    }
+}
+
+bool
+AudioFileWriter::closeRecordFile(InstrumentId id, AudioFileId &returnedId)
+{
+    if (!m_files[id].first)
+        return false;
+
+    returnedId = m_files[id].first->getId();
+    m_files[id].second->setStatus(RecordableAudioFile::DEFUNCT);
+
+#ifdef DEBUG_WRITER
+
+    std::cerr << "AudioFileWriter::closeRecordFile: instrument " << id << " file set defunct (file ID is " << returnedId << ")" << std::endl;
+#endif
+
+    // Don't reset the file pointers here; that will be done in the
+    // next call to kick().  Doesn't really matter when that happens,
+    // but let's encourage it to happen soon just for certainty.
+    signal();
+
+    return true;
+}
+
+bool
+AudioFileWriter::haveRecordFileOpen(InstrumentId id)
+{
+    InstrumentId instrumentBase;
+    int instrumentCount;
+    m_driver->getAudioInstrumentNumbers(instrumentBase, instrumentCount);
+
+    if (id < instrumentBase || id >= instrumentBase + instrumentCount) {
+        return false;
+    }
+
+    return (m_files[id].first &&
+            (m_files[id].second->getStatus() != RecordableAudioFile::DEFUNCT));
+}
+
+bool
+AudioFileWriter::haveRecordFilesOpen()
+{
+    InstrumentId instrumentBase;
+    int instrumentCount;
+    m_driver->getAudioInstrumentNumbers(instrumentBase, instrumentCount);
+
+    for (InstrumentId id = instrumentBase; id < instrumentBase + instrumentCount; ++id) {
+
+        if (m_files[id].first &&
+                (m_files[id].second->getStatus() != RecordableAudioFile::DEFUNCT)) {
+#ifdef DEBUG_WRITER
+            std::cerr << "AudioFileWriter::haveRecordFilesOpen: found open record file for instrument " << id << std::endl;
+#endif
+
+            return true;
+        }
+    }
+#ifdef DEBUG_WRITER
+    std::cerr << "AudioFileWriter::haveRecordFilesOpen: nope" << std::endl;
+#endif
+
+    return false;
+}
+
+void
+AudioFileWriter::kick(bool wantLock)
+{
+    if (wantLock)
+        getLock();
+
+    InstrumentId instrumentBase;
+    int instrumentCount;
+    m_driver->getAudioInstrumentNumbers(instrumentBase, instrumentCount);
+
+    for (InstrumentId id = instrumentBase;
+            id < instrumentBase + instrumentCount; ++id) {
+
+        if (!m_files[id].first)
+            continue;
+
+        RecordableAudioFile *raf = m_files[id].second;
+
+        if (raf->getStatus() == RecordableAudioFile::DEFUNCT) {
+
+#ifdef DEBUG_WRITER
+            std::cerr << "AudioFileWriter::kick: found defunct file on instrument " << id << std::endl;
+#endif
+
+            m_files[id].first = 0;
+            delete raf; // also deletes the AudioFile
+            m_files[id].second = 0;
+
+        } else {
+#ifdef DEBUG_WRITER
+            std::cerr << "AudioFileWriter::kick: writing file on instrument " << id << std::endl;
+#endif
+
+            raf->write();
+        }
+    }
+
+    if (wantLock)
+        releaseLock();
+}
+
+void
+AudioFileWriter::threadRun()
+{
+    while (!m_exiting) {
+
+        kick(false);
+
+        RealTime t = m_driver->getAudioWriteBufferLength();
+        t = t / 2;
+        if (t < RealTime(0, 10000000))
+            t = RealTime(0, 10000000); // 10ms minimum
+
+        struct timeval now;
+        gettimeofday(&now, 0);
+        t = t + RealTime(now.tv_sec, now.tv_usec * 1000);
+
+        struct timespec timeout;
+        timeout.tv_sec = t.sec;
+        timeout.tv_nsec = t.nsec;
+
+        pthread_cond_timedwait(&m_condition, &m_lock, &timeout);
+        pthread_testcancel();
+    }
+}
+
+
+}
+
diff --git a/src/sound/AudioProcess.h b/src/sound/AudioProcess.h
new file mode 100644
index 0000000..b517bc9
--- /dev/null
+++ b/src/sound/AudioProcess.h
@@ -0,0 +1,390 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _AUDIO_PROCESS_H_
+#define _AUDIO_PROCESS_H_
+
+#include "SoundDriver.h"
+#include "Instrument.h"
+#include "RealTime.h"
+#include "RingBuffer.h"
+#include "RunnablePluginInstance.h"
+#include "AudioPlayQueue.h"
+#include "RecordableAudioFile.h"
+
+namespace Rosegarden
+{
+
+class AudioThread
+{
+public:
+    typedef float sample_t;
+
+    AudioThread(std::string name, // for diagnostics
+                SoundDriver *driver,
+                unsigned int sampleRate);
+
+    virtual ~AudioThread();
+
+    // This is to be called by the owning class after construction.
+    void run();
+
+    // This is to be called by the owning class to cause the thread to
+    // exit and clean up, before destruction.
+    void terminate();
+
+    bool running() const { return m_running; }
+
+    int getLock();
+    int tryLock();
+    int releaseLock();
+    void signal();
+
+protected:
+    virtual void threadRun() = 0;
+    virtual int getPriority() { return 0; }
+
+    std::string       m_name;
+
+    SoundDriver      *m_driver;
+    unsigned int      m_sampleRate;
+
+    pthread_t         m_thread;
+    pthread_mutex_t   m_lock;
+    pthread_cond_t    m_condition;
+    bool              m_running;
+    volatile bool     m_exiting;
+
+private:
+    static void *staticThreadRun(void *arg);
+    static void  staticThreadCleanup(void *arg);
+};
+    
+
+class AudioInstrumentMixer;
+
+class AudioBussMixer : public AudioThread
+{
+public:
+    AudioBussMixer(SoundDriver *driver,
+                   AudioInstrumentMixer *instrumentMixer,
+                   unsigned int sampleRate,
+                   unsigned int blockSize);
+
+    virtual ~AudioBussMixer();
+
+    void kick(bool wantLock = true, bool signalInstrumentMixer = true);
+    
+    /**
+     * Prebuffer.  This should be called only when the transport is
+     * not running.  This also calls fillBuffers on the instrument
+     * mixer.
+     */
+    void fillBuffers(const RealTime &currentTime);
+
+    /**
+     * Empty and discard buffer contents.
+     */
+    void emptyBuffers();
+
+    int getBussCount() {
+        return m_bussCount;
+    }
+
+    /**
+     * A buss is "dormant" if every readable sample on every one of
+     * its buffers is zero.  It can therefore be safely skipped during
+     * playback.
+     */
+    bool isBussDormant(int buss) {
+        return m_bufferMap[buss].dormant;
+    }
+
+    /**
+     * Busses are currently always stereo.
+     */
+    RingBuffer<sample_t> *getRingBuffer(int buss, unsigned int channel) {
+        if (channel < m_bufferMap[buss].buffers.size()) {
+            return m_bufferMap[buss].buffers[channel];
+        } else {
+            return 0;
+        }
+    }
+
+    /// For call from MappedStudio.  Pan is in range -100.0 -> 100.0
+    void setBussLevels(int buss, float dB, float pan);
+
+    /// For call regularly from anywhere in a non-RT thread
+    void updateInstrumentConnections();
+
+protected:
+    virtual void threadRun();
+
+    void processBlocks();
+    void generateBuffers();
+
+    AudioInstrumentMixer   *m_instrumentMixer;
+    unsigned int            m_blockSize;
+    int                     m_bussCount;
+
+    std::vector<sample_t *> m_processBuffers;
+
+    struct BufferRec
+    {
+        BufferRec() : dormant(true), buffers(), instruments(),
+                      gainLeft(0.0), gainRight(0.0) { }
+        ~BufferRec();
+
+        bool dormant;
+
+        std::vector<RingBuffer<sample_t> *> buffers;
+        std::vector<bool> instruments; // index is instrument id minus base
+
+        float gainLeft;
+        float gainRight;
+    };
+
+    typedef std::map<int, BufferRec> BufferMap;
+    BufferMap m_bufferMap;
+};                 
+
+
+class AudioFileReader;
+class AudioFileWriter;
+
+class AudioInstrumentMixer : public AudioThread
+{
+public:
+    typedef std::vector<RunnablePluginInstance *> PluginList;
+    typedef std::map<InstrumentId, PluginList> PluginMap;
+    typedef std::map<InstrumentId, RunnablePluginInstance *> SynthPluginMap;
+
+    AudioInstrumentMixer(SoundDriver *driver,
+                         AudioFileReader *fileReader,
+                         unsigned int sampleRate,
+                         unsigned int blockSize);
+
+    virtual ~AudioInstrumentMixer();
+
+    void kick(bool wantLock = true);
+
+    void setBussMixer(AudioBussMixer *mixer) { m_bussMixer = mixer; }
+
+    void setPlugin(InstrumentId id, int position, QString identifier);
+    void removePlugin(InstrumentId id, int position);
+    void removeAllPlugins();
+
+    void setPluginPortValue(InstrumentId id, int position,
+                            unsigned int port, float value);
+    float getPluginPortValue(InstrumentId id, int position,
+                             unsigned int port);
+
+    void setPluginBypass(InstrumentId, int position, bool bypass);
+
+    QStringList getPluginPrograms(InstrumentId, int);
+    QString getPluginProgram(InstrumentId, int);
+    QString getPluginProgram(InstrumentId, int, int, int);
+    unsigned long getPluginProgram(InstrumentId, int, QString);
+    void setPluginProgram(InstrumentId, int, QString);
+
+    QString configurePlugin(InstrumentId, int, QString, QString);
+
+    void resetAllPlugins(bool discardEvents = false);
+    void discardPluginEvents();
+    void destroyAllPlugins();
+
+    RunnablePluginInstance *getSynthPlugin(InstrumentId id) { return m_synths[id]; }
+
+    /**
+     * Return the plugins intended for a particular buss.  (By coincidence,
+     * this will also work for instruments, but it's not to be relied on.)
+     * It's purely by historical accident that the instrument mixer happens
+     * to hold buss plugins as well -- this could do with being refactored.
+     */
+    PluginList &getBussPlugins(unsigned int bussId) { return m_plugins[bussId]; }
+
+    /**
+     * Return the total of the plugin latencies for a given instrument
+     * or buss id.
+     */
+    size_t getPluginLatency(unsigned int id);
+
+    /**
+     * Prebuffer.  This should be called only when the transport is
+     * not running. 
+     */
+    void fillBuffers(const RealTime &currentTime);
+    
+    /**
+     * Ensure plugins etc have enough buffers.  This is also done by
+     * fillBuffers and only needs to be called here if the extra work
+     * involved in fillBuffers is not desirable.
+     */
+    void allocateBuffers();
+
+    /**
+     * Empty and discard buffer contents.
+     */
+    void emptyBuffers(RealTime currentTime = RealTime::zeroTime);
+
+    /**
+     * An instrument is "empty" if it has no audio files, synths or
+     * plugins assigned to it, and so cannot generate sound.  Empty
+     * instruments can safely be ignored during playback.
+     */
+    bool isInstrumentEmpty(InstrumentId id) {
+        return m_bufferMap[id].empty;
+    }
+
+    /**
+     * An instrument is "dormant" if every readable sample on every
+     * one of its buffers is zero.  Dormant instruments can safely be
+     * skipped rather than mixed during playback, but they should not
+     * be ignored (unless also empty).
+     */
+    bool isInstrumentDormant(InstrumentId id) {
+        return m_bufferMap[id].dormant;
+    }
+
+    /**
+     * We always have at least two channels (and hence buffers) by
+     * this point, because even on a mono instrument we still have a
+     * Pan setting which will have been applied by the time we get to
+     * these buffers.
+     */
+    RingBuffer<sample_t, 2> *getRingBuffer(InstrumentId id, unsigned int channel) {
+        if (channel < m_bufferMap[id].buffers.size()) {
+            return m_bufferMap[id].buffers[channel];
+        } else {
+            return 0;
+        }
+    }
+
+    /// For call from MappedStudio.  Pan is in range -100.0 -> 100.0
+    void setInstrumentLevels(InstrumentId instrument, float dB, float pan);
+
+    /// For call regularly from anywhere in a non-RT thread
+    void updateInstrumentMuteStates();
+
+protected:
+    virtual void threadRun();
+
+    virtual int getPriority() { return 3; }
+
+    void processBlocks(bool &readSomething);
+    void processEmptyBlocks(InstrumentId id);
+    bool processBlock(InstrumentId id, PlayableAudioFile **, size_t, bool &readSomething);
+    void generateBuffers();
+
+    AudioFileReader  *m_fileReader;
+    AudioBussMixer   *m_bussMixer;
+    unsigned int      m_blockSize;
+
+    // The plugin data structures will all be pre-sized and so of
+    // fixed size during normal run time; this will allow us to add
+    // and edit plugins without locking.
+    RunnablePluginInstance *getPluginInstance(InstrumentId, int);
+    PluginMap m_plugins;
+    SynthPluginMap m_synths;
+
+    // maintain the same number of these as the maximum number of
+    // channels on any audio instrument
+    std::vector<sample_t *> m_processBuffers;
+
+    struct BufferRec
+    {
+        BufferRec() : empty(true), dormant(true), zeroFrames(0),
+                      filledTo(RealTime::zeroTime), channels(2),
+                      buffers(), gainLeft(0.0), gainRight(0.0), volume(0.0),
+                      muted(false) { }
+        ~BufferRec();
+
+        bool empty;
+        bool dormant;
+        size_t zeroFrames;
+
+        RealTime filledTo;
+        size_t channels;
+        std::vector<RingBuffer<sample_t, 2> *> buffers;
+
+        float gainLeft;
+        float gainRight;
+        float volume;
+        bool muted;
+    };
+
+    typedef std::map<InstrumentId, BufferRec> BufferMap;
+    BufferMap m_bufferMap;
+};
+
+
+class AudioFileReader : public AudioThread
+{
+public:
+    AudioFileReader(SoundDriver *driver,
+                    unsigned int sampleRate);
+
+    virtual ~AudioFileReader();
+
+    bool kick(bool wantLock = true);
+
+    /**
+     * Prebuffer.  This should be called only when the transport is
+     * not running. 
+     */
+    void fillBuffers(const RealTime &currentTime);
+
+protected:
+    virtual void threadRun();
+};
+
+
+class AudioFileWriter : public AudioThread
+{
+public:
+    AudioFileWriter(SoundDriver *driver,
+                    unsigned int sampleRate);
+
+    virtual ~AudioFileWriter();
+
+    void kick(bool wantLock = true);
+
+    bool openRecordFile(InstrumentId id, const std::string &fileName);
+    bool closeRecordFile(InstrumentId id, AudioFileId &returnedId);
+
+    bool haveRecordFileOpen(InstrumentId id);
+    bool haveRecordFilesOpen();
+    
+    void write(InstrumentId id, const sample_t *, int channel, size_t samples);
+
+protected:
+    virtual void threadRun();
+
+    typedef std::pair<AudioFile *, RecordableAudioFile *> FilePair;
+    typedef std::map<InstrumentId, FilePair> FileMap;
+    FileMap m_files;
+};
+
+
+}
+
+#endif
+
diff --git a/src/sound/AudioTimeStretcher.cpp b/src/sound/AudioTimeStretcher.cpp
new file mode 100644
index 0000000..392693e
--- /dev/null
+++ b/src/sound/AudioTimeStretcher.cpp
@@ -0,0 +1,667 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Sonic Visualiser
+    An audio file viewer and annotation editor.
+    Centre for Digital Music, Queen Mary, University of London.
+    This file copyright 2006 Chris Cannam and QMUL.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "AudioTimeStretcher.h"
+
+#include <iostream>
+#include <fstream>
+#include <cassert>
+#include <cstring>
+
+namespace Rosegarden 
+{
+
+static double mod(double x, double y) { return x - (y * floor(x / y)); }
+static float modf(float x, float y) { return x - (y * floorf(x / y)); }
+
+static double princarg(double a) { return mod(a + M_PI, -2 * M_PI) + M_PI; }
+static float princargf(float a) { return modf(a + M_PI, -2 * M_PI) + M_PI; }
+
+
+//#define DEBUG_AUDIO_TIME_STRETCHER 1
+
+AudioTimeStretcher::AudioTimeStretcher(size_t sampleRate,
+                                       size_t channels,
+                                       float ratio,
+                                       bool sharpen,
+                                       size_t maxOutputBlockSize) :
+    m_sampleRate(sampleRate),
+    m_channels(channels),
+    m_maxOutputBlockSize(maxOutputBlockSize),
+    m_ratio(ratio),
+    m_sharpen(sharpen),
+    m_totalCount(0),
+    m_transientCount(0),
+    m_n2sum(0),
+    m_n2total(0),
+    m_adjustCount(50)
+{
+    pthread_mutex_t initialisingMutex = PTHREAD_MUTEX_INITIALIZER;
+    memcpy(&m_mutex, &initialisingMutex, sizeof(pthread_mutex_t));
+
+    initialise();
+}
+
+AudioTimeStretcher::~AudioTimeStretcher()
+{
+    std::cerr << "AudioTimeStretcher::~AudioTimeStretcher" << std::endl;
+
+    std::cerr << "AudioTimeStretcher::~AudioTimeStretcher: actual ratio = " << (m_totalCount > 0 ? (float (m_n2total) / float(m_totalCount * m_n1)) : 1.f) << ", ideal = " << m_ratio << ", nominal = " << getRatio() << ")" << std::endl;
+
+    cleanup();
+    
+    pthread_mutex_destroy(&m_mutex);
+}
+
+void
+AudioTimeStretcher::initialise()
+{
+    std::cerr << "AudioTimeStretcher::initialise" << std::endl;
+
+    calculateParameters();
+        
+    m_analysisWindow = new SampleWindow<float>(SampleWindow<float>::Hanning, m_wlen);
+    m_synthesisWindow = new SampleWindow<float>(SampleWindow<float>::Hanning, m_wlen);
+
+    m_prevPhase = new float *[m_channels];
+    m_prevAdjustedPhase = new float *[m_channels];
+
+    m_prevTransientMag = (float *)fftwf_malloc(sizeof(float) * (m_wlen / 2 + 1));
+    m_prevTransientScore = 0;
+    m_prevTransient = false;
+
+    m_tempbuf = (float *)fftwf_malloc(sizeof(float) * m_wlen);
+
+    m_time = new float *[m_channels];
+    m_freq = new fftwf_complex *[m_channels];
+    m_plan = new fftwf_plan[m_channels];
+    m_iplan = new fftwf_plan[m_channels];
+
+    m_inbuf = new RingBuffer<float> *[m_channels];
+    m_outbuf = new RingBuffer<float> *[m_channels];
+    m_mashbuf = new float *[m_channels];
+
+    m_modulationbuf = (float *)fftwf_malloc(sizeof(float) * m_wlen);
+        
+    for (size_t c = 0; c < m_channels; ++c) {
+
+        m_prevPhase[c] = (float *)fftwf_malloc(sizeof(float) * (m_wlen / 2 + 1));
+        m_prevAdjustedPhase[c] = (float *)fftwf_malloc(sizeof(float) * (m_wlen / 2 + 1));
+
+        m_time[c] = (float *)fftwf_malloc(sizeof(float) * m_wlen);
+        m_freq[c] = (fftwf_complex *)fftwf_malloc(sizeof(fftwf_complex) *
+                                                  (m_wlen / 2 + 1));
+        
+        m_plan[c] = fftwf_plan_dft_r2c_1d(m_wlen, m_time[c], m_freq[c], FFTW_ESTIMATE);
+        m_iplan[c] = fftwf_plan_dft_c2r_1d(m_wlen, m_freq[c], m_time[c], FFTW_ESTIMATE);
+
+        m_outbuf[c] = new RingBuffer<float>
+            ((m_maxOutputBlockSize + m_wlen) * 2);
+        m_inbuf[c] = new RingBuffer<float>
+            (lrintf(m_outbuf[c]->getSize() / m_ratio) + m_wlen);
+
+        std::cerr << "making inbuf size " << m_inbuf[c]->getSize() << " (outbuf size is " << m_outbuf[c]->getSize() << ", ratio " << m_ratio << ")" << std::endl;
+
+           
+        m_mashbuf[c] = (float *)fftwf_malloc(sizeof(float) * m_wlen);
+        
+        for (size_t i = 0; i < m_wlen; ++i) {
+            m_mashbuf[c][i] = 0.0;
+        }
+
+        for (size_t i = 0; i <= m_wlen/2; ++i) {
+            m_prevPhase[c][i] = 0.0;
+            m_prevAdjustedPhase[c][i] = 0.0;
+        }
+    }
+
+    for (size_t i = 0; i < m_wlen; ++i) {
+        m_modulationbuf[i] = 0.0;
+    }
+
+    for (size_t i = 0; i <= m_wlen/2; ++i) {
+        m_prevTransientMag[i] = 0.0;
+    }
+}
+
+void
+AudioTimeStretcher::calculateParameters()
+{
+    std::cerr << "AudioTimeStretcher::calculateParameters" << std::endl;
+
+    m_wlen = 1024;
+
+    //!!! In transient sharpening mode, we need to pick the window
+    //length so as to be more or less fixed in audio duration (i.e. we
+    //need to exploit the sample rate)
+
+    //!!! have to work out the relationship between wlen and transient
+    //threshold
+
+    if (m_ratio < 1) {
+        if (m_ratio < 0.4) {
+            m_n1 = 1024;
+            m_wlen = 2048;
+        } else if (m_ratio < 0.8) {
+            m_n1 = 512;
+        } else {
+            m_n1 = 256;
+        }
+        if (shouldSharpen()) {
+            m_wlen = 2048;
+        }
+        m_n2 = lrintf(m_n1 * m_ratio);
+    } else {
+        if (m_ratio > 2) {
+            m_n2 = 512;
+            m_wlen = 4096; 
+        } else if (m_ratio > 1.6) {
+            m_n2 = 384;
+            m_wlen = 2048;
+        } else {
+            m_n2 = 256;
+        }
+        if (shouldSharpen()) {
+            if (m_wlen < 2048) m_wlen = 2048;
+        }
+        m_n1 = lrintf(m_n2 / m_ratio);
+        if (m_n1 == 0) {
+            m_n1 = 1;
+            m_n2 = m_ratio;
+        }
+    }
+
+    m_transientThreshold = lrintf(m_wlen / 4.5);
+
+    m_totalCount = 0;
+    m_transientCount = 0;
+    m_n2sum = 0;
+    m_n2total = 0;
+    m_n2list.clear();
+
+    std::cerr << "AudioTimeStretcher: channels = " << m_channels
+              << ", ratio = " << m_ratio
+              << ", n1 = " << m_n1 << ", n2 = " << m_n2 << ", wlen = "
+              << m_wlen << ", max = " << m_maxOutputBlockSize << std::endl;
+//              << ", outbuflen = " << m_outbuf[0]->getSize() << std::endl;
+}
+
+void
+AudioTimeStretcher::cleanup()
+{
+    std::cerr << "AudioTimeStretcher::cleanup" << std::endl;
+
+    for (size_t c = 0; c < m_channels; ++c) {
+
+        fftwf_destroy_plan(m_plan[c]);
+        fftwf_destroy_plan(m_iplan[c]);
+
+        fftwf_free(m_time[c]);
+        fftwf_free(m_freq[c]);
+
+        fftwf_free(m_mashbuf[c]);
+        fftwf_free(m_prevPhase[c]);
+        fftwf_free(m_prevAdjustedPhase[c]);
+
+        delete m_inbuf[c];
+        delete m_outbuf[c];
+    }
+
+    fftwf_free(m_tempbuf);
+    fftwf_free(m_modulationbuf);
+    fftwf_free(m_prevTransientMag);
+
+    delete[] m_prevPhase;
+    delete[] m_prevAdjustedPhase;
+    delete[] m_inbuf;
+    delete[] m_outbuf;
+    delete[] m_mashbuf;
+    delete[] m_time;
+    delete[] m_freq;
+    delete[] m_plan;
+    delete[] m_iplan;
+
+    delete m_analysisWindow;
+    delete m_synthesisWindow;
+}	
+
+void
+AudioTimeStretcher::setRatio(float ratio)
+{
+    pthread_mutex_lock(&m_mutex);
+
+    size_t formerWlen = m_wlen;
+    m_ratio = ratio;
+
+    std::cerr << "AudioTimeStretcher::setRatio: new ratio " << ratio
+              << std::endl;
+
+    calculateParameters();
+
+    if (m_wlen == formerWlen) {
+
+        // This is the only container whose size depends on m_ratio
+
+        RingBuffer<float> **newin = new RingBuffer<float> *[m_channels];
+
+        size_t formerSize = m_inbuf[0]->getSize();
+        size_t newSize = lrintf(m_outbuf[0]->getSize() / m_ratio) + m_wlen;
+
+        std::cerr << "resizing inbuf from " << formerSize << " to "
+                  << newSize << " (outbuf size is " << m_outbuf[0]->getSize() << ", ratio " << m_ratio << ")" << std::endl;
+
+        if (formerSize != newSize) {
+
+            size_t ready = m_inbuf[0]->getReadSpace();
+
+            for (size_t c = 0; c < m_channels; ++c) {
+                newin[c] = new RingBuffer<float>(newSize);
+            }
+
+            if (ready > 0) {
+
+                size_t copy = std::min(ready, newSize);
+                float *tmp = new float[ready];
+
+                for (size_t c = 0; c < m_channels; ++c) {
+                    m_inbuf[c]->read(tmp, ready);
+                    newin[c]->write(tmp + ready - copy, copy);
+                }
+                
+                delete[] tmp;
+            }
+            
+            for (size_t c = 0; c < m_channels; ++c) {
+                delete m_inbuf[c];
+            }
+            
+            delete[] m_inbuf;
+            m_inbuf = newin;
+        }
+
+    } else {
+        
+        std::cerr << "wlen changed" << std::endl;
+        cleanup();
+        initialise();
+    }
+
+    pthread_mutex_unlock(&m_mutex);
+}
+
+size_t
+AudioTimeStretcher::getProcessingLatency() const
+{
+    return getWindowSize() - getInputIncrement();
+}
+
+size_t
+AudioTimeStretcher::getRequiredInputSamples() const
+{
+    size_t rv;
+    pthread_mutex_lock(&m_mutex);
+
+    if (m_inbuf[0]->getReadSpace() >= m_wlen) rv = 0;
+    else rv = m_wlen - m_inbuf[0]->getReadSpace();
+
+    pthread_mutex_unlock(&m_mutex);
+    return rv;
+}
+
+void
+AudioTimeStretcher::putInput(float **input, size_t samples)
+{
+    pthread_mutex_lock(&m_mutex);
+
+    // We need to add samples from input to our internal buffer.  When
+    // we have m_windowSize samples in the buffer, we can process it,
+    // move the samples back by m_n1 and write the output onto our
+    // internal output buffer.  If we have (samples * ratio) samples
+    // in that, we can write m_n2 of them back to output and return
+    // (otherwise we have to write zeroes).
+
+    // When we process, we write m_wlen to our fixed output buffer
+    // (m_mashbuf).  We then pull out the first m_n2 samples from that
+    // buffer, push them into the output ring buffer, and shift
+    // m_mashbuf left by that amount.
+
+    // The processing latency is then m_wlen - m_n2.
+
+    size_t consumed = 0;
+
+    while (consumed < samples) {
+
+	size_t writable = m_inbuf[0]->getWriteSpace();
+	writable = std::min(writable, samples - consumed);
+
+	if (writable == 0) {
+#ifdef DEBUG_AUDIO_TIME_STRETCHER
+	    std::cerr << "WARNING: AudioTimeStretcher::putInput: writable == 0 (inbuf has " << m_inbuf[0]->getReadSpace() << " samples available for reading, space for " << m_inbuf[0]->getWriteSpace() << " more)" << std::endl;
+#endif
+            if (m_inbuf[0]->getReadSpace() < m_wlen ||
+                m_outbuf[0]->getWriteSpace() < m_n2) {
+                std::cerr << "WARNING: AudioTimeStretcher::putInput: Inbuf has " << m_inbuf[0]->getReadSpace() << ", outbuf has space for " << m_outbuf[0]->getWriteSpace() << " (n2 = " << m_n2 << ", wlen = " << m_wlen << "), won't be able to process" << std::endl;
+                break;
+            }
+	} else {
+
+#ifdef DEBUG_AUDIO_TIME_STRETCHER
+            std::cerr << "writing " << writable << " from index " << consumed << " to inbuf, consumed will be " << consumed + writable << std::endl;
+#endif
+
+            for (size_t c = 0; c < m_channels; ++c) {
+                m_inbuf[c]->write(input[c] + consumed, writable);
+            }
+            consumed += writable;
+        }
+
+	while (m_inbuf[0]->getReadSpace() >= m_wlen &&
+	       m_outbuf[0]->getWriteSpace() >= m_n2) {
+
+	    // We know we have at least m_wlen samples available
+	    // in m_inbuf.  We need to peek m_wlen of them for
+	    // processing, and then read m_n1 to advance the read
+	    // pointer.
+            
+            for (size_t c = 0; c < m_channels; ++c) {
+
+                size_t got = m_inbuf[c]->peek(m_tempbuf, m_wlen);
+                assert(got == m_wlen);
+
+                analyseBlock(c, m_tempbuf);
+            }
+
+            bool transient = false;
+            if (shouldSharpen()) transient = isTransient();
+
+            size_t n2 = m_n2;
+
+            if (transient) {
+                n2 = m_n1;
+            }
+
+            ++m_totalCount;
+            if (transient) ++m_transientCount;
+
+            m_n2sum += n2;
+            m_n2total += n2;
+
+            if (m_totalCount > 50 && m_transientCount < m_totalCount) {
+
+                int fixed = m_transientCount * m_n1;
+
+                float idealTotal = m_totalCount * m_n1 * m_ratio;
+                float idealSquashy = idealTotal - fixed;
+
+                float squashyCount = m_totalCount - m_transientCount;
+                
+                float fn2 = idealSquashy / squashyCount;
+
+                n2 = int(fn2);
+
+                float remainder = fn2 - n2;
+                if (drand48() < remainder) ++n2;
+
+#ifdef DEBUG_AUDIO_TIME_STRETCHER
+                if (n2 != m_n2) {
+                    std::cerr << m_n2 << " -> " << n2 << " (ideal = " << (idealSquashy / squashyCount) << ")" << std::endl;
+                }
+#endif
+            }
+
+            for (size_t c = 0; c < m_channels; ++c) {
+
+                synthesiseBlock(c, m_mashbuf[c],
+                                c == 0 ? m_modulationbuf : 0,
+                                m_prevTransient ? m_n1 : m_n2);
+
+
+#ifdef DEBUG_AUDIO_TIME_STRETCHER
+                std::cerr << "writing first " << m_n2 << " from mashbuf, skipping " << m_n1 << " on inbuf " << std::endl;
+#endif
+                m_inbuf[c]->skip(m_n1);
+
+                for (size_t i = 0; i < n2; ++i) {
+                    if (m_modulationbuf[i] > 0.f) {
+                        m_mashbuf[c][i] /= m_modulationbuf[i];
+                    }
+                }
+
+                m_outbuf[c]->write(m_mashbuf[c], n2);
+
+                for (size_t i = 0; i < m_wlen - n2; ++i) {
+                    m_mashbuf[c][i] = m_mashbuf[c][i + n2];
+                }
+
+                for (size_t i = m_wlen - n2; i < m_wlen; ++i) {
+                    m_mashbuf[c][i] = 0.0f;
+                }
+            }
+
+            m_prevTransient = transient;
+
+            for (size_t i = 0; i < m_wlen - n2; ++i) {
+                m_modulationbuf[i] = m_modulationbuf[i + n2];
+	    }
+
+	    for (size_t i = m_wlen - n2; i < m_wlen; ++i) {
+                m_modulationbuf[i] = 0.0f;
+	    }
+
+            if (!transient) m_n2 = n2;
+	}
+
+
+#ifdef DEBUG_AUDIO_TIME_STRETCHER
+	std::cerr << "loop ended: inbuf read space " << m_inbuf[0]->getReadSpace() << ", outbuf write space " << m_outbuf[0]->getWriteSpace() << std::endl;
+#endif
+    }
+
+#ifdef DEBUG_AUDIO_TIME_STRETCHER
+    std::cerr << "AudioTimeStretcher::putInput returning" << std::endl;
+#endif
+
+    pthread_mutex_unlock(&m_mutex);
+
+//    std::cerr << "ratio: nominal: " << getRatio() << " actual: "
+//              << m_total2 << "/" << m_total1 << " = " << float(m_total2) / float(m_total1) << " ideal: " << m_ratio << std::endl;
+}
+
+size_t
+AudioTimeStretcher::getAvailableOutputSamples() const
+{
+    pthread_mutex_lock(&m_mutex);
+
+    size_t rv = m_outbuf[0]->getReadSpace();
+
+    pthread_mutex_unlock(&m_mutex);
+    return rv;
+}
+
+void
+AudioTimeStretcher::getOutput(float **output, size_t samples)
+{
+    pthread_mutex_lock(&m_mutex);
+
+    if (m_outbuf[0]->getReadSpace() < samples) {
+	std::cerr << "WARNING: AudioTimeStretcher::getOutput: not enough data (yet?) (" << m_outbuf[0]->getReadSpace() << " < " << samples << ")" << std::endl;
+	size_t fill = samples - m_outbuf[0]->getReadSpace();
+        for (size_t c = 0; c < m_channels; ++c) {
+            for (size_t i = 0; i < fill; ++i) {
+                output[c][i] = 0.0;
+            }
+            m_outbuf[c]->read(output[c] + fill, m_outbuf[c]->getReadSpace());
+        }
+    } else {
+#ifdef DEBUG_AUDIO_TIME_STRETCHER
+	std::cerr << "enough data - writing " << samples << " from outbuf" << std::endl;
+#endif
+        for (size_t c = 0; c < m_channels; ++c) {
+            m_outbuf[c]->read(output[c], samples);
+        }
+    }
+
+#ifdef DEBUG_AUDIO_TIME_STRETCHER
+    std::cerr << "AudioTimeStretcher::getOutput returning" << std::endl;
+#endif
+
+    pthread_mutex_unlock(&m_mutex);
+}
+
+void
+AudioTimeStretcher::analyseBlock(size_t c, float *buf)
+{
+    size_t i;
+
+    // buf contains m_wlen samples
+
+#ifdef DEBUG_AUDIO_TIME_STRETCHER
+    std::cerr << "AudioTimeStretcher::analyseBlock (channel " << c << ")" << std::endl;
+#endif
+
+    m_analysisWindow->cut(buf);
+
+    for (i = 0; i < m_wlen/2; ++i) {
+	float temp = buf[i];
+	buf[i] = buf[i + m_wlen/2];
+	buf[i + m_wlen/2] = temp;
+    }
+
+    for (i = 0; i < m_wlen; ++i) {
+	m_time[c][i] = buf[i];
+    }
+
+    fftwf_execute(m_plan[c]); // m_time -> m_freq
+}
+
+bool
+AudioTimeStretcher::isTransient()
+{
+    int count = 0;
+
+    for (size_t i = 0; i <= m_wlen/2; ++i) {
+
+        float real = 0.f, imag = 0.f;
+
+        for (size_t c = 0; c < m_channels; ++c) {
+            real += m_freq[c][i][0];
+            imag += m_freq[c][i][1];
+        }
+
+        float sqrmag = (real * real + imag * imag);
+
+        if (m_prevTransientMag[i] > 0.f) {
+            float diff = 10.f * log10f(sqrmag / m_prevTransientMag[i]);
+            if (diff > 3.f) ++count;
+        }
+
+        m_prevTransientMag[i] = sqrmag;
+    }
+
+    bool isTransient = false;
+
+//    if (count > m_transientThreshold &&
+//        count > m_prevTransientScore * 1.2) {
+    if (count > m_prevTransientScore &&
+        count > m_transientThreshold &&
+        count - m_prevTransientScore > m_wlen / 20) {
+        isTransient = true;
+
+#ifdef DEBUG_AUDIO_TIME_STRETCHER
+        std::cerr << "isTransient (count = " << count << ", prev = " << m_prevTransientScore << ", diff = " << count - m_prevTransientScore << ", ratio = " << (m_totalCount > 0 ? (float (m_n2sum) / float(m_totalCount * m_n1)) : 1.f) << ", ideal = " << m_ratio << ", nominal = " << getRatio() << ")" << std::endl;
+//    } else {
+//        std::cerr << " !transient (count = " << count << ", prev = " << m_prevTransientScore << ", diff = " << count - m_prevTransientScore << ")" << std::endl;
+#endif
+    }
+
+    m_prevTransientScore = count;
+
+    return isTransient;
+}
+
+void
+AudioTimeStretcher::synthesiseBlock(size_t c,
+                                    float *out,
+                                    float *modulation,
+                                    size_t lastStep)
+{
+    bool unchanged = (lastStep == m_n1);
+
+    for (size_t i = 0; i <= m_wlen/2; ++i) {
+		
+        float phase = princargf(atan2f(m_freq[c][i][1], m_freq[c][i][0]));
+        float adjustedPhase = phase;
+
+//        float binfreq = float(m_sampleRate * i) / m_wlen;
+
+        if (!unchanged) {
+
+            float mag = sqrtf(m_freq[c][i][0] * m_freq[c][i][0] +
+                              m_freq[c][i][1] * m_freq[c][i][1]);
+
+            float omega = (2 * M_PI * m_n1 * i) / m_wlen;
+	
+            float expectedPhase = m_prevPhase[c][i] + omega;
+
+            float phaseError = princargf(phase - expectedPhase);
+
+            float phaseIncrement = (omega + phaseError) / m_n1;
+            
+            adjustedPhase = m_prevAdjustedPhase[c][i] +
+                lastStep * phaseIncrement;
+            
+            float real = mag * cosf(adjustedPhase);
+            float imag = mag * sinf(adjustedPhase);
+            m_freq[c][i][0] = real;
+            m_freq[c][i][1] = imag;
+        }
+
+        m_prevPhase[c][i] = phase;
+        m_prevAdjustedPhase[c][i] = adjustedPhase;
+    }
+
+    fftwf_execute(m_iplan[c]); // m_freq -> m_time, inverse fft
+
+    for (size_t i = 0; i < m_wlen/2; ++i) {
+        float temp = m_time[c][i];
+        m_time[c][i] = m_time[c][i + m_wlen/2];
+        m_time[c][i + m_wlen/2] = temp;
+    }
+    
+    for (size_t i = 0; i < m_wlen; ++i) {
+        m_time[c][i] = m_time[c][i] / m_wlen;
+    }
+
+    m_synthesisWindow->cut(m_time[c]);
+
+    for (size_t i = 0; i < m_wlen; ++i) {
+        out[i] += m_time[c][i];
+    }
+
+    if (modulation) {
+
+        float area = m_analysisWindow->getArea();
+
+        for (size_t i = 0; i < m_wlen; ++i) {
+            float val = m_synthesisWindow->getValue(i);
+            modulation[i] += val * area;
+        }
+    }
+}
+
+
+
+}
+
diff --git a/src/sound/AudioTimeStretcher.h b/src/sound/AudioTimeStretcher.h
new file mode 100644
index 0000000..c5d0170
--- /dev/null
+++ b/src/sound/AudioTimeStretcher.h
@@ -0,0 +1,221 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rosegarden
+    A MIDI and audio sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <richard.bown@ferventsoftware.com>
+
+    The moral rights of Guillaume Laurent, Chris Cannam, and Richard
+    Bown to claim authorship of this work have been asserted.
+
+    Other copyrights also apply to some parts of this work.  Please
+    see the AUTHORS file and individual file headers for details.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+/*
+    This file is derived from
+
+    Sonic Visualiser
+    An audio file viewer and annotation editor.
+    Centre for Digital Music, Queen Mary, University of London.
+    This file copyright 2006 Chris Cannam and QMUL.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _AUDIO_TIME_STRETCHER_H_
+#define _AUDIO_TIME_STRETCHER_H_
+
+#include "SampleWindow.h"
+#include "RingBuffer.h"
+
+#include <fftw3.h>
+#include <pthread.h>
+#include <list>
+
+namespace Rosegarden
+{
+
+/**
+ * A time stretcher that alters the performance speed of audio,
+ * preserving pitch.
+ *
+ * This is based on the straightforward phase vocoder with phase
+ * unwrapping (as in e.g. the DAFX book pp275-), with optional
+ * percussive transient detection to avoid smearing percussive notes
+ * and resynchronise phases, and adding a stream API for real-time
+ * use.  Principles and methods from Chris Duxbury, AES 2002 and 2004
+ * thesis; Emmanuel Ravelli, DAFX 2005; Dan Barry, ISSC 2005 on
+ * percussion detection; code by Chris Cannam.
+ */
+
+class AudioTimeStretcher
+{
+public:
+    AudioTimeStretcher(size_t sampleRate,
+                       size_t channels,
+                       float ratio,
+                       bool sharpen,
+                       size_t maxOutputBlockSize);
+    virtual ~AudioTimeStretcher();
+
+    /**
+     * Return the number of samples that would need to be added via
+     * putInput in order to provoke the time stretcher into doing some
+     * time stretching and making more output samples available.
+     * This will be an estimate, if transient sharpening is on; the 
+     * caller may need to do the put/get/test cycle more than once.
+     */
+    size_t getRequiredInputSamples() const;
+
+    /**
+     * Put (and possibly process) a given number of input samples.
+     * Number should usually equal the value returned from
+     * getRequiredInputSamples().
+     */
+    void putInput(float **input, size_t samples);
+
+    /**
+     * Get the number of processed samples ready for reading.
+     */
+    size_t getAvailableOutputSamples() const;
+
+    /**
+     * Get some processed samples.
+     */
+    void getOutput(float **output, size_t samples);
+
+    //!!! and reset?
+
+    /**
+     * Change the time stretch ratio.
+     */
+    void setRatio(float ratio);
+
+    /**
+     * Get the hop size for input.
+     */
+    size_t getInputIncrement() const { return m_n1; }
+
+    /**
+     * Get the hop size for output.
+     */
+    size_t getOutputIncrement() const { return m_n2; }
+
+    /**
+     * Get the window size for FFT processing.
+     */
+    size_t getWindowSize() const { return m_wlen; }
+
+    /**
+     * Get the stretch ratio.
+     */
+    float getRatio() const { return float(m_n2) / float(m_n1); }
+
+    /**
+     * Return whether this time stretcher will attempt to sharpen transients.
+     */
+    bool getSharpening() const { return m_sharpen; }
+
+    /**
+     * Return the number of channels for this time stretcher.
+     */
+    size_t getChannelCount() const { return m_channels; }
+
+    /**
+     * Get the latency added by the time stretcher, in sample frames.
+     * This will be exact if transient sharpening is off, or approximate
+     * if it is on.
+     */
+    size_t getProcessingLatency() const;
+
+protected:
+    /**
+     * Process a single phase vocoder frame from "in" into
+     * m_freq[channel].
+     */
+    void analyseBlock(size_t channel, float *in); // into m_freq[channel]
+
+    /**
+     * Examine m_freq[0..m_channels-1] and return whether a percussive
+     * transient is found.
+     */
+    bool isTransient(); 
+
+    /**
+     * Resynthesise from m_freq[channel] adding in to "out",
+     * adjusting phases on the basis of a prior step size of lastStep.
+     * Also add the window shape in to the modulation array (if
+     * present) -- for use in ensuring the output has the correct
+     * magnitude afterwards.
+     */
+    void synthesiseBlock(size_t channel, float *out, float *modulation,
+                         size_t lastStep);
+
+    void initialise();
+    void calculateParameters();
+    void cleanup();
+
+    bool shouldSharpen() {
+        return m_sharpen && (m_ratio > 0.25);
+    }
+
+    size_t m_sampleRate;
+    size_t m_channels;
+    size_t m_maxOutputBlockSize;
+    float m_ratio;
+    bool m_sharpen;
+    size_t m_n1;
+    size_t m_n2;
+    size_t m_wlen;
+    SampleWindow<float> *m_analysisWindow;
+    SampleWindow<float> *m_synthesisWindow;
+
+    int m_totalCount;
+    int m_transientCount;
+
+    int m_n2sum;
+    int m_n2total;
+    std::list<int> m_n2list;
+    int m_adjustCount;
+
+    float **m_prevPhase;
+    float **m_prevAdjustedPhase;
+
+    float *m_prevTransientMag;
+    int  m_prevTransientScore;
+    int  m_transientThreshold;
+    bool m_prevTransient;
+
+    float *m_tempbuf;
+    float **m_time;
+    fftwf_complex **m_freq;
+    fftwf_plan *m_plan;
+    fftwf_plan *m_iplan;
+    
+    RingBuffer<float> **m_inbuf;
+    RingBuffer<float> **m_outbuf;
+    float **m_mashbuf;
+    float *m_modulationbuf;
+
+    mutable pthread_mutex_t m_mutex;
+};
+
+}
+
+
+#endif
diff --git a/src/sound/Audit.cpp b/src/sound/Audit.cpp
new file mode 100644
index 0000000..25a6c8b
--- /dev/null
+++ b/src/sound/Audit.cpp
@@ -0,0 +1,30 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "Audit.h"
+
+namespace Rosegarden
+{
+
+std::string Audit::m_audit;
+
+}
+
diff --git a/src/sound/Audit.h b/src/sound/Audit.h
new file mode 100644
index 0000000..4e0a20e
--- /dev/null
+++ b/src/sound/Audit.h
@@ -0,0 +1,60 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _AUDIT_H_
+#define _AUDIT_H_
+
+#include <string>
+#include <iostream>
+
+#if (__GNUC__ < 3)
+#include <strstream>
+#define stringstream strstream
+#else
+#include <sstream>
+#endif
+
+// A staggeringly simple-minded audit trail implementation.
+
+namespace Rosegarden {
+
+class Audit : public std::stringstream
+{
+public:
+    Audit() { }
+
+    virtual ~Audit() {
+#if (__GNUC__ < 3)
+        *this << std::ends;
+#endif
+        std::cerr << str();
+        m_audit += str();
+    }
+
+    static std::string getAudit() { return m_audit; }
+
+protected:
+    static std::string m_audit;
+};
+
+}
+
+#endif
diff --git a/src/sound/BWFAudioFile.cpp b/src/sound/BWFAudioFile.cpp
new file mode 100644
index 0000000..c38820f
--- /dev/null
+++ b/src/sound/BWFAudioFile.cpp
@@ -0,0 +1,171 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "BWFAudioFile.h"
+#include "RealTime.h"
+
+#if (__GNUC__ < 3)
+#include <strstream>
+#define stringstream strstream
+#else
+#include <sstream>
+#endif
+
+using std::cout;
+using std::cerr;
+using std::endl;
+
+
+namespace Rosegarden
+{
+
+BWFAudioFile::BWFAudioFile(const unsigned int &id,
+                           const std::string &name,
+                           const std::string &fileName):
+        RIFFAudioFile(id, name, fileName)
+{
+    m_type = WAV;
+
+}
+
+BWFAudioFile::BWFAudioFile(const std::string &fileName,
+                           unsigned int channels = 1,
+                           unsigned int sampleRate = 48000,
+                           unsigned int bytesPerSecond = 6000,
+                           unsigned int bytesPerFrame = 2,
+                           unsigned int bitsPerSample = 16):
+        RIFFAudioFile(0, "", fileName)
+{
+    m_type = WAV;
+    m_bitsPerSample = bitsPerSample;
+    m_sampleRate = sampleRate;
+    m_bytesPerSecond = bytesPerSecond;
+    m_bytesPerFrame = bytesPerFrame;
+    m_channels = channels;
+}
+
+BWFAudioFile::~BWFAudioFile()
+{}
+
+bool
+BWFAudioFile::open()
+{
+    // if already open
+    if (m_inFile && (*m_inFile))
+        return true;
+
+    m_inFile = new std::ifstream(m_fileName.c_str(),
+                                 std::ios::in | std::ios::binary);
+
+    if (!(*m_inFile)) {
+        m_type = UNKNOWN;
+        return false;
+    }
+
+    // Get the file size and store it for comparison later
+    //
+    m_fileSize = m_fileInfo->size();
+
+    try {
+        parseHeader();
+    } catch (BadSoundFileException s) {
+        //throw(s);
+        return false;
+    }
+
+    return true;
+}
+
+// Open the file for writing, write out the header and move
+// to the data chunk to accept samples.  We fill in all the
+// totals when we close().
+//
+bool
+BWFAudioFile::write()
+{
+    // close if we're open
+    if (m_outFile) {
+        m_outFile->close();
+        delete m_outFile;
+    }
+
+    // open for writing
+    m_outFile = new std::ofstream(m_fileName.c_str(),
+                                  std::ios::out | std::ios::binary);
+
+    if (!(*m_outFile))
+        return false;
+
+    // write out format header chunk and prepare for sample writing
+    //
+    writeFormatChunk();
+
+    return true;
+}
+
+void
+BWFAudioFile::close()
+{
+    if (m_outFile == 0)
+        return ;
+
+    m_outFile->seekp(0, std::ios::end);
+    unsigned int totalSize = m_outFile->tellp();
+
+    // seek to first length position
+    m_outFile->seekp(4, std::ios::beg);
+
+    // write complete file size minus 8 bytes to here
+    putBytes(m_outFile, getLittleEndianFromInteger(totalSize - 8, 4));
+
+    // reseek from start forward 40
+    m_outFile->seekp(40, std::ios::beg);
+
+    // write the data chunk size to end
+    putBytes(m_outFile, getLittleEndianFromInteger(totalSize - 44, 4));
+
+    m_outFile->close();
+
+    delete m_outFile;
+    m_outFile = 0;
+}
+
+// Set the AudioFile meta data according to WAV file format specification.
+//
+void
+BWFAudioFile::parseHeader()
+{
+    // Read the format chunk and populate the file data.  A plain WAV
+    // file only has this chunk.  Exceptions tumble through.
+    //
+    readFormatChunk();
+
+}
+
+std::streampos
+BWFAudioFile::getDataOffset()
+{
+    return 0;
+}
+
+
+
+}
diff --git a/src/sound/BWFAudioFile.h b/src/sound/BWFAudioFile.h
new file mode 100644
index 0000000..d6717aa
--- /dev/null
+++ b/src/sound/BWFAudioFile.h
@@ -0,0 +1,94 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+
+// Specialisation of a RIFF file - the WAV defines a format chunk
+// holding audio file meta data and a data chunk with interleaved
+// sample bytes.
+//
+
+#include "RIFFAudioFile.h"
+
+
+#ifndef _BWFAUDIOFILE_H_
+#define _BWFAUDIOFILE_H_
+
+namespace Rosegarden
+{
+
+class BWFAudioFile : public RIFFAudioFile
+{
+public:
+    BWFAudioFile(const unsigned int &id,
+                 const std::string &name,
+                 const std::string &fileName);
+
+    BWFAudioFile(const std::string &fileName,
+                  unsigned int channels,
+                  unsigned int sampleRate,
+                  unsigned int bytesPerSecond,
+                  unsigned int bytesPerFrame,
+                  unsigned int bitsPerSample);
+
+    ~BWFAudioFile();
+
+    // Override these methods for the WAV
+    //
+    virtual bool open();
+    virtual bool write();
+    virtual void close();
+
+    // Get all header information
+    //
+    void parseHeader();
+
+    //
+    // 
+    //virtual std::vector<float> getPreview(const RealTime &resolution);
+
+    // Offset to start of sample data
+    //
+    virtual std::streampos getDataOffset();
+
+    // Peak file name
+    //
+    virtual std::string getPeakFilename()
+        { return (m_fileName + std::string(".pk")); }
+
+
+    //!!! NOT IMPLEMENTED YET
+    // 
+    virtual bool decode(const unsigned char *sourceData,
+                        size_t sourceBytes,
+                        size_t targetSampleRate,
+                        size_t targetChannels,
+                        size_t targetFrames,
+                        std::vector<float *> &targetData,
+                        bool addToResultBuffers = false) { return false; }
+
+protected:
+
+};
+
+}
+
+
+#endif // _BWFUDIOFILE_H_
diff --git a/src/sound/ControlBlock.cpp b/src/sound/ControlBlock.cpp
new file mode 100644
index 0000000..d0bb6a8
--- /dev/null
+++ b/src/sound/ControlBlock.cpp
@@ -0,0 +1,181 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include <cstring>
+
+#include "ControlBlock.h"
+
+namespace Rosegarden
+{
+
+ControlBlock::ControlBlock(unsigned int maxTrackId)
+        : m_maxTrackId(maxTrackId),
+        m_solo(false),
+        m_routing(true),
+        m_thruFilter(0),
+        m_recordFilter(0),
+        m_selectedTrack(0)
+{
+    m_metronomeInfo.muted = true;
+    m_metronomeInfo.instrumentId = 0;
+    for (unsigned int i = 0; i < CONTROLBLOCK_MAX_NB_TRACKS; ++i) {
+        m_trackInfo[i].muted = true;
+        m_trackInfo[i].deleted = true;
+        m_trackInfo[i].armed = true;
+        m_trackInfo[i].instrumentId = 0;
+    }
+}
+
+ControlBlock::ControlBlock()
+{
+    // DO NOT initialize anything - this ctor is meant to be used by
+    // the sequencer, through a placement new over an mmapped file.
+}
+
+void ControlBlock::updateTrackData(Track* t)
+{
+    if (t) {
+        setInstrumentForTrack(t->getId(), t->getInstrument());
+        setTrackArmed(t->getId(), t->isArmed());
+        setTrackMuted(t->getId(), t->isMuted());
+        setTrackDeleted(t->getId(), false);
+        setTrackChannelFilter(t->getId(), t->getMidiInputChannel());
+        setTrackDeviceFilter(t->getId(), t->getMidiInputDevice());
+        if (t->getId() > m_maxTrackId)
+            m_maxTrackId = t->getId();
+    }
+}
+
+void ControlBlock::setInstrumentForTrack(TrackId trackId, InstrumentId instId)
+{
+    if (trackId < CONTROLBLOCK_MAX_NB_TRACKS)
+        m_trackInfo[trackId].instrumentId = instId;
+}
+
+InstrumentId ControlBlock::getInstrumentForTrack(TrackId trackId) const
+{
+    if (trackId < CONTROLBLOCK_MAX_NB_TRACKS)
+        return m_trackInfo[trackId].instrumentId;
+    return 0;
+}
+
+void ControlBlock::setTrackMuted(TrackId trackId, bool mute)
+{
+    if (trackId < CONTROLBLOCK_MAX_NB_TRACKS)
+        m_trackInfo[trackId].muted = mute;
+}
+
+bool ControlBlock::isTrackMuted(TrackId trackId) const
+{
+    if (trackId < CONTROLBLOCK_MAX_NB_TRACKS)
+        return m_trackInfo[trackId].muted;
+    return true;
+}
+
+void ControlBlock::setTrackArmed(TrackId trackId, bool armed)
+{
+    if (trackId < CONTROLBLOCK_MAX_NB_TRACKS)
+        m_trackInfo[trackId].armed = armed;
+}
+
+bool ControlBlock::isTrackArmed(TrackId trackId) const
+{
+    if (trackId < CONTROLBLOCK_MAX_NB_TRACKS)
+        return m_trackInfo[trackId].armed;
+    return false;
+}
+
+void ControlBlock::setTrackDeleted(TrackId trackId, bool deleted)
+{
+    if (trackId < CONTROLBLOCK_MAX_NB_TRACKS)
+        m_trackInfo[trackId].deleted = deleted;
+}
+
+bool ControlBlock::isTrackDeleted(TrackId trackId) const
+{
+    if (trackId < CONTROLBLOCK_MAX_NB_TRACKS)
+        return m_trackInfo[trackId].deleted;
+    return true;
+}
+
+void ControlBlock::setTrackChannelFilter(TrackId trackId, char channel)
+{
+    if (trackId < CONTROLBLOCK_MAX_NB_TRACKS)
+        m_trackInfo[trackId].channelFilter = channel;
+}
+
+char ControlBlock::getTrackChannelFilter(TrackId trackId) const
+{
+    if (trackId < CONTROLBLOCK_MAX_NB_TRACKS)
+        return m_trackInfo[trackId].channelFilter;
+    return -1;
+}
+
+void ControlBlock::setTrackDeviceFilter(TrackId trackId, DeviceId device)
+{
+    if (trackId < CONTROLBLOCK_MAX_NB_TRACKS)
+        m_trackInfo[trackId].deviceFilter = device;
+}
+
+DeviceId ControlBlock::getTrackDeviceFilter(TrackId trackId) const
+{
+    if (trackId < CONTROLBLOCK_MAX_NB_TRACKS)
+        return m_trackInfo[trackId].deviceFilter;
+    return Device::ALL_DEVICES;
+}
+
+bool ControlBlock::isInstrumentMuted(InstrumentId instrumentId) const
+{
+    for (unsigned int i = 0; i <= m_maxTrackId; ++i) {
+        if (m_trackInfo[i].instrumentId == instrumentId &&
+                !m_trackInfo[i].deleted && !m_trackInfo[i].muted)
+            return false;
+    }
+    return true;
+}
+
+bool ControlBlock::isInstrumentUnused(InstrumentId instrumentId) const
+{
+    for (unsigned int i = 0; i <= m_maxTrackId; ++i) {
+        if (m_trackInfo[i].instrumentId == instrumentId &&
+                !m_trackInfo[i].deleted)
+            return false;
+    }
+    return true;
+}
+
+InstrumentId ControlBlock::getInstrumentForEvent(unsigned int dev, unsigned int chan)
+{
+    for (unsigned int i = 0; i <= m_maxTrackId; ++i) {
+        if (!m_trackInfo[i].deleted && m_trackInfo[i].armed) {
+            if (((m_trackInfo[i].deviceFilter == Device::ALL_DEVICES) ||
+                    (m_trackInfo[i].deviceFilter == dev)) &&
+                    ((m_trackInfo[i].channelFilter == -1) ||
+                     (m_trackInfo[i].channelFilter == chan)))
+                return m_trackInfo[i].instrumentId;
+        }
+    }
+    // There is not a matching filter, return the selected track instrument
+    return getInstrumentForTrack(getSelectedTrack());
+}
+
+
+}
diff --git a/src/sound/ControlBlock.h b/src/sound/ControlBlock.h
new file mode 100644
index 0000000..6db0ee5
--- /dev/null
+++ b/src/sound/ControlBlock.h
@@ -0,0 +1,128 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _CONTROLBLOCK_H_
+#define _CONTROLBLOCK_H_
+
+#include "MidiProgram.h"
+#include "Track.h"
+
+namespace Rosegarden 
+{
+
+/**
+ * ONLY PUT PLAIN DATA HERE - NO POINTERS EVER
+ */
+struct TrackInfo 
+{
+    bool deleted;
+    bool muted;
+    bool armed;
+    char channelFilter;
+    DeviceId deviceFilter;
+    InstrumentId instrumentId;
+};
+
+#define CONTROLBLOCK_MAX_NB_TRACKS 1024 // can't be a symbol
+
+/**
+ * Sequencer control block, mmapped by the GUI
+ */
+class ControlBlock
+{
+public:
+    /// ctor for GUI
+    ControlBlock(unsigned int maxTrackId);
+
+    /// ctor for sequencer - all data is read from mmapped file
+    ControlBlock();
+
+    unsigned int getMaxTrackId() const { return m_maxTrackId; }
+    void updateTrackData(Track*);
+
+    void setInstrumentForTrack(TrackId trackId, InstrumentId);
+    InstrumentId getInstrumentForTrack(TrackId trackId) const;
+
+    void setTrackArmed(TrackId trackId, bool);
+    bool isTrackArmed(TrackId trackId) const;
+
+    void setTrackMuted(TrackId trackId, bool);
+    bool isTrackMuted(TrackId trackId) const;
+
+    void setTrackDeleted(TrackId trackId, bool);
+    bool isTrackDeleted(TrackId trackId) const;
+    
+    void setTrackChannelFilter(TrackId trackId, char);
+    char getTrackChannelFilter(TrackId trackId) const;
+    
+    void setTrackDeviceFilter(TrackId trackId, DeviceId);
+    DeviceId getTrackDeviceFilter(TrackId trackId) const;
+    
+    bool isInstrumentMuted(InstrumentId instrumentId) const;
+    bool isInstrumentUnused(InstrumentId instrumentId) const;
+
+    void setInstrumentForMetronome(InstrumentId instId) { m_metronomeInfo.instrumentId = instId; }
+    InstrumentId getInstrumentForMetronome() const      { return m_metronomeInfo.instrumentId; }
+
+    void setMetronomeMuted(bool mute) { m_metronomeInfo.muted = mute; }
+    bool isMetronomeMuted() const     { return m_metronomeInfo.muted; }
+
+    bool isSolo() const      { return m_solo; }
+    void setSolo(bool value) { m_solo = value; }
+    TrackId getSelectedTrack() const     { return m_selectedTrack; }
+    void setSelectedTrack(TrackId track) { m_selectedTrack = track; }
+
+    void setThruFilter(MidiFilter filter) { m_thruFilter = filter; }
+    MidiFilter getThruFilter() const { return m_thruFilter; }
+
+    void setRecordFilter(MidiFilter filter) { m_recordFilter = filter; }
+    MidiFilter getRecordFilter() const { return m_recordFilter; }
+    
+    void setMidiRoutingEnabled(bool enabled) { m_routing = enabled; }
+    bool isMidiRoutingEnabled() const { return m_routing; } 
+    
+    /**
+     * Gets an InstrumentId for the given DeviceId and Channel. If there
+     * is an armed track having a matching device and channel filters, 
+     * this method returns the instrument assigned to the track, even if 
+     * there are more tracks matching the same filters. If there is not a
+     * single match, it returns the instrument assigned to the selected
+     * track.
+     */
+    InstrumentId getInstrumentForEvent(unsigned int dev, 
+                                       unsigned int chan);
+
+protected:
+    //--------------- Data members ---------------------------------
+    // PUT ONLY PLAIN DATA HERE - NO POINTERS EVER
+    unsigned int m_maxTrackId;
+    bool m_solo;
+    bool m_routing;
+    MidiFilter m_thruFilter;
+    MidiFilter m_recordFilter;
+    TrackId m_selectedTrack;
+    TrackInfo m_metronomeInfo;
+    TrackInfo m_trackInfo[CONTROLBLOCK_MAX_NB_TRACKS]; // should be high enough for the moment
+};
+
+}
+
+#endif
diff --git a/src/sound/DSSIPluginFactory.cpp b/src/sound/DSSIPluginFactory.cpp
new file mode 100644
index 0000000..8447450
--- /dev/null
+++ b/src/sound/DSSIPluginFactory.cpp
@@ -0,0 +1,396 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "DSSIPluginFactory.h"
+#include <iostream>
+#include <cstdlib>
+
+#ifdef HAVE_DSSI
+
+#include <dlfcn.h>
+#include "AudioPluginInstance.h"
+#include "DSSIPluginInstance.h"
+#include "MappedStudio.h"
+#include "PluginIdentifier.h"
+
+#ifdef HAVE_LIBLRDF
+#include "lrdf.h"
+#endif // HAVE_LIBLRDF
+
+namespace Rosegarden
+{
+
+DSSIPluginFactory::DSSIPluginFactory() :
+        LADSPAPluginFactory()
+{
+    // nothing else to do
+}
+
+DSSIPluginFactory::~DSSIPluginFactory()
+{
+    // nothing else to do here either
+}
+
+void
+DSSIPluginFactory::enumeratePlugins(MappedObjectPropertyList &list)
+{
+    for (std::vector<QString>::iterator i = m_identifiers.begin();
+            i != m_identifiers.end(); ++i) {
+
+        const DSSI_Descriptor *ddesc = getDSSIDescriptor(*i);
+        if (!ddesc)
+            continue;
+
+        const LADSPA_Descriptor *descriptor = ddesc->LADSPA_Plugin;
+        if (!descriptor)
+            continue;
+
+        //	std::cerr << "DSSIPluginFactory::enumeratePlugins: Name " << (descriptor->Name ? descriptor->Name : "NONE" ) << std::endl;
+
+        list.push_back(*i);
+        list.push_back(descriptor->Name);
+        list.push_back(QString("%1").arg(descriptor->UniqueID));
+        list.push_back(descriptor->Label);
+        list.push_back(descriptor->Maker);
+        list.push_back(descriptor->Copyright);
+        list.push_back((ddesc->run_synth || ddesc->run_multiple_synths) ? "true" : "false");
+        list.push_back(ddesc->run_multiple_synths ? "true" : "false");
+        list.push_back(m_taxonomy[descriptor->UniqueID]);
+        list.push_back(QString("%1").arg(descriptor->PortCount));
+
+        for (unsigned long p = 0; p < descriptor->PortCount; ++p) {
+
+            int type = 0;
+            if (LADSPA_IS_PORT_CONTROL(descriptor->PortDescriptors[p])) {
+                type |= PluginPort::Control;
+            } else {
+                type |= PluginPort::Audio;
+            }
+            if (LADSPA_IS_PORT_INPUT(descriptor->PortDescriptors[p])) {
+                type |= PluginPort::Input;
+            } else {
+                type |= PluginPort::Output;
+            }
+
+            list.push_back(QString("%1").arg(p));
+            list.push_back(descriptor->PortNames[p]);
+            list.push_back(QString("%1").arg(type));
+            list.push_back(QString("%1").arg(getPortDisplayHint(descriptor, p)));
+            list.push_back(QString("%1").arg(getPortMinimum(descriptor, p)));
+            list.push_back(QString("%1").arg(getPortMaximum(descriptor, p)));
+            list.push_back(QString("%1").arg(getPortDefault(descriptor, p)));
+        }
+    }
+
+    unloadUnusedLibraries();
+}
+
+
+void
+DSSIPluginFactory::populatePluginSlot(QString identifier, MappedPluginSlot &slot)
+{
+    const LADSPA_Descriptor *descriptor = getLADSPADescriptor(identifier);
+    if (!descriptor)
+        return ;
+
+    if (descriptor) {
+
+        slot.setProperty(MappedPluginSlot::Label, descriptor->Label);
+        slot.setProperty(MappedPluginSlot::PluginName, descriptor->Name);
+        slot.setProperty(MappedPluginSlot::Author, descriptor->Maker);
+        slot.setProperty(MappedPluginSlot::Copyright, descriptor->Copyright);
+        slot.setProperty(MappedPluginSlot::PortCount, descriptor->PortCount);
+        slot.setProperty(MappedPluginSlot::Category, m_taxonomy[descriptor->UniqueID]);
+
+        slot.destroyChildren();
+
+        for (unsigned long i = 0; i < descriptor->PortCount; i++) {
+
+            if (LADSPA_IS_PORT_CONTROL(descriptor->PortDescriptors[i]) &&
+                    LADSPA_IS_PORT_INPUT(descriptor->PortDescriptors[i])) {
+
+                MappedStudio *studio = dynamic_cast<MappedStudio *>(slot.getParent());
+                if (!studio) {
+                    std::cerr << "WARNING: DSSIPluginFactory::populatePluginSlot: can't find studio" << std::endl;
+                    return ;
+                }
+
+                MappedPluginPort *port =
+                    dynamic_cast<MappedPluginPort *>
+                    (studio->createObject(MappedObject::PluginPort));
+
+                slot.addChild(port);
+                port->setParent(&slot);
+
+                port->setProperty(MappedPluginPort::PortNumber, i);
+                port->setProperty(MappedPluginPort::Name,
+                                  descriptor->PortNames[i]);
+                port->setProperty(MappedPluginPort::Maximum,
+                                  getPortMaximum(descriptor, i));
+                port->setProperty(MappedPluginPort::Minimum,
+                                  getPortMinimum(descriptor, i));
+                port->setProperty(MappedPluginPort::Default,
+                                  getPortDefault(descriptor, i));
+                port->setProperty(MappedPluginPort::DisplayHint,
+                                  getPortDisplayHint(descriptor, i));
+            }
+        }
+    }
+
+    //!!! leak here if the plugin is not instantiated too...?
+}
+
+RunnablePluginInstance *
+DSSIPluginFactory::instantiatePlugin(QString identifier,
+                                     int instrument,
+                                     int position,
+                                     unsigned int sampleRate,
+                                     unsigned int blockSize,
+                                     unsigned int channels)
+{
+    const DSSI_Descriptor *descriptor = getDSSIDescriptor(identifier);
+
+    if (descriptor) {
+
+        DSSIPluginInstance *instance =
+            new DSSIPluginInstance
+            (this, instrument, identifier, position, sampleRate, blockSize, channels,
+             descriptor);
+
+        m_instances.insert(instance);
+
+        return instance;
+    }
+
+    return 0;
+}
+
+
+const DSSI_Descriptor *
+DSSIPluginFactory::getDSSIDescriptor(QString identifier)
+{
+    QString type, soname, label;
+    PluginIdentifier::parseIdentifier(identifier, type, soname, label);
+
+    if (m_libraryHandles.find(soname) == m_libraryHandles.end()) {
+        loadLibrary(soname);
+        if (m_libraryHandles.find(soname) == m_libraryHandles.end()) {
+            std::cerr << "WARNING: DSSIPluginFactory::getDSSIDescriptor: loadLibrary failed for " << soname << std::endl;
+            return 0;
+        }
+    }
+
+    void *libraryHandle = m_libraryHandles[soname];
+
+    DSSI_Descriptor_Function fn = (DSSI_Descriptor_Function)
+                                  dlsym(libraryHandle, "dssi_descriptor");
+
+    if (!fn) {
+        std::cerr << "WARNING: DSSIPluginFactory::getDSSIDescriptor: No descriptor function in library " << soname << std::endl;
+        return 0;
+    }
+
+    const DSSI_Descriptor *descriptor = 0;
+
+    int index = 0;
+    while ((descriptor = fn(index))) {
+        if (descriptor->LADSPA_Plugin->Label == label)
+            return descriptor;
+        ++index;
+    }
+
+    std::cerr << "WARNING: DSSIPluginFactory::getDSSIDescriptor: No such plugin as " << label << " in library " << soname << std::endl;
+
+    return 0;
+}
+
+const LADSPA_Descriptor *
+DSSIPluginFactory::getLADSPADescriptor(QString identifier)
+{
+    const DSSI_Descriptor *dssiDescriptor = getDSSIDescriptor(identifier);
+    if (dssiDescriptor)
+        return dssiDescriptor->LADSPA_Plugin;
+    else
+        return 0;
+}
+
+
+std::vector<QString>
+DSSIPluginFactory::getPluginPath()
+{
+    std::vector<QString> pathList;
+    std::string path;
+
+    char *cpath = getenv("DSSI_PATH");
+    if (cpath)
+        path = cpath;
+
+    if (path == "") {
+        path = "/usr/local/lib/dssi:/usr/lib/dssi";
+        char *home = getenv("HOME");
+        if (home)
+            path = std::string(home) + "/.dssi:" + path;
+    }
+
+    std::string::size_type index = 0, newindex = 0;
+
+    while ((newindex = path.find(':', index)) < path.size()) {
+        pathList.push_back(path.substr(index, newindex - index).c_str());
+        index = newindex + 1;
+    }
+
+    pathList.push_back(path.substr(index).c_str());
+
+    return pathList;
+}
+
+
+#ifdef HAVE_LIBLRDF
+std::vector<QString>
+DSSIPluginFactory::getLRDFPath(QString &baseUri)
+{
+    std::vector<QString> pathList = getPluginPath();
+    std::vector<QString> lrdfPaths;
+
+    lrdfPaths.push_back("/usr/local/share/dssi/rdf");
+    lrdfPaths.push_back("/usr/share/dssi/rdf");
+
+    lrdfPaths.push_back("/usr/local/share/ladspa/rdf");
+    lrdfPaths.push_back("/usr/share/ladspa/rdf");
+
+    for (std::vector<QString>::iterator i = pathList.begin();
+            i != pathList.end(); ++i) {
+        lrdfPaths.push_back(*i + "/rdf");
+    }
+
+#ifdef DSSI_BASE
+    baseUri = DSSI_BASE;
+#else
+
+    baseUri = "http://dssi.sourceforge.net/ontology#";
+#endif
+
+    return lrdfPaths;
+}
+#endif
+
+
+void
+DSSIPluginFactory::discoverPlugins(QString soName)
+{
+    void *libraryHandle = dlopen(soName.data(), RTLD_LAZY);
+
+    if (!libraryHandle) {
+        std::cerr << "WARNING: DSSIPluginFactory::discoverPlugins: couldn't dlopen "
+        << soName << " - " << dlerror() << std::endl;
+        return ;
+    }
+
+    DSSI_Descriptor_Function fn = (DSSI_Descriptor_Function)
+                                  dlsym(libraryHandle, "dssi_descriptor");
+
+    if (!fn) {
+        std::cerr << "WARNING: DSSIPluginFactory::discoverPlugins: No descriptor function in " << soName << std::endl;
+        return ;
+    }
+
+    const DSSI_Descriptor *descriptor = 0;
+
+    int index = 0;
+    while ((descriptor = fn(index))) {
+
+        const LADSPA_Descriptor * ladspaDescriptor = descriptor->LADSPA_Plugin;
+        if (!ladspaDescriptor) {
+            std::cerr << "WARNING: DSSIPluginFactory::discoverPlugins: No LADSPA descriptor for plugin " << index << " in " << soName << std::endl;
+            ++index;
+            continue;
+        }
+
+#ifdef HAVE_LIBLRDF
+        char *def_uri = 0;
+        lrdf_defaults *defs = 0;
+
+        QString category = m_taxonomy[ladspaDescriptor->UniqueID];
+
+        if (category == "" && ladspaDescriptor->Name != 0) {
+            std::string name = ladspaDescriptor->Name;
+            if (name.length() > 4 &&
+                    name.substr(name.length() - 4) == " VST") {
+                if (descriptor->run_synth || descriptor->run_multiple_synths) {
+                    category = "VST instruments";
+                } else {
+                    category = "VST effects";
+                }
+                m_taxonomy[ladspaDescriptor->UniqueID] = category;
+            }
+        }
+
+        //	std::cerr << "Plugin id is " << ladspaDescriptor->UniqueID
+        //		  << ", category is \"" << (category ? category : QString("(none)"))
+        //		  << "\", name is " << ladspaDescriptor->Name
+        //		  << ", label is " << ladspaDescriptor->Label
+        //		  << std::endl;
+
+        def_uri = lrdf_get_default_uri(ladspaDescriptor->UniqueID);
+        if (def_uri) {
+            defs = lrdf_get_setting_values(def_uri);
+        }
+
+        int controlPortNumber = 1;
+
+        for (unsigned long i = 0; i < ladspaDescriptor->PortCount; i++) {
+
+            if (LADSPA_IS_PORT_CONTROL(ladspaDescriptor->PortDescriptors[i])) {
+
+                if (def_uri && defs) {
+
+                    for (int j = 0; j < defs->count; j++) {
+                        if (defs->items[j].pid == controlPortNumber) {
+                            //			    std::cerr << "Default for this port (" << defs->items[j].pid << ", " << defs->items[j].label << ") is " << defs->items[j].value << "; applying this to port number " << i << " with name " << ladspaDescriptor->PortNames[i] << std::endl;
+                            m_portDefaults[ladspaDescriptor->UniqueID][i] =
+                                defs->items[j].value;
+                        }
+                    }
+                }
+
+                ++controlPortNumber;
+            }
+        }
+#endif // HAVE_LIBLRDF
+
+        QString identifier = PluginIdentifier::createIdentifier
+                             ("dssi", soName, ladspaDescriptor->Label);
+        m_identifiers.push_back(identifier);
+
+        ++index;
+    }
+
+    if (dlclose(libraryHandle) != 0) {
+        std::cerr << "WARNING: DSSIPluginFactory::discoverPlugins - can't unload " << libraryHandle << std::endl;
+        return ;
+    }
+}
+
+
+}
+
+#endif // HAVE_DSSI
+
diff --git a/src/sound/DSSIPluginFactory.h b/src/sound/DSSIPluginFactory.h
new file mode 100644
index 0000000..8c1bd7c
--- /dev/null
+++ b/src/sound/DSSIPluginFactory.h
@@ -0,0 +1,72 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _DSSI_PLUGIN_FACTORY_H_
+#define _DSSI_PLUGIN_FACTORY_H_
+
+#ifdef HAVE_DSSI
+
+#include "LADSPAPluginFactory.h"
+#include <dssi.h>
+
+namespace Rosegarden
+{
+
+class DSSIPluginInstance;
+
+class DSSIPluginFactory : public LADSPAPluginFactory
+{
+public:
+    virtual ~DSSIPluginFactory();
+
+    virtual void enumeratePlugins(MappedObjectPropertyList &list);
+
+    virtual void populatePluginSlot(QString identifier, MappedPluginSlot &slot);
+
+    virtual RunnablePluginInstance *instantiatePlugin(QString identifier,
+                                                      int instrumentId,
+                                                      int position,
+                                                      unsigned int sampleRate,
+                                                      unsigned int blockSize,
+                                                      unsigned int channels);
+
+protected:
+    DSSIPluginFactory();
+    friend class PluginFactory;
+
+    virtual std::vector<QString> getPluginPath();
+
+#ifdef HAVE_LIBLRDF
+    virtual std::vector<QString> getLRDFPath(QString &baseUri);
+#endif
+
+    virtual void discoverPlugins(QString soName);
+
+    virtual const LADSPA_Descriptor *getLADSPADescriptor(QString identifier);
+    virtual const DSSI_Descriptor *getDSSIDescriptor(QString identifier);
+};
+
+}
+
+#endif
+
+#endif
+
diff --git a/src/sound/DSSIPluginInstance.cpp b/src/sound/DSSIPluginInstance.cpp
new file mode 100644
index 0000000..2ceb0df
--- /dev/null
+++ b/src/sound/DSSIPluginInstance.cpp
@@ -0,0 +1,1208 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include <iostream>
+#include <cassert>
+#include <cstdlib>
+
+#include "DSSIPluginInstance.h"
+#include "PluginIdentifier.h"
+#include "LADSPAPluginFactory.h"
+
+#ifdef HAVE_DSSI
+
+//#define DEBUG_DSSI 1
+//#define DEBUG_DSSI_PROCESS 1
+
+namespace Rosegarden
+{
+
+#define EVENT_BUFFER_SIZE 1023
+
+DSSIPluginInstance::GroupMap DSSIPluginInstance::m_groupMap;
+snd_seq_event_t **DSSIPluginInstance::m_groupLocalEventBuffers = 0;
+size_t DSSIPluginInstance::m_groupLocalEventBufferCount = 0;
+Scavenger<ScavengerArrayWrapper<snd_seq_event_t *> > DSSIPluginInstance::m_bufferScavenger(2, 10);
+
+
+DSSIPluginInstance::DSSIPluginInstance(PluginFactory *factory,
+                                       InstrumentId instrument,
+                                       QString identifier,
+                                       int position,
+                                       unsigned long sampleRate,
+                                       size_t blockSize,
+                                       int idealChannelCount,
+                                       const DSSI_Descriptor* descriptor) :
+        RunnablePluginInstance(factory, identifier),
+        m_instrument(instrument),
+        m_position(position),
+        m_descriptor(descriptor),
+        m_programCacheValid(false),
+        m_eventBuffer(EVENT_BUFFER_SIZE),
+        m_blockSize(blockSize),
+        m_idealChannelCount(idealChannelCount),
+        m_sampleRate(sampleRate),
+        m_latencyPort(0),
+        m_run(false),
+        m_runSinceReset(false),
+        m_bypassed(false),
+        m_grouped(false)
+{
+    pthread_mutex_t initialisingMutex = PTHREAD_MUTEX_INITIALIZER;
+    memcpy(&m_processLock, &initialisingMutex, sizeof(pthread_mutex_t));
+
+#ifdef DEBUG_DSSI
+
+    std::cerr << "DSSIPluginInstance::DSSIPluginInstance(" << identifier << ")"
+    << std::endl;
+#endif
+
+    init();
+
+    m_inputBuffers = new sample_t * [m_audioPortsIn.size()];
+    m_outputBuffers = new sample_t * [m_outputBufferCount];
+
+    for (size_t i = 0; i < m_audioPortsIn.size(); ++i) {
+        m_inputBuffers[i] = new sample_t[blockSize];
+    }
+    for (size_t i = 0; i < m_outputBufferCount; ++i) {
+        m_outputBuffers[i] = new sample_t[blockSize];
+    }
+
+    m_ownBuffers = true;
+
+    m_pending.lsb = m_pending.msb = m_pending.program = -1;
+
+    instantiate(sampleRate);
+    if (isOK()) {
+        connectPorts();
+        activate();
+        initialiseGroupMembership();
+    }
+}
+
+DSSIPluginInstance::DSSIPluginInstance(PluginFactory *factory,
+                                       InstrumentId instrument,
+                                       QString identifier,
+                                       int position,
+                                       unsigned long sampleRate,
+                                       size_t blockSize,
+                                       sample_t **inputBuffers,
+                                       sample_t **outputBuffers,
+                                       const DSSI_Descriptor* descriptor) :
+        RunnablePluginInstance(factory, identifier),
+        m_instrument(instrument),
+        m_position(position),
+        m_descriptor(descriptor),
+        m_eventBuffer(EVENT_BUFFER_SIZE),
+        m_blockSize(blockSize),
+        m_inputBuffers(inputBuffers),
+        m_outputBuffers(outputBuffers),
+        m_ownBuffers(false),
+        m_idealChannelCount(0),
+        m_sampleRate(sampleRate),
+        m_latencyPort(0),
+        m_run(false),
+        m_runSinceReset(false),
+        m_bypassed(false),
+        m_grouped(false)
+{
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance::DSSIPluginInstance[buffers supplied](" << identifier << ")"
+    << std::endl;
+#endif
+
+    init();
+
+    m_pending.lsb = m_pending.msb = m_pending.program = -1;
+
+    instantiate(sampleRate);
+    if (isOK()) {
+        connectPorts();
+        activate();
+        if (m_descriptor->run_multiple_synths) {
+            m_grouped = true;
+            initialiseGroupMembership();
+        }
+    }
+}
+
+
+void
+DSSIPluginInstance::init()
+{
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance::init" << std::endl;
+#endif
+
+    // Discover ports numbers and identities
+    //
+    const LADSPA_Descriptor *descriptor = m_descriptor->LADSPA_Plugin;
+
+    for (unsigned long i = 0; i < descriptor->PortCount; ++i) {
+        if (LADSPA_IS_PORT_AUDIO(descriptor->PortDescriptors[i])) {
+            if (LADSPA_IS_PORT_INPUT(descriptor->PortDescriptors[i])) {
+                m_audioPortsIn.push_back(i);
+            } else {
+                m_audioPortsOut.push_back(i);
+            }
+        } else
+            if (LADSPA_IS_PORT_CONTROL(descriptor->PortDescriptors[i])) {
+                if (LADSPA_IS_PORT_INPUT(descriptor->PortDescriptors[i])) {
+
+                    LADSPA_Data *data = new LADSPA_Data(0.0);
+
+                    m_controlPortsIn.push_back(std::pair<unsigned long, LADSPA_Data*>
+                                               (i, data));
+
+                    m_backupControlPortsIn.push_back(0.0);
+                    m_portChangedSinceProgramChange.push_back(false);
+
+                } else {
+                    LADSPA_Data *data = new LADSPA_Data(0.0);
+                    m_controlPortsOut.push_back(
+                        std::pair<unsigned long, LADSPA_Data*>(i, data));
+                    if (!strcmp(descriptor->PortNames[i], "latency") ||
+                            !strcmp(descriptor->PortNames[i], "_latency")) {
+#ifdef DEBUG_DSSI
+                        std::cerr << "Wooo! We have a latency port!" << std::endl;
+#endif
+
+                        m_latencyPort = data;
+                    }
+                }
+            }
+#ifdef DEBUG_DSSI
+            else
+                std::cerr << "DSSIPluginInstance::DSSIPluginInstance - "
+                << "unrecognised port type" << std::endl;
+#endif
+
+    }
+
+    m_outputBufferCount = std::max(m_idealChannelCount, m_audioPortsOut.size());
+}
+
+size_t
+DSSIPluginInstance::getLatency()
+{
+#ifdef DEBUG_DSSI 
+    //    std::cerr << "DSSIPluginInstance::getLatency(): m_latencyPort " << m_latencyPort << ", m_run " << m_run << std::endl;
+#endif
+
+    if (m_latencyPort) {
+        if (!m_run) {
+            for (int i = 0; i < getAudioInputCount(); ++i) {
+                for (int j = 0; j < m_blockSize; ++j) {
+                    m_inputBuffers[i][j] = 0.f;
+                }
+            }
+            run(RealTime::zeroTime);
+	}
+#ifdef DEBUG_DSSI
+
+        std::cerr << "DSSIPluginInstance::getLatency(): latency is " << (size_t)(*m_latencyPort + 0.1) << std::endl;
+#endif
+
+        return (size_t)(*m_latencyPort + 0.1);
+    }
+    return 0;
+}
+
+void
+DSSIPluginInstance::silence()
+{
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance::silence: m_run " << m_run << ", m_runSinceReset " << m_runSinceReset << std::endl;
+#endif
+
+    if (m_run && !m_runSinceReset) {
+        return ;
+    }
+    if (m_instanceHandle != 0) {
+        deactivate();
+        activate();
+    }
+    m_runSinceReset = false;
+}
+
+void
+DSSIPluginInstance::discardEvents()
+{
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance::discardEvents" << std::endl;
+#endif
+
+    m_eventBuffer.reset();
+}
+
+void
+DSSIPluginInstance::setIdealChannelCount(size_t channels)
+{
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance::setIdealChannelCount: channel count "
+    << channels << " (was " << m_idealChannelCount << ")" << std::endl;
+#endif
+
+    if (channels == m_idealChannelCount) {
+        silence();
+        return ;
+    }
+
+    if (m_instanceHandle != 0) {
+        deactivate();
+    }
+
+    m_idealChannelCount = channels;
+
+    if (channels > m_outputBufferCount) {
+
+        for (size_t i = 0; i < m_outputBufferCount; ++i) {
+            delete[] m_outputBuffers[i];
+        }
+
+        delete[] m_outputBuffers;
+
+        m_outputBufferCount = channels;
+
+        m_outputBuffers = new sample_t * [m_outputBufferCount];
+
+        for (size_t i = 0; i < m_outputBufferCount; ++i) {
+            m_outputBuffers[i] = new sample_t[m_blockSize];
+        }
+
+        connectPorts();
+    }
+
+    if (m_instanceHandle != 0) {
+        activate();
+    }
+}
+
+void
+DSSIPluginInstance::detachFromGroup()
+{
+    if (!m_grouped)
+        return ;
+    m_groupMap[m_identifier].erase(this);
+    m_grouped = false;
+}
+
+void
+DSSIPluginInstance::initialiseGroupMembership()
+{
+    if (!m_descriptor->run_multiple_synths) {
+        m_grouped = false;
+        return ;
+    }
+
+    //!!! GroupMap is not actually thread-safe.
+
+    size_t pluginsInGroup = m_groupMap[m_identifier].size();
+
+    if (++pluginsInGroup > m_groupLocalEventBufferCount) {
+
+        size_t nextBufferCount = pluginsInGroup * 2;
+
+        snd_seq_event_t **eventLocalBuffers = new snd_seq_event_t * [nextBufferCount];
+
+        for (size_t i = 0; i < m_groupLocalEventBufferCount; ++i) {
+            eventLocalBuffers[i] = m_groupLocalEventBuffers[i];
+        }
+        for (size_t i = m_groupLocalEventBufferCount; i < nextBufferCount; ++i) {
+            eventLocalBuffers[i] = new snd_seq_event_t[EVENT_BUFFER_SIZE];
+        }
+
+        if (m_groupLocalEventBuffers) {
+            m_bufferScavenger.claim(new ScavengerArrayWrapper<snd_seq_event_t *>
+                                    (m_groupLocalEventBuffers));
+        }
+
+        m_groupLocalEventBuffers = eventLocalBuffers;
+        m_groupLocalEventBufferCount = nextBufferCount;
+    }
+
+    m_grouped = true;
+    m_groupMap[m_identifier].insert(this);
+}
+
+DSSIPluginInstance::~DSSIPluginInstance()
+{
+//    std::cerr << "DSSIPluginInstance::~DSSIPluginInstance" << std::endl;
+
+    detachFromGroup();
+
+    if (m_instanceHandle != 0) {
+        deactivate();
+    }
+
+    cleanup();
+
+    for (unsigned int i = 0; i < m_controlPortsIn.size(); ++i)
+        delete m_controlPortsIn[i].second;
+
+    for (unsigned int i = 0; i < m_controlPortsOut.size(); ++i)
+        delete m_controlPortsOut[i].second;
+
+    m_controlPortsIn.clear();
+    m_controlPortsOut.clear();
+
+    if (m_ownBuffers) {
+        for (size_t i = 0; i < m_audioPortsIn.size(); ++i) {
+            delete[] m_inputBuffers[i];
+        }
+        for (size_t i = 0; i < m_outputBufferCount; ++i) {
+            delete[] m_outputBuffers[i];
+        }
+
+        delete[] m_inputBuffers;
+        delete[] m_outputBuffers;
+    }
+
+    m_audioPortsIn.clear();
+    m_audioPortsOut.clear();
+}
+
+
+void
+DSSIPluginInstance::instantiate(unsigned long sampleRate)
+{
+#ifdef DEBUG_DSSI
+    std::cout << "DSSIPluginInstance::instantiate - plugin unique id = "
+    << m_descriptor->LADSPA_Plugin->UniqueID << std::endl;
+#endif
+
+    if (!m_descriptor)
+        return ;
+
+    const LADSPA_Descriptor *descriptor = m_descriptor->LADSPA_Plugin;
+
+    if (!descriptor->instantiate) {
+        std::cerr << "Bad plugin: plugin id " << descriptor->UniqueID
+        << ":" << descriptor->Label
+        << " has no instantiate method!" << std::endl;
+        return ;
+    }
+
+    m_instanceHandle = descriptor->instantiate(descriptor, sampleRate);
+
+    if (m_instanceHandle) {
+
+        if (m_descriptor->get_midi_controller_for_port) {
+
+            for (unsigned long i = 0; i < descriptor->PortCount; ++i) {
+
+                if (LADSPA_IS_PORT_CONTROL(descriptor->PortDescriptors[i]) &&
+                        LADSPA_IS_PORT_INPUT(descriptor->PortDescriptors[i])) {
+
+                    int controller = m_descriptor->get_midi_controller_for_port
+                                     (m_instanceHandle, i);
+
+                    if (controller != 0 && controller != 32 &&
+                            DSSI_IS_CC(controller)) {
+
+                        m_controllerMap[DSSI_CC_NUMBER(controller)] = i;
+                    }
+                }
+            }
+        }
+    }
+}
+
+void
+DSSIPluginInstance::checkProgramCache()
+{
+    if (m_programCacheValid)
+        return ;
+    m_cachedPrograms.clear();
+
+#ifdef DEBUG_DSSI
+
+    std::cerr << "DSSIPluginInstance::checkProgramCache" << std::endl;
+#endif
+
+    if (!m_descriptor || !m_descriptor->get_program) {
+        m_programCacheValid = true;
+        return ;
+    }
+
+    unsigned long index = 0;
+    const DSSI_Program_Descriptor *programDescriptor;
+    while ((programDescriptor = m_descriptor->get_program(m_instanceHandle, index))) {
+        ++index;
+        ProgramDescriptor d;
+        d.bank = programDescriptor->Bank;
+        d.program = programDescriptor->Program;
+        d.name = QString("%1. %2").arg(index).arg(programDescriptor->Name);
+        m_cachedPrograms.push_back(d);
+    }
+
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance::checkProgramCache: have " << m_cachedPrograms.size() << " programs" << std::endl;
+#endif
+
+    m_programCacheValid = true;
+}
+
+QStringList
+DSSIPluginInstance::getPrograms()
+{
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance::getPrograms" << std::endl;
+#endif
+
+    if (!m_descriptor)
+        return QStringList();
+
+    checkProgramCache();
+
+    QStringList programs;
+
+    for (std::vector<ProgramDescriptor>::iterator i = m_cachedPrograms.begin();
+            i != m_cachedPrograms.end(); ++i) {
+        programs.push_back(i->name);
+    }
+
+    return programs;
+}
+
+QString
+DSSIPluginInstance::getProgram(int bank, int program)
+{
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance::getProgram(" << bank << "," << program << ")" << std::endl;
+#endif
+
+    if (!m_descriptor)
+        return QString();
+
+    checkProgramCache();
+
+    for (std::vector<ProgramDescriptor>::iterator i = m_cachedPrograms.begin();
+            i != m_cachedPrograms.end(); ++i) {
+        if (i->bank == bank && i->program == program)
+            return i->name;
+    }
+
+    return QString();
+}
+
+unsigned long
+DSSIPluginInstance::getProgram(QString name)
+{
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance::getProgram(" << name << ")" << std::endl;
+#endif
+
+    if (!m_descriptor)
+        return 0;
+
+    checkProgramCache();
+
+    unsigned long rv;
+
+    for (std::vector<ProgramDescriptor>::iterator i = m_cachedPrograms.begin();
+            i != m_cachedPrograms.end(); ++i) {
+        if (i->name == name) {
+            rv = i->bank;
+            rv = (rv << 16) + i->program;
+            return rv;
+        }
+    }
+
+    return 0;
+}
+
+QString
+DSSIPluginInstance::getCurrentProgram()
+{
+    return m_program;
+}
+
+void
+DSSIPluginInstance::selectProgram(QString program)
+{
+    selectProgramAux(program, true);
+}
+
+void
+DSSIPluginInstance::selectProgramAux(QString program, bool backupPortValues)
+{
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance[" << this << "]::selectProgram(" << program << ", " << backupPortValues << ")" << std::endl;
+#endif
+
+    if (!m_descriptor)
+        return ;
+
+    checkProgramCache();
+
+    if (!m_descriptor->select_program)
+        return ;
+
+    bool found = false;
+    unsigned long bankNo = 0, programNo = 0;
+
+    for (std::vector<ProgramDescriptor>::iterator i = m_cachedPrograms.begin();
+            i != m_cachedPrograms.end(); ++i) {
+
+        if (i->name == program) {
+
+            bankNo = i->bank;
+            programNo = i->program;
+            found = true;
+
+#ifdef DEBUG_DSSI
+
+            std::cerr << "DSSIPluginInstance::selectProgram(" << program << "): found at bank " << bankNo << ", program " << programNo << std::endl;
+#endif
+
+            break;
+        }
+    }
+
+    if (!found)
+        return ;
+    m_program = program;
+
+    // DSSI select_program is an audio context call
+    pthread_mutex_lock(&m_processLock);
+    m_descriptor->select_program(m_instanceHandle, bankNo, programNo);
+    pthread_mutex_unlock(&m_processLock);
+
+#ifdef DEBUG_DSSI
+
+    std::cerr << "DSSIPluginInstance::selectProgram(" << program << "): made select_program(" << bankNo << "," << programNo << ") call" << std::endl;
+#endif
+
+    if (backupPortValues) {
+        for (size_t i = 0; i < m_backupControlPortsIn.size(); ++i) {
+            m_backupControlPortsIn[i] = *m_controlPortsIn[i].second;
+            m_portChangedSinceProgramChange[i] = false;
+        }
+    }
+}
+
+void
+DSSIPluginInstance::activate()
+{
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance[" << this << "]::activate" << std::endl;
+#endif
+
+    if (!m_descriptor || !m_descriptor->LADSPA_Plugin->activate)
+        return ;
+    m_descriptor->LADSPA_Plugin->activate(m_instanceHandle);
+
+    for (size_t i = 0; i < m_backupControlPortsIn.size(); ++i) {
+        if (m_portChangedSinceProgramChange[i]) {
+#ifdef DEBUG_DSSI
+            std::cerr << "DSSIPluginInstance::activate: setting port " << m_controlPortsIn[i].first << " to " << m_backupControlPortsIn[i] << std::endl;
+#endif
+
+            *m_controlPortsIn[i].second = m_backupControlPortsIn[i];
+        }
+    }
+
+    if (m_program) {
+#ifdef DEBUG_DSSI
+        std::cerr << "DSSIPluginInstance::activate: restoring program " << m_program << std::endl;
+#endif
+
+        selectProgramAux(m_program, false);
+
+        for (size_t i = 0; i < m_backupControlPortsIn.size(); ++i) {
+            if (m_portChangedSinceProgramChange[i]) {
+#ifdef DEBUG_DSSI
+                std::cerr << "DSSIPluginInstance::activate: setting port " << m_controlPortsIn[i].first << " to " << m_backupControlPortsIn[i] << std::endl;
+#endif
+
+                *m_controlPortsIn[i].second = m_backupControlPortsIn[i];
+            }
+        }
+    }
+}
+
+void
+DSSIPluginInstance::connectPorts()
+{
+    if (!m_descriptor || !m_descriptor->LADSPA_Plugin->connect_port)
+        return ;
+#ifdef DEBUG_DSSI
+
+    std::cerr << "DSSIPluginInstance::connectPorts: " << m_audioPortsIn.size()
+    << " audio ports in, " << m_audioPortsOut.size() << " out, "
+    << m_outputBufferCount << " output buffers" << std::endl;
+#endif
+
+    assert(sizeof(LADSPA_Data) == sizeof(float));
+    assert(sizeof(sample_t) == sizeof(float));
+
+    int inbuf = 0, outbuf = 0;
+
+    for (unsigned int i = 0; i < m_audioPortsIn.size(); ++i) {
+        m_descriptor->LADSPA_Plugin->connect_port
+        (m_instanceHandle,
+         m_audioPortsIn[i],
+         (LADSPA_Data *)m_inputBuffers[inbuf]);
+        ++inbuf;
+    }
+
+    for (unsigned int i = 0; i < m_audioPortsOut.size(); ++i) {
+        m_descriptor->LADSPA_Plugin->connect_port
+        (m_instanceHandle,
+         m_audioPortsOut[i],
+         (LADSPA_Data *)m_outputBuffers[outbuf]);
+        ++outbuf;
+    }
+
+    for (unsigned int i = 0; i < m_controlPortsIn.size(); ++i) {
+        m_descriptor->LADSPA_Plugin->connect_port
+        (m_instanceHandle,
+         m_controlPortsIn[i].first,
+         m_controlPortsIn[i].second);
+    }
+
+    for (unsigned int i = 0; i < m_controlPortsOut.size(); ++i) {
+        m_descriptor->LADSPA_Plugin->connect_port
+        (m_instanceHandle,
+         m_controlPortsOut[i].first,
+         m_controlPortsOut[i].second);
+    }
+}
+
+void
+DSSIPluginInstance::setPortValue(unsigned int portNumber, float value)
+{
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance[" << this << "]::setPortValue(" << portNumber << ") to " << value << std::endl;
+#endif
+
+    for (unsigned int i = 0; i < m_controlPortsIn.size(); ++i) {
+        if (m_controlPortsIn[i].first == portNumber) {
+            LADSPAPluginFactory *f = dynamic_cast<LADSPAPluginFactory *>(m_factory);
+            if (f) {
+                if (value < f->getPortMinimum(m_descriptor->LADSPA_Plugin, portNumber)) {
+                    value = f->getPortMinimum(m_descriptor->LADSPA_Plugin, portNumber);
+                }
+                if (value > f->getPortMaximum(m_descriptor->LADSPA_Plugin, portNumber)) {
+                    value = f->getPortMaximum(m_descriptor->LADSPA_Plugin, portNumber);
+                }
+            }
+            (*m_controlPortsIn[i].second) = value;
+            m_backupControlPortsIn[i] = value;
+            m_portChangedSinceProgramChange[i] = true;
+        }
+    }
+}
+
+void
+DSSIPluginInstance::setPortValueFromController(unsigned int port, int cv)
+{
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance::setPortValueFromController(" << port << ") to " << cv << std::endl;
+#endif
+
+    const LADSPA_Descriptor *p = m_descriptor->LADSPA_Plugin;
+    LADSPA_PortRangeHintDescriptor d = p->PortRangeHints[port].HintDescriptor;
+    LADSPA_Data lb = p->PortRangeHints[port].LowerBound;
+    LADSPA_Data ub = p->PortRangeHints[port].UpperBound;
+
+    float value = (float)cv;
+
+    if (!LADSPA_IS_HINT_BOUNDED_BELOW(d)) {
+        if (!LADSPA_IS_HINT_BOUNDED_ABOVE(d)) {
+            /* unbounded: might as well leave the value alone. */
+        } else {
+            /* bounded above only. just shift the range. */
+            value = ub - 127.0f + value;
+        }
+    } else {
+        if (!LADSPA_IS_HINT_BOUNDED_ABOVE(d)) {
+            /* bounded below only. just shift the range. */
+            value = lb + value;
+        } else {
+            /* bounded both ends.  more interesting. */
+            /* XXX !!! todo: fill in logarithmic, sample rate &c */
+            value = lb + ((ub - lb) * value / 127.0f);
+        }
+    }
+
+    setPortValue(port, value);
+}
+
+float
+DSSIPluginInstance::getPortValue(unsigned int portNumber)
+{
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance::getPortValue(" << portNumber << ")" << std::endl;
+#endif
+
+    for (unsigned int i = 0; i < m_controlPortsIn.size(); ++i) {
+        if (m_controlPortsIn[i].first == portNumber) {
+            return (*m_controlPortsIn[i].second);
+        }
+    }
+
+    return 0.0;
+}
+
+QString
+DSSIPluginInstance::configure(QString key,
+                              QString value)
+{
+    if (!m_descriptor || !m_descriptor->configure)
+        return QString();
+
+    if (key == PluginIdentifier::RESERVED_PROJECT_DIRECTORY_KEY) {
+#ifdef DSSI_PROJECT_DIRECTORY_KEY
+        key = DSSI_PROJECT_DIRECTORY_KEY;
+#else
+
+        return QString();
+#endif
+
+    }
+
+
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance::configure(" << key << "," << value << ")" << std::endl;
+#endif
+
+    char *message = m_descriptor->configure(m_instanceHandle, key.data(), value.data());
+
+    m_programCacheValid = false;
+
+    QString qm;
+
+    // Ignore return values from reserved key configuration calls such
+    // as project directory
+#ifdef DSSI_RESERVED_CONFIGURE_PREFIX
+
+    if (key.startsWith(DSSI_RESERVED_CONFIGURE_PREFIX)) {
+        return qm;
+    }
+#endif
+
+    if (message) {
+        if (m_descriptor->LADSPA_Plugin && m_descriptor->LADSPA_Plugin->Label) {
+            qm = QString(m_descriptor->LADSPA_Plugin->Label) + ": ";
+        }
+        qm = qm + message;
+        free(message);
+    }
+
+    return qm;
+}
+
+void
+DSSIPluginInstance::sendEvent(const RealTime &eventTime,
+                              const void *e)
+{
+    snd_seq_event_t *event = (snd_seq_event_t *)e;
+#ifdef DEBUG_DSSI_PROCESS
+
+    std::cerr << "DSSIPluginInstance::sendEvent at " << eventTime << std::endl;
+#endif
+
+    snd_seq_event_t ev(*event);
+
+    ev.time.time.tv_sec = eventTime.sec;
+    ev.time.time.tv_nsec = eventTime.nsec;
+
+    // DSSI doesn't use MIDI channels, it uses run_multiple_synths instead.
+    ev.data.note.channel = 0;
+
+    m_eventBuffer.write(&ev, 1);
+}
+
+bool
+DSSIPluginInstance::handleController(snd_seq_event_t *ev)
+{
+    int controller = ev->data.control.param;
+
+#ifdef DEBUG_DSSI_PROCESS
+
+    std::cerr << "DSSIPluginInstance::handleController " << controller << std::endl;
+#endif
+
+    if (controller == 0) { // bank select MSB
+
+        m_pending.msb = ev->data.control.value;
+
+    } else if (controller == 32) { // bank select LSB
+
+        m_pending.lsb = ev->data.control.value;
+
+    } else if (controller > 0 && controller < 128) {
+
+        if (m_controllerMap.find(controller) != m_controllerMap.end()) {
+            int port = m_controllerMap[controller];
+            setPortValueFromController(port, ev->data.control.value);
+        } else {
+            return true; // pass through to plugin
+        }
+    }
+
+    return false;
+}
+
+void
+DSSIPluginInstance::run(const RealTime &blockTime)
+{
+    static snd_seq_event_t localEventBuffer[EVENT_BUFFER_SIZE];
+    int evCount = 0;
+
+    bool needLock = false;
+    if (m_descriptor->select_program)
+        needLock = true;
+
+    if (needLock) {
+        if (pthread_mutex_trylock(&m_processLock) != 0) {
+            for (size_t ch = 0; ch < m_audioPortsOut.size(); ++ch) {
+                memset(m_outputBuffers[ch], 0, m_blockSize * sizeof(sample_t));
+            }
+            return ;
+        }
+    }
+
+    if (m_grouped) {
+        runGrouped(blockTime);
+        goto done;
+    }
+
+    if (!m_descriptor || !m_descriptor->run_synth) {
+        m_eventBuffer.skip(m_eventBuffer.getReadSpace());
+        if (m_descriptor->LADSPA_Plugin->run) {
+            m_descriptor->LADSPA_Plugin->run(m_instanceHandle, m_blockSize);
+        } else {
+            for (size_t ch = 0; ch < m_audioPortsOut.size(); ++ch) {
+                memset(m_outputBuffers[ch], 0, m_blockSize * sizeof(sample_t));
+            }
+        }
+        m_run = true;
+        m_runSinceReset = true;
+        if (needLock)
+            pthread_mutex_unlock(&m_processLock);
+        return ;
+    }
+
+#ifdef DEBUG_DSSI_PROCESS
+    std::cerr << "DSSIPluginInstance::run(" << blockTime << ")" << std::endl;
+#endif
+
+#ifdef DEBUG_DSSI_PROCESS
+
+    if (m_eventBuffer.getReadSpace() > 0) {
+        std::cerr << "DSSIPluginInstance::run: event buffer has "
+        << m_eventBuffer.getReadSpace() << " event(s) in it" << std::endl;
+    }
+#endif
+
+    while (m_eventBuffer.getReadSpace() > 0) {
+
+        snd_seq_event_t *ev = localEventBuffer + evCount;
+        *ev = m_eventBuffer.peek();
+        bool accept = true;
+
+        RealTime evTime(ev->time.time.tv_sec, ev->time.time.tv_nsec);
+
+        int frameOffset = 0;
+        if (evTime > blockTime) {
+            frameOffset = RealTime::realTime2Frame(evTime - blockTime, m_sampleRate);
+        }
+
+#ifdef DEBUG_DSSI_PROCESS
+        std::cerr << "DSSIPluginInstance::run: evTime " << evTime << ", blockTime " << blockTime << ", frameOffset " << frameOffset
+        << ", blockSize " << m_blockSize << std::endl;
+        std::cerr << "Type: " << int(ev->type) << ", pitch: " << int(ev->data.note.note) << ", velocity: " << int(ev->data.note.velocity) << std::endl;
+#endif
+
+        if (frameOffset >= int(m_blockSize))
+            break;
+        if (frameOffset < 0)
+            frameOffset = 0;
+
+        ev->time.tick = frameOffset;
+        m_eventBuffer.skip(1);
+
+        if (ev->type == SND_SEQ_EVENT_CONTROLLER) {
+            accept = handleController(ev);
+        } else if (ev->type == SND_SEQ_EVENT_PGMCHANGE) {
+            m_pending.program = ev->data.control.value;
+            accept = false;
+        }
+
+        if (accept) {
+            if (++evCount >= EVENT_BUFFER_SIZE)
+                break;
+        }
+    }
+
+    if (m_pending.program >= 0 && m_descriptor->select_program) {
+
+        int program = m_pending.program;
+        int bank = m_pending.lsb + 128 * m_pending.msb;
+
+#ifdef DEBUG_DSSI
+
+        std::cerr << "DSSIPluginInstance::run: making select_program(" << bank << "," << program << " call" << std::endl;
+#endif
+
+        m_pending.lsb = m_pending.msb = m_pending.program = -1;
+        m_descriptor->select_program(m_instanceHandle, bank, program);
+
+#ifdef DEBUG_DSSI
+
+        std::cerr << "DSSIPluginInstance::run: made select_program(" << bank << "," << program << " call" << std::endl;
+#endif
+
+    }
+
+#ifdef DEBUG_DSSI_PROCESS
+    std::cerr << "DSSIPluginInstance::run: running with " << evCount << " events"
+    << std::endl;
+#endif
+
+    m_descriptor->run_synth(m_instanceHandle, m_blockSize,
+                            localEventBuffer, evCount);
+
+#ifdef DEBUG_DSSI_PROCESS 
+    //    for (int i = 0; i < m_blockSize; ++i) {
+    //	std::cout << m_outputBuffers[0][i] << " ";
+    //	if (i % 8 == 0) std::cout << std::endl;
+    //    }
+#endif
+
+done:
+    if (needLock)
+        pthread_mutex_unlock(&m_processLock);
+
+    if (m_audioPortsOut.size() == 0) {
+        // copy inputs to outputs
+        for (size_t ch = 0; ch < m_idealChannelCount; ++ch) {
+            size_t sch = ch % m_audioPortsIn.size();
+            for (size_t i = 0; i < m_blockSize; ++i) {
+                m_outputBuffers[ch][i] = m_inputBuffers[sch][i];
+            }
+        }
+    } else if (m_idealChannelCount < m_audioPortsOut.size()) {
+        if (m_idealChannelCount == 1) {
+            // mix down to mono
+            for (size_t ch = 1; ch < m_audioPortsOut.size(); ++ch) {
+                for (size_t i = 0; i < m_blockSize; ++i) {
+                    m_outputBuffers[0][i] += m_outputBuffers[ch][i];
+                }
+            }
+        }
+    } else if (m_idealChannelCount > m_audioPortsOut.size()) {
+        // duplicate
+        for (size_t ch = m_audioPortsOut.size(); ch < m_idealChannelCount; ++ch) {
+            size_t sch = (ch - m_audioPortsOut.size()) % m_audioPortsOut.size();
+            for (size_t i = 0; i < m_blockSize; ++i) {
+                m_outputBuffers[ch][i] = m_outputBuffers[sch][i];
+            }
+        }
+    }
+
+    m_lastRunTime = blockTime;
+    m_run = true;
+    m_runSinceReset = true;
+}
+
+void
+DSSIPluginInstance::runGrouped(const RealTime &blockTime)
+{
+    // If something else in our group has just been called for this
+    // block time (but we haven't) then we should just write out the
+    // results and return; if we have just been called for this block
+    // time or nothing else in the group has been, we should run the
+    // whole group.
+
+    bool needRun = true;
+
+    PluginSet &s = m_groupMap[m_identifier];
+
+#ifdef DEBUG_DSSI_PROCESS
+
+    std::cerr << "DSSIPluginInstance::runGrouped(" << blockTime << "): this is " << this << "; " << s.size() << " elements in m_groupMap[" << m_identifier << "]" << std::endl;
+#endif
+
+    if (m_lastRunTime != blockTime) {
+        for (PluginSet::iterator i = s.begin(); i != s.end(); ++i) {
+            DSSIPluginInstance *instance = *i;
+            if (instance != this && instance->m_lastRunTime == blockTime) {
+#ifdef DEBUG_DSSI_PROCESS
+                std::cerr << "DSSIPluginInstance::runGrouped(" << blockTime << "): plugin " << instance << " has already been run" << std::endl;
+#endif
+
+                needRun = false;
+            }
+        }
+    }
+
+    if (!needRun) {
+#ifdef DEBUG_DSSI_PROCESS
+        std::cerr << "DSSIPluginInstance::runGrouped(" << blockTime << "): already run, returning" << std::endl;
+#endif
+
+        return ;
+    }
+
+#ifdef DEBUG_DSSI_PROCESS
+    std::cerr << "DSSIPluginInstance::runGrouped(" << blockTime << "): I'm the first, running" << std::endl;
+#endif
+
+    size_t index = 0;
+    unsigned long *counts = (unsigned long *)
+                            alloca(m_groupLocalEventBufferCount * sizeof(unsigned long));
+    LADSPA_Handle *instances = (LADSPA_Handle *)
+                               alloca(m_groupLocalEventBufferCount * sizeof(LADSPA_Handle));
+
+    for (PluginSet::iterator i = s.begin(); i != s.end(); ++i) {
+
+        if (index >= m_groupLocalEventBufferCount)
+            break;
+
+        DSSIPluginInstance *instance = *i;
+        counts[index] = 0;
+        instances[index] = instance->m_instanceHandle;
+
+#ifdef DEBUG_DSSI_PROCESS
+
+        std::cerr << "DSSIPluginInstance::runGrouped(" << blockTime << "): running " << instance << std::endl;
+#endif
+
+        if (instance->m_pending.program >= 0 &&
+                instance->m_descriptor->select_program) {
+            int program = instance->m_pending.program;
+            int bank = instance->m_pending.lsb + 128 * instance->m_pending.msb;
+            instance->m_pending.lsb = instance->m_pending.msb = instance->m_pending.program = -1;
+            instance->m_descriptor->select_program
+            (instance->m_instanceHandle, bank, program);
+        }
+
+        while (instance->m_eventBuffer.getReadSpace() > 0) {
+
+            snd_seq_event_t *ev = m_groupLocalEventBuffers[index] + counts[index];
+            *ev = instance->m_eventBuffer.peek();
+            bool accept = true;
+
+            RealTime evTime(ev->time.time.tv_sec, ev->time.time.tv_nsec);
+
+            int frameOffset = 0;
+            if (evTime > blockTime) {
+                frameOffset = RealTime::realTime2Frame(evTime - blockTime, m_sampleRate);
+            }
+
+#ifdef DEBUG_DSSI_PROCESS
+            std::cerr << "DSSIPluginInstance::runGrouped: evTime " << evTime << ", frameOffset " << frameOffset
+            << ", block size " << m_blockSize << std::endl;
+#endif
+
+            if (frameOffset >= int(m_blockSize))
+                break;
+            if (frameOffset < 0)
+                frameOffset = 0;
+
+            ev->time.tick = frameOffset;
+            instance->m_eventBuffer.skip(1);
+
+            if (ev->type == SND_SEQ_EVENT_CONTROLLER) {
+                accept = instance->handleController(ev);
+            } else if (ev->type == SND_SEQ_EVENT_PGMCHANGE) {
+                instance->m_pending.program = ev->data.control.value;
+                accept = false;
+            }
+
+            if (accept) {
+                if (++counts[index] >= EVENT_BUFFER_SIZE)
+                    break;
+            }
+        }
+
+        ++index;
+    }
+
+    m_descriptor->run_multiple_synths(index,
+                                      instances,
+                                      m_blockSize,
+                                      m_groupLocalEventBuffers,
+                                      counts);
+}
+
+
+void
+DSSIPluginInstance::deactivate()
+{
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance::deactivate " << m_identifier << std::endl;
+#endif
+
+    if (!m_descriptor || !m_descriptor->LADSPA_Plugin->deactivate)
+        return ;
+
+    for (size_t i = 0; i < m_backupControlPortsIn.size(); ++i) {
+        m_backupControlPortsIn[i] = *m_controlPortsIn[i].second;
+    }
+
+    m_descriptor->LADSPA_Plugin->deactivate(m_instanceHandle);
+#ifdef DEBUG_DSSI
+
+    std::cerr << "DSSIPluginInstance::deactivate " << m_identifier << " done" << std::endl;
+#endif
+
+    m_bufferScavenger.scavenge();
+}
+
+void
+DSSIPluginInstance::cleanup()
+{
+#ifdef DEBUG_DSSI
+    std::cerr << "DSSIPluginInstance::cleanup " << m_identifier << std::endl;
+#endif
+
+    if (!m_descriptor)
+        return ;
+
+    if (!m_descriptor->LADSPA_Plugin->cleanup) {
+        std::cerr << "Bad plugin: plugin id "
+        << m_descriptor->LADSPA_Plugin->UniqueID
+        << ":" << m_descriptor->LADSPA_Plugin->Label
+        << " has no cleanup method!" << std::endl;
+        return ;
+    }
+
+    m_descriptor->LADSPA_Plugin->cleanup(m_instanceHandle);
+    m_instanceHandle = 0;
+#ifdef DEBUG_DSSI
+
+    std::cerr << "DSSIPluginInstance::cleanup " << m_identifier << " done" << std::endl;
+#endif
+}
+
+
+
+}
+
+#endif // HAVE_DSSI
+
+
diff --git a/src/sound/DSSIPluginInstance.h b/src/sound/DSSIPluginInstance.h
new file mode 100644
index 0000000..eca6327
--- /dev/null
+++ b/src/sound/DSSIPluginInstance.h
@@ -0,0 +1,193 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _DSSIPLUGININSTANCE_H_
+#define _DSSIPLUGININSTANCE_H_
+
+#include <vector>
+#include <set>
+#include <map>
+#include <qstring.h>
+#include "Instrument.h"
+
+#ifdef HAVE_DSSI
+
+#include <dssi.h>
+#include "RingBuffer.h"
+#include "RunnablePluginInstance.h"
+#include "Scavenger.h"
+#include <pthread.h>
+
+namespace Rosegarden
+{
+
+class DSSIPluginInstance : public RunnablePluginInstance
+{
+public:
+    virtual ~DSSIPluginInstance();
+
+    virtual bool isOK() const { return m_instanceHandle != 0; }
+
+    InstrumentId getInstrument() const { return m_instrument; }
+    virtual QString getIdentifier() const { return m_identifier; }
+    int getPosition() const { return m_position; }
+
+    virtual void run(const RealTime &);
+
+    virtual void setPortValue(unsigned int portNumber, float value);
+    virtual float getPortValue(unsigned int portNumber);
+    virtual QString configure(QString key, QString value);
+    virtual void sendEvent(const RealTime &eventTime,
+                           const void *event);
+
+    virtual size_t getBufferSize() { return m_blockSize; }
+    virtual size_t getAudioInputCount() { return m_audioPortsIn.size(); }
+    virtual size_t getAudioOutputCount() { return m_idealChannelCount; }
+    virtual sample_t **getAudioInputBuffers() { return m_inputBuffers; }
+    virtual sample_t **getAudioOutputBuffers() { return m_outputBuffers; }
+
+    virtual QStringList getPrograms();
+    virtual QString getCurrentProgram();
+    virtual QString getProgram(int bank, int program);
+    virtual unsigned long getProgram(QString name);
+    virtual void selectProgram(QString program);
+
+    virtual bool isBypassed() const { return m_bypassed; }
+    virtual void setBypassed(bool bypassed) { m_bypassed = bypassed; }
+
+    virtual size_t getLatency();
+
+    virtual void silence();
+    virtual void discardEvents();
+    virtual void setIdealChannelCount(size_t channels); // may re-instantiate
+
+    virtual bool isInGroup() const { return m_grouped; }
+    virtual void detachFromGroup();
+
+protected:
+    // To be constructed only by DSSIPluginFactory
+    friend class DSSIPluginFactory;
+
+    // Constructor that creates the buffers internally
+    // 
+    DSSIPluginInstance(PluginFactory *factory,
+                       InstrumentId instrument,
+                       QString identifier,
+                       int position,
+                       unsigned long sampleRate,
+                       size_t blockSize,
+                       int idealChannelCount,
+                       const DSSI_Descriptor* descriptor);
+    
+    // Constructor that uses shared buffers
+    // 
+    DSSIPluginInstance(PluginFactory *factory,
+                       InstrumentId instrument,
+                       QString identifier,
+                       int position,
+                       unsigned long sampleRate,
+                       size_t blockSize,
+                       sample_t **inputBuffers,
+                       sample_t **outputBuffers,
+                       const DSSI_Descriptor* descriptor);
+
+    void init();
+    void instantiate(unsigned long sampleRate);
+    void cleanup();
+    void activate();
+    void deactivate();
+    void connectPorts();
+
+    bool handleController(snd_seq_event_t *ev);
+    void setPortValueFromController(unsigned int portNumber, int controlValue);
+    void selectProgramAux(QString program, bool backupPortValues);
+    void checkProgramCache();
+
+    void initialiseGroupMembership();
+    void runGrouped(const RealTime &);
+
+    InstrumentId   m_instrument;
+    int                        m_position;
+    LADSPA_Handle              m_instanceHandle;
+    const DSSI_Descriptor     *m_descriptor;
+
+    std::vector<std::pair<unsigned long, LADSPA_Data*> > m_controlPortsIn;
+    std::vector<std::pair<unsigned long, LADSPA_Data*> > m_controlPortsOut;
+
+    std::vector<LADSPA_Data>  m_backupControlPortsIn;
+    std::vector<bool>  m_portChangedSinceProgramChange;
+
+    std::map<int, int>        m_controllerMap;
+
+    std::vector<int>          m_audioPortsIn;
+    std::vector<int>          m_audioPortsOut;
+
+    struct ProgramControl {
+        int msb;
+        int lsb;
+        int program;
+    };
+    ProgramControl m_pending;
+
+    struct ProgramDescriptor {
+        int bank;
+        int program;
+        QString name;
+    };
+    std::vector<ProgramDescriptor> m_cachedPrograms;
+    bool m_programCacheValid;
+
+    RingBuffer<snd_seq_event_t> m_eventBuffer;
+
+    size_t                    m_blockSize;
+    sample_t                **m_inputBuffers;
+    sample_t                **m_outputBuffers;
+    bool                      m_ownBuffers;
+    size_t                    m_idealChannelCount;
+    size_t                    m_outputBufferCount;
+    size_t                    m_sampleRate;
+    float                    *m_latencyPort;
+
+    bool                      m_run;
+    bool                      m_runSinceReset;
+    
+    bool                      m_bypassed;
+    QString                   m_program;
+    bool                      m_grouped;
+    RealTime                  m_lastRunTime;
+
+    pthread_mutex_t           m_processLock;
+
+    typedef std::set<DSSIPluginInstance *> PluginSet;
+    typedef std::map<QString, PluginSet> GroupMap;
+    static GroupMap m_groupMap;
+    static snd_seq_event_t **m_groupLocalEventBuffers;
+    static size_t m_groupLocalEventBufferCount;
+
+    static Scavenger<ScavengerArrayWrapper<snd_seq_event_t *> > m_bufferScavenger;
+};
+
+};
+
+#endif // HAVE_DSSI
+
+#endif // _DSSIPLUGININSTANCE_H_
+
diff --git a/src/sound/DummyDriver.h b/src/sound/DummyDriver.h
new file mode 100644
index 0000000..838e7bd
--- /dev/null
+++ b/src/sound/DummyDriver.h
@@ -0,0 +1,166 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include "SoundDriver.h"
+
+// An empty sound driver for when we don't want sound support
+// but still want to build the sequencer.
+//
+
+#ifndef _DUMMYDRIVER_H_
+#define _DUMMYDRIVER_H_
+
+namespace Rosegarden
+{
+
+class DummyDriver : public SoundDriver
+{
+public:
+    DummyDriver(MappedStudio *studio):
+        SoundDriver(studio, std::string("DummyDriver - no sound")) { }
+    DummyDriver(MappedStudio *studio, const std::string & name):
+        SoundDriver(studio, std::string("DummyDriver: " + name)) { }
+    virtual ~DummyDriver() { }
+
+    virtual bool initialise()  { m_recordComposition.clear(); return true; }
+    virtual void initialisePlayback(const RealTime & /*position*/) { }
+    virtual void stopPlayback() { }
+    virtual void punchOut() { }
+    virtual void resetPlayback(const RealTime & /*old position*/,
+                               const RealTime & /*position*/) { }
+    virtual void allNotesOff()  { }
+    
+    virtual RealTime getSequencerTime() { return RealTime(0, 0);}
+
+    virtual MappedComposition* getMappedComposition()
+        { return &m_recordComposition;}
+
+    virtual void processEventsOut(const MappedComposition & /*mC*/) { }
+
+    virtual void processEventsOut(const MappedComposition &,
+                                  const RealTime &,
+                                  const RealTime &) { }
+
+    // Activate a recording state
+    //
+    virtual bool record(RecordStatus /*recordStatus*/,
+                        const std::vector<InstrumentId> */*armedInstruments = 0*/,
+                        const std::vector<QString> */*audioFileNames = 0*/)
+        { return false; }
+
+    // Process anything that's pending
+    //
+    virtual void processPending() { }
+
+    // Sample rate
+    //
+    virtual unsigned int getSampleRate() const { return 0; }
+
+    // Return the last recorded audio level
+    //
+    virtual float getLastRecordedAudioLevel() { return 0.0; }
+
+    // Plugin instance management
+    //
+    virtual void setPluginInstance(InstrumentId /*id*/,
+                                   QString /*pluginIdent*/,
+                                   int /*position*/) { }
+
+    virtual void removePluginInstance(InstrumentId /*id*/,
+                                      int /*position*/) { }
+
+    virtual void removePluginInstances() { }
+
+    virtual void setPluginInstancePortValue(InstrumentId /*id*/,
+                                            int /*position*/,
+                                            unsigned long /*portNumber*/,
+                                            float /*value*/) { }
+
+    virtual float getPluginInstancePortValue(InstrumentId ,
+                                             int ,
+                                             unsigned long ) { return 0; }
+
+    virtual void setPluginInstanceBypass(InstrumentId /*id*/,
+                                         int /*position*/,
+                                         bool /*value*/) { }
+
+    virtual QStringList getPluginInstancePrograms(InstrumentId ,
+                                                  int ) { return QStringList(); }
+
+    virtual QString getPluginInstanceProgram(InstrumentId,
+                                             int ) { return QString(); }
+
+    virtual QString getPluginInstanceProgram(InstrumentId,
+                                             int,
+                                             int,
+                                             int) { return QString(); }
+
+    virtual unsigned long getPluginInstanceProgram(InstrumentId,
+                                                   int ,
+                                                   QString) { return 0; }
+    
+    virtual void setPluginInstanceProgram(InstrumentId,
+                                          int ,
+                                          QString ) { }
+
+    virtual QString configurePlugin(InstrumentId,
+                                    int,
+                                    QString ,
+                                    QString ) { return QString(); }
+
+    virtual void setAudioBussLevels(int ,
+                                    float ,
+                                    float ) { }
+
+    virtual void setAudioInstrumentLevels(InstrumentId,
+                                          float,
+                                          float) { }
+
+    virtual bool checkForNewClients() { return false; }
+
+    virtual void setLoop(const RealTime &/*loopStart*/,
+                         const RealTime &/*loopEnd*/) { }
+
+    virtual std::vector<PlayableAudioFile*> getPlayingAudioFiles()
+        { return std::vector<PlayableAudioFile*>(); }
+
+    virtual void getAudioInstrumentNumbers(InstrumentId &i, int &n) {
+        i = 0; n = 0;
+    }
+    virtual void getSoftSynthInstrumentNumbers(InstrumentId &i, int &n) {
+        i = 0; n = 0;
+    }
+
+    virtual void claimUnwantedPlugin(void *plugin) { }
+    virtual void scavengePlugins() { }
+
+    virtual bool areClocksRunning() const { return true; }
+
+protected:
+    virtual void processMidiOut(const MappedComposition & /*mC*/,
+                                const RealTime &, const RealTime &) { }
+    virtual void generateInstruments()  { }
+
+};
+
+}
+
+#endif // _DUMMYDRIVER_H_
+
diff --git a/src/sound/ExternalTransport.h b/src/sound/ExternalTransport.h
new file mode 100644
index 0000000..f40a5a2
--- /dev/null
+++ b/src/sound/ExternalTransport.h
@@ -0,0 +1,67 @@
+// -*- c-basic-offset: 4 -*-
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _EXTERNAL_TRANSPORT_H_
+#define _EXTERNAL_TRANSPORT_H_
+
+namespace Rosegarden {
+
+/**
+ * Simple interface that we can pass to low-level audio code and on
+ * which it can call back when something external requests a transport
+ * change.  The callback is asynchronous, and there's a method for the
+ * low-level code to use to find out whether its request has finished
+ * synchronising yet.
+ *
+ * (Each of the transportXX functions returns a token which can then
+ * be passed to isTransportSyncComplete.)
+ */
+
+class ExternalTransport
+{
+public:
+    typedef unsigned long TransportToken;
+
+    enum TransportRequest {
+        TransportNoChange,
+        TransportStop,
+        TransportStart,
+        TransportPlay,
+        TransportRecord,
+        TransportJumpToTime, // time arg required
+        TransportStartAtTime, // time arg required
+        TransportStopAtTime // time arg required
+    };
+
+    virtual TransportToken transportChange(TransportRequest) = 0;
+    virtual TransportToken transportJump(TransportRequest, RealTime) = 0;
+
+    virtual bool isTransportSyncComplete(TransportToken token) = 0;
+
+    // The value returned here is a constant (within the context of a
+    // particular ExternalTransport object) that is guaranteed never
+    // to be returned by any of the transport request methods.
+    virtual TransportToken getInvalidTransportToken() const = 0;
+};
+
+}
+
+#endif
+
diff --git a/src/sound/JackDriver.cpp b/src/sound/JackDriver.cpp
new file mode 100644
index 0000000..24eb6fe
--- /dev/null
+++ b/src/sound/JackDriver.cpp
@@ -0,0 +1,2480 @@
+
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "JackDriver.h"
+#include "AlsaDriver.h"
+#include "MappedStudio.h"
+#include "AudioProcess.h"
+#include "Profiler.h"
+#include "AudioLevel.h"
+#include "Audit.h"
+#include "PluginFactory.h"
+
+#ifdef HAVE_ALSA
+#ifdef HAVE_LIBJACK
+
+//#define DEBUG_JACK_DRIVER 1
+//#define DEBUG_JACK_TRANSPORT 1
+//#define DEBUG_JACK_PROCESS 1
+//#define DEBUG_JACK_XRUN 1
+
+namespace Rosegarden
+{
+
+#if (defined(DEBUG_JACK_DRIVER) || defined(DEBUG_JACK_PROCESS) || defined(DEBUG_JACK_TRANSPORT))
+static unsigned long framesThisPlay = 0;
+static RealTime startTime;
+#endif
+
+JackDriver::JackDriver(AlsaDriver *alsaDriver) :
+        m_client(0),
+        m_bufferSize(0),
+        m_sampleRate(0),
+        m_tempOutBuffer(0),
+        m_jackTransportEnabled(false),
+        m_jackTransportMaster(false),
+        m_waiting(false),
+        m_waitingState(JackTransportStopped),
+        m_waitingToken(0),
+        m_ignoreProcessTransportCount(0),
+        m_bussMixer(0),
+        m_instrumentMixer(0),
+        m_fileReader(0),
+        m_fileWriter(0),
+        m_alsaDriver(alsaDriver),
+        m_masterLevel(1.0),
+        m_directMasterAudioInstruments(0L),
+        m_directMasterSynthInstruments(0L),
+        m_haveAsyncAudioEvent(false),
+        m_kickedOutAt(0),
+        m_framesProcessed(0),
+        m_ok(false)
+{
+    assert(sizeof(sample_t) == sizeof(float));
+    initialise();
+}
+
+JackDriver::~JackDriver()
+{
+#ifdef DEBUG_JACK_DRIVER
+    std::cerr << "JackDriver::~JackDriver" << std::endl;
+#endif
+
+    m_ok = false; // prevent any more work in process()
+
+    if (m_client) {
+#ifdef DEBUG_JACK_DRIVER
+        std::cerr << "JackDriver::shutdown - deactivating JACK client"
+        << std::endl;
+#endif
+
+        if (jack_deactivate(m_client)) {
+            std::cerr << "JackDriver::shutdown - deactivation failed"
+            << std::endl;
+	}
+    }
+
+#ifdef DEBUG_JACK_DRIVER
+    std::cerr << "JackDriver::~JackDriver: terminating buss mixer" << std::endl;
+#endif
+
+    AudioBussMixer *bussMixer = m_bussMixer;
+    m_bussMixer = 0;
+    if (bussMixer)
+        bussMixer->terminate();
+
+#ifdef DEBUG_JACK_DRIVER
+    std::cerr << "JackDriver::~JackDriver: terminating instrument mixer" << std::endl;
+#endif
+
+    AudioInstrumentMixer *instrumentMixer = m_instrumentMixer;
+    m_instrumentMixer = 0;
+    if (instrumentMixer) {
+        instrumentMixer->terminate();
+        instrumentMixer->destroyAllPlugins();
+    }
+
+#ifdef DEBUG_JACK_DRIVER
+    std::cerr << "JackDriver::~JackDriver: terminating file reader" << std::endl;
+#endif
+
+    AudioFileReader *reader = m_fileReader;
+    m_fileReader = 0;
+    if (reader)
+        reader->terminate();
+
+#ifdef DEBUG_JACK_DRIVER
+    std::cerr << "JackDriver::~JackDriver: terminating file writer" << std::endl;
+#endif
+
+    AudioFileWriter *writer = m_fileWriter;
+    m_fileWriter = 0;
+    if (writer)
+        writer->terminate();
+
+    if (m_client) {
+
+#ifdef DEBUG_JACK_DRIVER
+        std::cerr << "JackDriver::shutdown - tearing down JACK client"
+        << std::endl;
+#endif
+
+        for (unsigned int i = 0; i < m_inputPorts.size(); ++i) {
+#ifdef DEBUG_JACK_DRIVER
+            std::cerr << "unregistering input " << i << std::endl;
+#endif
+
+            if (jack_port_unregister(m_client, m_inputPorts[i])) {
+                std::cerr << "JackDriver::shutdown - "
+                << "can't unregister input port " << i + 1
+                << std::endl;
+            }
+        }
+
+        for (unsigned int i = 0; i < m_outputSubmasters.size(); ++i) {
+#ifdef DEBUG_JACK_DRIVER
+            std::cerr << "unregistering output sub " << i << std::endl;
+#endif
+
+            if (jack_port_unregister(m_client, m_outputSubmasters[i])) {
+                std::cerr << "JackDriver::shutdown - "
+                << "can't unregister output submaster " << i + 1 << std::endl;
+            }
+        }
+
+        for (unsigned int i = 0; i < m_outputMonitors.size(); ++i) {
+#ifdef DEBUG_JACK_DRIVER
+            std::cerr << "unregistering output mon " << i << std::endl;
+#endif
+
+            if (jack_port_unregister(m_client, m_outputMonitors[i])) {
+                std::cerr << "JackDriver::shutdown - "
+                << "can't unregister output monitor " << i + 1 << std::endl;
+            }
+        }
+
+        for (unsigned int i = 0; i < m_outputMasters.size(); ++i) {
+#ifdef DEBUG_JACK_DRIVER
+            std::cerr << "unregistering output master " << i << std::endl;
+#endif
+
+            if (jack_port_unregister(m_client, m_outputMasters[i])) {
+                std::cerr << "JackDriver::shutdown - "
+                << "can't unregister output master " << i + 1 << std::endl;
+            }
+        }
+
+#ifdef DEBUG_JACK_DRIVER
+        std::cerr << "closing client" << std::endl;
+#endif
+
+        jack_client_close(m_client);
+        std::cerr << "done" << std::endl;
+        m_client = 0;
+    }
+
+#ifdef DEBUG_JACK_DRIVER
+    std::cerr << "JackDriver: deleting mixers etc" << std::endl;
+#endif
+
+    delete bussMixer;
+    delete instrumentMixer;
+    delete reader;
+    delete writer;
+
+#ifdef DEBUG_JACK_DRIVER
+
+    std::cerr << "JackDriver::~JackDriver exiting" << std::endl;
+#endif
+}
+
+void
+JackDriver::initialise(bool reinitialise)
+{
+    m_ok = false;
+
+    Audit audit;
+    audit << std::endl;
+
+    std::string jackClientName = "rosegarden";
+
+    // attempt connection to JACK server
+    //
+    if ((m_client = jack_client_new(jackClientName.c_str())) == 0) {
+        audit << "JackDriver::initialiseAudio - "
+        << "JACK server not running"
+        << std::endl;
+        return ;
+    }
+
+    InstrumentId instrumentBase;
+    int instrumentCount;
+    m_alsaDriver->getAudioInstrumentNumbers(instrumentBase, instrumentCount);
+    for (InstrumentId id = instrumentBase;
+            id < instrumentBase + instrumentCount; ++id) {
+        // prefill so that we can refer to the map without a lock (as
+        // the number of instruments won't change)
+        m_recordInputs[id] = RecordInputDesc(1000, -1, 0.0);
+    }
+
+    // set callbacks
+    //
+    jack_set_process_callback(m_client, jackProcessStatic, this);
+    jack_set_buffer_size_callback(m_client, jackBufferSize, this);
+    jack_set_sample_rate_callback(m_client, jackSampleRate, this);
+    jack_on_shutdown(m_client, jackShutdown, this);
+    jack_set_xrun_callback(m_client, jackXRun, this);
+    jack_set_sync_callback(m_client, jackSyncCallback, this);
+
+    // get and report the sample rate and buffer size
+    //
+    m_sampleRate = jack_get_sample_rate(m_client);
+    m_bufferSize = jack_get_buffer_size(m_client);
+
+    audit << "JackDriver::initialiseAudio - JACK sample rate = "
+    << m_sampleRate << "Hz, buffer size = " << m_bufferSize
+    << std::endl;
+
+    PluginFactory::setSampleRate(m_sampleRate);
+
+    // Get the initial buffer size before we activate the client
+    //
+
+    if (!reinitialise) {
+
+        // create processing buffer(s)
+        //
+        m_tempOutBuffer = new sample_t[m_bufferSize];
+
+        audit << "JackDriver::initialiseAudio - "
+        << "creating disk thread" << std::endl;
+
+        m_fileReader = new AudioFileReader(m_alsaDriver, m_sampleRate);
+        m_fileWriter = new AudioFileWriter(m_alsaDriver, m_sampleRate);
+        m_instrumentMixer = new AudioInstrumentMixer
+                            (m_alsaDriver, m_fileReader, m_sampleRate, m_bufferSize);
+        m_bussMixer = new AudioBussMixer
+                      (m_alsaDriver, m_instrumentMixer, m_sampleRate, m_bufferSize);
+        m_instrumentMixer->setBussMixer(m_bussMixer);
+
+        // We run the file reader whatever, but we only run the other
+        // threads (instrument mixer, buss mixer, file writer) when we
+        // actually need them.  (See updateAudioData and createRecordFile.)
+        m_fileReader->run();
+    }
+
+    // Create and connect the default numbers of ports.  We always create
+    // one stereo pair each of master and monitor outs, and then we create
+    // record ins, fader outs and submaster outs according to the user's
+    // preferences.  Since we don't know the user's preferences yet, we'll
+    // start by creating one pair of record ins and no fader or submaster
+    // outs.
+    //
+    m_outputMasters.clear();
+    m_outputMonitors.clear();
+    m_outputSubmasters.clear();
+    m_outputInstruments.clear();
+    m_inputPorts.clear();
+
+    if (!createMainOutputs()) { // one stereo pair master, one pair monitor
+        audit << "JackDriver::initialise - "
+        << "failed to create main outputs!" << std::endl;
+        return ;
+    }
+
+    if (!createRecordInputs(1)) {
+        audit << "JackDriver::initialise - "
+        << "failed to create record inputs!" << std::endl;
+        return ;
+    }
+
+    if (jack_activate(m_client)) {
+        audit << "JackDriver::initialise - "
+        << "client activation failed" << std::endl;
+        return ;
+    }
+
+    // Now set up the default connections.
+
+    std::string playback_1, playback_2;
+
+    const char **ports =
+        jack_get_ports(m_client, NULL, NULL,
+                       JackPortIsPhysical | JackPortIsInput);
+
+    if (ports) {
+        if (ports[0])
+            playback_1 = std::string(ports[0]);
+        if (ports[1])
+            playback_2 = std::string(ports[1]);
+
+        // count ports
+        unsigned int i = 0;
+        for (i = 0; ports[i]; i++)
+            ;
+        audit << "JackDriver::initialiseAudio - "
+        << "found " << i << " JACK physical outputs"
+        << std::endl;
+    } else
+        audit << "JackDriver::initialiseAudio - "
+        << "no JACK physical outputs found"
+        << std::endl;
+    free(ports);
+
+    if (playback_1 != "") {
+        audit << "JackDriver::initialiseAudio - "
+        << "connecting from "
+        << "\"" << jack_port_name(m_outputMasters[0])
+        << "\" to \"" << playback_1.c_str() << "\""
+        << std::endl;
+
+        // connect our client up to the ALSA ports - first left output
+        //
+        if (jack_connect(m_client, jack_port_name(m_outputMasters[0]),
+                         playback_1.c_str())) {
+            audit << "JackDriver::initialiseAudio - "
+            << "cannot connect to JACK output port" << std::endl;
+            return ;
+        }
+
+        /*
+                if (jack_connect(m_client, jack_port_name(m_outputMonitors[0]),
+                                 playback_1.c_str()))
+                {
+                    audit << "JackDriver::initialiseAudio - "
+        		  << "cannot connect to JACK output port" << std::endl;
+        	    return;
+                }
+        */
+    }
+
+    if (playback_2 != "") {
+        audit << "JackDriver::initialiseAudio - "
+        << "connecting from "
+        << "\"" << jack_port_name(m_outputMasters[1])
+        << "\" to \"" << playback_2.c_str() << "\""
+        << std::endl;
+
+        if (jack_connect(m_client, jack_port_name(m_outputMasters[1]),
+                         playback_2.c_str())) {
+            audit << "JackDriver::initialiseAudio - "
+            << "cannot connect to JACK output port" << std::endl;
+        }
+
+        /*
+                if (jack_connect(m_client, jack_port_name(m_outputMonitors[1]),
+                                 playback_2.c_str()))
+                {
+                    audit << "JackDriver::initialiseAudio - "
+        		  << "cannot connect to JACK output port" << std::endl;
+                }
+        */
+    }
+
+
+    std::string capture_1, capture_2;
+
+    ports =
+        jack_get_ports(m_client, NULL, NULL,
+                       JackPortIsPhysical | JackPortIsOutput);
+
+    if (ports) {
+        if (ports[0])
+            capture_1 = std::string(ports[0]);
+        if (ports[1])
+            capture_2 = std::string(ports[1]);
+
+        // count ports
+        unsigned int i = 0;
+        for (i = 0; ports[i]; i++)
+            ;
+        audit << "JackDriver::initialiseAudio - "
+        << "found " << i << " JACK physical inputs"
+        << std::endl;
+    } else
+        audit << "JackDriver::initialiseAudio - "
+        << "no JACK physical inputs found"
+        << std::endl;
+    free(ports);
+
+    if (capture_1 != "") {
+
+        audit << "JackDriver::initialiseAudio - "
+        << "connecting from "
+        << "\"" << capture_1.c_str()
+        << "\" to \"" << jack_port_name(m_inputPorts[0]) << "\""
+        << std::endl;
+
+        if (jack_connect(m_client, capture_1.c_str(),
+                         jack_port_name(m_inputPorts[0]))) {
+            audit << "JackDriver::initialiseAudio - "
+            << "cannot connect to JACK input port" << std::endl;
+        }
+    }
+
+    if (capture_2 != "") {
+
+        audit << "JackDriver::initialiseAudio - "
+        << "connecting from "
+        << "\"" << capture_2.c_str()
+        << "\" to \"" << jack_port_name(m_inputPorts[1]) << "\""
+        << std::endl;
+
+        if (jack_connect(m_client, capture_2.c_str(),
+                         jack_port_name(m_inputPorts[1]))) {
+            audit << "JackDriver::initialiseAudio - "
+            << "cannot connect to JACK input port" << std::endl;
+        }
+    }
+
+    audit << "JackDriver::initialiseAudio - "
+    << "initialised JACK audio subsystem"
+    << std::endl;
+
+    m_ok = true;
+}
+
+bool
+JackDriver::createMainOutputs()
+{
+    if (!m_client)
+        return false;
+
+    jack_port_t *port = jack_port_register
+                        (m_client, "master out L",
+                         JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
+    if (!port)
+        return false;
+    m_outputMasters.push_back(port);
+
+    port = jack_port_register
+           (m_client, "master out R",
+            JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
+    if (!port)
+        return false;
+    m_outputMasters.push_back(port);
+
+    port = jack_port_register
+           (m_client, "record monitor out L",
+            JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
+    if (!port)
+        return false;
+    m_outputMonitors.push_back(port);
+
+    port = jack_port_register
+           (m_client, "record monitor out R",
+            JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
+    if (!port)
+        return false;
+    m_outputMonitors.push_back(port);
+
+    return true;
+}
+
+bool
+JackDriver::createFaderOutputs(int audioPairs, int synthPairs)
+{
+    if (!m_client)
+        return false;
+
+    int pairs = audioPairs + synthPairs;
+    int pairsNow = m_outputInstruments.size() / 2;
+    if (pairs == pairsNow)
+        return true;
+
+    for (int i = pairsNow; i < pairs; ++i) {
+
+        char namebuffer[22];
+        jack_port_t *port;
+
+        if (i < audioPairs) {
+            snprintf(namebuffer, 21, "audio fader %d out L", i + 1);
+        } else {
+            snprintf(namebuffer, 21, "synth fader %d out L", i - audioPairs + 1);
+        }
+
+        port = jack_port_register(m_client,
+                                  namebuffer,
+                                  JACK_DEFAULT_AUDIO_TYPE,
+                                  JackPortIsOutput,
+                                  0);
+        if (!port)
+            return false;
+        m_outputInstruments.push_back(port);
+
+        if (i < audioPairs) {
+            snprintf(namebuffer, 21, "audio fader %d out R", i + 1);
+        } else {
+            snprintf(namebuffer, 21, "synth fader %d out R", i - audioPairs + 1);
+        }
+
+        port = jack_port_register(m_client,
+                                  namebuffer,
+                                  JACK_DEFAULT_AUDIO_TYPE,
+                                  JackPortIsOutput,
+                                  0);
+        if (!port)
+            return false;
+        m_outputInstruments.push_back(port);
+    }
+
+    while ((int)m_outputInstruments.size() > pairs * 2) {
+        std::vector<jack_port_t *>::iterator itr = m_outputInstruments.end();
+        --itr;
+        jack_port_unregister(m_client, *itr);
+        m_outputInstruments.erase(itr);
+    }
+
+    return true;
+}
+
+bool
+JackDriver::createSubmasterOutputs(int pairs)
+{
+    if (!m_client)
+        return false;
+
+    int pairsNow = m_outputSubmasters.size() / 2;
+    if (pairs == pairsNow)
+        return true;
+
+    for (int i = pairsNow; i < pairs; ++i) {
+
+        char namebuffer[22];
+        jack_port_t *port;
+
+        snprintf(namebuffer, 21, "submaster %d out L", i + 1);
+        port = jack_port_register(m_client,
+                                  namebuffer,
+                                  JACK_DEFAULT_AUDIO_TYPE,
+                                  JackPortIsOutput,
+                                  0);
+        if (!port)
+            return false;
+        m_outputSubmasters.push_back(port);
+
+        snprintf(namebuffer, 21, "submaster %d out R", i + 1);
+        port = jack_port_register(m_client,
+                                  namebuffer,
+                                  JACK_DEFAULT_AUDIO_TYPE,
+                                  JackPortIsOutput,
+                                  0);
+        if (!port)
+            return false;
+        m_outputSubmasters.push_back(port);
+    }
+
+    while ((int)m_outputSubmasters.size() > pairs * 2) {
+        std::vector<jack_port_t *>::iterator itr = m_outputSubmasters.end();
+        --itr;
+        jack_port_unregister(m_client, *itr);
+        m_outputSubmasters.erase(itr);
+    }
+
+    return true;
+}
+
+bool
+JackDriver::createRecordInputs(int pairs)
+{
+    if (!m_client)
+        return false;
+
+    int pairsNow = m_inputPorts.size() / 2;
+    if (pairs == pairsNow)
+        return true;
+
+    for (int i = pairsNow; i < pairs; ++i) {
+
+        char namebuffer[22];
+        jack_port_t *port;
+
+        snprintf(namebuffer, 21, "record in %d L", i + 1);
+        port = jack_port_register(m_client,
+                                  namebuffer,
+                                  JACK_DEFAULT_AUDIO_TYPE,
+                                  JackPortIsInput,
+                                  0);
+        if (!port)
+            return false;
+        m_inputPorts.push_back(port);
+
+        snprintf(namebuffer, 21, "record in %d R", i + 1);
+        port = jack_port_register(m_client,
+                                  namebuffer,
+                                  JACK_DEFAULT_AUDIO_TYPE,
+                                  JackPortIsInput,
+                                  0);
+        if (!port)
+            return false;
+        m_inputPorts.push_back(port);
+    }
+
+    while ((int)m_outputSubmasters.size() > pairs * 2) {
+        std::vector<jack_port_t *>::iterator itr = m_outputSubmasters.end();
+        --itr;
+        jack_port_unregister(m_client, *itr);
+        m_outputSubmasters.erase(itr);
+    }
+
+    return true;
+}
+
+
+void
+JackDriver::setAudioPorts(bool faderOuts, bool submasterOuts)
+{
+    if (!m_client)
+        return ;
+
+    Audit audit;
+#ifdef DEBUG_JACK_DRIVER
+
+    std::cerr << "JackDriver::setAudioPorts(" << faderOuts << "," << submasterOuts << ")" << std::endl;
+#endif
+
+    if (!m_client) {
+        std::cerr << "JackDriver::setAudioPorts(" << faderOuts << "," << submasterOuts << "): no client yet" << std::endl;
+        return ;
+    }
+
+    if (faderOuts) {
+        InstrumentId instrumentBase;
+        int audioInstruments;
+        int synthInstruments;
+        m_alsaDriver->getAudioInstrumentNumbers(instrumentBase, audioInstruments);
+        m_alsaDriver->getSoftSynthInstrumentNumbers(instrumentBase, synthInstruments);
+        if (!createFaderOutputs(audioInstruments, synthInstruments)) {
+            m_ok = false;
+            audit << "Failed to create fader outs!" << std::endl;
+            return ;
+        }
+    } else {
+        createFaderOutputs(0, 0);
+    }
+
+    if (submasterOuts) {
+
+        // one fewer than returned here, because the master has a buss object too
+        if (!createSubmasterOutputs
+                (m_alsaDriver->getMappedStudio()->getObjectCount
+                 (MappedObject::AudioBuss) - 1)) {
+            m_ok = false;
+            audit << "Failed to create submaster outs!" << std::endl;
+            return ;
+        }
+
+    } else {
+        createSubmasterOutputs(0);
+    }
+}
+
+RealTime
+JackDriver::getAudioPlayLatency() const
+{
+    if (!m_client)
+        return RealTime::zeroTime;
+
+    jack_nframes_t latency =
+        jack_port_get_total_latency(m_client, m_outputMasters[0]);
+
+    return RealTime::frame2RealTime(latency, m_sampleRate);
+}
+
+RealTime
+JackDriver::getAudioRecordLatency() const
+{
+    if (!m_client)
+        return RealTime::zeroTime;
+
+    jack_nframes_t latency =
+        jack_port_get_total_latency(m_client, m_inputPorts[0]);
+
+    return RealTime::frame2RealTime(latency, m_sampleRate);
+}
+
+RealTime
+JackDriver::getInstrumentPlayLatency(InstrumentId id) const
+{
+    if (m_instrumentLatencies.find(id) == m_instrumentLatencies.end()) {
+        return RealTime::zeroTime;
+    } else {
+        return m_instrumentLatencies.find(id)->second;
+    }
+}
+
+RealTime
+JackDriver::getMaximumPlayLatency() const
+{
+    return m_maxInstrumentLatency;
+}
+
+int
+JackDriver::jackProcessStatic(jack_nframes_t nframes, void *arg)
+{
+    JackDriver *inst = static_cast<JackDriver*>(arg);
+    if (inst)
+        return inst->jackProcess(nframes);
+    else
+        return 0;
+}
+
+int
+JackDriver::jackProcess(jack_nframes_t nframes)
+{
+    if (!m_ok || !m_client) {
+#ifdef DEBUG_JACK_PROCESS
+        std::cerr << "JackDriver::jackProcess: not OK" << std::endl;
+#endif
+
+        return 0;
+    }
+
+    if (!m_bussMixer) {
+#ifdef DEBUG_JACK_PROCESS
+        std::cerr << "JackDriver::jackProcess: no buss mixer" << std::endl;
+#endif
+
+        return jackProcessEmpty(nframes);
+    }
+
+    if (m_alsaDriver->areClocksRunning()) {
+        m_alsaDriver->checkTimerSync(m_framesProcessed);
+    } else {
+        m_alsaDriver->checkTimerSync(0);
+    }
+
+    bool lowLatencyMode = m_alsaDriver->getLowLatencyMode();
+    bool clocksRunning = m_alsaDriver->areClocksRunning();
+    bool playing = m_alsaDriver->isPlaying();
+    bool asyncAudio = m_haveAsyncAudioEvent;
+
+#ifdef DEBUG_JACK_PROCESS
+
+    Profiler profiler("jackProcess", true);
+#else
+#ifdef DEBUG_JACK_XRUN
+
+    Profiler profiler("jackProcess", false);
+#endif
+#endif
+
+    if (lowLatencyMode) {
+        if (clocksRunning) {
+            if (playing || asyncAudio) {
+
+                if (m_instrumentMixer->tryLock() == 0) {
+                    m_instrumentMixer->kick(false);
+                    m_instrumentMixer->releaseLock();
+                    //#ifdef DEBUG_JACK_PROCESS
+                } else {
+                    std::cerr << "JackDriver::jackProcess: no instrument mixer lock available" << std::endl;
+                    //#endif
+                }
+                if (m_bussMixer->getBussCount() > 0) {
+                    if (m_bussMixer->tryLock() == 0) {
+                        m_bussMixer->kick(false, false);
+                        m_bussMixer->releaseLock();
+                        //#ifdef DEBUG_JACK_PROCESS
+                    } else {
+                        std::cerr << "JackDriver::jackProcess: no buss mixer lock available" << std::endl;
+                        //#endif
+                    }
+                }
+            }
+        }
+    }
+
+    if (jack_cpu_load(m_client) > 97.0) {
+        reportFailure(MappedEvent::FailureCPUOverload);
+        return jackProcessEmpty(nframes);
+    }
+
+#ifdef DEBUG_JACK_PROCESS
+    Profiler profiler2("jackProcess post mix", true);
+#else
+#ifdef DEBUG_JACK_XRUN
+
+    Profiler profiler2("jackProcess post mix", false);
+#endif
+#endif
+
+    SequencerDataBlock *sdb = m_alsaDriver->getSequencerDataBlock();
+
+    jack_position_t position;
+    jack_transport_state_t state = JackTransportRolling;
+    bool doneRecord = false;
+
+    int ignoreCount = m_ignoreProcessTransportCount;
+    if (ignoreCount > 0)
+        --m_ignoreProcessTransportCount;
+
+    InstrumentId audioInstrumentBase;
+    int audioInstruments;
+    m_alsaDriver->getAudioInstrumentNumbers(audioInstrumentBase, audioInstruments);
+
+    if (m_jackTransportEnabled) {
+
+        state = jack_transport_query(m_client, &position);
+
+#ifdef DEBUG_JACK_PROCESS
+
+        std::cerr << "JackDriver::jackProcess: JACK transport state is " << state << std::endl;
+#endif
+
+        if (state == JackTransportStopped) {
+            if (playing && clocksRunning && !m_waiting) {
+                ExternalTransport *transport =
+                    m_alsaDriver->getExternalTransportControl();
+                if (transport) {
+#ifdef DEBUG_JACK_TRANSPORT
+                    std::cerr << "JackDriver::jackProcess: JACK transport stopped externally at " << position.frame << std::endl;
+#endif
+
+                    m_waitingToken =
+                        transport->transportJump
+                        (ExternalTransport::TransportStopAtTime,
+                         RealTime::frame2RealTime(position.frame,
+                                                  position.frame_rate));
+                }
+            } else if (clocksRunning) {
+                if (!asyncAudio) {
+#ifdef DEBUG_JACK_PROCESS
+                    std::cerr << "JackDriver::jackProcess: no interesting async events" << std::endl;
+#endif
+                    // do this before record monitor, otherwise we lose monitor out
+                    jackProcessEmpty(nframes);
+                }
+
+                // for monitoring:
+                int rv = 0;
+                for (InstrumentId id = audioInstrumentBase;
+                        id < audioInstrumentBase + audioInstruments; ++id) {
+                    int irv = jackProcessRecord(id, nframes, 0, 0, clocksRunning);
+                    if (irv != 0)
+                        rv = irv;
+                }
+                doneRecord = true;
+
+                if (!asyncAudio) {
+                    return rv;
+                }
+
+            } else {
+                return jackProcessEmpty(nframes);
+            }
+        } else if (state == JackTransportStarting) {
+            return jackProcessEmpty(nframes);
+        } else if (state != JackTransportRolling) {
+            std::cerr << "JackDriver::jackProcess: unexpected JACK transport state " << state << std::endl;
+        }
+    }
+
+    if (state == JackTransportRolling) { // also covers not-on-transport case
+        if (m_waiting) {
+            if (ignoreCount > 0) {
+#ifdef DEBUG_JACK_TRANSPORT
+                std::cerr << "JackDriver::jackProcess: transport rolling, but we're ignoring it (count = " << ignoreCount << ")" << std::endl;
+#endif
+
+            } else {
+#ifdef DEBUG_JACK_TRANSPORT
+                std::cerr << "JackDriver::jackProcess: transport rolling, telling ALSA driver to go!" << std::endl;
+#endif
+
+                m_alsaDriver->startClocksApproved();
+                m_waiting = false;
+            }
+        }
+
+#ifdef DEBUG_JACK_PROCESS
+        std::cerr << "JackDriver::jackProcess (rolling or not on JACK transport)" << std::endl;
+#endif
+
+        if (!clocksRunning) {
+#ifdef DEBUG_JACK_PROCESS
+            std::cerr << "JackDriver::jackProcess: clocks stopped" << std::endl;
+#endif
+
+            return jackProcessEmpty(nframes);
+
+        } else if (!playing) {
+#ifdef DEBUG_JACK_PROCESS
+            std::cerr << "JackDriver::jackProcess: not playing" << std::endl;
+#endif
+
+            if (!asyncAudio) {
+#ifdef DEBUG_JACK_PROCESS
+                std::cerr << "JackDriver::jackProcess: no interesting async events" << std::endl;
+#endif
+                // do this before record monitor, otherwise we lose monitor out
+                jackProcessEmpty(nframes);
+            }
+
+            // for monitoring:
+            int rv = 0;
+            for (InstrumentId id = audioInstrumentBase;
+                    id < audioInstrumentBase + audioInstruments; ++id) {
+                int irv = jackProcessRecord(id, nframes, 0, 0, clocksRunning);
+                if (irv != 0)
+                    rv = irv;
+            }
+            doneRecord = true;
+
+            if (!asyncAudio) {
+                return rv;
+            }
+        }
+    }
+
+#ifdef DEBUG_JACK_PROCESS
+    Profiler profiler3("jackProcess post transport", true);
+#else
+#ifdef DEBUG_JACK_XRUN
+
+    Profiler profiler3("jackProcess post transport", false);
+#endif
+#endif
+
+    InstrumentId synthInstrumentBase;
+    int synthInstruments;
+    m_alsaDriver->getSoftSynthInstrumentNumbers(synthInstrumentBase, synthInstruments);
+
+    // We always have the master out
+
+    sample_t *master[2] = {
+                              static_cast<sample_t *>
+                              (jack_port_get_buffer(m_outputMasters[0], nframes)),
+                              static_cast<sample_t *>
+                              (jack_port_get_buffer(m_outputMasters[1], nframes))
+                          };
+
+    memset(master[0], 0, nframes * sizeof(sample_t));
+    memset(master[1], 0, nframes * sizeof(sample_t));
+
+    // Reset monitor outs (if present) here prior to mixing
+
+    if (m_outputMonitors.size() > 0) {
+        sample_t *buffer = static_cast<sample_t *>
+                           (jack_port_get_buffer(m_outputMonitors[0], nframes));
+        if (buffer)
+            memset(buffer, 0, nframes * sizeof(sample_t));
+    }
+
+    if (m_outputMonitors.size() > 1) {
+        sample_t *buffer = static_cast<sample_t *>
+                           (jack_port_get_buffer(m_outputMonitors[1], nframes));
+        if (buffer)
+            memset(buffer, 0, nframes * sizeof(sample_t));
+    }
+
+    int bussCount = m_bussMixer->getBussCount();
+
+    // If we have any busses, then we just mix from them (but we still
+    // need to keep ourselves up to date by reading and monitoring the
+    // instruments).  If we have no busses, mix direct from instruments.
+
+    for (int buss = 0; buss < bussCount; ++buss) {
+
+        sample_t *submaster[2] = { 0, 0 };
+        sample_t peak[2] = { 0.0, 0.0 };
+
+        if ((int)m_outputSubmasters.size() > buss * 2 + 1) {
+            submaster[0] = static_cast<sample_t *>
+                           (jack_port_get_buffer(m_outputSubmasters[buss * 2], nframes));
+            submaster[1] = static_cast<sample_t *>
+                           (jack_port_get_buffer(m_outputSubmasters[buss * 2 + 1], nframes));
+        }
+
+        if (!submaster[0])
+            submaster[0] = m_tempOutBuffer;
+        if (!submaster[1])
+            submaster[1] = m_tempOutBuffer;
+
+        for (int ch = 0; ch < 2; ++ch) {
+
+            RingBuffer<AudioBussMixer::sample_t> *rb =
+                m_bussMixer->getRingBuffer(buss, ch);
+
+            if (!rb || m_bussMixer->isBussDormant(buss)) {
+                if (rb)
+                    rb->skip(nframes);
+                if (submaster[ch])
+                    memset(submaster[ch], 0, nframes * sizeof(sample_t));
+            } else {
+                size_t actual = rb->read(submaster[ch], nframes);
+                if (actual < nframes) {
+                    reportFailure(MappedEvent::FailureBussMixUnderrun);
+                }
+                for (size_t i = 0; i < nframes; ++i) {
+                    sample_t sample = submaster[ch][i];
+                    if (sample > peak[ch])
+                        peak[ch] = sample;
+                    master[ch][i] += sample;
+                }
+            }
+        }
+
+        if (sdb) {
+            LevelInfo info;
+            info.level = AudioLevel::multiplier_to_fader
+                         (peak[0], 127, AudioLevel::LongFader);
+            info.levelRight = AudioLevel::multiplier_to_fader
+                              (peak[1], 127, AudioLevel::LongFader);
+
+            sdb->setSubmasterLevel(buss, info);
+        }
+
+        for (InstrumentId id = audioInstrumentBase;
+                id < audioInstrumentBase + audioInstruments; ++id) {
+            if (buss + 1 == m_recordInputs[id].input) {
+                jackProcessRecord(id, nframes, submaster[0], submaster[1], clocksRunning);
+            }
+        }
+    }
+
+#ifdef DEBUG_JACK_PROCESS
+    std::cerr << "JackDriver::jackProcess: have " << audioInstruments << " audio and " << synthInstruments << " synth instruments and " << bussCount << " busses" << std::endl;
+#endif
+
+    bool allInstrumentsDormant = true;
+    static RealTime dormantTime = RealTime::zeroTime;
+
+    for (int i = 0; i < audioInstruments + synthInstruments; ++i) {
+
+        InstrumentId id;
+        if (i < audioInstruments)
+            id = audioInstrumentBase + i;
+        else
+            id = synthInstrumentBase + (i - audioInstruments);
+
+        if (m_instrumentMixer->isInstrumentEmpty(id))
+            continue;
+
+        sample_t *instrument[2] = { 0, 0 };
+        sample_t peak[2] = { 0.0, 0.0 };
+
+        if (int(m_outputInstruments.size()) > i * 2 + 1) {
+            instrument[0] = static_cast<sample_t *>
+                            (jack_port_get_buffer(m_outputInstruments[i * 2], nframes));
+            instrument[1] = static_cast<sample_t *>
+                            (jack_port_get_buffer(m_outputInstruments[i * 2 + 1], nframes));
+        }
+
+        if (!instrument[0])
+            instrument[0] = m_tempOutBuffer;
+        if (!instrument[1])
+            instrument[1] = m_tempOutBuffer;
+
+        for (int ch = 0; ch < 2; ++ch) {
+
+            // We always need to read from an instrument's ring buffer
+            // to keep the instrument moving along, as well as for
+            // monitoring.  If the instrument is connected straight to
+            // the master, then we also need to mix from it.  (We have
+            // that information cached courtesy of updateAudioData.)
+
+            bool directToMaster = false;
+            if (i < audioInstruments) {
+                directToMaster = (m_directMasterAudioInstruments & (1 << i));
+            } else {
+                directToMaster = (m_directMasterSynthInstruments &
+                                  (1 << (i - audioInstruments)));
+            }
+
+#ifdef DEBUG_JACK_PROCESS
+            if (id == 1000 || id == 10000) {
+                std::cerr << "JackDriver::jackProcess: instrument id " << id << ", base " << audioInstrumentBase << ", direct masters " << m_directMasterAudioInstruments << ": " << directToMaster << std::endl;
+            }
+#endif
+
+            RingBuffer<AudioInstrumentMixer::sample_t, 2> *rb =
+                m_instrumentMixer->getRingBuffer(id, ch);
+
+            if (!rb || m_instrumentMixer->isInstrumentDormant(id)) {
+#ifdef DEBUG_JACK_PROCESS
+                if (id == 1000 || id == 10000) {
+                    if (rb) {
+                        std::cerr << "JackDriver::jackProcess: instrument " << id << " dormant" << std::endl;
+                    } else {
+                        std::cerr << "JackDriver::jackProcess: instrument " << id << " has no ring buffer for channel " << ch << std::endl;
+                    }
+                }
+#endif
+                if (rb)
+                    rb->skip(nframes);
+                if (instrument[ch])
+                    memset(instrument[ch], 0, nframes * sizeof(sample_t));
+
+            } else {
+
+                allInstrumentsDormant = false;
+
+                size_t actual = rb->read(instrument[ch], nframes);
+
+#ifdef DEBUG_JACK_PROCESS
+
+                if (id == 1000) {
+                    std::cerr << "JackDriver::jackProcess: read " << actual << " of " << nframes << " frames for instrument " << id << " channel " << ch << std::endl;
+                }
+#endif
+
+                if (actual < nframes) {
+
+                    std::cerr << "JackDriver::jackProcess: read " << actual << " of " << nframes << " frames for " << id << " ch " << ch << " (pl " << playing << ", cl " << clocksRunning << ", aa " << asyncAudio << ")" << std::endl;
+
+                    reportFailure(MappedEvent::FailureMixUnderrun);
+                }
+                for (size_t f = 0; f < nframes; ++f) {
+                    sample_t sample = instrument[ch][f];
+                    if (sample > peak[ch])
+                        peak[ch] = sample;
+                    if (directToMaster)
+                        master[ch][f] += sample;
+                }
+            }
+
+            // If the instrument is connected straight to master we
+            // also need to skip() on the buss mixer's reader for it,
+            // otherwise it'll block because the buss mixer isn't
+            // needing to read it.
+
+            if (rb && directToMaster) {
+                rb->skip(nframes, 1); // 1 is the buss mixer's reader (magic)
+            }
+        }
+
+        if (sdb) {
+            LevelInfo info;
+            info.level = AudioLevel::multiplier_to_fader
+                         (peak[0], 127, AudioLevel::LongFader);
+            info.levelRight = AudioLevel::multiplier_to_fader
+                              (peak[1], 127, AudioLevel::LongFader);
+
+            sdb->setInstrumentLevel(id, info);
+        }
+    }
+
+    if (asyncAudio) {
+        if (!allInstrumentsDormant) {
+            dormantTime = RealTime::zeroTime;
+        } else {
+            dormantTime = dormantTime +
+                          RealTime::frame2RealTime(m_bufferSize, m_sampleRate);
+            if (dormantTime > RealTime(10, 0)) {
+                std::cerr << "JackDriver: dormantTime = " << dormantTime << ", resetting m_haveAsyncAudioEvent" << std::endl;
+                m_haveAsyncAudioEvent = false;
+            }
+        }
+    }
+
+    // Get master fader levels.  There's no pan on the master.
+    float gain = AudioLevel::dB_to_multiplier(m_masterLevel);
+    float masterPeak[2] = { 0.0, 0.0 };
+
+    for (int ch = 0; ch < 2; ++ch) {
+        for (size_t i = 0; i < nframes; ++i) {
+            sample_t sample = master[ch][i] * gain;
+            if (sample > masterPeak[ch])
+                masterPeak[ch] = sample;
+            master[ch][i] = sample;
+        }
+    }
+
+    if (sdb) {
+        LevelInfo info;
+        info.level = AudioLevel::multiplier_to_fader
+                     (masterPeak[0], 127, AudioLevel::LongFader);
+        info.levelRight = AudioLevel::multiplier_to_fader
+                          (masterPeak[1], 127, AudioLevel::LongFader);
+
+        sdb->setMasterLevel(info);
+    }
+
+    for (InstrumentId id = audioInstrumentBase;
+            id < audioInstrumentBase + audioInstruments; ++id) {
+        if (m_recordInputs[id].input == 0) {
+            jackProcessRecord(id, nframes, master[0], master[1], clocksRunning);
+        } else if (m_recordInputs[id].input < 1000) { // buss, already done
+            // nothing
+        } else if (!doneRecord) {
+            jackProcessRecord(id, nframes, 0, 0, clocksRunning);
+        }
+    }
+
+    if (playing) {
+        if (!lowLatencyMode) {
+            if (m_bussMixer->getBussCount() == 0) {
+                m_instrumentMixer->signal();
+            } else {
+                m_bussMixer->signal();
+            }
+        }
+    }
+
+    m_framesProcessed += nframes;
+
+#if (defined(DEBUG_JACK_DRIVER) || defined(DEBUG_JACK_PROCESS) || defined(DEBUG_JACK_TRANSPORT))
+
+    framesThisPlay += nframes; //!!!
+#endif
+#ifdef DEBUG_JACK_PROCESS
+
+    std::cerr << "JackDriver::jackProcess: " << nframes << " frames, " << framesThisPlay << " this play, " << m_framesProcessed << " total" << std::endl;
+#endif
+
+    return 0;
+}
+
+int
+JackDriver::jackProcessEmpty(jack_nframes_t nframes)
+{
+    sample_t *buffer;
+
+#ifdef DEBUG_JACK_PROCESS
+
+    std::cerr << "JackDriver::jackProcessEmpty" << std::endl;
+#endif
+
+    buffer = static_cast<sample_t *>
+             (jack_port_get_buffer(m_outputMasters[0], nframes));
+    if (buffer)
+        memset(buffer, 0, nframes * sizeof(sample_t));
+
+    buffer = static_cast<sample_t *>
+             (jack_port_get_buffer(m_outputMasters[1], nframes));
+    if (buffer)
+        memset(buffer, 0, nframes * sizeof(sample_t));
+
+    buffer = static_cast<sample_t *>
+             (jack_port_get_buffer(m_outputMonitors[0], nframes));
+    if (buffer)
+        memset(buffer, 0, nframes * sizeof(sample_t));
+
+    buffer = static_cast<sample_t *>
+             (jack_port_get_buffer(m_outputMonitors[1], nframes));
+    if (buffer)
+        memset(buffer, 0, nframes * sizeof(sample_t));
+
+    for (unsigned int i = 0; i < m_outputSubmasters.size(); ++i) {
+        buffer = static_cast<sample_t *>
+                 (jack_port_get_buffer(m_outputSubmasters[i], nframes));
+        if (buffer)
+            memset(buffer, 0, nframes * sizeof(sample_t));
+    }
+
+    for (unsigned int i = 0; i < m_outputInstruments.size(); ++i) {
+        buffer = static_cast<sample_t *>
+                 (jack_port_get_buffer(m_outputInstruments[i], nframes));
+        if (buffer)
+            memset(buffer, 0, nframes * sizeof(sample_t));
+    }
+
+    m_framesProcessed += nframes;
+
+#if (defined(DEBUG_JACK_DRIVER) || defined(DEBUG_JACK_PROCESS) || defined(DEBUG_JACK_TRANSPORT))
+
+    framesThisPlay += nframes;
+#endif
+#ifdef DEBUG_JACK_PROCESS
+
+    std::cerr << "JackDriver::jackProcess: " << nframes << " frames, " << framesThisPlay << " this play, " << m_framesProcessed << " total" << std::endl;
+#endif
+
+    return 0;
+}
+
+int
+JackDriver::jackProcessRecord(InstrumentId id,
+                              jack_nframes_t nframes,
+                              sample_t *sourceBufferLeft,
+                              sample_t *sourceBufferRight,
+                              bool clocksRunning)
+{
+#ifdef DEBUG_JACK_PROCESS
+    Profiler profiler("jackProcessRecord", true);
+#else
+#ifdef DEBUG_JACK_XRUN
+
+    Profiler profiler("jackProcessRecord", false);
+#endif
+#endif
+
+    SequencerDataBlock *sdb = m_alsaDriver->getSequencerDataBlock();
+    bool wroteSomething = false;
+    sample_t peakLeft = 0.0, peakRight = 0.0;
+
+#ifdef DEBUG_JACK_PROCESS
+
+    std::cerr << "JackDriver::jackProcessRecord(" << id << "): clocksRunning " << clocksRunning << std::endl;
+#endif
+
+    // Get input buffers
+    //
+    sample_t *inputBufferLeft = 0, *inputBufferRight = 0;
+
+    int recInput = m_recordInputs[id].input;
+
+    int channel = m_recordInputs[id].channel;
+    int channels = (channel == -1 ? 2 : 1);
+    if (channels == 2)
+        channel = 0;
+
+    float level = m_recordInputs[id].level;
+
+    if (sourceBufferLeft) {
+
+#ifdef DEBUG_JACK_PROCESS
+        std::cerr << "JackDriver::jackProcessRecord(" << id << "): buss input provided" << std::endl;
+#endif
+
+        inputBufferLeft = sourceBufferLeft;
+        if (sourceBufferRight)
+            inputBufferRight = sourceBufferRight;
+
+    } else if (recInput < 1000) {
+
+#ifdef DEBUG_JACK_PROCESS
+        std::cerr << "JackDriver::jackProcessRecord(" << id << "): no known input" << std::endl;
+#endif
+
+        return 0;
+
+    } else {
+
+#ifdef DEBUG_JACK_PROCESS
+        std::cerr << "JackDriver::jackProcessRecord(" << id << "): record input " << recInput << std::endl;
+#endif
+
+        int input = recInput - 1000;
+
+        int port = input * channels + channel;
+        int portRight = input * channels + 1;
+
+        if (port < int(m_inputPorts.size())) {
+            inputBufferLeft = static_cast<sample_t*>
+                              (jack_port_get_buffer(m_inputPorts[port], nframes));
+        }
+
+        if (channels == 2 && portRight < int(m_inputPorts.size())) {
+            inputBufferRight = static_cast<sample_t*>
+                               (jack_port_get_buffer(m_inputPorts[portRight], nframes));
+        }
+    }
+
+    float gain = AudioLevel::dB_to_multiplier(level);
+
+    if (m_alsaDriver->getRecordStatus() == RECORD_ON &&
+            clocksRunning &&
+            m_fileWriter->haveRecordFileOpen(id)) {
+
+#ifdef DEBUG_JACK_PROCESS
+        std::cerr << "JackDriver::jackProcessRecord(" << id << "): recording" << std::endl;
+#endif
+
+        memset(m_tempOutBuffer, 0, nframes * sizeof(sample_t));
+
+        if (inputBufferLeft) {
+            for (size_t i = 0; i < nframes; ++i) {
+                sample_t sample = inputBufferLeft[i] * gain;
+                if (sample > peakLeft)
+                    peakLeft = sample;
+                m_tempOutBuffer[i] = sample;
+            }
+
+            if (m_outputMonitors.size() > 0) {
+                sample_t *buf =
+                    static_cast<sample_t *>
+                    (jack_port_get_buffer(m_outputMonitors[0], nframes));
+                if (buf) {
+                    for (size_t i = 0; i < nframes; ++i) {
+                        buf[i] += m_tempOutBuffer[i];
+                    }
+                }
+            }
+
+            m_fileWriter->write(id, m_tempOutBuffer, 0, nframes);
+        }
+
+        if (channels == 2) {
+
+            if (inputBufferRight) {
+                for (size_t i = 0; i < nframes; ++i) {
+                    sample_t sample = inputBufferRight[i] * gain;
+                    if (sample > peakRight)
+                        peakRight = sample;
+                    m_tempOutBuffer[i] = sample;
+                }
+                if (m_outputMonitors.size() > 1) {
+                    sample_t *buf =
+                        static_cast<sample_t *>
+                        (jack_port_get_buffer(m_outputMonitors[1], nframes));
+                    if (buf) {
+                        for (size_t i = 0; i < nframes; ++i) {
+                            buf[i] += m_tempOutBuffer[i];
+                        }
+                    }
+                }
+            }
+
+            m_fileWriter->write(id, m_tempOutBuffer, 1, nframes);
+        }
+
+        wroteSomething = true;
+
+    } else {
+
+        // want peak levels and monitors anyway, even if not recording
+
+#ifdef DEBUG_JACK_PROCESS
+        std::cerr << "JackDriver::jackProcessRecord(" << id << "): monitoring only" << std::endl;
+#endif
+
+        if (inputBufferLeft) {
+
+            sample_t *buf = 0;
+            if (m_outputMonitors.size() > 0) {
+                buf = static_cast<sample_t *>
+                      (jack_port_get_buffer(m_outputMonitors[0], nframes));
+            }
+
+            for (size_t i = 0; i < nframes; ++i) {
+                sample_t sample = inputBufferLeft[i] * gain;
+                if (sample > peakLeft)
+                    peakLeft = sample;
+                if (buf)
+                    buf[i] = sample;
+            }
+
+            if (channels == 2 && inputBufferRight) {
+
+                buf = 0;
+                if (m_outputMonitors.size() > 1) {
+                    buf = static_cast<sample_t *>
+                          (jack_port_get_buffer(m_outputMonitors[1], nframes));
+                }
+
+                for (size_t i = 0; i < nframes; ++i) {
+                    sample_t sample = inputBufferRight[i] * gain;
+                    if (sample > peakRight)
+                        peakRight = sample;
+                    if (buf)
+                        buf[i] = sample;
+                }
+            }
+        }
+    }
+
+    if (channels < 2)
+        peakRight = peakLeft;
+
+    if (sdb) {
+        LevelInfo info;
+        info.level = AudioLevel::multiplier_to_fader
+                     (peakLeft, 127, AudioLevel::LongFader);
+        info.levelRight = AudioLevel::multiplier_to_fader
+                          (peakRight, 127, AudioLevel::LongFader);
+        sdb->setInstrumentRecordLevel(id, info);
+    }
+
+    if (wroteSomething) {
+        m_fileWriter->signal();
+    }
+
+    return 0;
+}
+
+
+int
+JackDriver::jackSyncCallback(jack_transport_state_t state,
+                             jack_position_t *position,
+                             void *arg)
+{
+    JackDriver *inst = (JackDriver *)arg;
+    if (!inst)
+        return true; // or rather, return "huh?"
+
+    inst->m_alsaDriver->checkTimerSync(0); // reset, as not processing
+
+    if (!inst->m_jackTransportEnabled)
+        return true; // ignore
+
+    ExternalTransport *transport =
+        inst->m_alsaDriver->getExternalTransportControl();
+    if (!transport)
+        return true;
+
+#ifdef DEBUG_JACK_TRANSPORT
+
+    std::cerr << "JackDriver::jackSyncCallback: state " << state << " [" << (state == 0 ? "stopped" : state == 1 ? "rolling" : state == 2 ? "looping" : state == 3 ? "starting" : "unknown") << "], frame " << position->frame << ", waiting " << inst->m_waiting << ", playing " << inst->m_alsaDriver->isPlaying() << std::endl;
+
+    std::cerr << "JackDriver::jackSyncCallback: m_waitingState " << inst->m_waitingState << ", unique_1 " << position->unique_1 << ", unique_2 " << position->unique_2 << std::endl;
+
+    std::cerr << "JackDriver::jackSyncCallback: rate " << position->frame_rate << ", bar " << position->bar << ", beat " << position->beat << ", tick " << position->tick << ", bpm " << position->beats_per_minute << std::endl;
+
+#endif
+
+    ExternalTransport::TransportRequest request =
+        ExternalTransport::TransportNoChange;
+
+    if (inst->m_alsaDriver->isPlaying()) {
+
+        if (state == JackTransportStarting) {
+            request = ExternalTransport::TransportJumpToTime;
+        } else if (state == JackTransportStopped) {
+            request = ExternalTransport::TransportStop;
+        }
+
+    } else {
+
+        if (state == JackTransportStarting) {
+            request = ExternalTransport::TransportStartAtTime;
+        } else if (state == JackTransportStopped) {
+            request = ExternalTransport::TransportNoChange;
+        }
+    }
+
+    if (!inst->m_waiting || inst->m_waitingState != state) {
+
+        if (request == ExternalTransport::TransportJumpToTime ||
+                request == ExternalTransport::TransportStartAtTime) {
+
+            RealTime rt = RealTime::frame2RealTime(position->frame,
+                                                   position->frame_rate);
+
+#ifdef DEBUG_JACK_TRANSPORT
+
+            std::cerr << "JackDriver::jackSyncCallback: Requesting jump to " << rt << std::endl;
+#endif
+
+            inst->m_waitingToken = transport->transportJump(request, rt);
+
+#ifdef DEBUG_JACK_TRANSPORT
+
+            std::cerr << "JackDriver::jackSyncCallback: My token is " << inst->m_waitingToken << std::endl;
+#endif
+
+        } else if (request == ExternalTransport::TransportStop) {
+
+#ifdef DEBUG_JACK_TRANSPORT
+            std::cerr << "JackDriver::jackSyncCallback: Requesting state change to " << request << std::endl;
+#endif
+
+            inst->m_waitingToken = transport->transportChange(request);
+
+#ifdef DEBUG_JACK_TRANSPORT
+
+            std::cerr << "JackDriver::jackSyncCallback: My token is " << inst->m_waitingToken << std::endl;
+#endif
+
+        } else if (request == ExternalTransport::TransportNoChange) {
+
+#ifdef DEBUG_JACK_TRANSPORT
+            std::cerr << "JackDriver::jackSyncCallback: Requesting no state change!" << std::endl;
+#endif
+
+            inst->m_waitingToken = transport->transportChange(request);
+
+#ifdef DEBUG_JACK_TRANSPORT
+
+            std::cerr << "JackDriver::jackSyncCallback: My token is " << inst->m_waitingToken << std::endl;
+#endif
+
+        }
+
+        inst->m_waiting = true;
+        inst->m_waitingState = state;
+
+#ifdef DEBUG_JACK_TRANSPORT
+
+        std::cerr << "JackDriver::jackSyncCallback: Setting waiting to " << inst->m_waiting << " and waiting state to " << inst->m_waitingState << " (request was " << request << ")" << std::endl;
+#endif
+
+        return 0;
+
+    } else {
+
+        if (transport->isTransportSyncComplete(inst->m_waitingToken)) {
+#ifdef DEBUG_JACK_TRANSPORT
+            std::cerr << "JackDriver::jackSyncCallback: Sync complete" << std::endl;
+#endif
+
+            return 1;
+        } else {
+#ifdef DEBUG_JACK_TRANSPORT
+            std::cerr << "JackDriver::jackSyncCallback: Sync not complete" << std::endl;
+#endif
+
+            return 0;
+        }
+    }
+}
+
+bool
+JackDriver::relocateTransportInternal(bool alsoStart)
+{
+    if (!m_client)
+        return true;
+
+#ifdef DEBUG_JACK_TRANSPORT
+
+    const char *fn = (alsoStart ?
+                      "JackDriver::startTransport" :
+                      "JackDriver::relocateTransport");
+#endif
+
+#ifdef DEBUG_JACK_TRANSPORT
+
+    std::cerr << fn << std::endl;
+#else
+#ifdef DEBUG_JACK_DRIVER
+
+    std::cerr << "JackDriver::relocateTransportInternal" << std::endl;
+#endif
+#endif
+
+    // m_waiting is true if we are waiting for the JACK transport
+    // to finish a change of state.
+
+    if (m_jackTransportEnabled) {
+
+        // If on the transport, we never return true here -- instead
+        // the JACK process calls startClocksApproved() to signal to
+        // the ALSA driver that it's time to go.  But we do use this
+        // to manage our JACK transport state requests.
+
+        // Where did this request come from?  Are we just responding
+        // to an external sync?
+
+        ExternalTransport *transport =
+            m_alsaDriver->getExternalTransportControl();
+
+        if (transport) {
+            if (transport->isTransportSyncComplete(m_waitingToken)) {
+
+                // Nope, this came from Rosegarden
+
+#ifdef DEBUG_JACK_TRANSPORT
+                std::cerr << fn << ": asking JACK transport to start, setting wait state" << std::endl;
+#endif
+
+                m_waiting = true;
+                m_waitingState = JackTransportStarting;
+
+                long frame = RealTime::realTime2Frame
+                             (m_alsaDriver->getSequencerTime(), m_sampleRate);
+
+                if (frame < 0) {
+                    // JACK Transport doesn't support preroll and
+                    // can't set transport position to before zero
+                    // (frame count is unsigned), so there's no very
+                    // satisfactory fix for what to do for count-in
+                    // bars.  Let's just start at zero instead.
+                    jack_transport_locate(m_client, 0);
+                } else {
+                    jack_transport_locate(m_client, frame);
+                }
+
+                if (alsoStart) {
+                    jack_transport_start(m_client);
+                    m_ignoreProcessTransportCount = 1;
+                } else {
+                    m_ignoreProcessTransportCount = 2;
+                }
+            } else {
+#ifdef DEBUG_JACK_TRANSPORT
+                std::cerr << fn << ": waiting already" << std::endl;
+#endif
+
+            }
+        }
+        return false;
+    }
+
+#if (defined(DEBUG_JACK_DRIVER) || defined(DEBUG_JACK_PROCESS) || defined(DEBUG_JACK_TRANSPORT))
+    framesThisPlay = 0; //!!!
+    struct timeval tv;
+    (void)gettimeofday(&tv, 0);
+    startTime = RealTime(tv.tv_sec, tv.tv_usec * 1000); //!!!
+#endif
+#ifdef DEBUG_JACK_TRANSPORT
+
+    std::cerr << fn << ": not on JACK transport, accepting right away" << std::endl;
+#endif
+
+    return true;
+}
+
+bool
+JackDriver::startTransport()
+{
+    return relocateTransportInternal(true);
+}
+
+bool
+JackDriver::relocateTransport()
+{
+
+    return relocateTransportInternal(false);
+}
+
+void
+JackDriver::stopTransport()
+{
+    if (!m_client)
+        return ;
+
+    std::cerr << "JackDriver::stopTransport: resetting m_haveAsyncAudioEvent" << std::endl;
+    m_haveAsyncAudioEvent = false;
+
+#ifdef DEBUG_JACK_TRANSPORT
+
+    struct timeval tv;
+    (void)gettimeofday(&tv, 0);
+    RealTime endTime = RealTime(tv.tv_sec, tv.tv_usec * 1000); //!!!
+    std::cerr << "\nJackDriver::stop: frames this play: " << framesThisPlay << ", elapsed " << (endTime - startTime) << std::endl;
+#endif
+
+    if (m_jackTransportEnabled) {
+
+        // Where did this request come from?  Is this a result of our
+        // sync to a transport that has in fact already stopped?
+
+        ExternalTransport *transport =
+            m_alsaDriver->getExternalTransportControl();
+
+        if (transport) {
+            if (transport->isTransportSyncComplete(m_waitingToken)) {
+
+                // No, we have no outstanding external requests; this
+                // must have genuinely been requested from within
+                // Rosegarden, so:
+
+#ifdef DEBUG_JACK_TRANSPORT
+                std::cerr << "JackDriver::stop: internal request, asking JACK transport to stop" << std::endl;
+#endif
+
+                jack_transport_stop(m_client);
+
+            } else {
+                // Nothing to do
+
+#ifdef DEBUG_JACK_TRANSPORT
+                std::cerr << "JackDriver::stop: external request, JACK transport is already stopped" << std::endl;
+#endif
+
+            }
+        }
+    }
+
+    if (m_instrumentMixer)
+        m_instrumentMixer->resetAllPlugins(true); // discard events too
+}
+
+
+// Pick up any change of buffer size
+//
+int
+JackDriver::jackBufferSize(jack_nframes_t nframes, void *arg)
+{
+    JackDriver *inst = static_cast<JackDriver*>(arg);
+
+#ifdef DEBUG_JACK_DRIVER
+
+    std::cerr << "JackDriver::jackBufferSize - buffer size changed to "
+    << nframes << std::endl;
+#endif
+
+    inst->m_bufferSize = nframes;
+
+    // Recreate our temporary mix buffers to the new size
+    //
+    //!!! need buffer size change callbacks on plugins (so long as they
+    // have internal buffers) and the mix manager, with locks acquired
+    // appropriately
+
+    delete [] inst->m_tempOutBuffer;
+    inst->m_tempOutBuffer = new sample_t[inst->m_bufferSize];
+
+    return 0;
+}
+
+// Sample rate change
+//
+int
+JackDriver::jackSampleRate(jack_nframes_t nframes, void *arg)
+{
+    JackDriver *inst = static_cast<JackDriver*>(arg);
+
+#ifdef DEBUG_JACK_DRIVER
+
+    std::cerr << "JackDriver::jackSampleRate - sample rate changed to "
+    << nframes << std::endl;
+#endif
+
+    inst->m_sampleRate = nframes;
+
+    return 0;
+}
+
+void
+JackDriver::jackShutdown(void *arg)
+{
+#ifdef DEBUG_JACK_DRIVER
+    std::cerr << "JackDriver::jackShutdown() - callback received - "
+    << "informing GUI" << std::endl;
+#endif
+
+#ifdef DEBUG_JACK_XRUN
+
+    std::cerr << "JackDriver::jackShutdown" << std::endl;
+    Profiles::getInstance()->dump();
+#endif
+
+    JackDriver *inst = static_cast<JackDriver*>(arg);
+    inst->m_ok = false;
+    inst->m_kickedOutAt = time(0);
+    inst->reportFailure(MappedEvent::FailureJackDied);
+}
+
+int
+JackDriver::jackXRun(void *arg)
+{
+#ifdef DEBUG_JACK_DRIVER
+    std::cerr << "JackDriver::jackXRun" << std::endl;
+#endif
+
+#ifdef DEBUG_JACK_XRUN
+
+    std::cerr << "JackDriver::jackXRun" << std::endl;
+    Profiles::getInstance()->dump();
+#endif
+
+    // Report to GUI
+    //
+    JackDriver *inst = static_cast<JackDriver*>(arg);
+    inst->reportFailure(MappedEvent::FailureXRuns);
+
+    return 0;
+}
+
+
+void
+
+JackDriver::restoreIfRestorable()
+{
+    if (m_kickedOutAt == 0)
+        return ;
+
+    if (m_client) {
+        jack_client_close(m_client);
+        std::cerr << "closed client" << std::endl;
+        m_client = 0;
+    }
+
+    time_t now = time(0);
+
+    if (now < m_kickedOutAt || now >= m_kickedOutAt + 3) {
+
+        if (m_instrumentMixer)
+            m_instrumentMixer->resetAllPlugins(true);
+        std::cerr << "reset plugins" << std::endl;
+
+        initialise(true);
+
+        if (m_ok) {
+            reportFailure(MappedEvent::FailureJackRestart);
+        } else {
+            reportFailure(MappedEvent::FailureJackRestartFailed);
+        }
+
+        m_kickedOutAt = 0;
+    }
+}
+
+void
+JackDriver::prepareAudio()
+{
+    if (!m_instrumentMixer)
+        return ;
+
+    // This is used when restarting clocks after repositioning, but
+    // when not actually playing (yet).  We need to do things like
+    // regenerating the processing buffers here.  prebufferAudio()
+    // also does all of this, but rather more besides.
+
+    m_instrumentMixer->allocateBuffers();
+    m_instrumentMixer->resetAllPlugins(false);
+}
+
+void
+JackDriver::prebufferAudio()
+{
+    if (!m_instrumentMixer)
+        return ;
+
+    // We want this to happen when repositioning during playback, and
+    // stopTransport no longer happens then, so we call it from here.
+    // NB. Don't want to discard events here as this is called after
+    // pushing events to the soft synth queues at startup
+    m_instrumentMixer->resetAllPlugins(false);
+
+#ifdef DEBUG_JACK_DRIVER
+
+    std::cerr << "JackDriver::prebufferAudio: sequencer time is "
+    << m_alsaDriver->getSequencerTime() << std::endl;
+#endif
+
+    RealTime sliceStart = getNextSliceStart(m_alsaDriver->getSequencerTime());
+
+    m_fileReader->fillBuffers(sliceStart);
+
+    if (m_bussMixer->getBussCount() > 0) {
+        m_bussMixer->fillBuffers(sliceStart); // also calls on m_instrumentMixer
+    } else {
+        m_instrumentMixer->fillBuffers(sliceStart);
+    }
+}
+
+void
+JackDriver::kickAudio()
+{
+#ifdef DEBUG_JACK_PROCESS
+    std::cerr << "JackDriver::kickAudio" << std::endl;
+#endif
+
+    if (m_fileReader)
+        m_fileReader->kick();
+    if (m_instrumentMixer)
+        m_instrumentMixer->kick();
+    if (m_bussMixer)
+        m_bussMixer->kick();
+    if (m_fileWriter)
+        m_fileWriter->kick();
+}
+
+void
+JackDriver::updateAudioData()
+{
+    if (!m_ok || !m_client)
+        return ;
+
+#ifdef DEBUG_JACK_DRIVER 
+    //    std::cerr << "JackDriver::updateAudioData starting" << std::endl;
+#endif
+
+    MappedAudioBuss *mbuss =
+        m_alsaDriver->getMappedStudio()->getAudioBuss(0);
+
+    if (mbuss) {
+        float level = 0.0;
+        (void)mbuss->getProperty(MappedAudioBuss::Level, level);
+        m_masterLevel = level;
+    }
+
+    unsigned long directMasterAudioInstruments = 0L;
+    unsigned long directMasterSynthInstruments = 0L;
+
+    InstrumentId audioInstrumentBase;
+    int audioInstruments;
+    m_alsaDriver->getAudioInstrumentNumbers(audioInstrumentBase, audioInstruments);
+
+    InstrumentId synthInstrumentBase;
+    int synthInstruments;
+    m_alsaDriver->getSoftSynthInstrumentNumbers(synthInstrumentBase, synthInstruments);
+
+    RealTime jackLatency = getAudioPlayLatency();
+    RealTime maxLatency = RealTime::zeroTime;
+
+    for (int i = 0; i < audioInstruments + synthInstruments; ++i) {
+
+        InstrumentId id;
+        if (i < audioInstruments)
+            id = audioInstrumentBase + i;
+        else
+            id = synthInstrumentBase + (i - audioInstruments);
+
+        MappedAudioFader *fader = m_alsaDriver->getMappedStudio()->getAudioFader(id);
+        if (!fader)
+            continue;
+
+        float f = 2;
+        (void)fader->getProperty(MappedAudioFader::Channels, f);
+        int channels = (int)f;
+
+        int inputChannel = -1;
+        if (channels == 1) {
+            float f = 0;
+            (void)fader->getProperty(MappedAudioFader::InputChannel, f);
+            inputChannel = (int)f;
+        }
+
+        float level = 0.0;
+        (void)fader->getProperty(MappedAudioFader::FaderRecordLevel, level);
+
+        // Like in base/Instrument.h, we use numbers < 1000 to
+        // mean buss numbers and >= 1000 to mean record ins
+        // when recording the record input number.
+
+        MappedObjectValueList connections = fader->getConnections
+                                            (MappedConnectableObject::In);
+        int input = 1000;
+
+        if (connections.empty()) {
+
+            std::cerr << "No connections in for record instrument "
+            << (id) << " (mapped id " << fader->getId() << ")" << std::endl;
+
+            // oh dear.
+            input = 1000;
+
+        } else if (*connections.begin() == mbuss->getId()) {
+
+            input = 0;
+
+        } else {
+
+            MappedObject *obj = m_alsaDriver->getMappedStudio()->
+                                getObjectById(MappedObjectId(*connections.begin()));
+
+            if (!obj) {
+
+                std::cerr << "No such object as " << *connections.begin() << std::endl;
+                input = 1000;
+            } else if (obj->getType() == MappedObject::AudioBuss) {
+                input = (int)((MappedAudioBuss *)obj)->getBussId();
+            } else if (obj->getType() == MappedObject::AudioInput) {
+                input = (int)((MappedAudioInput *)obj)->getInputNumber()
+                        + 1000;
+            } else {
+                std::cerr << "Object " << *connections.begin() << " is not buss or input" << std::endl;
+                input = 1000;
+            }
+        }
+
+        if (m_recordInputs[id].input != input) {
+            std::cerr << "Changing record input for instrument "
+            << id << " to " << input << std::endl;
+        }
+        m_recordInputs[id] = RecordInputDesc(input, inputChannel, level);
+
+        size_t pluginLatency = 0;
+        bool empty = m_instrumentMixer->isInstrumentEmpty(id);
+
+        if (!empty) {
+            pluginLatency = m_instrumentMixer->getPluginLatency(id);
+        }
+
+        // If we find the object is connected to no output, or to buss
+        // number 0 (the master), then we set the bit appropriately.
+
+        connections = fader->getConnections(MappedConnectableObject::Out);
+
+        if (connections.empty() || (*connections.begin() == mbuss->getId())) {
+            if (i < audioInstruments) {
+                directMasterAudioInstruments |= (1 << i);
+            } else {
+                directMasterSynthInstruments |= (1 << (i - audioInstruments));
+            }
+        } else if (!empty) {
+            pluginLatency +=
+                m_instrumentMixer->getPluginLatency((unsigned int) * connections.begin());
+        }
+
+        if (empty) {
+            m_instrumentLatencies[id] = RealTime::zeroTime;
+        } else {
+            m_instrumentLatencies[id] = jackLatency +
+                                        RealTime::frame2RealTime(pluginLatency, m_sampleRate);
+            if (m_instrumentLatencies[id] > maxLatency) {
+                maxLatency = m_instrumentLatencies[id];
+            }
+        }
+    }
+
+    m_maxInstrumentLatency = maxLatency;
+    m_directMasterAudioInstruments = directMasterAudioInstruments;
+    m_directMasterSynthInstruments = directMasterSynthInstruments;
+    m_maxInstrumentLatency = maxLatency;
+
+    int inputs = m_alsaDriver->getMappedStudio()->
+                 getObjectCount(MappedObject::AudioInput);
+
+    if (m_client) {
+        // this will return with no work if the inputs are already correct:
+        createRecordInputs(inputs);
+    }
+
+    m_bussMixer->updateInstrumentConnections();
+    m_instrumentMixer->updateInstrumentMuteStates();
+
+    if (m_bussMixer->getBussCount() == 0 || m_alsaDriver->getLowLatencyMode()) {
+        if (m_bussMixer->running()) {
+            m_bussMixer->terminate();
+        }
+    } else {
+        if (!m_bussMixer->running()) {
+            m_bussMixer->run();
+        }
+    }
+
+    if (m_alsaDriver->getLowLatencyMode()) {
+        if (m_instrumentMixer->running()) {
+            m_instrumentMixer->terminate();
+        }
+    } else {
+        if (!m_instrumentMixer->running()) {
+            m_instrumentMixer->run();
+        }
+    }
+
+#ifdef DEBUG_JACK_DRIVER 
+    //    std::cerr << "JackDriver::updateAudioData exiting" << std::endl;
+#endif
+}
+
+void
+JackDriver::setAudioBussLevels(int bussNo, float dB, float pan)
+{
+    if (m_bussMixer) {
+        m_bussMixer->setBussLevels(bussNo, dB, pan);
+    }
+}
+
+void
+JackDriver::setAudioInstrumentLevels(InstrumentId instrument, float dB, float pan)
+{
+    if (m_instrumentMixer) {
+        m_instrumentMixer->setInstrumentLevels(instrument, dB, pan);
+    }
+}
+
+RealTime
+JackDriver::getNextSliceStart(const RealTime &now) const
+{
+    jack_nframes_t frame;
+    bool neg = false;
+
+    if (now < RealTime::zeroTime) {
+        neg = true;
+        frame = RealTime::realTime2Frame(RealTime::zeroTime - now, m_sampleRate);
+    } else {
+        frame = RealTime::realTime2Frame(now, m_sampleRate);
+    }
+
+    jack_nframes_t rounded = frame;
+    rounded /= m_bufferSize;
+    rounded *= m_bufferSize;
+
+    RealTime roundrt;
+
+    if (rounded == frame)
+        roundrt = RealTime::frame2RealTime(rounded, m_sampleRate);
+    else if (neg)
+        roundrt = RealTime::frame2RealTime(rounded - m_bufferSize, m_sampleRate);
+    else
+        roundrt = RealTime::frame2RealTime(rounded + m_bufferSize, m_sampleRate);
+
+    if (neg)
+        roundrt = RealTime::zeroTime - roundrt;
+
+    return roundrt;
+}
+
+
+int
+JackDriver::getAudioQueueLocks()
+{
+    // We have to lock the mixers first, because the mixers can try to
+    // lock the disk manager from within a locked section -- so if we
+    // locked the disk manager first we would risk deadlock when
+    // trying to acquire the instrument mixer lock
+
+    int rv = 0;
+    if (m_bussMixer) {
+#ifdef DEBUG_JACK_DRIVER
+        std::cerr << "JackDriver::getAudioQueueLocks: trying to lock buss mixer" << std::endl;
+#endif
+
+        rv = m_bussMixer->getLock();
+        if (rv)
+            return rv;
+    }
+    if (m_instrumentMixer) {
+#ifdef DEBUG_JACK_DRIVER
+        std::cerr << "JackDriver::getAudioQueueLocks: ok, now trying for instrument mixer" << std::endl;
+#endif
+
+        rv = m_instrumentMixer->getLock();
+        if (rv)
+            return rv;
+    }
+    if (m_fileReader) {
+#ifdef DEBUG_JACK_DRIVER
+        std::cerr << "JackDriver::getAudioQueueLocks: ok, now trying for disk reader" << std::endl;
+#endif
+
+        rv = m_fileReader->getLock();
+        if (rv)
+            return rv;
+    }
+    if (m_fileWriter) {
+#ifdef DEBUG_JACK_DRIVER
+        std::cerr << "JackDriver::getAudioQueueLocks: ok, now trying for disk writer" << std::endl;
+#endif
+
+        rv = m_fileWriter->getLock();
+    }
+#ifdef DEBUG_JACK_DRIVER
+    std::cerr << "JackDriver::getAudioQueueLocks: ok" << std::endl;
+#endif
+
+    return rv;
+}
+
+int
+JackDriver::tryAudioQueueLocks()
+{
+    int rv = 0;
+    if (m_bussMixer) {
+        rv = m_bussMixer->tryLock();
+        if (rv)
+            return rv;
+    }
+    if (m_instrumentMixer) {
+        rv = m_instrumentMixer->tryLock();
+        if (rv) {
+            if (m_bussMixer) {
+                m_bussMixer->releaseLock();
+            }
+        }
+    }
+    if (m_fileReader) {
+        rv = m_fileReader->tryLock();
+        if (rv) {
+            if (m_instrumentMixer) {
+                m_instrumentMixer->releaseLock();
+            }
+            if (m_bussMixer) {
+                m_bussMixer->releaseLock();
+            }
+        }
+    }
+    if (m_fileWriter) {
+        rv = m_fileWriter->tryLock();
+        if (rv) {
+            if (m_fileReader) {
+                m_fileReader->releaseLock();
+            }
+            if (m_instrumentMixer) {
+                m_instrumentMixer->releaseLock();
+            }
+            if (m_bussMixer) {
+                m_bussMixer->releaseLock();
+            }
+        }
+    }
+    return rv;
+}
+
+int
+JackDriver::releaseAudioQueueLocks()
+{
+    int rv = 0;
+#ifdef DEBUG_JACK_DRIVER
+
+    std::cerr << "JackDriver::releaseAudioQueueLocks" << std::endl;
+#endif
+
+    if (m_fileWriter)
+        rv = m_fileWriter->releaseLock();
+    if (m_fileReader)
+        rv = m_fileReader->releaseLock();
+    if (m_instrumentMixer)
+        rv = m_instrumentMixer->releaseLock();
+    if (m_bussMixer)
+        rv = m_bussMixer->releaseLock();
+    return rv;
+}
+
+
+void
+JackDriver::setPluginInstance(InstrumentId id, QString identifier,
+                              int position)
+{
+    if (m_instrumentMixer) {
+        m_instrumentMixer->setPlugin(id, position, identifier);
+    }
+    if (!m_alsaDriver->isPlaying()) {
+        prebufferAudio(); // to ensure the plugin's ringbuffers are generated
+    }
+}
+
+void
+JackDriver::removePluginInstance(InstrumentId id, int position)
+{
+    if (m_instrumentMixer)
+        m_instrumentMixer->removePlugin(id, position);
+}
+
+void
+JackDriver::removePluginInstances()
+{
+    if (m_instrumentMixer)
+        m_instrumentMixer->removeAllPlugins();
+}
+
+void
+JackDriver::setPluginInstancePortValue(InstrumentId id, int position,
+                                       unsigned long portNumber,
+                                       float value)
+{
+    if (m_instrumentMixer)
+        m_instrumentMixer->setPluginPortValue(id, position, portNumber, value);
+}
+
+float
+JackDriver::getPluginInstancePortValue(InstrumentId id, int position,
+                                       unsigned long portNumber)
+{
+    if (m_instrumentMixer)
+        return m_instrumentMixer->getPluginPortValue(id, position, portNumber);
+    return 0;
+}
+
+void
+JackDriver::setPluginInstanceBypass(InstrumentId id, int position, bool value)
+{
+    if (m_instrumentMixer)
+        m_instrumentMixer->setPluginBypass(id, position, value);
+}
+
+QStringList
+JackDriver::getPluginInstancePrograms(InstrumentId id, int position)
+{
+    if (m_instrumentMixer)
+        return m_instrumentMixer->getPluginPrograms(id, position);
+    return QStringList();
+}
+
+QString
+JackDriver::getPluginInstanceProgram(InstrumentId id, int position)
+{
+    if (m_instrumentMixer)
+        return m_instrumentMixer->getPluginProgram(id, position);
+    return QString();
+}
+
+QString
+JackDriver::getPluginInstanceProgram(InstrumentId id, int position,
+                                     int bank, int program)
+{
+    if (m_instrumentMixer)
+        return m_instrumentMixer->getPluginProgram(id, position, bank, program);
+    return QString();
+}
+
+unsigned long
+JackDriver::getPluginInstanceProgram(InstrumentId id, int position, QString name)
+{
+    if (m_instrumentMixer)
+        return m_instrumentMixer->getPluginProgram(id, position, name);
+    return 0;
+}
+
+void
+JackDriver::setPluginInstanceProgram(InstrumentId id, int position, QString program)
+{
+    if (m_instrumentMixer)
+        m_instrumentMixer->setPluginProgram(id, position, program);
+}
+
+QString
+JackDriver::configurePlugin(InstrumentId id, int position, QString key, QString value)
+{
+    if (m_instrumentMixer)
+        return m_instrumentMixer->configurePlugin(id, position, key, value);
+    return QString();
+}
+
+RunnablePluginInstance *
+JackDriver::getSynthPlugin(InstrumentId id)
+{
+    if (m_instrumentMixer)
+        return m_instrumentMixer->getSynthPlugin(id);
+    else
+        return 0;
+}
+
+void
+JackDriver::clearSynthPluginEvents()
+{
+    if (!m_instrumentMixer) return;
+
+#ifdef DEBUG_JACK_DRIVER
+    std::cerr << "JackDriver::clearSynthPluginEvents" << std::endl;
+#endif
+
+    m_instrumentMixer->discardPluginEvents();
+}
+
+bool
+JackDriver::openRecordFile(InstrumentId id,
+                           const std::string &filename)
+{
+    if (m_fileWriter) {
+        if (!m_fileWriter->running()) {
+            m_fileWriter->run();
+        }
+        return m_fileWriter->openRecordFile(id, filename);
+    } else {
+        std::cerr << "JackDriver::openRecordFile: No file writer available!" << std::endl;
+        return false;
+    }
+}
+
+bool
+JackDriver::closeRecordFile(InstrumentId id,
+                            AudioFileId &returnedId)
+{
+    if (m_fileWriter) {
+        return m_fileWriter->closeRecordFile(id, returnedId);
+        if (m_fileWriter->running() && !m_fileWriter->haveRecordFilesOpen()) {
+            m_fileWriter->terminate();
+        }
+    } else
+        return false;
+}
+
+
+void
+JackDriver::reportFailure(MappedEvent::FailureCode code)
+{
+    if (m_alsaDriver)
+        m_alsaDriver->reportFailure(code);
+}
+
+
+}
+
+#endif // HAVE_LIBJACK
+#endif // HAVE_ALSA
diff --git a/src/sound/JackDriver.h b/src/sound/JackDriver.h
new file mode 100644
index 0000000..b46080d
--- /dev/null
+++ b/src/sound/JackDriver.h
@@ -0,0 +1,297 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _JACKDRIVER_H_
+#define _JACKDRIVER_H_
+
+#ifdef HAVE_ALSA
+#ifdef HAVE_LIBJACK
+
+#include "RunnablePluginInstance.h"
+#include <jack/jack.h>
+#include "SoundDriver.h"
+#include "Instrument.h"
+#include "RealTime.h"
+#include "ExternalTransport.h"
+#include <qstringlist.h>
+
+namespace Rosegarden
+{
+
+class AlsaDriver;
+class AudioBussMixer;
+class AudioInstrumentMixer;
+class AudioFileReader;
+class AudioFileWriter;
+
+class JackDriver
+{
+public:
+    // convenience
+    typedef jack_default_audio_sample_t sample_t;
+
+    JackDriver(AlsaDriver *alsaDriver);
+    virtual ~JackDriver();
+
+    bool isOK() const { return m_ok; }
+
+    bool isTransportEnabled() { return m_jackTransportEnabled; }
+    bool isTransportMaster () { return m_jackTransportMaster; }
+
+    void setTransportEnabled(bool e) { m_jackTransportEnabled = e; }
+    void setTransportMaster (bool m) { m_jackTransportMaster  = m; }
+
+    // These methods call back on the sound driver if necessary to
+    // establish the current transport location to start at or
+    // relocate to.  startTransport and relocateTransport return true
+    // if they have completed and the sound driver can safely call
+    // startClocks; false if the sound driver should wait for the JACK
+    // driver to call back on startClocksApproved before starting.
+    bool startTransport();
+    bool relocateTransport();
+    void stopTransport();
+
+    RealTime getAudioPlayLatency() const;
+    RealTime getAudioRecordLatency() const;
+    RealTime getInstrumentPlayLatency(InstrumentId) const;
+    RealTime getMaximumPlayLatency() const;
+
+    // Plugin instance management
+    //
+    virtual void setPluginInstance(InstrumentId id,
+                                   QString identifier,
+                                   int position);
+
+    virtual void removePluginInstance(InstrumentId id, int position);
+
+    // Remove all plugin instances
+    //
+    virtual void removePluginInstances();
+
+    virtual void setPluginInstancePortValue(InstrumentId id,
+                                            int position,
+                                            unsigned long portNumber,
+                                            float value);
+
+    virtual float getPluginInstancePortValue(InstrumentId id,
+                                             int position,
+                                             unsigned long portNumber);
+
+    virtual void setPluginInstanceBypass(InstrumentId id,
+                                         int position,
+                                         bool value);
+
+    virtual QStringList getPluginInstancePrograms(InstrumentId id,
+                                                  int position); 
+
+    virtual QString getPluginInstanceProgram(InstrumentId id,
+                                             int position); 
+  
+    virtual QString getPluginInstanceProgram(InstrumentId id,
+                                             int position,
+                                             int bank,
+                                             int program); 
+  
+    virtual unsigned long getPluginInstanceProgram(InstrumentId id,
+                                                   int position,
+                                                   QString name);
+  
+    virtual void setPluginInstanceProgram(InstrumentId id,
+                                          int position,
+                                          QString program);
+
+    virtual QString configurePlugin(InstrumentId id,
+                                    int position, 
+                                    QString key, QString value);
+
+    virtual RunnablePluginInstance *getSynthPlugin(InstrumentId id);
+
+    virtual void clearSynthPluginEvents(); // when stopping
+
+    virtual unsigned int getSampleRate() const { return m_sampleRate; }
+    virtual unsigned int getBufferSize() const { return m_bufferSize; }
+
+    // A new audio file for storage of our recorded samples - the
+    // file stays open so we can append samples at will.  We must
+    // explicitly close the file eventually though to make sure
+    // the integrity is correct (sample sizes must be written).
+    //
+    bool openRecordFile(InstrumentId id,
+                        const std::string &fileName);
+    bool closeRecordFile(InstrumentId id,
+                         AudioFileId &returnedId);
+
+    // Set or change the number of audio inputs and outputs.
+    // The first of these is slightly misnamed -- the submasters
+    // argument controls the number of busses, not ports (which
+    // may or may not exist depending on the setAudioPorts call).
+    //
+    void setAudioPorts(bool faderOuts, bool submasterOuts);
+
+    // Locks used by the disk thread and mix thread.  The AlsaDriver
+    // should hold these locks whenever it wants to modify its audio
+    // play queue -- at least when adding or removing files or
+    // resetting status; it doesn't need to hold the locks when
+    // incrementing their statuses or simply reading them.
+    //
+    int getAudioQueueLocks();
+    int tryAudioQueueLocks();
+    int releaseAudioQueueLocks();
+
+    void prepareAudio(); // when repositioning etc
+    void prebufferAudio(); // when starting playback (incorporates prepareAudio)
+    void kickAudio(); // for paranoia only
+
+    // Because we don't want to do any lookups that might involve
+    // locking etc from within the JACK process thread, we instead
+    // call this regularly from the ALSA driver thread -- it looks up
+    // various bits of data such as the master fader and monitoring
+    // levels, number of inputs etc and either processes them or
+    // writes them into simple records in the JACK driver for process
+    // to read.  Actually quite a lot of work.
+    //
+    void updateAudioData();
+
+    // Similarly, set data on the buss mixer to avoid the buss mixer
+    // having to call back on the mapped studio to discover it
+    // 
+    void setAudioBussLevels(int bussNo, float dB, float pan);
+
+    // Likewise for instrument mixer
+    // 
+    void setAudioInstrumentLevels(InstrumentId instrument, float dB, float pan);
+
+    // Called from AlsaDriver to indicate that an async MIDI event is
+    // being sent to a soft synth.  JackDriver uses this to suggest
+    // that it needs to start processing soft synths, if it wasn't
+    // already.  It will switch this off again itself when things
+    // fall silent.
+    // 
+    void setHaveAsyncAudioEvent() { m_haveAsyncAudioEvent = true; }
+
+    RealTime getNextSliceStart(const RealTime &now) const;
+
+    // For audit purposes only.
+    size_t getFramesProcessed() const { return m_framesProcessed; }
+
+    // Reinitialise if we've been kicked off JACK -- if we can
+    // 
+    void restoreIfRestorable();
+
+    // Report back to GUI via the AlsaDriver
+    //
+    void reportFailure(MappedEvent::FailureCode code);
+
+protected:
+
+    // static methods for JACK process thread:
+    static int   jackProcessStatic(jack_nframes_t nframes, void *arg);
+    static int   jackBufferSize(jack_nframes_t nframes, void *arg);
+    static int   jackSampleRate(jack_nframes_t nframes, void *arg);
+    static void  jackShutdown(void *arg);
+    static int   jackXRun(void *);
+
+    // static JACK transport callbacks
+    static int   jackSyncCallback(jack_transport_state_t,
+                                  jack_position_t *, void *);
+    static int   jackTimebaseCallback(jack_transport_state_t,
+                                      jack_nframes_t,
+                                      jack_position_t *,
+                                      int,
+                                      void *);
+
+    // jackProcessStatic delegates to this
+    int          jackProcess(jack_nframes_t nframes);
+    int          jackProcessRecord(InstrumentId id,
+                                   jack_nframes_t nframes,
+                                   sample_t *, sample_t *, bool);
+    int          jackProcessEmpty(jack_nframes_t nframes);
+
+    // other helper methods:
+
+    void initialise(bool reinitialise = false);
+
+    bool createMainOutputs();
+    bool createFaderOutputs(int audioPairs, int synthPairs);
+    bool createSubmasterOutputs(int pairs);
+    bool createRecordInputs(int pairs);
+
+    bool relocateTransportInternal(bool alsoStart);
+
+    // data members:
+
+    jack_client_t               *m_client;
+
+    std::vector<jack_port_t *>   m_inputPorts;
+    std::vector<jack_port_t *>   m_outputInstruments;
+    std::vector<jack_port_t *>   m_outputSubmasters;
+    std::vector<jack_port_t *>   m_outputMonitors;
+    std::vector<jack_port_t *>   m_outputMasters;
+
+    jack_nframes_t               m_bufferSize;
+    jack_nframes_t               m_sampleRate;
+
+    sample_t                    *m_tempOutBuffer;
+
+    bool                         m_jackTransportEnabled;
+    bool                         m_jackTransportMaster;
+
+    bool                         m_waiting;
+    jack_transport_state_t       m_waitingState;
+    ExternalTransport::TransportToken m_waitingToken;
+    int                          m_ignoreProcessTransportCount;
+
+    AudioBussMixer              *m_bussMixer;
+    AudioInstrumentMixer        *m_instrumentMixer;
+    AudioFileReader             *m_fileReader;
+    AudioFileWriter             *m_fileWriter;
+    AlsaDriver                  *m_alsaDriver;
+
+    float                        m_masterLevel;
+    unsigned long                m_directMasterAudioInstruments; // bitmap
+    unsigned long                m_directMasterSynthInstruments;
+    std::map<InstrumentId, RealTime> m_instrumentLatencies;
+    RealTime                     m_maxInstrumentLatency;
+    bool                         m_haveAsyncAudioEvent;
+
+    struct RecordInputDesc {
+        int   input;
+        int   channel;
+        float level;
+        RecordInputDesc(int i = 1000, int c = -1, float l = 0.0f) :
+            input(i), channel(c), level(l) { }
+    };
+    typedef std::map<InstrumentId, RecordInputDesc> RecordInputMap;
+    RecordInputMap               m_recordInputs;
+
+    time_t                       m_kickedOutAt;
+    size_t                       m_framesProcessed;
+    bool                         m_ok;
+};
+
+
+}
+
+#endif
+#endif
+
+#endif
+
diff --git a/src/sound/LADSPAPluginFactory.cpp b/src/sound/LADSPAPluginFactory.cpp
new file mode 100644
index 0000000..2a4a4ea
--- /dev/null
+++ b/src/sound/LADSPAPluginFactory.cpp
@@ -0,0 +1,841 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "LADSPAPluginFactory.h"
+#include <iostream>
+#include <cstdlib>
+
+#ifdef HAVE_LADSPA
+
+#include <dlfcn.h>
+#include <qdir.h>
+#include <cmath>
+
+#include "AudioPluginInstance.h"
+#include "LADSPAPluginInstance.h"
+#include "MappedStudio.h"
+#include "PluginIdentifier.h"
+
+#ifdef HAVE_LIBLRDF
+#include "lrdf.h"
+#endif // HAVE_LIBLRDF
+
+#include <kdebug.h>
+
+namespace Rosegarden
+{
+
+LADSPAPluginFactory::LADSPAPluginFactory()
+{}
+
+LADSPAPluginFactory::~LADSPAPluginFactory()
+{
+    for (std::set
+                <RunnablePluginInstance *>::iterator i = m_instances.begin();
+                i != m_instances.end(); ++i) {
+            (*i)->setFactory(0);
+            delete *i;
+        }
+    m_instances.clear();
+    unloadUnusedLibraries();
+}
+
+const std::vector<QString> &
+LADSPAPluginFactory::getPluginIdentifiers() const
+{
+    return m_identifiers;
+}
+
+void
+LADSPAPluginFactory::enumeratePlugins(MappedObjectPropertyList &list)
+{
+    for (std::vector<QString>::iterator i = m_identifiers.begin();
+            i != m_identifiers.end(); ++i) {
+
+        const LADSPA_Descriptor *descriptor = getLADSPADescriptor(*i);
+
+        if (!descriptor) {
+            std::cerr << "WARNING: LADSPAPluginFactory::enumeratePlugins: couldn't get descriptor for identifier " << *i << std::endl;
+            continue;
+        }
+
+//	std::cerr << "Enumerating plugin identifier " << *i << std::endl;
+
+        list.push_back(*i);
+        list.push_back(descriptor->Name);
+        list.push_back(QString("%1").arg(descriptor->UniqueID));
+        list.push_back(descriptor->Label);
+        list.push_back(descriptor->Maker);
+        list.push_back(descriptor->Copyright);
+        list.push_back("false"); // is synth
+        list.push_back("false"); // is grouped
+
+        if (m_taxonomy.find(descriptor->UniqueID) != m_taxonomy.end() &&
+                m_taxonomy[descriptor->UniqueID] != "") {
+//            		std::cerr << "LADSPAPluginFactory: cat for " << *i<< " found in taxonomy as " << m_taxonomy[descriptor->UniqueID] << std::endl;
+            list.push_back(m_taxonomy[descriptor->UniqueID]);
+
+        } else if (m_fallbackCategories.find(*i) !=
+                   m_fallbackCategories.end()) {
+            list.push_back(m_fallbackCategories[*i]);
+//            		std::cerr << "LADSPAPluginFactory: cat for " << *i  <<" found in fallbacks as " << m_fallbackCategories[*i] << std::endl;
+
+        } else {
+            list.push_back("");
+//            		std::cerr << "LADSPAPluginFactory: cat for " << *i << " not found (despite having " << m_fallbackCategories.size() << " fallbacks)" << std::endl;
+
+        }
+
+        list.push_back(QString("%1").arg(descriptor->PortCount));
+
+        for (unsigned long p = 0; p < descriptor->PortCount; ++p) {
+
+            int type = 0;
+            if (LADSPA_IS_PORT_CONTROL(descriptor->PortDescriptors[p])) {
+                type |= PluginPort::Control;
+            } else {
+                type |= PluginPort::Audio;
+            }
+            if (LADSPA_IS_PORT_INPUT(descriptor->PortDescriptors[p])) {
+                type |= PluginPort::Input;
+            } else {
+                type |= PluginPort::Output;
+            }
+
+            list.push_back(QString("%1").arg(p));
+            list.push_back(descriptor->PortNames[p]);
+            list.push_back(QString("%1").arg(type));
+            list.push_back(QString("%1").arg(getPortDisplayHint(descriptor, p)));
+            list.push_back(QString("%1").arg(getPortMinimum(descriptor, p)));
+            list.push_back(QString("%1").arg(getPortMaximum(descriptor, p)));
+            list.push_back(QString("%1").arg(getPortDefault(descriptor, p)));
+        }
+    }
+
+    unloadUnusedLibraries();
+}
+
+
+void
+LADSPAPluginFactory::populatePluginSlot(QString identifier, MappedPluginSlot &slot)
+{
+    const LADSPA_Descriptor *descriptor = getLADSPADescriptor(identifier);
+
+    if (descriptor) {
+
+        slot.setProperty(MappedPluginSlot::Label, descriptor->Label);
+        slot.setProperty(MappedPluginSlot::PluginName, descriptor->Name);
+        slot.setProperty(MappedPluginSlot::Author, descriptor->Maker);
+        slot.setProperty(MappedPluginSlot::Copyright, descriptor->Copyright);
+        slot.setProperty(MappedPluginSlot::PortCount, descriptor->PortCount);
+
+        if (m_taxonomy.find(descriptor->UniqueID) != m_taxonomy.end() &&
+                m_taxonomy[descriptor->UniqueID] != "") {
+            //		std::cerr << "LADSPAPluginFactory: cat for " << identifier<< " found in taxonomy as " << m_taxonomy[descriptor->UniqueID] << std::endl;
+            slot.setProperty(MappedPluginSlot::Category,
+                             m_taxonomy[descriptor->UniqueID]);
+
+        } else if (m_fallbackCategories.find(identifier) !=
+                   m_fallbackCategories.end()) {
+            //		std::cerr << "LADSPAPluginFactory: cat for " << identifier  <<" found in fallbacks as " << m_fallbackCategories[identifier] << std::endl;
+            slot.setProperty(MappedPluginSlot::Category,
+                             m_fallbackCategories[identifier]);
+
+        } else {
+            //		std::cerr << "LADSPAPluginFactory: cat for " << identifier << " not found (despite having " << m_fallbackCategories.size() << " fallbacks)" << std::endl;
+            slot.setProperty(MappedPluginSlot::Category, "");
+        }
+
+        slot.destroyChildren();
+
+        for (unsigned long i = 0; i < descriptor->PortCount; i++) {
+
+            if (LADSPA_IS_PORT_CONTROL(descriptor->PortDescriptors[i]) &&
+                    LADSPA_IS_PORT_INPUT(descriptor->PortDescriptors[i])) {
+
+                MappedStudio *studio = dynamic_cast<MappedStudio *>(slot.getParent());
+                if (!studio) {
+                    std::cerr << "WARNING: LADSPAPluginFactory::populatePluginSlot: can't find studio" << std::endl;
+                    return ;
+                }
+
+                MappedPluginPort *port =
+                    dynamic_cast<MappedPluginPort *>
+                    (studio->createObject(MappedObject::PluginPort));
+
+                slot.addChild(port);
+                port->setParent(&slot);
+
+                port->setProperty(MappedPluginPort::PortNumber, i);
+                port->setProperty(MappedPluginPort::Name,
+                                  descriptor->PortNames[i]);
+                port->setProperty(MappedPluginPort::Maximum,
+                                  getPortMaximum(descriptor, i));
+                port->setProperty(MappedPluginPort::Minimum,
+                                  getPortMinimum(descriptor, i));
+                port->setProperty(MappedPluginPort::Default,
+                                  getPortDefault(descriptor, i));
+                port->setProperty(MappedPluginPort::DisplayHint,
+                                  getPortDisplayHint(descriptor, i));
+            }
+        }
+    }
+
+    //!!! leak here if the plugin is not instantiated too...?
+}
+
+MappedObjectValue
+LADSPAPluginFactory::getPortMinimum(const LADSPA_Descriptor *descriptor, int port)
+{
+    LADSPA_PortRangeHintDescriptor d =
+        descriptor->PortRangeHints[port].HintDescriptor;
+
+    MappedObjectValue minimum = 0.0;
+
+    if (LADSPA_IS_HINT_BOUNDED_BELOW(d)) {
+        MappedObjectValue lb = descriptor->PortRangeHints[port].LowerBound;
+        minimum = lb;
+    } else if (LADSPA_IS_HINT_BOUNDED_ABOVE(d)) {
+        MappedObjectValue ub = descriptor->PortRangeHints[port].UpperBound;
+        minimum = std::min(0.f, ub - 1.f);
+    }
+
+    if (LADSPA_IS_HINT_SAMPLE_RATE(d)) {
+        minimum *= m_sampleRate;
+    }
+
+    if (LADSPA_IS_HINT_LOGARITHMIC(d)) {
+        if (minimum == 0.f) minimum = 1.f;
+    }
+
+    return minimum;
+}
+
+MappedObjectValue
+LADSPAPluginFactory::getPortMaximum(const LADSPA_Descriptor *descriptor, int port)
+{
+    LADSPA_PortRangeHintDescriptor d =
+        descriptor->PortRangeHints[port].HintDescriptor;
+
+    MappedObjectValue maximum = 1.0;
+
+//    std::cerr << "LADSPAPluginFactory::getPortMaximum(" << port << ")" << std::endl;
+//    std::cerr << "bounded above: " << LADSPA_IS_HINT_BOUNDED_ABOVE(d) << std::endl;
+
+    if (LADSPA_IS_HINT_BOUNDED_ABOVE(d)) {
+        MappedObjectValue ub = descriptor->PortRangeHints[port].UpperBound;
+        maximum = ub;
+    } else {
+        MappedObjectValue lb = descriptor->PortRangeHints[port].LowerBound;
+	if (LADSPA_IS_HINT_LOGARITHMIC(d)) {
+	    if (lb == 0.f) lb = 1.f;
+	    maximum = lb * 100.f;
+	} else {
+	    if (lb == 1.f) maximum = 10.f;
+	    else maximum = lb + 10;
+	}
+    }
+
+    if (LADSPA_IS_HINT_SAMPLE_RATE(d)) {
+//	std::cerr << "note: port has sample rate hint" << std::endl;
+        maximum *= m_sampleRate;
+    }
+
+//    std::cerr << "maximum: " << maximum << std::endl;
+//    if (LADSPA_IS_HINT_LOGARITHMIC(d)) {
+//	std::cerr << "note: port is logarithmic" << std::endl;
+//    }
+//    std::cerr << "note: minimum is reported as " << getPortMinimum(descriptor, port) << " (from bounded = " << LADSPA_IS_HINT_BOUNDED_BELOW(d) << ", bound = " << descriptor->PortRangeHints[port].LowerBound << ")" << std::endl;
+
+    return maximum;
+}
+
+MappedObjectValue
+LADSPAPluginFactory::getPortDefault(const LADSPA_Descriptor *descriptor, int port)
+{
+    MappedObjectValue minimum = getPortMinimum(descriptor, port);
+    MappedObjectValue maximum = getPortMaximum(descriptor, port);
+    MappedObjectValue deft;
+
+    if (m_portDefaults.find(descriptor->UniqueID) !=
+            m_portDefaults.end()) {
+        if (m_portDefaults[descriptor->UniqueID].find(port) !=
+	    m_portDefaults[descriptor->UniqueID].end()) {
+
+            deft = m_portDefaults[descriptor->UniqueID][port];
+            if (deft < minimum) deft = minimum;
+            if (deft > maximum) deft = maximum;
+//	    std::cerr << "port " << port << ": default " << deft << " from defaults" << std::endl;
+            return deft;
+        }
+    }
+
+    LADSPA_PortRangeHintDescriptor d =
+        descriptor->PortRangeHints[port].HintDescriptor;
+
+    bool logarithmic = LADSPA_IS_HINT_LOGARITHMIC(d);
+
+    float logmin = 0, logmax = 0;
+    if (logarithmic) {
+        float thresh = powf(10, -10);
+        if (minimum < thresh) logmin = -10;
+        else logmin = log10f(minimum);
+        if (maximum < thresh) logmax = -10;
+        else logmax = log10f(maximum);
+    }
+
+    if (!LADSPA_IS_HINT_HAS_DEFAULT(d)) {
+
+        deft = minimum;
+
+    } else if (LADSPA_IS_HINT_DEFAULT_MINIMUM(d)) {
+
+	// See comment for DEFAULT_MAXIMUM below
+	if (!LADSPA_IS_HINT_BOUNDED_BELOW(d)) {
+	    deft = descriptor->PortRangeHints[port].LowerBound;
+	    if (LADSPA_IS_HINT_SAMPLE_RATE(d)) {
+		deft *= m_sampleRate;
+	    }
+//	    std::cerr << "default-minimum: " << deft << std::endl;
+	    if (deft < minimum || deft > maximum) deft = minimum;
+//	    std::cerr << "default-minimum: " << deft << std::endl;
+	} else {
+	    deft = minimum;
+	}
+
+    } else if (LADSPA_IS_HINT_DEFAULT_LOW(d)) {
+	
+	if (logarithmic) {
+	    deft = powf(10, logmin * 0.75 + logmax * 0.25);
+	} else {
+	    deft = minimum * 0.75 + maximum * 0.25;
+	}
+	
+    } else if (LADSPA_IS_HINT_DEFAULT_MIDDLE(d)) {
+	
+	if (logarithmic) {
+	    deft = powf(10, logmin * 0.5 + logmax * 0.5);
+	} else {
+	    deft = minimum * 0.5 + maximum * 0.5;
+	}
+	
+    } else if (LADSPA_IS_HINT_DEFAULT_HIGH(d)) {
+	
+	if (logarithmic) {
+	    deft = powf(10, logmin * 0.25 + logmax * 0.75);
+	} else {
+	    deft = minimum * 0.25 + maximum * 0.75;
+	}
+
+    } else if (LADSPA_IS_HINT_DEFAULT_MAXIMUM(d)) {
+
+	// CMT plugins employ this grossness (setting DEFAULT_MAXIMUM
+	// without BOUNDED_ABOVE and then using the UPPER_BOUND as the
+	// port default)
+	if (!LADSPA_IS_HINT_BOUNDED_ABOVE(d)) {
+	    deft = descriptor->PortRangeHints[port].UpperBound;
+	    if (LADSPA_IS_HINT_SAMPLE_RATE(d)) {
+		deft *= m_sampleRate;
+	    }
+//	    std::cerr << "default-maximum: " << deft << std::endl;
+	    if (deft < minimum || deft > maximum) deft = maximum;
+//	    std::cerr << "default-maximum: " << deft << std::endl;
+	} else {
+	    deft = maximum;
+	}
+
+    } else if (LADSPA_IS_HINT_DEFAULT_0(d)) {
+
+        deft = 0.0;
+
+    } else if (LADSPA_IS_HINT_DEFAULT_1(d)) {
+
+        deft = 1.0;
+
+    } else if (LADSPA_IS_HINT_DEFAULT_100(d)) {
+
+        deft = 100.0;
+
+    } else if (LADSPA_IS_HINT_DEFAULT_440(d)) {
+
+        deft = 440.0;
+
+    } else {
+
+        deft = minimum;
+    }
+
+//    std::cerr << "port " << port << " default = "<< deft << std::endl;
+
+    return deft;
+}
+
+int
+LADSPAPluginFactory::getPortDisplayHint(const LADSPA_Descriptor *descriptor, int port)
+{
+    LADSPA_PortRangeHintDescriptor d =
+        descriptor->PortRangeHints[port].HintDescriptor;
+    int hint = PluginPort::NoHint;
+
+    if (LADSPA_IS_HINT_TOGGLED(d))
+        hint |= PluginPort::Toggled;
+    if (LADSPA_IS_HINT_INTEGER(d))
+        hint |= PluginPort::Integer;
+    if (LADSPA_IS_HINT_LOGARITHMIC(d))
+        hint |= PluginPort::Logarithmic;
+
+    return hint;
+}
+
+
+RunnablePluginInstance *
+LADSPAPluginFactory::instantiatePlugin(QString identifier,
+                                       int instrument,
+                                       int position,
+                                       unsigned int sampleRate,
+                                       unsigned int blockSize,
+                                       unsigned int channels)
+{
+    const LADSPA_Descriptor *descriptor = getLADSPADescriptor(identifier);
+
+    if (descriptor) {
+
+        LADSPAPluginInstance *instance =
+            new LADSPAPluginInstance
+            (this, instrument, identifier, position, sampleRate, blockSize, channels,
+             descriptor);
+
+        m_instances.insert(instance);
+
+        return instance;
+    }
+
+    return 0;
+}
+
+void
+LADSPAPluginFactory::releasePlugin(RunnablePluginInstance *instance,
+                                   QString identifier)
+{
+    if (m_instances.find(instance) == m_instances.end()) {
+        std::cerr << "WARNING: LADSPAPluginFactory::releasePlugin: Not one of mine!"
+        << std::endl;
+        return ;
+    }
+
+    QString type, soname, label;
+    PluginIdentifier::parseIdentifier(identifier, type, soname, label);
+
+    m_instances.erase(m_instances.find(instance));
+
+    bool stillInUse = false;
+
+    for (std::set
+                <RunnablePluginInstance *>::iterator ii = m_instances.begin();
+                ii != m_instances.end(); ++ii) {
+            QString itype, isoname, ilabel;
+            PluginIdentifier::parseIdentifier((*ii)->getIdentifier(), itype, isoname, ilabel);
+            if (isoname == soname) {
+                //	    std::cerr << "LADSPAPluginFactory::releasePlugin: dll " << soname << " is still in use for plugin " << ilabel << std::endl;
+                stillInUse = true;
+                break;
+            }
+        }
+
+    if (!stillInUse) {
+        //	std::cerr << "LADSPAPluginFactory::releasePlugin: dll " << soname << " no longer in use, unloading" << std::endl;
+        unloadLibrary(soname);
+    }
+}
+
+const LADSPA_Descriptor *
+LADSPAPluginFactory::getLADSPADescriptor(QString identifier)
+{
+    QString type, soname, label;
+    PluginIdentifier::parseIdentifier(identifier, type, soname, label);
+
+    if (m_libraryHandles.find(soname) == m_libraryHandles.end()) {
+        loadLibrary(soname);
+        if (m_libraryHandles.find(soname) == m_libraryHandles.end()) {
+            std::cerr << "WARNING: LADSPAPluginFactory::getLADSPADescriptor: loadLibrary failed for " << soname << std::endl;
+            return 0;
+        }
+    }
+
+    void *libraryHandle = m_libraryHandles[soname];
+
+    LADSPA_Descriptor_Function fn = (LADSPA_Descriptor_Function)
+                                    dlsym(libraryHandle, "ladspa_descriptor");
+
+    if (!fn) {
+        std::cerr << "WARNING: LADSPAPluginFactory::getLADSPADescriptor: No descriptor function in library " << soname << std::endl;
+        return 0;
+    }
+
+    const LADSPA_Descriptor *descriptor = 0;
+
+    int index = 0;
+    while ((descriptor = fn(index))) {
+        if (descriptor->Label == label)
+            return descriptor;
+        ++index;
+    }
+
+    std::cerr << "WARNING: LADSPAPluginFactory::getLADSPADescriptor: No such plugin as " << label << " in library " << soname << std::endl;
+
+    return 0;
+}
+
+void
+LADSPAPluginFactory::loadLibrary(QString soName)
+{
+    void *libraryHandle = dlopen(soName.data(), RTLD_NOW);
+    if (libraryHandle)
+        m_libraryHandles[soName] = libraryHandle;
+}
+
+void
+LADSPAPluginFactory::unloadLibrary(QString soName)
+{
+    LibraryHandleMap::iterator li = m_libraryHandles.find(soName);
+    if (li != m_libraryHandles.end()) {
+        //	std::cerr << "unloading " << soName << std::endl;
+        dlclose(m_libraryHandles[soName]);
+        m_libraryHandles.erase(li);
+    }
+}
+
+void
+LADSPAPluginFactory::unloadUnusedLibraries()
+{
+    std::vector<QString> toUnload;
+
+    for (LibraryHandleMap::iterator i = m_libraryHandles.begin();
+            i != m_libraryHandles.end(); ++i) {
+
+        bool stillInUse = false;
+
+        for (std::set
+                    <RunnablePluginInstance *>::iterator ii = m_instances.begin();
+                    ii != m_instances.end(); ++ii) {
+
+                QString itype, isoname, ilabel;
+                PluginIdentifier::parseIdentifier((*ii)->getIdentifier(), itype, isoname, ilabel);
+                if (isoname == i->first) {
+                    stillInUse = true;
+                    break;
+                }
+            }
+
+        if (!stillInUse)
+            toUnload.push_back(i->first);
+    }
+
+    for (std::vector<QString>::iterator i = toUnload.begin();
+            i != toUnload.end(); ++i) {
+        unloadLibrary(*i);
+    }
+}
+
+
+// It is only later, after they've gone,
+// I realize they have delivered a letter.
+// It's a letter from my wife.  "What are you doing
+// there?" my wife asks.  "Are you drinking?"
+// I study the postmark for hours.  Then it, too, begins to fade.
+// I hope someday to forget all this.
+
+
+std::vector<QString>
+LADSPAPluginFactory::getPluginPath()
+{
+    std::vector<QString> pathList;
+    std::string path;
+
+    char *cpath = getenv("LADSPA_PATH");
+    if (cpath)
+        path = cpath;
+
+    if (path == "") {
+        path = "/usr/local/lib/ladspa:/usr/lib/ladspa";
+        char *home = getenv("HOME");
+        if (home)
+            path = std::string(home) + "/.ladspa:" + path;
+    }
+
+    std::string::size_type index = 0, newindex = 0;
+
+    while ((newindex = path.find(':', index)) < path.size()) {
+        pathList.push_back(path.substr(index, newindex - index).c_str());
+        index = newindex + 1;
+    }
+
+    pathList.push_back(path.substr(index).c_str());
+
+    return pathList;
+}
+
+
+#ifdef HAVE_LIBLRDF
+std::vector<QString>
+LADSPAPluginFactory::getLRDFPath(QString &baseUri)
+{
+    std::vector<QString> pathList = getPluginPath();
+    std::vector<QString> lrdfPaths;
+
+    lrdfPaths.push_back("/usr/local/share/ladspa/rdf");
+    lrdfPaths.push_back("/usr/share/ladspa/rdf");
+
+    for (std::vector<QString>::iterator i = pathList.begin();
+            i != pathList.end(); ++i) {
+        lrdfPaths.push_back(*i + "/rdf");
+    }
+
+    baseUri = LADSPA_BASE;
+    return lrdfPaths;
+}
+#endif
+
+void
+LADSPAPluginFactory::discoverPlugins()
+{
+    std::vector<QString> pathList = getPluginPath();
+
+    std::cerr << "LADSPAPluginFactory::discoverPlugins - "
+    	      << "discovering plugins; path is ";
+    for (std::vector<QString>::iterator i = pathList.begin();
+            i != pathList.end(); ++i) {
+        std::cerr << "[" << *i << "] ";
+    }
+    std::cerr << std::endl;
+
+//    std::cerr << "LADSPAPluginFactory::discoverPlugins - "
+//    	      << "trace is ";
+//    std::cerr << kdBacktrace() << std::endl;
+
+#ifdef HAVE_LIBLRDF
+    // Initialise liblrdf and read the description files
+    //
+    lrdf_init();
+
+    QString baseUri;
+    std::vector<QString> lrdfPaths = getLRDFPath(baseUri);
+
+    bool haveSomething = false;
+
+    for (size_t i = 0; i < lrdfPaths.size(); ++i) {
+        QDir dir(lrdfPaths[i], "*.rdf;*.rdfs");
+        for (unsigned int j = 0; j < dir.count(); ++j) {
+            if (!lrdf_read_file(QString("file:" + lrdfPaths[i] + "/" + dir[j]).data())) {
+                //		std::cerr << "LADSPAPluginFactory: read RDF file " << (lrdfPaths[i] + "/" + dir[j]) << std::endl;
+                haveSomething = true;
+            }
+        }
+    }
+
+    if (haveSomething) {
+        generateTaxonomy(baseUri + "Plugin", "");
+    }
+#endif // HAVE_LIBLRDF
+
+    generateFallbackCategories();
+
+    for (std::vector<QString>::iterator i = pathList.begin();
+            i != pathList.end(); ++i) {
+
+        QDir pluginDir(*i, "*.so");
+
+        for (unsigned int j = 0; j < pluginDir.count(); ++j) {
+            discoverPlugins(QString("%1/%2").arg(*i).arg(pluginDir[j]));
+        }
+    }
+
+#ifdef HAVE_LIBLRDF
+    // Cleanup after the RDF library
+    //
+    lrdf_cleanup();
+#endif // HAVE_LIBLRDF
+
+    std::cerr << "LADSPAPluginFactory::discoverPlugins - done" << std::endl;
+}
+
+void
+LADSPAPluginFactory::discoverPlugins(QString soName)
+{
+    void *libraryHandle = dlopen(soName.data(), RTLD_LAZY);
+
+    if (!libraryHandle) {
+        std::cerr << "WARNING: LADSPAPluginFactory::discoverPlugins: couldn't dlopen "
+        << soName << " - " << dlerror() << std::endl;
+        return ;
+    }
+
+    LADSPA_Descriptor_Function fn = (LADSPA_Descriptor_Function)
+                                    dlsym(libraryHandle, "ladspa_descriptor");
+
+    if (!fn) {
+        std::cerr << "WARNING: LADSPAPluginFactory::discoverPlugins: No descriptor function in " << soName << std::endl;
+        return ;
+    }
+
+    const LADSPA_Descriptor *descriptor = 0;
+
+    int index = 0;
+    while ((descriptor = fn(index))) {
+
+#ifdef HAVE_LIBLRDF
+        char * def_uri = 0;
+        lrdf_defaults *defs = 0;
+
+        QString category = m_taxonomy[descriptor->UniqueID];
+
+        if (category == "" && descriptor->Name != 0) {
+            std::string name = descriptor->Name;
+            if (name.length() > 4 &&
+                    name.substr(name.length() - 4) == " VST") {
+                category = "VST effects";
+                m_taxonomy[descriptor->UniqueID] = category;
+            }
+        }
+
+//        	std::cerr << "Plugin id is " << descriptor->UniqueID
+//        		  << ", category is \"" << (category ? category : QString("(none)"))
+//        		  << "\", name is " << descriptor->Name
+//        		  << ", label is " << descriptor->Label
+//        		  << std::endl;
+
+        def_uri = lrdf_get_default_uri(descriptor->UniqueID);
+        if (def_uri) {
+            defs = lrdf_get_setting_values(def_uri);
+        }
+
+        int controlPortNumber = 1;
+
+        for (unsigned long i = 0; i < descriptor->PortCount; i++) {
+
+            if (LADSPA_IS_PORT_CONTROL(descriptor->PortDescriptors[i])) {
+
+                if (def_uri && defs) {
+
+                    for (int j = 0; j < defs->count; j++) {
+                        if (defs->items[j].pid == controlPortNumber) {
+                            //			    std::cerr << "Default for this port (" << defs->items[j].pid << ", " << defs->items[j].label << ") is " << defs->items[j].value << "; applying this to port number " << i << " with name " << descriptor->PortNames[i] << std::endl;
+                            m_portDefaults[descriptor->UniqueID][i] =
+                                defs->items[j].value;
+                        }
+                    }
+                }
+
+                ++controlPortNumber;
+            }
+        }
+#endif // HAVE_LIBLRDF
+
+        QString identifier = PluginIdentifier::createIdentifier
+                             ("ladspa", soName, descriptor->Label);
+//	std::cerr << "Added plugin identifier " << identifier << std::endl;
+        m_identifiers.push_back(identifier);
+
+        ++index;
+    }
+
+    if (dlclose(libraryHandle) != 0) {
+        std::cerr << "WARNING: LADSPAPluginFactory::discoverPlugins - can't unload " << libraryHandle << std::endl;
+        return ;
+    }
+}
+
+void
+LADSPAPluginFactory::generateFallbackCategories()
+{
+    std::vector<QString> pluginPath = getPluginPath();
+    std::vector<QString> path;
+
+    for (size_t i = 0; i < pluginPath.size(); ++i) {
+        if (pluginPath[i].contains("/lib/")) {
+            QString p(pluginPath[i]);
+            p.replace("/lib/", "/share/");
+            path.push_back(p);
+            //	    std::cerr << "LADSPAPluginFactory::generateFallbackCategories: path element " << p << std::endl;
+        }
+        path.push_back(pluginPath[i]);
+        //	std::cerr << "LADSPAPluginFactory::generateFallbackCategories: path element " << pluginPath[i] << std::endl;
+    }
+
+    for (size_t i = 0; i < path.size(); ++i) {
+
+        QDir dir(path[i], "*.cat");
+
+//        	std::cerr << "LADSPAPluginFactory::generateFallbackCategories: directory " << path[i] << " has " << dir.count() << " .cat files" << std::endl;
+        for (unsigned int j = 0; j < dir.count(); ++j) {
+
+            QFile file(path[i] + "/" + dir[j]);
+
+            //	    std::cerr << "LADSPAPluginFactory::generateFallbackCategories: about to open " << (path[i] + "/" + dir[j]) << std::endl;
+
+            if (file.open(IO_ReadOnly)) {
+                //		    std::cerr << "...opened" << std::endl;
+                QTextStream stream(&file);
+                QString line;
+
+                while (!stream.eof()) {
+                    line = stream.readLine();
+                    //		    std::cerr << "line is: \"" << line << "\"" << std::endl;
+                    QString id = line.section("::", 0, 0);
+                    QString cat = line.section("::", 1, 1);
+                    m_fallbackCategories[id] = cat;
+//                    		    std::cerr << "set id \"" << id << "\" to cat \"" << cat << "\"" << std::endl;
+                }
+            }
+        }
+    }
+}
+
+void
+LADSPAPluginFactory::generateTaxonomy(QString uri, QString base)
+{
+#ifdef HAVE_LIBLRDF
+    lrdf_uris *uris = lrdf_get_instances(uri.data());
+
+    if (uris != NULL) {
+        for (int i = 0; i < uris->count; ++i) {
+            m_taxonomy[lrdf_get_uid(uris->items[i])] = base;
+        }
+        lrdf_free_uris(uris);
+    }
+
+    uris = lrdf_get_subclasses(uri.data());
+
+    if (uris != NULL) {
+        for (int i = 0; i < uris->count; ++i) {
+            char *label = lrdf_get_label(uris->items[i]);
+            generateTaxonomy(uris->items[i],
+                             base + (base.length() > 0 ? " > " : "") + label);
+        }
+        lrdf_free_uris(uris);
+    }
+#endif
+}
+
+}
+
+#endif // HAVE_LADSPA
+
diff --git a/src/sound/LADSPAPluginFactory.h b/src/sound/LADSPAPluginFactory.h
new file mode 100644
index 0000000..a5ec368
--- /dev/null
+++ b/src/sound/LADSPAPluginFactory.h
@@ -0,0 +1,104 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _LADSPA_PLUGIN_FACTORY_H_
+#define _LADSPA_PLUGIN_FACTORY_H_
+
+#ifdef HAVE_LADSPA
+
+#include "PluginFactory.h"
+#include <ladspa.h>
+
+#include <vector>
+#include <map>
+#include <set>
+#include <qstring.h>
+
+namespace Rosegarden
+{
+
+class LADSPAPluginInstance;
+
+class LADSPAPluginFactory : public PluginFactory
+{
+public:
+    virtual ~LADSPAPluginFactory();
+
+    virtual void discoverPlugins();
+
+    virtual const std::vector<QString> &getPluginIdentifiers() const;
+
+    virtual void enumeratePlugins(MappedObjectPropertyList &list);
+
+    virtual void populatePluginSlot(QString identifier, MappedPluginSlot &slot);
+
+    virtual RunnablePluginInstance *instantiatePlugin(QString identifier,
+                                                      int instrumentId,
+                                                      int position,
+                                                      unsigned int sampleRate,
+                                                      unsigned int blockSize,
+                                                      unsigned int channels);
+
+    MappedObjectValue getPortMinimum(const LADSPA_Descriptor *, int port);
+    MappedObjectValue getPortMaximum(const LADSPA_Descriptor *, int port);
+    MappedObjectValue getPortDefault(const LADSPA_Descriptor *, int port);
+    int getPortDisplayHint(const LADSPA_Descriptor *, int port);
+
+protected:
+    LADSPAPluginFactory();
+    friend class PluginFactory;
+
+    virtual std::vector<QString> getPluginPath();
+
+#ifdef HAVE_LIBLRDF
+    virtual std::vector<QString> getLRDFPath(QString &baseUri);
+#endif
+
+    virtual void discoverPlugins(QString soName);
+    virtual void generateTaxonomy(QString uri, QString base);
+    virtual void generateFallbackCategories();
+
+    virtual void releasePlugin(RunnablePluginInstance *, QString);
+
+    virtual const LADSPA_Descriptor *getLADSPADescriptor(QString identifier);
+
+    void loadLibrary(QString soName);
+    void unloadLibrary(QString soName);
+    void unloadUnusedLibraries();
+
+    std::vector<QString> m_identifiers;
+
+    std::map<unsigned long, QString> m_taxonomy;
+    std::map<QString, QString> m_fallbackCategories;
+    std::map<unsigned long, std::map<int, float> > m_portDefaults;
+
+    std::set<RunnablePluginInstance *> m_instances;
+
+    typedef std::map<QString, void *> LibraryHandleMap;
+    LibraryHandleMap m_libraryHandles;
+};
+
+}
+
+#endif
+
+#endif
+
diff --git a/src/sound/LADSPAPluginInstance.cpp b/src/sound/LADSPAPluginInstance.cpp
new file mode 100644
index 0000000..e2b8890
--- /dev/null
+++ b/src/sound/LADSPAPluginInstance.cpp
@@ -0,0 +1,435 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+ 
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+ 
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+ 
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include <iostream>
+#include <cassert>
+
+#include "LADSPAPluginInstance.h"
+#include "LADSPAPluginFactory.h"
+
+#ifdef HAVE_LADSPA
+
+//#define DEBUG_LADSPA 1
+
+namespace Rosegarden
+{
+
+
+LADSPAPluginInstance::LADSPAPluginInstance(PluginFactory *factory,
+        InstrumentId instrument,
+        QString identifier,
+        int position,
+        unsigned long sampleRate,
+        size_t blockSize,
+        int idealChannelCount,
+        const LADSPA_Descriptor* descriptor) :
+        RunnablePluginInstance(factory, identifier),
+        m_instrument(instrument),
+        m_position(position),
+        m_instanceCount(0),
+        m_descriptor(descriptor),
+        m_blockSize(blockSize),
+        m_sampleRate(sampleRate),
+        m_latencyPort(0),
+        m_run(false),
+        m_bypassed(false)
+{
+    init(idealChannelCount);
+
+    m_inputBuffers = new sample_t * [m_instanceCount * m_audioPortsIn.size()];
+    m_outputBuffers = new sample_t * [m_instanceCount * m_audioPortsOut.size()];
+
+    for (size_t i = 0; i < m_instanceCount * m_audioPortsIn.size(); ++i) {
+        m_inputBuffers[i] = new sample_t[blockSize];
+    }
+    for (size_t i = 0; i < m_instanceCount * m_audioPortsOut.size(); ++i) {
+        m_outputBuffers[i] = new sample_t[blockSize];
+    }
+
+    m_ownBuffers = true;
+
+    instantiate(sampleRate);
+    if (isOK()) {
+        connectPorts();
+        activate();
+    }
+}
+
+LADSPAPluginInstance::LADSPAPluginInstance(PluginFactory *factory,
+        InstrumentId instrument,
+        QString identifier,
+        int position,
+        unsigned long sampleRate,
+        size_t blockSize,
+        sample_t **inputBuffers,
+        sample_t **outputBuffers,
+        const LADSPA_Descriptor* descriptor) :
+        RunnablePluginInstance(factory, identifier),
+        m_instrument(instrument),
+        m_position(position),
+        m_instanceCount(0),
+        m_descriptor(descriptor),
+        m_blockSize(blockSize),
+        m_inputBuffers(inputBuffers),
+        m_outputBuffers(outputBuffers),
+        m_ownBuffers(false),
+        m_sampleRate(sampleRate),
+        m_latencyPort(0),
+        m_run(false),
+        m_bypassed(false)
+{
+    init();
+
+    instantiate(sampleRate);
+    if (isOK()) {
+        connectPorts();
+        activate();
+    }
+}
+
+
+void
+LADSPAPluginInstance::init(int idealChannelCount)
+{
+#ifdef DEBUG_LADSPA
+    std::cerr << "LADSPAPluginInstance::init(" << idealChannelCount << "): plugin has "
+    << m_descriptor->PortCount << " ports" << std::endl;
+#endif
+
+    // Discover ports numbers and identities
+    //
+    for (unsigned long i = 0; i < m_descriptor->PortCount; ++i) {
+        if (LADSPA_IS_PORT_AUDIO(m_descriptor->PortDescriptors[i])) {
+            if (LADSPA_IS_PORT_INPUT(m_descriptor->PortDescriptors[i])) {
+#ifdef DEBUG_LADSPA
+                std::cerr << "LADSPAPluginInstance::init: port " << i << " is audio in" << std::endl;
+#endif
+
+                m_audioPortsIn.push_back(i);
+            } else {
+#ifdef DEBUG_LADSPA
+                std::cerr << "LADSPAPluginInstance::init: port " << i << " is audio out" << std::endl;
+#endif
+
+                m_audioPortsOut.push_back(i);
+            }
+        } else
+            if (LADSPA_IS_PORT_CONTROL(m_descriptor->PortDescriptors[i])) {
+                if (LADSPA_IS_PORT_INPUT(m_descriptor->PortDescriptors[i])) {
+#ifdef DEBUG_LADSPA
+                    std::cerr << "LADSPAPluginInstance::init: port " << i << " is control in" << std::endl;
+#endif
+
+                    LADSPA_Data *data = new LADSPA_Data(0.0);
+                    m_controlPortsIn.push_back(
+                        std::pair<unsigned long, LADSPA_Data*>(i, data));
+                } else {
+#ifdef DEBUG_LADSPA
+                    std::cerr << "LADSPAPluginInstance::init: port " << i << " is control out" << std::endl;
+#endif
+
+                    LADSPA_Data *data = new LADSPA_Data(0.0);
+                    m_controlPortsOut.push_back(
+                        std::pair<unsigned long, LADSPA_Data*>(i, data));
+                    if (!strcmp(m_descriptor->PortNames[i], "latency") ||
+                            !strcmp(m_descriptor->PortNames[i], "_latency")) {
+#ifdef DEBUG_LADSPA
+                        std::cerr << "Wooo! We have a latency port!" << std::endl;
+#endif
+
+                        m_latencyPort = data;
+                    }
+                }
+            }
+#ifdef DEBUG_LADSPA
+            else
+                std::cerr << "LADSPAPluginInstance::init - "
+                << "unrecognised port type" << std::endl;
+#endif
+
+    }
+
+    m_instanceCount = 1;
+
+    if (idealChannelCount > 0) {
+        if (m_audioPortsIn.size() == 1) {
+            // mono plugin: duplicate it if need be
+            m_instanceCount = idealChannelCount;
+        }
+    }
+}
+
+size_t
+LADSPAPluginInstance::getLatency()
+{
+    if (m_latencyPort) {
+        if (!m_run) {
+            for (int i = 0; i < getAudioInputCount(); ++i) {
+                for (int j = 0; j < m_blockSize; ++j) {
+                    m_inputBuffers[i][j] = 0.f;
+                }
+            }
+            run(RealTime::zeroTime);
+	}
+        return *m_latencyPort;
+    }
+    return 0;
+}
+
+void
+LADSPAPluginInstance::silence()
+{
+    if (isOK()) {
+        deactivate();
+        activate();
+    }
+}
+
+void
+LADSPAPluginInstance::setIdealChannelCount(size_t channels)
+{
+    if (m_audioPortsIn.size() != 1 || channels == m_instanceCount) {
+        silence();
+        return ;
+    }
+
+    if (isOK()) {
+        deactivate();
+    }
+
+    //!!! don't we need to reallocate inputBuffers and outputBuffers?
+
+    cleanup();
+    m_instanceCount = channels;
+    instantiate(m_sampleRate);
+    if (isOK()) {
+        connectPorts();
+        activate();
+    }
+}
+
+
+LADSPAPluginInstance::~LADSPAPluginInstance()
+{
+#ifdef DEBUG_LADSPA
+    std::cerr << "LADSPAPluginInstance::~LADSPAPluginInstance" << std::endl;
+#endif
+
+    if (m_instanceHandles.size() != 0) { // "isOK()"
+        deactivate();
+    }
+
+    cleanup();
+
+    for (unsigned int i = 0; i < m_controlPortsIn.size(); ++i)
+        delete m_controlPortsIn[i].second;
+
+    for (unsigned int i = 0; i < m_controlPortsOut.size(); ++i)
+        delete m_controlPortsOut[i].second;
+
+    m_controlPortsIn.clear();
+    m_controlPortsOut.clear();
+
+    if (m_ownBuffers) {
+        for (size_t i = 0; i < m_audioPortsIn.size(); ++i) {
+            delete[] m_inputBuffers[i];
+        }
+        for (size_t i = 0; i < m_audioPortsOut.size(); ++i) {
+            delete[] m_outputBuffers[i];
+        }
+
+        delete[] m_inputBuffers;
+        delete[] m_outputBuffers;
+    }
+
+    m_audioPortsIn.clear();
+    m_audioPortsOut.clear();
+}
+
+
+void
+LADSPAPluginInstance::instantiate(unsigned long sampleRate)
+{
+#ifdef DEBUG_LADSPA
+    std::cout << "LADSPAPluginInstance::instantiate - plugin unique id = "
+    << m_descriptor->UniqueID << std::endl;
+#endif
+
+    if (!m_descriptor)
+        return ;
+
+    if (!m_descriptor->instantiate) {
+        std::cerr << "Bad plugin: plugin id " << m_descriptor->UniqueID
+        << ":" << m_descriptor->Label
+        << " has no instantiate method!" << std::endl;
+        return ;
+    }
+
+    for (int i = 0; i < m_instanceCount; ++i) {
+        m_instanceHandles.push_back
+        (m_descriptor->instantiate(m_descriptor, sampleRate));
+    }
+}
+
+void
+LADSPAPluginInstance::activate()
+{
+    if (!m_descriptor || !m_descriptor->activate)
+        return ;
+
+    for (std::vector<LADSPA_Handle>::iterator hi = m_instanceHandles.begin();
+            hi != m_instanceHandles.end(); ++hi) {
+        m_descriptor->activate(*hi);
+    }
+}
+
+void
+LADSPAPluginInstance::connectPorts()
+{
+    if (!m_descriptor || !m_descriptor->connect_port)
+        return ;
+
+    assert(sizeof(LADSPA_Data) == sizeof(float));
+    assert(sizeof(sample_t) == sizeof(float));
+
+    int inbuf = 0, outbuf = 0;
+
+    for (std::vector<LADSPA_Handle>::iterator hi = m_instanceHandles.begin();
+            hi != m_instanceHandles.end(); ++hi) {
+
+        for (unsigned int i = 0; i < m_audioPortsIn.size(); ++i) {
+            m_descriptor->connect_port(*hi,
+                                       m_audioPortsIn[i],
+                                       (LADSPA_Data *)m_inputBuffers[inbuf]);
+            ++inbuf;
+        }
+
+        for (unsigned int i = 0; i < m_audioPortsOut.size(); ++i) {
+            m_descriptor->connect_port(*hi,
+                                       m_audioPortsOut[i],
+                                       (LADSPA_Data *)m_outputBuffers[outbuf]);
+            ++outbuf;
+        }
+
+        // If there is more than one instance, they all share the same
+        // control port ins (and outs, for the moment, because we
+        // don't actually do anything with the outs anyway -- but they
+        // do have to be connected as the plugin can't know if they're
+        // not and will write to them anyway).
+
+        for (unsigned int i = 0; i < m_controlPortsIn.size(); ++i) {
+            m_descriptor->connect_port(*hi,
+                                       m_controlPortsIn[i].first,
+                                       m_controlPortsIn[i].second);
+        }
+
+        for (unsigned int i = 0; i < m_controlPortsOut.size(); ++i) {
+            m_descriptor->connect_port(*hi,
+                                       m_controlPortsOut[i].first,
+                                       m_controlPortsOut[i].second);
+        }
+    }
+}
+
+void
+LADSPAPluginInstance::setPortValue(unsigned int portNumber, float value)
+{
+    for (unsigned int i = 0; i < m_controlPortsIn.size(); ++i) {
+        if (m_controlPortsIn[i].first == portNumber) {
+            LADSPAPluginFactory *f = dynamic_cast<LADSPAPluginFactory *>(m_factory);
+            if (f) {
+                if (value < f->getPortMinimum(m_descriptor, portNumber)) {
+                    value = f->getPortMinimum(m_descriptor, portNumber);
+                }
+                if (value > f->getPortMaximum(m_descriptor, portNumber)) {
+                    value = f->getPortMaximum(m_descriptor, portNumber);
+                }
+            }
+            (*m_controlPortsIn[i].second) = value;
+        }
+    }
+}
+
+float
+LADSPAPluginInstance::getPortValue(unsigned int portNumber)
+{
+    for (unsigned int i = 0; i < m_controlPortsIn.size(); ++i) {
+        if (m_controlPortsIn[i].first == portNumber) {
+            return (*m_controlPortsIn[i].second);
+        }
+    }
+
+    return 0.0;
+}
+
+void
+LADSPAPluginInstance::run(const RealTime &)
+{
+    if (!m_descriptor || !m_descriptor->run)
+        return ;
+
+    for (std::vector<LADSPA_Handle>::iterator hi = m_instanceHandles.begin();
+            hi != m_instanceHandles.end(); ++hi) {
+        m_descriptor->run(*hi, m_blockSize);
+    }
+
+    m_run = true;
+}
+
+void
+LADSPAPluginInstance::deactivate()
+{
+    if (!m_descriptor || !m_descriptor->deactivate)
+        return ;
+
+    for (std::vector<LADSPA_Handle>::iterator hi = m_instanceHandles.begin();
+            hi != m_instanceHandles.end(); ++hi) {
+        m_descriptor->deactivate(*hi);
+    }
+}
+
+void
+LADSPAPluginInstance::cleanup()
+{
+    if (!m_descriptor)
+        return ;
+
+    if (!m_descriptor->cleanup) {
+        std::cerr << "Bad plugin: plugin id " << m_descriptor->UniqueID
+        << ":" << m_descriptor->Label
+        << " has no cleanup method!" << std::endl;
+        return ;
+    }
+
+    for (std::vector<LADSPA_Handle>::iterator hi = m_instanceHandles.begin();
+            hi != m_instanceHandles.end(); ++hi) {
+        m_descriptor->cleanup(*hi);
+    }
+
+    m_instanceHandles.clear();
+}
+
+
+
+}
+
+#endif // HAVE_LADSPA
+
+
diff --git a/src/sound/LADSPAPluginInstance.h b/src/sound/LADSPAPluginInstance.h
new file mode 100644
index 0000000..9654cfb
--- /dev/null
+++ b/src/sound/LADSPAPluginInstance.h
@@ -0,0 +1,137 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include <vector>
+#include <set>
+#include <qstring.h>
+#include "Instrument.h"
+
+#ifndef _LADSPAPLUGININSTANCE_H_
+#define _LADSPAPLUGININSTANCE_H_
+
+#ifdef HAVE_LADSPA
+
+#include <ladspa.h>
+#include "RunnablePluginInstance.h"
+
+namespace Rosegarden
+{
+
+// LADSPA plugin instance.  LADSPA is a variable block size API, but
+// for one reason and another it's more convenient to use a fixed
+// block size in this wrapper.
+//
+class LADSPAPluginInstance : public RunnablePluginInstance
+{
+public:
+    virtual ~LADSPAPluginInstance();
+
+    virtual bool isOK() const { return m_instanceHandles.size() != 0; }
+
+    InstrumentId getInstrument() const { return m_instrument; }
+    virtual QString getIdentifier() const { return m_identifier; }
+    int getPosition() const { return m_position; }
+
+    virtual void run(const RealTime &rt);
+
+    virtual void setPortValue(unsigned int portNumber, float value);
+    virtual float getPortValue(unsigned int portNumber);
+
+    virtual size_t getBufferSize() { return m_blockSize; }
+    virtual size_t getAudioInputCount() { return m_instanceCount * m_audioPortsIn.size(); }
+    virtual size_t getAudioOutputCount() { return m_instanceCount * m_audioPortsOut.size(); }
+    virtual sample_t **getAudioInputBuffers() { return m_inputBuffers; }
+    virtual sample_t **getAudioOutputBuffers() { return m_outputBuffers; }
+
+    virtual bool isBypassed() const { return m_bypassed; }
+    virtual void setBypassed(bool bypassed) { m_bypassed = bypassed; }
+
+    virtual size_t getLatency();
+
+    virtual void silence();
+    virtual void setIdealChannelCount(size_t channels); // may re-instantiate
+
+protected:
+    // To be constructed only by LADSPAPluginFactory
+    friend class LADSPAPluginFactory;
+
+    // Constructor that creates the buffers internally
+    // 
+    LADSPAPluginInstance(PluginFactory *factory,
+                         InstrumentId instrument,
+                         QString identifier,
+                         int position,
+                         unsigned long sampleRate,
+                         size_t blockSize,
+                         int idealChannelCount,
+                         const LADSPA_Descriptor* descriptor);
+
+    // Constructor that uses shared buffers
+    // 
+    LADSPAPluginInstance(PluginFactory *factory,
+                         InstrumentId instrument,
+                         QString identifier,
+                         int position,
+                         unsigned long sampleRate,
+                         size_t blockSize,
+                         sample_t **inputBuffers,
+                         sample_t **outputBuffers,
+                         const LADSPA_Descriptor* descriptor);
+
+    void init(int idealChannelCount = 0);
+    void instantiate(unsigned long sampleRate);
+    void cleanup();
+    void activate();
+    void deactivate();
+
+    // Connection of data (and behind the scenes control) ports
+    //
+    void connectPorts();
+    
+    InstrumentId   m_instrument;
+    int                        m_position;
+    std::vector<LADSPA_Handle> m_instanceHandles;
+    size_t                     m_instanceCount;
+    const LADSPA_Descriptor   *m_descriptor;
+
+    std::vector<std::pair<unsigned long, LADSPA_Data*> > m_controlPortsIn;
+    std::vector<std::pair<unsigned long, LADSPA_Data*> > m_controlPortsOut;
+
+    std::vector<int>          m_audioPortsIn;
+    std::vector<int>          m_audioPortsOut;
+
+    size_t                    m_blockSize;
+    sample_t                **m_inputBuffers;
+    sample_t                **m_outputBuffers;
+    bool                      m_ownBuffers;
+    size_t                    m_sampleRate;
+    float                    *m_latencyPort;
+    bool                      m_run;
+    
+    bool                      m_bypassed;
+};
+
+}
+
+#endif // HAVE_LADSPA
+
+#endif // _LADSPAPLUGININSTANCE_H_
+
diff --git a/src/sound/MP3AudioFile.cpp b/src/sound/MP3AudioFile.cpp
new file mode 100644
index 0000000..700169f
--- /dev/null
+++ b/src/sound/MP3AudioFile.cpp
@@ -0,0 +1,329 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+
+#include "MP3AudioFile.h"
+
+#ifdef HAVE_LIBMAD
+
+#include <mad.h>
+
+namespace Rosegarden
+{
+
+
+/*
+ * This is a private message structure. A generic pointer to this structure
+ * is passed to each of the callback functions. Put here any data you need
+ * to access from within the callbacks.
+ */
+
+struct player
+{
+    unsigned char const *start;
+    unsigned long length;
+    //int default_driver;
+    //ao_device *device;
+    //ao_sample_format format;
+    //class SoundTouch *touch;
+};
+
+
+
+MP3AudioFile::MP3AudioFile(const unsigned int &id,
+                           const std::string &name,
+                           const std::string &fileName):
+        AudioFile(id, name, fileName)
+{
+    m_type = MP3;
+}
+
+
+MP3AudioFile::MP3AudioFile(const std::string &fileName,
+                           unsigned int /*channels*/,
+                           unsigned int /*sampleRate*/,
+                           unsigned int /*bytesPerSecond*/,
+                           unsigned int /*bytesPerSample*/,
+                           unsigned int /*bitsPerSample*/):
+        AudioFile(0, std::string(""), fileName)
+{
+    m_type = MP3;
+}
+
+
+MP3AudioFile::~MP3AudioFile()
+{}
+
+bool
+MP3AudioFile::open()
+{
+    // if already open
+    if (m_inFile && (*m_inFile))
+        return true;
+
+    m_inFile = new std::ifstream(m_fileName.c_str(),
+                                 std::ios::in | std::ios::binary);
+
+    if (!(*m_inFile)) {
+        m_type = UNKNOWN;
+        return false;
+    }
+
+    // Get the file size and store it for comparison later
+    m_fileSize = m_fileInfo->size();
+
+    try {
+        parseHeader();
+    } catch (BadSoundFileException s) {
+        throw(s);
+    }
+
+    return true;
+}
+
+bool
+MP3AudioFile::write()
+{
+    return false;
+}
+
+void
+MP3AudioFile::close()
+{}
+
+void
+MP3AudioFile::parseHeader()
+{
+    const std::string MP3_TAG("TAG");
+    if (m_inFile == 0)
+        return ;
+
+    // store size conveniently
+    m_fileSize = m_fileInfo->size();
+
+    if (m_fileSize == 0) {
+        std::string mess = std::string("\"") + m_fileName +
+                           std::string("\" is empty - invalid MP3 file");
+        throw(mess);
+    }
+
+    // seek to beginning
+    m_inFile->seekg(0, std::ios::beg);
+
+    // get some header information
+    //
+    const int bufferLength = 3096;
+    std::string hS = getBytes(bufferLength);
+    bool foundMP3 = false;
+
+    for (unsigned int i = 0; i < hS.length() - 1; ++i) {
+        if ((hS[i] & 0xff) == 0xff && (hS[i + 1] & 0xe0) == 0xe0) {
+            foundMP3 = true;
+            break;
+        }
+    }
+
+    if (foundMP3 == false || (int)hS.length() < bufferLength) {
+        std::string mess = std::string("\"") + m_fileName +
+                           std::string("\" doesn't appear to be a valid MP3 file");
+        throw(mess);
+    }
+
+    // guess most likely values - these are reset during decoding
+    m_channels = 2;
+    m_sampleRate = 44100;
+
+    mad_synth synth;
+    mad_frame frame;
+    mad_stream stream;
+
+    mad_synth_init(&synth);
+    mad_stream_init(&stream);
+    mad_frame_init(&frame);
+
+    /*
+    mad_stream_buffer(&stream, hS.data(), hS.length());
+
+    if (mad_header_decode(&frame.header, &stream) == -1)
+    {
+        throw("Can't decode header");
+    }
+
+    mad_frame_decode(&frame, &stream);
+
+    m_sampleRate = frame.header.samplerate;
+
+    mad_synth_frame(&synth, &frame);
+    struct mad_pcm *pcm = &synth.pcm;
+
+    m_channels = pcm->channels;
+    */
+
+    /*
+        struct player player;
+        struct mad_decoder decoder;
+        struct stat stat;
+        void *fdm;
+        int result;
+     
+        if (fstat(fd, &stat) == -1 ||
+            stat.st_size == 0)
+          return 0;
+     
+        fdm = mmap(0, stat.st_size, PROT_READ, MAP_SHARED, fd, 0);
+        if (fdm == MAP_FAILED) {
+          fprintf(stderr, "mmap failed, aborting...\n");
+          return 0;
+        }
+     
+        player.start = (unsigned char *)fdm;
+        player.length = stat.st_size;
+        player.default_driver = ao_default_driver_id();
+        player.device = NULL;
+        player.touch = new SoundTouch;
+        player.touch->setTempo(tempo);
+        player.touch->setPitch(pitch);
+        mad_decoder_init(&decoder, &player,
+                         input, 0 , 0 , process_output,
+                         decode_error, 0);
+     
+        result = mad_decoder_run(&decoder, MAD_DECODER_MODE_SYNC);
+        mad_decoder_finish(&decoder);
+        delete player.touch;
+        ao_close(player.device);
+        if (munmap((void *)player.start, stat.st_size) == -1)
+            return 4;
+     
+        return result;
+    */
+}
+
+std::streampos
+MP3AudioFile::getDataOffset()
+{
+    return 0;
+}
+
+bool
+MP3AudioFile::scanTo(const RealTime & /*time*/)
+{
+    return false;
+}
+
+bool
+MP3AudioFile::scanTo(std::ifstream * /*file*/, const RealTime & /*time*/)
+{
+    return false;
+}
+
+
+// Scan forward in a file by a certain amount of time
+//
+bool
+MP3AudioFile::scanForward(const RealTime & /*time*/)
+{
+    return false;
+}
+
+bool
+MP3AudioFile::scanForward(std::ifstream * /*file*/, const RealTime & /*time*/)
+{
+    return false;
+}
+
+
+// Return a number of samples - caller will have to
+// de-interleave n-channel samples themselves.
+//
+std::string
+MP3AudioFile::getSampleFrames(std::ifstream * /*file*/,
+                              unsigned int /*frames*/)
+{
+    return "";
+}
+
+unsigned int
+MP3AudioFile::getSampleFrames(std::ifstream * /*file*/,
+                              char * /* buf */,
+                              unsigned int /*frames*/)
+{
+    return 0;
+}
+
+std::string
+MP3AudioFile::getSampleFrames(unsigned int /*frames*/)
+{
+    return "";
+}
+
+
+// Return a number of (possibly) interleaved samples
+// over a time slice from current file pointer position.
+//
+std::string
+MP3AudioFile::getSampleFrameSlice(std::ifstream * /*file*/,
+                                  const RealTime & /*time*/)
+{
+    return "";
+}
+
+std::string
+MP3AudioFile::getSampleFrameSlice(const RealTime & /*time*/)
+{
+    return "";
+}
+
+
+// Append a string of samples to an already open (for writing)
+// audio file.
+//
+bool
+MP3AudioFile::appendSamples(const std::string & /*buffer*/)
+{
+    return false;
+}
+
+bool
+MP3AudioFile::appendSamples(const char * /*buffer*/, unsigned int)
+{
+    return false;
+}
+
+
+// Get the length of the sample in Seconds/Microseconds
+//
+RealTime
+MP3AudioFile::getLength()
+{
+    return RealTime(0, 0);
+}
+
+void
+MP3AudioFile::printStats()
+{}
+
+
+
+
+
+}
+
+#endif // HAVE_LIBMAD
diff --git a/src/sound/MP3AudioFile.h b/src/sound/MP3AudioFile.h
new file mode 100644
index 0000000..2423cd9
--- /dev/null
+++ b/src/sound/MP3AudioFile.h
@@ -0,0 +1,128 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _MP3AUDIOFILE_H_
+#define _MP3AUDIOFILE_H_
+
+#ifdef HAVE_LIBMAD
+
+#include "AudioFile.h"
+
+namespace Rosegarden
+{
+
+class MP3AudioFile : public AudioFile
+{
+public:
+    MP3AudioFile(const unsigned int &id,
+                 const std::string &name,
+                 const std::string &fileName);
+
+    MP3AudioFile(const std::string &fileName,
+                  unsigned int channels,
+                  unsigned int sampleRate,
+                  unsigned int bytesPerSecond,
+                  unsigned int bytesPerSample,
+                  unsigned int bitsPerSample);
+
+    ~MP3AudioFile();
+
+    // Override these methods for the WAV
+    //
+    virtual bool open();
+    virtual bool write();
+    virtual void close();
+
+    // Show the information we have on this file
+    //
+    virtual void printStats();
+
+    // Get all header information
+    //
+    void parseHeader();
+
+    // Offset to start of sample data
+    //
+    virtual std::streampos getDataOffset();
+
+    // Peak file name
+    //
+    virtual std::string getPeakFilename()
+        { return (m_fileName + std::string(".pk")); }
+
+    // scan time was valid and successful.
+    //
+    virtual bool scanTo(const RealTime &time);
+    virtual bool scanTo(std::ifstream *file, const RealTime &time);
+
+    // Scan forward in a file by a certain amount of time
+    //
+    virtual bool scanForward(const RealTime &time);
+    virtual bool scanForward(std::ifstream *file, const RealTime &time);
+
+    // Return a number of samples - caller will have to
+    // de-interleave n-channel samples themselves.
+    //
+    virtual std::string getSampleFrames(std::ifstream *file,
+                                        unsigned int frames);
+    virtual unsigned int getSampleFrames(std::ifstream *file,
+                                         char *buf,
+                                         unsigned int frames);
+    virtual std::string getSampleFrames(unsigned int frames);
+
+    // Return a number of (possibly) interleaved samples
+    // over a time slice from current file pointer position.
+    //
+    virtual std::string getSampleFrameSlice(std::ifstream *file,
+                                            const RealTime &time);
+    virtual std::string getSampleFrameSlice(const RealTime &time);
+
+    // Append a string of samples to an already open (for writing)
+    // audio file.
+    //
+    virtual bool appendSamples(const std::string &buffer);
+    virtual bool appendSamples(const char *buffer, unsigned int samples);
+
+    // Get the length of the sample in Seconds/Microseconds
+    //
+    virtual RealTime getLength();
+
+    virtual unsigned int getBytesPerFrame() { return 0; }
+
+
+    //!!! NOT IMPLEMENTED YET
+    // 
+    virtual bool decode(const unsigned char *sourceData,
+                        size_t sourceBytes,
+                        size_t targetSampleRate,
+                        size_t targetChannels,
+                        size_t targetFrames,
+                        std::vector<float *> &targetData,
+                        bool addToResultBuffers = false) { return false; }
+
+};
+
+}
+
+#endif // HAVE_LIBMAD
+
+#endif // _MP3AUDIOFILE_H_
+
diff --git a/src/sound/MappedCommon.h b/src/sound/MappedCommon.h
new file mode 100644
index 0000000..5ef5487
--- /dev/null
+++ b/src/sound/MappedCommon.h
@@ -0,0 +1,68 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#ifndef _MAPPEDCOMMON_H_
+#define _MAPPEDCOMMON_H_
+
+// Some Mapped types that gui and sound libraries use to communicate
+// plugin and Studio information.  Putting them here so we can change
+// MappedStudio regularly without having to rebuild the gui.
+//
+#include <vector>
+
+#include <qstring.h>
+#include <qdatastream.h>
+
+namespace Rosegarden
+{
+
+typedef int          MappedObjectId;
+typedef QString      MappedObjectProperty;
+typedef float        MappedObjectValue;
+
+// typedef QValueVector<MappedObjectProperty> MappedObjectPropertyList;
+// replaced with a std::vector<> for Qt2 compatibility
+
+typedef std::vector<MappedObjectId> MappedObjectIdList;
+typedef std::vector<MappedObjectProperty> MappedObjectPropertyList;
+typedef std::vector<MappedObjectValue> MappedObjectValueList;
+
+// The direction in which a port operates.
+//
+typedef enum
+{
+    ReadOnly,  // input port
+    WriteOnly, // output port
+    Duplex
+} PortDirection;
+
+QDataStream& operator>>(QDataStream& s, MappedObjectIdList&);
+QDataStream& operator<<(QDataStream&, const MappedObjectIdList&);
+
+QDataStream& operator>>(QDataStream& s, MappedObjectPropertyList&);
+QDataStream& operator<<(QDataStream&, const MappedObjectPropertyList&);
+
+QDataStream& operator>>(QDataStream& s, MappedObjectValueList&);
+QDataStream& operator<<(QDataStream&, const MappedObjectValueList&);
+
+}
+
+#endif // _MAPPEDCOMMON_H_
diff --git a/src/sound/MappedComposition.cpp b/src/sound/MappedComposition.cpp
new file mode 100644
index 0000000..975fccf
--- /dev/null
+++ b/src/sound/MappedComposition.cpp
@@ -0,0 +1,216 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+ 
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+ 
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+ 
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include <qdatastream.h>
+#include "MappedComposition.h"
+#include "MappedEvent.h"
+#include "SegmentPerformanceHelper.h"
+#include <iostream>
+
+namespace Rosegarden
+{
+
+using std::cerr;
+using std::cout;
+using std::endl;
+
+MappedComposition::~MappedComposition()
+{
+    clear();
+}
+
+// copy constructor
+MappedComposition::MappedComposition(const MappedComposition &mC):
+        std::multiset<MappedEvent *, MappedEvent::MappedEventCmp>()
+{
+    clear();
+
+    // deep copy
+    for (MappedComposition::const_iterator it = mC.begin(); it != mC.end(); it++)
+        this->insert(new MappedEvent(**it));
+
+}
+
+// Turn a MappedComposition into a QDataStream - MappedEvents can
+// stream themselves.
+//
+QDataStream&
+operator<<(QDataStream &dS, MappedComposition *mC)
+{
+    dS << int(mC->size());
+
+    for (MappedCompositionIterator it = mC->begin(); it != mC->end(); ++it )
+        dS << (*it);
+
+    return dS;
+}
+
+
+QDataStream&
+operator<<(QDataStream &dS, const MappedComposition &mC)
+{
+    dS << int(mC.size());
+
+    for (MappedComposition::const_iterator it = mC.begin(); it != mC.end(); ++it )
+        dS << (*it);
+
+    return dS;
+}
+
+
+// Turn a QDataStream into a MappedComposition
+//
+QDataStream&
+operator>>(QDataStream &dS, MappedComposition *mC)
+{
+    int sliceSize;
+    MappedEvent *mE;
+
+    dS >> sliceSize;
+
+    while (!dS.atEnd() && sliceSize) {
+        mE = new MappedEvent();
+        dS >> mE;
+
+        try {
+            mC->insert(mE);
+        } catch (...) {
+            ;
+        }
+
+        sliceSize--;
+
+    }
+
+#ifdef DEBUG_MAPPEDCOMPOSITION
+    if (sliceSize) {
+        cerr << "operator>> - wrong number of events received" << endl;
+    }
+#endif
+
+    return dS;
+}
+
+QDataStream&
+operator>>(QDataStream &dS, MappedComposition &mC)
+{
+    int sliceSize;
+    MappedEvent *mE;
+
+    dS >> sliceSize;
+
+    while (!dS.atEnd() && sliceSize) {
+        mE = new MappedEvent();
+
+        dS >> mE;
+
+        try {
+            mC.insert(mE);
+        } catch (...) {
+            ;
+        }
+
+        sliceSize--;
+
+    }
+
+#ifdef DEBUG_MAPPEDCOMPOSITION
+    if (sliceSize) {
+        cerr << "operator>> - wrong number of events received" << endl;
+    }
+#endif
+
+
+    return dS;
+}
+
+// Move the start time of this MappedComposition and all its events.
+// Actually - we have a special case for audio events at the moment..
+//
+//
+void
+MappedComposition::moveStartTime(const RealTime &mT)
+{
+    MappedCompositionIterator it;
+
+    for (it = this->begin(); it != this->end(); ++it) {
+        // Reset start time and duration
+        //
+        (*it)->setEventTime((*it)->getEventTime() + mT);
+        (*it)->setDuration((*it)->getDuration() - mT);
+
+        // For audio adjust the start index
+        //
+        if ((*it)->getType() == MappedEvent::Audio)
+            (*it)->setAudioStartMarker((*it)->getAudioStartMarker() + mT);
+    }
+
+    m_startTime = m_startTime + mT;
+    m_endTime = m_endTime + mT;
+
+}
+
+
+// Concatenate MappedComposition
+//
+MappedComposition&
+MappedComposition::operator+(const MappedComposition &mC)
+{
+    for (MappedComposition::const_iterator it = mC.begin(); it != mC.end(); it++)
+        this->insert(new MappedEvent(**it)); // deep copy
+
+    return *this;
+}
+
+// Assign (clear and deep copy)
+//
+MappedComposition&
+MappedComposition::operator=(const MappedComposition &mC)
+{
+    if (&mC == this)
+        return * this;
+
+    clear();
+
+    // deep copy
+    for (MappedComposition::const_iterator it = mC.begin(); it != mC.end(); it++)
+        this->insert(new MappedEvent(**it));
+
+    return *this;
+}
+
+void
+MappedComposition::clear()
+{
+    // Only clear if the events aren't persistent
+    //
+    for (MappedCompositionIterator it = this->begin(); it != this->end(); it++)
+        if (!(*it)->isPersistent())
+            delete (*it);
+
+    this->erase(this->begin(), this->end());
+}
+
+
+
+}
+
+
diff --git a/src/sound/MappedComposition.h b/src/sound/MappedComposition.h
new file mode 100644
index 0000000..bfa7c05
--- /dev/null
+++ b/src/sound/MappedComposition.h
@@ -0,0 +1,93 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+
+#ifndef _MAPPEDCOMPOSITION_H_
+#define _MAPPEDCOMPOSITION_H_
+
+
+// MappedComposition is used with MappedEvent to create a sequence
+// of MIDI ready events ready for playing.  The QDataStream operators
+// are a necessary part of the DCOP transmission process allowing 
+// the whole class to be serialized.  The core application is sent
+// a request specifying a time slice between given start and end
+// points which it fills with MappedEvents which are cut down
+// (sequencer suitable) versions of the core Events.
+//
+
+#include <Composition.h>
+#include "MappedEvent.h"
+#include <set>
+#include <qdatastream.h>
+
+namespace Rosegarden
+{
+
+class MappedComposition : public std::multiset<MappedEvent *,
+                          MappedEvent::MappedEventCmp>
+{
+public:
+    MappedComposition():m_startTime(0, 0), m_endTime(0, 0) {;}
+
+    MappedComposition(const RealTime &sT,
+                      const RealTime &eT):
+        m_startTime(sT), m_endTime(eT) {;}
+
+    MappedComposition(const MappedComposition &mC);
+
+    ~MappedComposition();
+
+    const RealTime getStartTime() const { return m_startTime; }
+    const RealTime getEndTime() const { return m_endTime; }
+    void setStartTime(const RealTime &sT) { m_startTime = sT; }
+    void setEndTime(const RealTime &eT) { m_endTime = eT; }
+
+    // When we're looping we want to be able to move the start
+    // time of MappedEvents around in this container
+    //
+    void moveStartTime(const RealTime &mT);
+
+    MappedComposition& operator+(const MappedComposition &mC);
+    MappedComposition& operator=(const MappedComposition &mC);
+
+    // This section is used for serialising this class over DCOP
+    //
+    //
+    friend QDataStream& operator>>(QDataStream &dS, MappedComposition *mC);
+    friend QDataStream& operator<<(QDataStream &dS, MappedComposition *mC);
+    friend QDataStream& operator>>(QDataStream &dS, MappedComposition &mC);
+    friend QDataStream& operator<<(QDataStream &dS, const MappedComposition &mC);
+
+    // Clear out
+    void clear();
+
+private:
+    RealTime m_startTime;
+    RealTime m_endTime;
+
+};
+
+typedef std::multiset<MappedEvent *, MappedEvent::MappedEventCmp>::iterator MappedCompositionIterator;
+
+
+}
+
+#endif // _MAPPEDCOMPOSITION_H_
diff --git a/src/sound/MappedDevice.cpp b/src/sound/MappedDevice.cpp
new file mode 100644
index 0000000..619be2a
--- /dev/null
+++ b/src/sound/MappedDevice.cpp
@@ -0,0 +1,250 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+ 
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+ 
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+ 
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include "MappedDevice.h"
+#include "MappedInstrument.h"
+#include <iostream>
+
+namespace Rosegarden
+{
+
+MappedDevice::MappedDevice():
+        std::vector<MappedInstrument*>(),
+        m_id(Device::NO_DEVICE),
+        m_type(Device::Midi),
+        m_name("Unconfigured device"),
+        m_connection(""),
+        m_direction(MidiDevice::Play),
+        m_recording(false)
+{}
+
+MappedDevice::MappedDevice(DeviceId id,
+                           Device::DeviceType type,
+                           std::string name,
+                           std::string connection):
+        std::vector<MappedInstrument*>(),
+        m_id(id),
+        m_type(type),
+        m_name(name),
+        m_connection(connection),
+        m_direction(MidiDevice::Play),
+        m_recording(false)
+{}
+
+MappedDevice::~MappedDevice()
+{}
+
+MappedDevice::MappedDevice(const MappedDevice &mD):
+        std::vector<MappedInstrument*>()
+{
+    clear();
+
+    for (MappedDeviceConstIterator it = mD.begin(); it != mD.end(); it++)
+        this->push_back(new MappedInstrument(**it));
+
+    m_id = mD.getId();
+    m_type = mD.getType();
+    m_name = mD.getName();
+    m_connection = mD.getConnection();
+    m_direction = mD.getDirection();
+    m_recording = mD.isRecording();
+}
+
+void
+MappedDevice::clear()
+{
+    MappedDeviceIterator it;
+
+    for (it = this->begin(); it != this->end(); it++)
+        delete (*it);
+
+    this->erase(this->begin(), this->end());
+}
+
+MappedDevice&
+MappedDevice::operator+(const MappedDevice &mD)
+{
+    for (MappedDeviceConstIterator it = mD.begin(); it != mD.end(); it++)
+        this->push_back(new MappedInstrument(**it));
+
+    return *this;
+}
+
+MappedDevice&
+MappedDevice::operator=(const MappedDevice &mD)
+{
+    if (&mD == this)
+        return * this;
+
+    clear();
+
+    for (MappedDeviceConstIterator it = mD.begin(); it != mD.end(); it++)
+        this->push_back(new MappedInstrument(**it));
+
+    m_id = mD.getId();
+    m_type = mD.getType();
+    m_name = mD.getName();
+    m_connection = mD.getConnection();
+    m_direction = mD.getDirection();
+    m_recording = mD.isRecording();
+
+    return *this;
+}
+
+
+QDataStream&
+operator>>(QDataStream &dS, MappedDevice *mD)
+{
+    int instruments = 0;
+    dS >> instruments;
+
+    MappedInstrument mI;
+    while (!dS.atEnd() && instruments) {
+        dS >> mI;
+        mD->push_back(new MappedInstrument(mI));
+        instruments--;
+    }
+
+    QString name;
+    unsigned int id, dType;
+    QString connection;
+    unsigned int direction;
+    unsigned int recording;
+
+    dS >> id;
+    dS >> dType;
+    dS >> name;
+    dS >> connection;
+    dS >> direction;
+    dS >> recording;
+    mD->setId(id);
+    mD->setType(Device::DeviceType(dType));
+    mD->setName(std::string(name.data()));
+    mD->setConnection(connection.data());
+    mD->setDirection(MidiDevice::DeviceDirection(direction));
+    mD->setRecording((bool)recording);
+
+#ifdef DEBUG_MAPPEDDEVICE
+
+    if (instruments) {
+        std::cerr << "MappedDevice::operator>> - "
+        << "wrong number of events received" << std::endl;
+    }
+#endif
+
+    return dS;
+}
+
+
+QDataStream&
+operator>>(QDataStream &dS, MappedDevice &mD)
+{
+    int instruments;
+    dS >> instruments;
+
+    MappedInstrument mI;
+
+    while (!dS.atEnd() && instruments) {
+        dS >> mI;
+        mD.push_back(new MappedInstrument(mI));
+        instruments--;
+    }
+
+    unsigned int id, dType;
+    QString name;
+    QString connection;
+    unsigned int direction;
+    unsigned int recording;
+
+    dS >> id;
+    dS >> dType;
+    dS >> name;
+    dS >> connection;
+    dS >> direction;
+    dS >> recording;
+    mD.setId(id);
+    mD.setType(Device::DeviceType(dType));
+    mD.setName(std::string(name.data()));
+    mD.setConnection(connection.data());
+    mD.setDirection(MidiDevice::DeviceDirection(direction));
+    mD.setRecording((bool)recording);
+
+#ifdef DEBUG_MAPPEDDEVICE
+
+    if (instruments) {
+        std::cerr << "MappedDevice::operator>> - "
+        << "wrong number of events received" << std::endl;
+    }
+#endif
+
+    return dS;
+}
+
+QDataStream&
+operator<<(QDataStream &dS, MappedDevice *mD)
+{
+    dS << (int)mD->size();
+
+    for (MappedDeviceIterator it = mD->begin(); it != mD->end(); it++)
+        dS << (*it);
+
+    dS << (unsigned int)(mD->getId());
+    dS << (int)(mD->getType());
+    dS << QString(mD->getName().c_str());
+    dS << QString(mD->getConnection().c_str());
+    dS << mD->getDirection();
+    dS << (unsigned int)(mD->isRecording());
+
+#ifdef DEBUG_MAPPEDDEVICE
+
+    std::cerr << "MappedDevice::operator>> - wrote \"" << mD->getConnection() << "\""
+    << std::endl;
+#endif
+
+    return dS;
+}
+
+QDataStream&
+operator<<(QDataStream &dS, const MappedDevice &mD)
+{
+    dS << (int)mD.size();
+
+    for (MappedDeviceConstIterator it = mD.begin(); it != mD.end(); it++)
+        dS << (*it);
+
+    dS << (unsigned int)(mD.getId());
+    dS << (int)(mD.getType());
+    dS << QString(mD.getName().c_str());
+    dS << QString(mD.getConnection().c_str());
+    dS << mD.getDirection();
+    dS << (unsigned int)(mD.isRecording());
+
+#ifdef DEBUG_MAPPEDDEVICE
+
+    std::cerr << "MappedDevice::operator>> - wrote \"" << mD.getConnection() << "\""
+    << std::endl;
+#endif
+
+    return dS;
+}
+
+}
+
diff --git a/src/sound/MappedDevice.h b/src/sound/MappedDevice.h
new file mode 100644
index 0000000..f53bb37
--- /dev/null
+++ b/src/sound/MappedDevice.h
@@ -0,0 +1,103 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include <string>
+#include <vector>
+
+#include <qdatastream.h>
+
+#include "Device.h"
+#include "MidiDevice.h"
+#include "MappedCommon.h"
+
+#ifndef _MAPPEDDEVICE_H_
+#define _MAPPEDDEVICE_H_
+
+// A DCOP wrapper to get MappedInstruments across to the GUI
+//
+
+namespace Rosegarden
+{
+
+class MappedInstrument;
+
+class MappedDevice : public std::vector<MappedInstrument*>
+{
+public:
+    MappedDevice();
+    MappedDevice(DeviceId id,
+                 Device::DeviceType type,
+                 std::string name,
+                 std::string connection = "");
+
+    MappedDevice(const MappedDevice &mD);
+    ~MappedDevice();
+ 
+    // Clear down
+    //
+    void clear();
+
+    MappedDevice& operator+(const MappedDevice &mD);
+    MappedDevice& operator=(const MappedDevice &mD);
+
+    friend QDataStream& operator>>(QDataStream &dS, MappedDevice *mD);
+    friend QDataStream& operator<<(QDataStream &dS, MappedDevice *mD);
+    friend QDataStream& operator>>(QDataStream &dS, MappedDevice &mD);
+    friend QDataStream& operator<<(QDataStream &dS, const MappedDevice &mD);
+
+    std::string getName() const { return m_name; }
+    void setName(const std::string &name) { m_name = name; }
+
+    DeviceId getId() const { return m_id; }
+    void setId(DeviceId id) { m_id = id; }
+
+    Device::DeviceType getType() const { return m_type; }
+    void setType(Device::DeviceType type) { m_type = type; }
+
+    std::string getConnection() const { return m_connection; }
+    void setConnection(std::string connection) { m_connection = connection; }
+
+    MidiDevice::DeviceDirection getDirection() const { return m_direction; }
+    void setDirection(MidiDevice::DeviceDirection direction) { m_direction = direction; }
+    
+    bool isRecording() const { return m_recording; }
+    void setRecording(bool recording) { m_recording = recording; }
+
+protected:
+
+    DeviceId            m_id;
+    Device::DeviceType  m_type;
+    std::string                     m_name;
+    std::string                     m_connection;
+    MidiDevice::DeviceDirection     m_direction;
+    bool                            m_recording;
+};
+
+typedef std::vector<MappedInstrument*>::const_iterator
+    MappedDeviceConstIterator;
+
+typedef std::vector<MappedInstrument*>::iterator
+    MappedDeviceIterator;
+
+}
+
+#endif // _MAPPEDDEVICE_H_
+
diff --git a/src/sound/MappedEvent.cpp b/src/sound/MappedEvent.cpp
new file mode 100644
index 0000000..9b4ccab
--- /dev/null
+++ b/src/sound/MappedEvent.cpp
@@ -0,0 +1,593 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+ 
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+ 
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+ 
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include <qdir.h>
+#include <qfile.h>
+#include <qfileinfo.h>
+
+#include <kstddirs.h>
+
+#include "MappedEvent.h"
+#include "BaseProperties.h"
+#include "Midi.h"
+#include "MidiTypes.h"
+
+#define DEBUG_MAPPEDEVENT 1
+
+namespace Rosegarden
+{
+
+MappedEvent::MappedEvent(InstrumentId id,
+                         const Event &e,
+                         const RealTime &eventTime,
+                         const RealTime &duration):
+        m_trackId(0),
+        m_instrument(id),
+        m_type(MidiNote),
+        m_data1(0),
+        m_data2(0),
+        m_eventTime(eventTime),
+        m_duration(duration),
+        m_audioStartMarker(0, 0),
+        m_dataBlockId(0),
+        m_isPersistent(false),
+        m_runtimeSegmentId( -1),
+        m_autoFade(false),
+        m_fadeInTime(RealTime::zeroTime),
+        m_fadeOutTime(RealTime::zeroTime),
+        m_recordedChannel(0),
+        m_recordedDevice(0)
+
+{
+    try {
+
+        // For each event type, we set the properties in a particular
+        // order: first the type, then whichever of data1 and data2 fits
+        // less well with its default value.  This way if one throws an
+        // exception for no data, we still have a good event with the
+        // defaults set.
+
+        if (e.isa(Note::EventType)) {
+            m_type = MidiNoteOneShot;
+            long v = MidiMaxValue;
+            e.get<Int>(BaseProperties::VELOCITY, v);
+            m_data2 = v;
+            m_data1 = e.get<Int>(BaseProperties::PITCH);
+        } else if (e.isa(PitchBend::EventType)) {
+            m_type = MidiPitchBend;
+            PitchBend pb(e);
+            m_data1 = pb.getMSB();
+            m_data2 = pb.getLSB();
+        } else if (e.isa(Controller::EventType)) {
+            m_type = MidiController;
+            Controller c(e);
+            m_data1 = c.getNumber();
+            m_data2 = c.getValue();
+        } else if (e.isa(ProgramChange::EventType)) {
+            m_type = MidiProgramChange;
+            ProgramChange pc(e);
+            m_data1 = pc.getProgram();
+        } else if (e.isa(KeyPressure::EventType)) {
+            m_type = MidiKeyPressure;
+            KeyPressure kp(e);
+            m_data1 = kp.getPitch();
+            m_data2 = kp.getPressure();
+        } else if (e.isa(ChannelPressure::EventType)) {
+            m_type = MidiChannelPressure;
+            ChannelPressure cp(e);
+            m_data1 = cp.getPressure();
+        } else if (e.isa(SystemExclusive::EventType)) {
+            m_type = MidiSystemMessage;
+            m_data1 = MIDI_SYSTEM_EXCLUSIVE;
+            SystemExclusive s(e);
+            std::string dataBlock = s.getRawData();
+            DataBlockRepository::getInstance()->registerDataBlockForEvent(dataBlock, this);
+        } else {
+            m_type = InvalidMappedEvent;
+        }
+    } catch (MIDIValueOutOfRange r) {
+
+#ifdef DEBUG_MAPPEDEVENT
+        std::cerr << "MIDI value out of range in MappedEvent ctor"
+        << std::endl;
+#else
+
+        ;
+#endif
+
+    } catch (Event::NoData d) {
+
+#ifdef DEBUG_MAPPEDEVENT
+        std::cerr << "Caught Event::NoData in MappedEvent ctor, message is:"
+        << std::endl << d.getMessage() << std::endl;
+#else
+
+        ;
+#endif
+
+    } catch (Event::BadType b) {
+
+#ifdef DEBUG_MAPPEDEVENT
+        std::cerr << "Caught Event::BadType in MappedEvent ctor, message is:"
+        << std::endl << b.getMessage() << std::endl;
+#else
+
+        ;
+#endif
+
+    } catch (SystemExclusive::BadEncoding e) {
+
+#ifdef DEBUG_MAPPEDEVENT
+        std::cerr << "Caught bad SysEx encoding in MappedEvent ctor"
+        << std::endl;
+#else
+
+        ;
+#endif
+
+    }
+}
+
+bool
+operator<(const MappedEvent &a, const MappedEvent &b)
+{
+    return a.getEventTime() < b.getEventTime();
+}
+
+MappedEvent&
+MappedEvent::operator=(const MappedEvent &mE)
+{
+    if (&mE == this)
+        return * this;
+
+    m_trackId = mE.getTrackId();
+    m_instrument = mE.getInstrument();
+    m_type = mE.getType();
+    m_data1 = mE.getData1();
+    m_data2 = mE.getData2();
+    m_eventTime = mE.getEventTime();
+    m_duration = mE.getDuration();
+    m_audioStartMarker = mE.getAudioStartMarker();
+    m_dataBlockId = mE.getDataBlockId();
+    m_runtimeSegmentId = mE.getRuntimeSegmentId();
+    m_autoFade = mE.isAutoFading();
+    m_fadeInTime = mE.getFadeInTime();
+    m_fadeOutTime = mE.getFadeOutTime();
+    m_recordedChannel = mE.getRecordedChannel();
+    m_recordedDevice = mE.getRecordedDevice();
+
+    return *this;
+}
+
+// Do we use this?  It looks dangerous so just commenting it out - rwb
+//
+//const size_t MappedEvent::streamedSize = 12 * sizeof(unsigned int);
+
+QDataStream&
+operator<<(QDataStream &dS, MappedEvent *mE)
+{
+    dS << (unsigned int)mE->getTrackId();
+    dS << (unsigned int)mE->getInstrument();
+    dS << (unsigned int)mE->getType();
+    dS << (unsigned int)mE->getData1();
+    dS << (unsigned int)mE->getData2();
+    dS << (unsigned int)mE->getEventTime().sec;
+    dS << (unsigned int)mE->getEventTime().nsec;
+    dS << (unsigned int)mE->getDuration().sec;
+    dS << (unsigned int)mE->getDuration().nsec;
+    dS << (unsigned int)mE->getAudioStartMarker().sec;
+    dS << (unsigned int)mE->getAudioStartMarker().nsec;
+    dS << (unsigned long)mE->getDataBlockId();
+    dS << mE->getRuntimeSegmentId();
+    dS << (unsigned int)mE->isAutoFading();
+    dS << (unsigned int)mE->getFadeInTime().sec;
+    dS << (unsigned int)mE->getFadeInTime().nsec;
+    dS << (unsigned int)mE->getFadeOutTime().sec;
+    dS << (unsigned int)mE->getFadeOutTime().nsec;
+    dS << (unsigned int)mE->getRecordedChannel();
+    dS << (unsigned int)mE->getRecordedDevice();
+
+    return dS;
+}
+
+QDataStream&
+operator<<(QDataStream &dS, const MappedEvent &mE)
+{
+    dS << (unsigned int)mE.getTrackId();
+    dS << (unsigned int)mE.getInstrument();
+    dS << (unsigned int)mE.getType();
+    dS << (unsigned int)mE.getData1();
+    dS << (unsigned int)mE.getData2();
+    dS << (unsigned int)mE.getEventTime().sec;
+    dS << (unsigned int)mE.getEventTime().nsec;
+    dS << (unsigned int)mE.getDuration().sec;
+    dS << (unsigned int)mE.getDuration().nsec;
+    dS << (unsigned int)mE.getAudioStartMarker().sec;
+    dS << (unsigned int)mE.getAudioStartMarker().nsec;
+    dS << (unsigned long)mE.getDataBlockId();
+    dS << mE.getRuntimeSegmentId();
+    dS << (unsigned int)mE.isAutoFading();
+    dS << (unsigned int)mE.getFadeInTime().sec;
+    dS << (unsigned int)mE.getFadeInTime().nsec;
+    dS << (unsigned int)mE.getFadeOutTime().sec;
+    dS << (unsigned int)mE.getFadeOutTime().nsec;
+    dS << (unsigned int)mE.getRecordedChannel();
+    dS << (unsigned int)mE.getRecordedDevice();
+
+    return dS;
+}
+
+QDataStream&
+operator>>(QDataStream &dS, MappedEvent *mE)
+{
+    unsigned int trackId = 0, instrument = 0, type = 0, data1 = 0, data2 = 0;
+    long eventTimeSec = 0, eventTimeNsec = 0, durationSec = 0, durationNsec = 0,
+                                           audioSec = 0, audioNsec = 0;
+    std::string dataBlock;
+    unsigned long dataBlockId = 0;
+    int runtimeSegmentId = -1;
+    unsigned int autoFade = 0,
+                            fadeInSec = 0, fadeInNsec = 0, fadeOutSec = 0, fadeOutNsec = 0,
+                                                        recordedChannel = 0, recordedDevice = 0;
+
+    dS >> trackId;
+    dS >> instrument;
+    dS >> type;
+    dS >> data1;
+    dS >> data2;
+    dS >> eventTimeSec;
+    dS >> eventTimeNsec;
+    dS >> durationSec;
+    dS >> durationNsec;
+    dS >> audioSec;
+    dS >> audioNsec;
+    dS >> dataBlockId;
+    dS >> runtimeSegmentId;
+    dS >> autoFade;
+    dS >> fadeInSec;
+    dS >> fadeInNsec;
+    dS >> fadeOutSec;
+    dS >> fadeOutNsec;
+    dS >> recordedChannel;
+    dS >> recordedDevice;
+
+    mE->setTrackId((TrackId)trackId);
+    mE->setInstrument((InstrumentId)instrument);
+    mE->setType((MappedEvent::MappedEventType)type);
+    mE->setData1((MidiByte)data1);
+    mE->setData2((MidiByte)data2);
+    mE->setEventTime(RealTime(eventTimeSec, eventTimeNsec));
+    mE->setDuration(RealTime(durationSec, durationNsec));
+    mE->setAudioStartMarker(RealTime(audioSec, audioNsec));
+    mE->setDataBlockId(dataBlockId);
+    mE->setRuntimeSegmentId(runtimeSegmentId);
+    mE->setAutoFade(autoFade);
+    mE->setFadeInTime(RealTime(fadeInSec, fadeInNsec));
+    mE->setFadeOutTime(RealTime(fadeOutSec, fadeOutNsec));
+    mE->setRecordedChannel(recordedChannel);
+    mE->setRecordedDevice(recordedDevice);
+
+    return dS;
+}
+
+QDataStream&
+operator>>(QDataStream &dS, MappedEvent &mE)
+{
+    unsigned int trackId = 0, instrument = 0, type = 0, data1 = 0, data2 = 0;
+    long eventTimeSec = 0, eventTimeNsec = 0, durationSec = 0, durationNsec = 0,
+                                           audioSec = 0, audioNsec = 0;
+    std::string dataBlock;
+    unsigned long dataBlockId = 0;
+    int runtimeSegmentId = -1;
+    unsigned int autoFade = 0,
+                            fadeInSec = 0, fadeInNsec = 0, fadeOutSec = 0, fadeOutNsec = 0,
+                                                        recordedChannel = 0, recordedDevice = 0;
+
+    dS >> trackId;
+    dS >> instrument;
+    dS >> type;
+    dS >> data1;
+    dS >> data2;
+    dS >> eventTimeSec;
+    dS >> eventTimeNsec;
+    dS >> durationSec;
+    dS >> durationNsec;
+    dS >> audioSec;
+    dS >> audioNsec;
+    dS >> dataBlockId;
+    dS >> runtimeSegmentId;
+    dS >> autoFade;
+    dS >> fadeInSec;
+    dS >> fadeInNsec;
+    dS >> fadeOutSec;
+    dS >> fadeOutNsec;
+    dS >> recordedChannel;
+    dS >> recordedDevice;
+
+    mE.setTrackId((TrackId)trackId);
+    mE.setInstrument((InstrumentId)instrument);
+    mE.setType((MappedEvent::MappedEventType)type);
+    mE.setData1((MidiByte)data1);
+    mE.setData2((MidiByte)data2);
+    mE.setEventTime(RealTime(eventTimeSec, eventTimeNsec));
+    mE.setDuration(RealTime(durationSec, durationNsec));
+    mE.setAudioStartMarker(RealTime(audioSec, audioNsec));
+    mE.setDataBlockId(dataBlockId);
+    mE.setRuntimeSegmentId(runtimeSegmentId);
+    mE.setAutoFade(autoFade);
+    mE.setFadeInTime(RealTime(fadeInSec, fadeInNsec));
+    mE.setFadeOutTime(RealTime(fadeOutSec, fadeOutNsec));
+    mE.setRecordedChannel(recordedChannel);
+    mE.setRecordedDevice(recordedDevice);
+
+    return dS;
+}
+
+void
+MappedEvent::addDataByte(MidiByte byte)
+{
+    DataBlockRepository::getInstance()->addDataByteForEvent(byte, this);
+}
+
+void
+MappedEvent::addDataString(const std::string& data)
+{
+    DataBlockRepository::getInstance()->addDataStringForEvent(data, this);
+}
+
+
+
+//--------------------------------------------------
+
+class DataBlockFile
+{
+public:
+    DataBlockFile(DataBlockRepository::blockid id);
+    ~DataBlockFile();
+
+    QString getFileName()
+    {
+        return m_fileName;
+    }
+
+    void addDataByte(MidiByte);
+    void addDataString(const std::string&);
+
+    void clear()
+    {
+        m_cleared = true;
+    }
+    bool exists();
+    void setData(const std::string&);
+    std::string getData();
+
+protected:
+    void prepareToWrite();
+    void prepareToRead();
+
+    //--------------- Data members ---------------------------------
+    QString m_fileName;
+    QFile m_file;
+    bool m_cleared;
+};
+
+DataBlockFile::DataBlockFile(DataBlockRepository::blockid id)
+        : m_fileName(KGlobal::dirs()->resourceDirs("tmp").first() + QString("/rosegarden_datablock_%1").arg(id)),
+        m_file(m_fileName),
+        m_cleared(false)
+{
+    //     std::cerr << "DataBlockFile " << m_fileName.latin1() << std::endl;
+}
+
+DataBlockFile::~DataBlockFile()
+{
+    if (m_cleared) {
+//        std::cerr << "~DataBlockFile : removing " << m_fileName.latin1() << std::endl;
+        QFile::remove
+            (m_fileName);
+    }
+
+}
+
+bool DataBlockFile::exists()
+{
+    return QFile::exists(m_fileName);
+}
+
+void DataBlockFile::setData(const std::string& s)
+{
+  //  std::cerr << "DataBlockFile::setData() : setting data to " << m_fileName << std::endl;
+    prepareToWrite();
+
+    QDataStream stream(&m_file);
+    stream.writeRawBytes(s.data(), s.length());
+}
+
+std::string DataBlockFile::getData()
+{
+    if (!exists())
+        return std::string();
+
+    prepareToRead();
+
+    QDataStream stream(&m_file);
+ //   std::cerr << "DataBlockFile::getData() : file size = " << m_file.size() << std::endl;
+    char* tmp = new char[m_file.size()];
+    stream.readRawBytes(tmp, m_file.size());
+    std::string res(tmp, m_file.size());
+    delete[] tmp;
+
+    return res;
+}
+
+void DataBlockFile::addDataByte(MidiByte byte)
+{
+    prepareToWrite();
+    m_file.putch(byte);
+}
+
+void DataBlockFile::addDataString(const std::string& s)
+{
+    prepareToWrite();
+    QDataStream stream(&m_file);
+    stream.writeRawBytes(s.data(), s.length());
+}
+
+void DataBlockFile::prepareToWrite()
+{
+ //   std::cerr << "DataBlockFile[" << m_fileName << "]: prepareToWrite" << std::endl;
+    if (!m_file.isWritable()) {
+        m_file.close();
+        m_file.open(IO_WriteOnly | IO_Append);
+        assert(m_file.isWritable());
+    }
+}
+
+void DataBlockFile::prepareToRead()
+{
+//    std::cerr << "DataBlockFile[" << m_fileName << "]: prepareToRead" << std::endl;
+    if (!m_file.isReadable()) {
+        m_file.close();
+        m_file.open(IO_ReadOnly);
+        assert(m_file.isReadable());
+    }
+}
+
+
+
+//--------------------------------------------------
+
+DataBlockRepository* DataBlockRepository::getInstance()
+{
+    if (!m_instance)
+        m_instance = new DataBlockRepository;
+    return m_instance;
+}
+
+std::string DataBlockRepository::getDataBlock(DataBlockRepository::blockid id)
+{
+    DataBlockFile dataBlockFile(id);
+
+    if (dataBlockFile.exists())
+        return dataBlockFile.getData();
+
+    return std::string();
+}
+
+
+std::string DataBlockRepository::getDataBlockForEvent(MappedEvent* e)
+{
+    blockid id = e->getDataBlockId();
+    if (id == 0) {
+   //     std::cerr << "WARNING: DataBlockRepository::getDataBlockForEvent called on event with data block id 0" << std::endl;
+        return "";
+    }
+    return getInstance()->getDataBlock(id);
+}
+
+void DataBlockRepository::setDataBlockForEvent(MappedEvent* e, const std::string& s)
+{
+    blockid id = e->getDataBlockId();
+    if (id == 0) {
+  //      std::cerr << "Creating new datablock for event" << std::endl;
+        getInstance()->registerDataBlockForEvent(s, e);
+    } else {
+   //     std::cerr << "Writing " << s.length() << " chars to file for datablock " << id << std::endl;
+        DataBlockFile dataBlockFile(id);
+        dataBlockFile.setData(s);
+    }
+}
+
+bool DataBlockRepository::hasDataBlock(DataBlockRepository::blockid id)
+{
+    return DataBlockFile(id).exists();
+}
+
+DataBlockRepository::blockid DataBlockRepository::registerDataBlock(const std::string& s)
+{
+    blockid id = 0;
+    while (id == 0 || DataBlockFile(id).exists())
+        id = (blockid)random();
+
+ //   std::cerr << "DataBlockRepository::registerDataBlock: " << s.length() << " chars, id is " << id << std::endl;
+
+    DataBlockFile dataBlockFile(id);
+    dataBlockFile.setData(s);
+
+    return id;
+}
+
+void DataBlockRepository::unregisterDataBlock(DataBlockRepository::blockid id)
+{
+    DataBlockFile dataBlockFile(id);
+
+    dataBlockFile.clear();
+}
+
+void DataBlockRepository::registerDataBlockForEvent(const std::string& s, MappedEvent* e)
+{
+    e->setDataBlockId(registerDataBlock(s));
+}
+
+void DataBlockRepository::unregisterDataBlockForEvent(MappedEvent* e)
+{
+    unregisterDataBlock(e->getDataBlockId());
+}
+
+
+DataBlockRepository::DataBlockRepository()
+{}
+
+void DataBlockRepository::clear()
+{
+#ifdef DEBUG_MAPPEDEVENT
+    std::cerr << "DataBlockRepository::clear()\n";
+#endif
+
+    // Erase all 'datablock_*' files
+    //
+    QString tmpPath = KGlobal::dirs()->resourceDirs("tmp").first();
+
+    QDir segmentsDir(tmpPath, "rosegarden_datablock_*");
+    for (unsigned int i = 0; i < segmentsDir.count(); ++i) {
+        QString segmentName = tmpPath + '/' + segmentsDir[i];
+        QFile::remove
+            (segmentName);
+    }
+}
+
+
+void DataBlockRepository::addDataByteForEvent(MidiByte byte, MappedEvent* e)
+{
+    DataBlockFile dataBlockFile(e->getDataBlockId());
+    dataBlockFile.addDataByte(byte);
+
+}
+
+void DataBlockRepository::addDataStringForEvent(const std::string& s, MappedEvent* e)
+{
+    DataBlockFile dataBlockFile(e->getDataBlockId());
+    dataBlockFile.addDataString(s);
+}
+
+DataBlockRepository* DataBlockRepository::m_instance = 0;
+
+}
diff --git a/src/sound/MappedEvent.h b/src/sound/MappedEvent.h
new file mode 100644
index 0000000..cc4e3f3
--- /dev/null
+++ b/src/sound/MappedEvent.h
@@ -0,0 +1,546 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include <qdatastream.h>
+
+#include "Composition.h" // for RealTime
+#include "Event.h"
+
+
+#ifndef _MAPPEDEVENT_H_
+#define _MAPPEDEVENT_H_
+
+// Used as a transformation stage between Composition, Events and output
+// at the Sequencer this class and MidiComposition eliminate the notion
+// of the Segment and Track for ease of Event access.  The MappedEvents
+// are ready for playing or routing through an Instrument or Effects
+// boxes.
+//
+// MappedEvents can also represent instructions for playback of audio
+// samples - if the m_type is Audio then the sequencer will attempt to
+// map the Pitch (m_data1) to the audio id.  Note that this limits us
+// to 256 audio files in the Composition unless we use a different
+// parameter for storing these IDs.
+// 
+// The MappedEvent/Instrument relationship is interesting - we don't
+// want to duplicate the entire Instrument at the Sequencer level as
+// it'd be messy and unnecessary.  Instead we use a MappedInstrument
+// which is just a very cut down Sequencer-side version of an Instrument.
+//
+// Some of these Events are unidirectional, some are bidirectional -
+// that is they only have a meaning in one direction (they are still
+// legal at either end).  They are broadcast in both directions using
+// the "getSequencerSlice" and "processAsync/Recorded" interfaces on
+// which the control messages can piggyback and eventually stripped out.
+//
+
+namespace Rosegarden
+{
+class MappedEvent;
+
+class DataBlockRepository
+{
+public:
+    friend class MappedEvent;
+    typedef unsigned long blockid;
+
+    static DataBlockRepository* getInstance();
+    static std::string getDataBlockForEvent(MappedEvent*);
+    static void setDataBlockForEvent(MappedEvent*, const std::string&);
+    /**
+     * Clear all block files
+     */
+    static void clear();
+    bool hasDataBlock(blockid);
+
+protected:
+    DataBlockRepository();
+
+    std::string getDataBlock(blockid);
+
+    void addDataByteForEvent(MidiByte byte, MappedEvent*);
+    void addDataStringForEvent(const std::string&, MappedEvent*);
+
+
+    blockid registerDataBlock(const std::string&);
+    void unregisterDataBlock(blockid);
+
+    void registerDataBlockForEvent(const std::string&, MappedEvent*);
+    void unregisterDataBlockForEvent(MappedEvent*);
+
+
+    //--------------- Data members ---------------------------------
+
+    static DataBlockRepository* m_instance;
+};
+
+
+class MappedEvent
+{
+public:
+    typedef enum
+    {
+        // INVALID
+        //
+        InvalidMappedEvent       = 0,
+
+        // Keep the MidiNotes bit flaggable so that filtering works
+        //
+        MidiNote                 = 1 << 0,
+        MidiNoteOneShot          = 1 << 1,  // doesn't need NOTE OFFs
+        MidiProgramChange        = 1 << 2,
+        MidiKeyPressure          = 1 << 3,
+        MidiChannelPressure      = 1 << 4,
+        MidiPitchBend            = 1 << 5,
+        MidiController           = 1 << 6,
+        MidiSystemMessage        = 1 << 7,
+
+        // Sent from the gui to play an audio file
+        Audio                    = 1 << 8,
+        // Sent from gui to cancel playing an audio file
+        AudioCancel              = 1 << 9,
+        // Sent to the gui with audio level on Instrument
+        AudioLevel               = 1 << 10,
+        // Sent to the gui to inform an audio file stopped
+        AudioStopped             = 1 << 11,
+        // The gui is clear to generate a preview for a new audio file
+        AudioGeneratePreview     = 1 << 12,
+
+        // Update Instruments - new ALSA client detected
+        SystemUpdateInstruments  = 1 << 13,
+        // Set RG as JACK master/slave
+        SystemJackTransport      = 1 << 14,
+        // Set RG as MMC master/slave
+        SystemMMCTransport       = 1 << 15,
+        // Set System Messages and MIDI Clock
+        SystemMIDIClock          = 1 << 16,
+        // Set Record device
+        SystemRecordDevice       = 1 << 17,
+        // Set Metronome device
+        SystemMetronomeDevice    = 1 << 18,
+        // Set Audio inputs/outputs: data1 num inputs, data2 num submasters
+        SystemAudioPortCounts    = 1 << 19,
+        // Set whether we create various Audio ports (data1 is an AudioOutMask)
+        SystemAudioPorts         = 1 << 20,
+        // Some failure has occurred: data1 contains FailureCode
+        SystemFailure            = 1 << 21,
+
+        // Time sig. event (from time sig. composition reference segment)
+        TimeSignature            = 1 << 22,
+        // Tempo event (from tempo composition reference segment)
+        Tempo                    = 1 << 23,
+        
+        // Panic function
+        Panic                    = 1 << 24,
+
+        // Set RG as MTC master/slave
+        SystemMTCTransport       = 1 << 25,
+        // Auto-connect sync outputs
+        SystemMIDISyncAuto       = 1 << 26,
+        // File format used for audio recording (data1 is 0=PCM,1=float)
+        SystemAudioFileFormat    = 1 << 27
+
+    } MappedEventType;
+
+    typedef enum
+    {
+        // These values are OR'd to produce the data2 field in a
+        // SystemAudioPorts event.
+        FaderOuts                = 1 << 0,
+        SubmasterOuts            = 1 << 1
+
+    } MappedEventAudioOutMask;
+
+    typedef enum
+    {
+        // JACK is having some xruns - warn the user maybe
+        FailureXRuns             = 0,
+        // JACK has died or kicked us out
+        FailureJackDied          = 1,
+        // Audio subsystem failed to read from disc fast enough
+        FailureDiscUnderrun      = 2,
+        // Audio subsystem failed to write to disc fast enough
+        FailureDiscOverrun       = 3,
+        // Audio subsystem failed to mix busses fast enough
+        FailureBussMixUnderrun   = 4,
+        // Audio subsystem failed to mix instruments fast enough
+        FailureMixUnderrun       = 5,
+        // Using a timer that has too low a resolution (e.g. 100Hz system timer)
+        WarningImpreciseTimer    = 6,
+        // Too much CPU time spent in audio processing -- risk of xruns and lockup
+        FailureCPUOverload       = 7,
+        // JACK kicked us out, but we've reconnected
+        FailureJackRestart       = 8,
+        // JACK kicked us out, and now the reconnection has failed
+        FailureJackRestartFailed = 9,
+        // A necessary ALSA call has returned an error code
+        FailureALSACallFailed    = 10,
+        // Using a timer that has too low a resolution, but RTC might work
+        WarningImpreciseTimerTryRTC = 11,
+    } FailureCode;      
+
+    MappedEvent(): m_trackId(0),
+                   m_instrument(0),
+                   m_type(MidiNote),
+                   m_data1(0),
+                   m_data2(0),
+                   m_eventTime(0, 0),
+                   m_duration(0, 0),
+                   m_audioStartMarker(0, 0),
+                   m_dataBlockId(0),
+                   m_isPersistent(false),
+                   m_runtimeSegmentId(-1),
+                   m_autoFade(false),
+                   m_fadeInTime(RealTime::zeroTime),
+                   m_fadeOutTime(RealTime::zeroTime),
+                   m_recordedChannel(0),
+                   m_recordedDevice(0) {}
+
+    // Construct from Events to Internal (MIDI) type MappedEvent
+    //
+    MappedEvent(const Event &e);
+
+    // Another Internal constructor from Events
+    MappedEvent(InstrumentId id,
+                const Event &e,
+                const RealTime &eventTime,
+                const RealTime &duration);
+
+    // A general MappedEvent constructor for any MappedEvent type
+    //
+    MappedEvent(InstrumentId id,
+                MappedEventType type,
+                MidiByte pitch,
+                MidiByte velocity,
+                const RealTime &absTime,
+                const RealTime &duration,
+                const RealTime &audioStartMarker):
+        m_trackId(0),
+        m_instrument(id),
+        m_type(type),
+        m_data1(pitch),
+        m_data2(velocity),
+        m_eventTime(absTime),
+        m_duration(duration),
+        m_audioStartMarker(audioStartMarker),
+        m_dataBlockId(0),
+        m_isPersistent(false),
+        m_runtimeSegmentId(-1),
+        m_autoFade(false),
+        m_fadeInTime(RealTime::zeroTime),
+        m_fadeOutTime(RealTime::zeroTime),
+        m_recordedChannel(0),
+        m_recordedDevice(0) {}
+
+    // Audio MappedEvent shortcut constructor
+    //
+    MappedEvent(InstrumentId id,
+                unsigned short audioID,
+                const RealTime &eventTime,
+                const RealTime &duration,
+                const RealTime &audioStartMarker):
+         m_trackId(0),
+         m_instrument(id),
+         m_type(Audio),
+         m_data1(audioID % 256),
+         m_data2(audioID / 256),
+         m_eventTime(eventTime),
+         m_duration(duration),
+         m_audioStartMarker(audioStartMarker),
+         m_dataBlockId(0),
+         m_isPersistent(false),
+         m_runtimeSegmentId(-1),
+         m_autoFade(false),
+         m_fadeInTime(RealTime::zeroTime),
+         m_fadeOutTime(RealTime::zeroTime),
+         m_recordedChannel(0),
+         m_recordedDevice(0) {}
+
+    // More generalised MIDI event containers for
+    // large and small events (one param, two param)
+    //
+    MappedEvent(InstrumentId id,
+                MappedEventType type,
+                MidiByte data1,
+                MidiByte data2):
+         m_trackId(0),
+         m_instrument(id),
+         m_type(type),
+         m_data1(data1),
+         m_data2(data2),
+         m_eventTime(RealTime(0, 0)),
+         m_duration(RealTime(0, 0)),
+         m_audioStartMarker(RealTime(0, 0)),
+         m_dataBlockId(0),
+         m_isPersistent(false),
+         m_runtimeSegmentId(-1),
+         m_autoFade(false),
+         m_fadeInTime(RealTime::zeroTime),
+         m_fadeOutTime(RealTime::zeroTime),
+         m_recordedChannel(0),
+         m_recordedDevice(0) {}
+
+    MappedEvent(InstrumentId id,
+                MappedEventType type,
+                MidiByte data1):
+        m_trackId(0),
+        m_instrument(id),
+        m_type(type),
+        m_data1(data1),
+        m_data2(0),
+        m_eventTime(RealTime(0, 0)),
+        m_duration(RealTime(0, 0)),
+        m_audioStartMarker(RealTime(0, 0)),
+        m_dataBlockId(0),
+        m_isPersistent(false),
+        m_runtimeSegmentId(-1),
+        m_autoFade(false),
+        m_fadeInTime(RealTime::zeroTime),
+        m_fadeOutTime(RealTime::zeroTime),
+        m_recordedChannel(0),
+        m_recordedDevice(0) {}
+
+
+    // Construct SysExs say
+    //
+    MappedEvent(InstrumentId id,
+                MappedEventType type):
+        m_trackId(0),
+        m_instrument(id),
+        m_type(type),
+        m_data1(0),
+        m_data2(0),
+        m_eventTime(RealTime(0, 0)),
+        m_duration(RealTime(0, 0)),
+        m_audioStartMarker(RealTime(0, 0)),
+        m_dataBlockId(0),
+        m_isPersistent(false),
+        m_runtimeSegmentId(-1),
+        m_autoFade(false),
+        m_fadeInTime(RealTime::zeroTime),
+        m_fadeOutTime(RealTime::zeroTime),
+        m_recordedChannel(0),
+        m_recordedDevice(0) {}
+
+    // Copy constructor
+    //
+    // Fix for 674731 by Pedro Lopez-Cabanillas (20030531)
+    MappedEvent(const MappedEvent &mE):
+        m_trackId(mE.getTrackId()),
+        m_instrument(mE.getInstrument()),
+        m_type(mE.getType()),
+        m_data1(mE.getData1()),
+        m_data2(mE.getData2()),
+        m_eventTime(mE.getEventTime()),
+        m_duration(mE.getDuration()),
+        m_audioStartMarker(mE.getAudioStartMarker()),
+        m_dataBlockId(mE.getDataBlockId()),
+        m_isPersistent(false),
+        m_runtimeSegmentId(mE.getRuntimeSegmentId()),
+        m_autoFade(mE.isAutoFading()),
+        m_fadeInTime(mE.getFadeInTime()),
+        m_fadeOutTime(mE.getFadeOutTime()),
+        m_recordedChannel(mE.getRecordedChannel()),
+        m_recordedDevice(mE.getRecordedDevice()) {}
+
+    // Copy from pointer
+    // Fix for 674731 by Pedro Lopez-Cabanillas (20030531)
+    MappedEvent(MappedEvent *mE):
+        m_trackId(mE->getTrackId()),
+        m_instrument(mE->getInstrument()),
+        m_type(mE->getType()),
+        m_data1(mE->getData1()),
+        m_data2(mE->getData2()),
+        m_eventTime(mE->getEventTime()),
+        m_duration(mE->getDuration()),
+        m_audioStartMarker(mE->getAudioStartMarker()),
+        m_dataBlockId(mE->getDataBlockId()),
+        m_isPersistent(false),
+        m_runtimeSegmentId(mE->getRuntimeSegmentId()),
+        m_autoFade(mE->isAutoFading()),
+        m_fadeInTime(mE->getFadeInTime()),
+        m_fadeOutTime(mE->getFadeOutTime()),
+        m_recordedChannel(mE->getRecordedChannel()),
+        m_recordedDevice(mE->getRecordedDevice()) {}
+
+    // Construct perhaps without initialising, for placement new or equivalent
+    MappedEvent(bool initialise) {
+        if (initialise) *this = MappedEvent();
+    }
+
+    // Event time
+    //
+    void setEventTime(const RealTime &a) { m_eventTime = a; }
+    RealTime getEventTime() const { return m_eventTime; }
+
+    // Duration
+    //
+    void setDuration(const RealTime &d) { m_duration = d; }
+    RealTime getDuration() const { return m_duration; }
+
+    // Instrument
+    void setInstrument(InstrumentId id) { m_instrument = id; }
+    InstrumentId getInstrument() const { return m_instrument; }
+
+    // Track
+    void setTrackId(TrackId id) { m_trackId = id; }
+    TrackId getTrackId() const { return m_trackId; }
+
+    MidiByte getPitch() const { return m_data1; }
+
+    // Keep pitch within MIDI limits
+    //
+    void setPitch(MidiByte p)
+    {
+        m_data1 = p;
+        if (m_data1 > MidiMaxValue) m_data1 = MidiMaxValue;
+    }
+
+    void setVelocity(MidiByte v) { m_data2 = v; }
+    MidiByte getVelocity() const { return m_data2; }
+
+    // And the trendy names for them
+    //
+    MidiByte getData1() const { return m_data1; }
+    MidiByte getData2() const { return m_data2; }
+    void setData1(MidiByte d1) { m_data1 = d1; }
+    void setData2(MidiByte d2) { m_data2 = d2; }
+
+    void setAudioID(unsigned short id) { m_data1 = id % 256; m_data2 = id / 256; }
+    int getAudioID() const { return m_data1 + 256 * m_data2; }
+
+    // A sample doesn't have to be played from the beginning.  When
+    // passing an Audio event this value may be set to indicate from
+    // where in the sample it should be played.  Duration is measured
+    // against total sounding length (not absolute position).
+    //
+    void setAudioStartMarker(const RealTime &aS)
+        { m_audioStartMarker = aS; }
+    RealTime getAudioStartMarker() const
+        { return m_audioStartMarker; }
+
+    MappedEventType getType() const { return m_type; }
+    void setType(const MappedEventType &value) { m_type = value; }
+
+    // Data block id
+    //
+    DataBlockRepository::blockid getDataBlockId() const { return m_dataBlockId; }
+    void setDataBlockId(DataBlockRepository::blockid dataBlockId) { m_dataBlockId = dataBlockId; }
+    
+    // How MappedEvents are ordered in the MappedComposition
+    //
+    struct MappedEventCmp
+    {
+        bool operator()(const MappedEvent *mE1, const MappedEvent *mE2) const
+        {
+            return *mE1 < *mE2;
+        }
+    };
+
+    friend bool operator<(const MappedEvent &a, const MappedEvent &b);
+
+    MappedEvent& operator=(const MappedEvent &mE);
+
+    friend QDataStream& operator>>(QDataStream &dS, MappedEvent *mE);
+    friend QDataStream& operator<<(QDataStream &dS, MappedEvent *mE);
+    friend QDataStream& operator>>(QDataStream &dS, MappedEvent &mE);
+    friend QDataStream& operator<<(QDataStream &dS, const MappedEvent &mE);
+
+    /// Add a single byte to the event's datablock (for SysExs)
+    void addDataByte(MidiByte byte);
+    /// Add several bytes to the event's datablock
+    void addDataString(const std::string& data);
+
+    void setPersistent(bool value) { m_isPersistent = value; }
+    bool isPersistent() const { return m_isPersistent; }
+
+    /// Size of a MappedEvent in a stream
+    static const size_t streamedSize;
+
+    // The runtime segment id of an audio file
+    //
+    int getRuntimeSegmentId() const { return m_runtimeSegmentId; }
+    void setRuntimeSegmentId(int id) { m_runtimeSegmentId = id; }
+
+    bool isAutoFading() const { return m_autoFade; }
+    void setAutoFade(bool value) { m_autoFade = value; }
+
+    RealTime getFadeInTime() const { return m_fadeInTime; }
+    void setFadeInTime(const RealTime &time)
+            { m_fadeInTime = time; }
+
+    RealTime getFadeOutTime() const { return m_fadeOutTime; }
+    void setFadeOutTime(const RealTime &time)
+            { m_fadeOutTime = time; }
+    
+    // Original event input channel as it was recorded
+    //
+    unsigned int getRecordedChannel() const { return m_recordedChannel; }
+    void setRecordedChannel(const unsigned int channel) 
+            { m_recordedChannel = channel; }
+            
+    // Original event record device as it was recorded
+    //
+    unsigned int getRecordedDevice() const { return m_recordedDevice; }
+    void setRecordedDevice(const unsigned int device) { m_recordedDevice = device; }
+
+private:
+    TrackId          m_trackId;
+    InstrumentId     m_instrument;
+    MappedEventType  m_type;
+    MidiByte         m_data1;
+    MidiByte         m_data2;
+    RealTime         m_eventTime;
+    RealTime         m_duration;
+    RealTime         m_audioStartMarker;
+
+    // Use this when we want to store something in addition to the
+    // other bytes in this type, e.g. System Exclusive.
+    //
+    DataBlockRepository::blockid m_dataBlockId;
+
+    // Should a MappedComposition try and delete this MappedEvent or
+    // if it persistent?
+    //
+    bool             m_isPersistent;
+
+
+    // Id of the segment that this (audio) event is derived from
+    //
+    int              m_runtimeSegmentId;
+
+    // Audio autofading
+    //
+    bool                  m_autoFade;
+    RealTime  m_fadeInTime;
+    RealTime  m_fadeOutTime;
+
+    // input event original data,
+    // stored as it was recorded
+    //
+    unsigned int          m_recordedChannel;
+    unsigned int          m_recordedDevice;
+};
+
+
+}
+
+#endif
diff --git a/src/sound/MappedInstrument.cpp b/src/sound/MappedInstrument.cpp
new file mode 100644
index 0000000..b353f78
--- /dev/null
+++ b/src/sound/MappedInstrument.cpp
@@ -0,0 +1,153 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+ 
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+ 
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+ 
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include "MappedInstrument.h"
+
+namespace Rosegarden
+{
+
+
+MappedInstrument::MappedInstrument():
+        m_type(Instrument::Midi),
+        m_channel(0),
+        m_id(0),
+        m_name(std::string("")),
+        m_audioChannels(0)
+{}
+
+MappedInstrument::MappedInstrument(Instrument::InstrumentType type,
+                                   MidiByte channel,
+                                   InstrumentId id):
+        m_type(type),
+        m_channel(channel),
+        m_id(id),
+        m_name(std::string("")),
+        m_audioChannels(0)
+{}
+
+MappedInstrument::MappedInstrument(Instrument::InstrumentType type,
+                                   MidiByte channel,
+                                   InstrumentId id,
+                                   const std::string &name,
+                                   DeviceId device):
+        m_type(type),
+        m_channel(channel),
+        m_id(id),
+        m_name(name),
+        m_device(device),
+        m_audioChannels(0)
+{}
+
+MappedInstrument::MappedInstrument(const Instrument &instr):
+        m_type(instr.getType()),
+        m_channel(instr.getMidiChannel()),
+        m_id(instr.getId()),
+        m_name(instr.getName()),
+        m_device((instr.getDevice())->getId()),
+        m_audioChannels(instr.getAudioChannels())
+{}
+
+MappedInstrument::MappedInstrument(Instrument *instr):
+        m_type(instr->getType()),
+        m_channel(instr->getMidiChannel()),
+        m_id(instr->getId()),
+        m_name(instr->getName()),
+        m_device(instr->getDevice()->getId()),
+        m_audioChannels(instr->getAudioChannels())
+{}
+
+QDataStream&
+operator>>(QDataStream &dS, MappedInstrument *mI)
+{
+    unsigned int type, channel, id, device, audioChannels;
+    QString name;
+
+    dS >> type;
+    dS >> channel;
+    dS >> id;
+    dS >> name;
+    dS >> device;
+    dS >> audioChannels;
+
+    mI->setType(Instrument::InstrumentType(type));
+    mI->setChannel(MidiByte(channel));
+    mI->setId(InstrumentId(id));
+    mI->setName(std::string(name.data()));
+    mI->setDevice(DeviceId(device));
+    mI->setAudioChannels(audioChannels);
+
+    return dS;
+}
+
+QDataStream&
+operator>>(QDataStream &dS, MappedInstrument &mI)
+{
+    unsigned int type, channel, id, device, audioChannels;
+    QString name;
+
+    dS >> type;
+    dS >> channel;
+    dS >> id;
+    dS >> name;
+    dS >> device;
+    dS >> audioChannels;
+
+    mI.setType(Instrument::InstrumentType(type));
+    mI.setChannel(MidiByte(channel));
+    mI.setId(InstrumentId(id));
+    mI.setName(std::string(name.data()));
+    mI.setDevice(DeviceId(device));
+    mI.setAudioChannels(audioChannels);
+
+    return dS;
+}
+
+QDataStream&
+operator<<(QDataStream &dS, MappedInstrument *mI)
+{
+    dS << (unsigned int)mI->getType();
+    dS << (unsigned int)mI->getChannel();
+    dS << (unsigned int)mI->getId();
+    ;
+    dS << QString(mI->getName().c_str());
+    dS << (unsigned int)mI->getDevice();
+    dS << (unsigned int)mI->getAudioChannels();
+
+    return dS;
+}
+
+
+QDataStream&
+operator<<(QDataStream &dS, const MappedInstrument &mI)
+{
+    dS << (unsigned int)mI.getType();
+    dS << (unsigned int)mI.getChannel();
+    dS << (unsigned int)mI.getId();
+    ;
+    dS << QString(mI.getName().c_str());
+    dS << (unsigned int)mI.getDevice();
+    dS << (unsigned int)mI.getAudioChannels();
+
+    return dS;
+}
+
+}
+
diff --git a/src/sound/MappedInstrument.h b/src/sound/MappedInstrument.h
new file mode 100644
index 0000000..49f0167
--- /dev/null
+++ b/src/sound/MappedInstrument.h
@@ -0,0 +1,106 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include "Instrument.h"
+#include "MappedDevice.h"
+#include "MappedCommon.h"
+
+#ifndef _MAPPEDINSTRUMENT_H_
+#define _MAPPEDINSTRUMENT_H_
+
+// A scaled-down version of an Instrument that we keep Sequencer
+// side.  IDs match with those on the GUI.
+//
+//
+
+namespace Rosegarden
+{
+
+class MappedInstrument
+{
+public:
+
+    MappedInstrument();
+
+    // GUI uses this constructor because it already knows
+    // the name of the Instrument
+    //
+    MappedInstrument(Instrument::InstrumentType type,
+                     MidiByte channel,
+                     InstrumentId id);
+
+    // Driver uses this constructor (because the gui will want
+    // to know the name)
+    //
+    MappedInstrument(Instrument::InstrumentType type,
+                     MidiByte channel,
+                     InstrumentId id,
+                     const std::string &name,
+                     DeviceId device);
+
+    // from instrument
+    MappedInstrument(const Instrument &instrument);
+    MappedInstrument(Instrument *instrument);
+
+    ~MappedInstrument() { ;}
+
+    void setId(InstrumentId id) { m_id = id; }
+    InstrumentId getId() const { return m_id; }
+
+    void setChannel(MidiByte channel) { m_channel = channel; }
+    MidiByte getChannel() const { return m_channel; }
+
+    void setType(Instrument::InstrumentType type) { m_type = type; }
+    Instrument::InstrumentType getType() const { return m_type; }
+
+    void setName(const std::string &name) { m_name = name; }
+    const std::string& getName() const { return m_name; }
+
+    void setDevice(DeviceId device) { m_device = device; }
+    DeviceId getDevice() const { return m_device; }
+
+    // How many audio channels we've got on this audio MappedInstrument
+    //
+    unsigned int getAudioChannels() const { return m_audioChannels; }
+    void setAudioChannels(unsigned int channels) { m_audioChannels = channels; }
+
+    friend QDataStream& operator>>(QDataStream &dS, MappedInstrument *mI);
+    friend QDataStream& operator<<(QDataStream &dS, MappedInstrument *mI);
+    friend QDataStream& operator>>(QDataStream &dS, MappedInstrument &mI);
+    friend QDataStream& operator<<(QDataStream &dS, const MappedInstrument &mI);
+
+private:
+
+    Instrument::InstrumentType  m_type;
+    MidiByte                    m_channel;
+    InstrumentId                m_id;
+    std::string                 m_name;
+    DeviceId                    m_device;
+
+    // If this is an audio MappedInstrument then how many channels
+    // are associated with it?
+    //
+    unsigned int                m_audioChannels;
+};
+
+}
+
+#endif // _MAPPEDINSTRUMENT_H_
diff --git a/src/sound/MappedRealTime.cpp b/src/sound/MappedRealTime.cpp
new file mode 100644
index 0000000..ba596fd
--- /dev/null
+++ b/src/sound/MappedRealTime.cpp
@@ -0,0 +1,62 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+ 
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+ 
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+ 
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include "MappedRealTime.h"
+
+namespace Rosegarden
+{
+
+QDataStream&
+operator>>(QDataStream &dS, MappedRealTime *mRT)
+{
+    dS >> mRT->sec;
+    dS >> mRT->nsec;
+    return dS;
+}
+
+QDataStream&
+operator<<(QDataStream &dS, MappedRealTime *mRT)
+{
+    dS << mRT->sec;
+    dS << mRT->nsec;
+    return dS;
+}
+
+QDataStream&
+operator>>(QDataStream &dS, MappedRealTime &mRT)
+{
+    dS >> mRT.sec;
+    dS >> mRT.nsec;
+    return dS;
+}
+
+
+QDataStream&
+operator<<(QDataStream &dS, const MappedRealTime &mRT)
+{
+    dS << mRT.sec;
+    dS << mRT.nsec;
+    return dS;
+}
+
+
+}
+
diff --git a/src/sound/MappedRealTime.h b/src/sound/MappedRealTime.h
new file mode 100644
index 0000000..ca2afd0
--- /dev/null
+++ b/src/sound/MappedRealTime.h
@@ -0,0 +1,56 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include <qdatastream.h>
+#include "RealTime.h"
+
+// Just a DCOP wrapper to RealTime
+//
+
+#ifndef _MAPPEDREALTIME_H_
+#define _MAPPEDREALTIME_H_
+
+namespace Rosegarden
+{
+
+class MappedRealTime : public RealTime
+{
+public:
+    MappedRealTime() : RealTime(0, 0) {;}
+    MappedRealTime(const RealTime &t) : RealTime(t.sec, t.nsec) {;}
+
+    // Return as RealTime
+    RealTime getRealTime() { return RealTime(sec, nsec); }
+
+    // DCOP datastream
+    //
+    friend QDataStream& operator>>(QDataStream &dS, MappedRealTime *mRT);
+    friend QDataStream& operator<<(QDataStream &dS, MappedRealTime *mRT);
+    friend QDataStream& operator>>(QDataStream &dS, MappedRealTime &mRT);
+    friend QDataStream& operator<<(QDataStream &dS, const MappedRealTime &mRT);
+
+
+};
+
+}
+
+#endif // _MAPPEDREALTIME_H_
+
diff --git a/src/sound/MappedStudio.cpp b/src/sound/MappedStudio.cpp
new file mode 100644
index 0000000..4b35122
--- /dev/null
+++ b/src/sound/MappedStudio.cpp
@@ -0,0 +1,1719 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include <iostream>
+
+#include "MappedStudio.h"
+#include "SoundDriver.h"
+#include "PluginFactory.h"
+
+#include <pthread.h> // for mutex
+
+//#define DEBUG_MAPPEDSTUDIO 1
+
+namespace Rosegarden
+{
+
+static pthread_mutex_t _mappedObjectContainerLock;
+
+#ifdef DEBUG_MAPPEDSTUDIO
+static int _approxLockCount = 0;
+#endif
+
+static inline void getLock(const char *file, int line)
+{
+#ifdef DEBUG_MAPPEDSTUDIO
+    std::cerr << "Acquiring MappedStudio container lock at " << file << ":" << line << ": count " << _approxLockCount++ << std::endl;
+#endif
+
+    pthread_mutex_lock(&_mappedObjectContainerLock);
+}
+
+static inline void releaseLock(const char *file, int line)
+{
+    pthread_mutex_unlock(&_mappedObjectContainerLock);
+#ifdef DEBUG_MAPPEDSTUDIO
+
+    std::cerr << "Released container lock at " << file << ":" << line << ": count " << --_approxLockCount << std::endl;
+#endif
+}
+
+#define GET_LOCK getLock(__FILE__,__LINE__)
+#define RELEASE_LOCK releaseLock(__FILE__,__LINE__)
+
+// These stream functions are stolen and adapted from Qt3 qvaluevector.h
+//
+// ** Copyright (C) 1992-2000 Trolltech AS.  All rights reserved.
+//
+QDataStream& operator>>(QDataStream& s, MappedObjectIdList& v)
+{
+    v.clear();
+    Q_UINT32 c;
+    s >> c;
+    v.resize(c);
+    for (Q_UINT32 i = 0; i < c; ++i) {
+        MappedObjectId t;
+        s >> t;
+        v[i] = t;
+    }
+    return s;
+}
+
+QDataStream& operator<<(QDataStream& s, const MappedObjectIdList& v)
+{
+    s << (Q_UINT32)v.size();
+    MappedObjectIdList::const_iterator it = v.begin();
+    for ( ; it != v.end(); ++it )
+        s << *it;
+    return s;
+}
+
+QDataStream& operator>>(QDataStream& s, MappedObjectPropertyList& v)
+{
+    v.clear();
+    Q_UINT32 c;
+    s >> c;
+    v.resize(c);
+    for (Q_UINT32 i = 0; i < c; ++i) {
+        MappedObjectProperty t;
+        s >> t;
+        v[i] = t;
+    }
+    return s;
+}
+
+QDataStream& operator<<(QDataStream& s, const MappedObjectPropertyList& v)
+{
+    s << (Q_UINT32)v.size();
+    MappedObjectPropertyList::const_iterator it = v.begin();
+    for ( ; it != v.end(); ++it )
+        s << *it;
+    return s;
+}
+
+QDataStream& operator>>(QDataStream& s, MappedObjectValueList& v)
+{
+    v.clear();
+    Q_UINT32 c;
+    s >> c;
+    v.resize(c);
+    for (Q_UINT32 i = 0; i < c; ++i) {
+        MappedObjectValue t;
+        s >> t;
+        v[i] = t;
+    }
+    return s;
+}
+
+QDataStream& operator<<(QDataStream& s, const MappedObjectValueList& v)
+{
+    s << (Q_UINT32)v.size();
+    MappedObjectValueList::const_iterator it = v.begin();
+    for ( ; it != v.end(); ++it )
+        s << *it;
+    return s;
+}
+
+// Define our object properties - these can be queried and set.
+//
+
+// General things
+//
+const MappedObjectProperty MappedObject::Name = "name";
+const MappedObjectProperty MappedObject::Instrument = "instrument";
+const MappedObjectProperty MappedObject::Position = "position";
+
+const MappedObjectProperty MappedConnectableObject::ConnectionsIn = "connectionsIn";
+const MappedObjectProperty MappedConnectableObject::ConnectionsOut = "connectionsOut";
+
+const MappedObjectProperty MappedAudioFader::Channels = "channels";
+const MappedObjectProperty MappedAudioFader::FaderLevel = "faderLevel";
+const MappedObjectProperty MappedAudioFader::FaderRecordLevel = "faderRecordLevel";
+const MappedObjectProperty MappedAudioFader::Pan = "pan";
+const MappedObjectProperty MappedAudioFader::InputChannel = "inputChannel";
+
+const MappedObjectProperty MappedAudioBuss::BussId = "bussId";
+const MappedObjectProperty MappedAudioBuss::Level = "level";
+const MappedObjectProperty MappedAudioBuss::Pan = "pan";
+
+const MappedObjectProperty MappedAudioInput::InputNumber = "inputNumber";
+
+const MappedObjectProperty MappedPluginSlot::Identifier = "identifier";
+const MappedObjectProperty MappedPluginSlot::PluginName = "pluginname";
+const MappedObjectProperty MappedPluginSlot::Label = "label";
+const MappedObjectProperty MappedPluginSlot::Author = "author";
+const MappedObjectProperty MappedPluginSlot::Copyright = "copyright";
+const MappedObjectProperty MappedPluginSlot::Category = "category";
+const MappedObjectProperty MappedPluginSlot::PortCount = "portcount";
+const MappedObjectProperty MappedPluginSlot::Ports = "ports";
+const MappedObjectProperty MappedPluginSlot::Instrument = "instrument";
+const MappedObjectProperty MappedPluginSlot::Position = "position";
+const MappedObjectProperty MappedPluginSlot::Bypassed = "bypassed";
+const MappedObjectProperty MappedPluginSlot::Programs = "programs";
+const MappedObjectProperty MappedPluginSlot::Program = "program";
+const MappedObjectProperty MappedPluginSlot::Configuration = "configuration";
+
+const MappedObjectProperty MappedPluginPort::PortNumber = "portnumber";
+const MappedObjectProperty MappedPluginPort::Name = "name";
+const MappedObjectProperty MappedPluginPort::Minimum = "minimum";
+const MappedObjectProperty MappedPluginPort::Maximum = "maximum";
+const MappedObjectProperty MappedPluginPort::Default = "default";
+const MappedObjectProperty MappedPluginPort::DisplayHint = "displayhint";
+const MappedObjectProperty MappedPluginPort::Value = "value";
+
+// --------- MappedObject ---------
+//
+
+void
+MappedObject::addChild(MappedObject *object)
+{
+    std::vector<MappedObject*>::iterator it = m_children.begin();
+    for (; it != m_children.end(); it++)
+        if ((*it) == object)
+            return ;
+
+    m_children.push_back(object);
+}
+
+void
+MappedObject::removeChild(MappedObject *object)
+{
+    std::vector<MappedObject*>::iterator it = m_children.begin();
+    for (; it != m_children.end(); it++) {
+        if ((*it) == object) {
+            m_children.erase(it);
+            return ;
+        }
+    }
+}
+
+// Return all child ids
+//
+MappedObjectPropertyList
+MappedObject::getChildren()
+{
+    MappedObjectPropertyList list;
+    std::vector<MappedObject*>::iterator it = m_children.begin();
+    for (; it != m_children.end(); it++)
+        list.push_back(QString("%1").arg((*it)->getId()));
+
+    return list;
+}
+
+
+// Return all child ids of a certain type
+//
+MappedObjectPropertyList
+MappedObject::getChildren(MappedObjectType type)
+{
+    MappedObjectPropertyList list;
+    std::vector<MappedObject*>::iterator it = m_children.begin();
+    for (; it != m_children.end(); it++) {
+        if ((*it)->getType() == type)
+            list.push_back(QString("%1").arg((*it)->getId()));
+    }
+
+    return list;
+}
+
+void
+MappedObject::destroyChildren()
+{
+    // remove references from the studio as well as from the object
+    MappedObject *studioObject = getParent();
+    while (!dynamic_cast<MappedStudio*>(studioObject))
+        studioObject = studioObject->getParent();
+
+    // see note in destroy() below
+
+    std::vector<MappedObject *> children = m_children;
+    m_children.clear();
+
+    std::vector<MappedObject *>::iterator it = children.begin();
+    for (; it != children.end(); it++)
+        (*it)->destroy(); // remove from studio and destroy
+}
+
+// Destroy this object and remove it from the studio and
+// do the same for all its children.
+//
+void
+MappedObject::destroy()
+{
+    MappedObject *studioObject = getParent();
+    while (!dynamic_cast<MappedStudio*>(studioObject))
+        studioObject = studioObject->getParent();
+
+    MappedStudio *studio = dynamic_cast<MappedStudio*>(studioObject);
+
+    // The destroy method on each child calls studio->clearObject,
+    // which calls back on the parent (in this case us) to remove the
+    // child.  (That's necessary for the case of destroying a plugin,
+    // where we need to remove it from its plugin manager -- etc.)  So
+    // we don't want to be iterating over m_children here, as it will
+    // change from under us.
+
+    std::vector<MappedObject *> children = m_children;
+    m_children.clear();
+
+    std::vector<MappedObject *>::iterator it = children.begin();
+    for (; it != children.end(); it++) {
+        (*it)->destroy();
+    }
+
+    (void)studio->clearObject(m_id);
+    delete this;
+}
+
+
+// ------- MappedStudio -------
+//
+
+MappedStudio::MappedStudio() :
+        MappedObject(0,
+                     "MappedStudio",
+                     Studio,
+                     0),
+        m_runningObjectId(1)
+{
+    pthread_mutexattr_t attr;
+    pthread_mutexattr_init(&attr);
+#ifdef HAVE_PTHREAD_MUTEX_RECURSIVE
+
+    pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE);
+#else
+#ifdef PTHREAD_MUTEX_RECURSIVE
+
+    pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE);
+#else
+
+    pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE_NP);
+#endif
+#endif
+
+    pthread_mutex_init(&_mappedObjectContainerLock, &attr);
+}
+
+MappedStudio::~MappedStudio()
+{
+#ifdef DEBUG_MAPPEDSTUDIO
+    std::cout << "MappedStudio::~MappedStudio" << std::endl;
+#endif
+
+    clear();
+}
+
+
+// Object factory
+//
+MappedObject*
+MappedStudio::createObject(MappedObjectType type)
+{
+    GET_LOCK;
+
+    MappedObject *mO = 0;
+
+    // Ensure we've got an empty slot
+    //
+    while (getObjectById(m_runningObjectId))
+        m_runningObjectId++;
+
+    mO = createObject(type, m_runningObjectId);
+
+    // If we've got a new object increase the running id
+    //
+    if (mO)
+        m_runningObjectId++;
+
+    RELEASE_LOCK;
+    return mO;
+}
+
+MappedObject*
+MappedStudio::createObject(MappedObjectType type,
+                           MappedObjectId id)
+{
+    GET_LOCK;
+
+    // fail if the object already exists and it's not zero
+    if (id != 0 && getObjectById(id)) {
+        RELEASE_LOCK;
+        return 0;
+    }
+
+    MappedObject *mO = 0;
+
+    if (type == MappedObject::AudioFader) {
+        mO = new MappedAudioFader(this,
+                                  id,
+                                  2); // channels
+
+        // push to the studio's child stack
+        addChild(mO);
+    } else if (type == MappedObject::AudioBuss) {
+        mO = new MappedAudioBuss(this,
+                                 id);
+
+        // push to the studio's child stack
+        addChild(mO);
+    } else if (type == MappedObject::AudioInput) {
+        mO = new MappedAudioInput(this,
+                                  id);
+
+        // push to the studio's child stack
+        addChild(mO);
+    } else if (type == MappedObject::PluginSlot) {
+        mO = new MappedPluginSlot(this,
+                                  id);
+        addChild(mO);
+    } else if (type == MappedObject::PluginPort) {
+        mO = new MappedPluginPort(this,
+                                  id);
+        // reset the port's parent after creation outside this method
+    }
+
+    // Insert
+    if (mO) {
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "Adding object " << id << " to category " << type << std::endl;
+#endif
+
+        m_objects[type][id] = mO;
+    }
+
+    RELEASE_LOCK;
+
+    return mO;
+}
+
+MappedObject*
+MappedStudio::getObjectOfType(MappedObjectType type)
+{
+    MappedObject *rv = 0;
+
+    GET_LOCK;
+
+    MappedObjectCategory &category = m_objects[type];
+    if (!category.empty())
+        rv = category.begin()->second;
+
+    RELEASE_LOCK;
+
+    return rv;
+}
+
+std::vector<MappedObject *>
+MappedStudio::getObjectsOfType(MappedObjectType type)
+{
+    std::vector<MappedObject *> rv;
+
+    GET_LOCK;
+
+    MappedObjectCategory &category = m_objects[type];
+
+    for (MappedObjectCategory::iterator i = category.begin();
+            i != category.end(); ++i) {
+        rv.push_back(i->second);
+    }
+
+    RELEASE_LOCK;
+
+    return rv;
+}
+
+unsigned int
+MappedStudio::getObjectCount(MappedObjectType type)
+{
+    unsigned int count = 0;
+
+    GET_LOCK;
+
+    MappedObjectCategory &category = m_objects[type];
+    count = category.size();
+
+    RELEASE_LOCK;
+
+    return count;
+}
+
+
+bool
+MappedStudio::destroyObject(MappedObjectId id)
+{
+    GET_LOCK;
+
+    MappedObject *obj = getObjectById(id);
+
+    bool rv = false;
+
+    if (obj) {
+        obj->destroy();
+        rv = true;
+    }
+
+    RELEASE_LOCK;
+
+    return rv;
+}
+
+bool
+MappedStudio::connectObjects(MappedObjectId mId1, MappedObjectId mId2)
+{
+    GET_LOCK;
+
+    bool rv = false;
+
+    // objects must exist and be of connectable types
+    MappedConnectableObject *obj1 =
+        dynamic_cast<MappedConnectableObject *>(getObjectById(mId1));
+    MappedConnectableObject *obj2 =
+        dynamic_cast<MappedConnectableObject *>(getObjectById(mId2));
+
+    if (obj1 && obj2) {
+        obj1->addConnection(MappedConnectableObject::Out, mId2);
+        obj2->addConnection(MappedConnectableObject::In, mId1);
+        rv = true;
+    }
+
+    RELEASE_LOCK;
+
+    return rv;
+}
+
+bool
+MappedStudio::disconnectObjects(MappedObjectId mId1, MappedObjectId mId2)
+{
+    GET_LOCK;
+
+    bool rv = false;
+
+    // objects must exist and be of connectable types
+    MappedConnectableObject *obj1 =
+        dynamic_cast<MappedConnectableObject *>(getObjectById(mId1));
+    MappedConnectableObject *obj2 =
+        dynamic_cast<MappedConnectableObject *>(getObjectById(mId2));
+
+    if (obj1 && obj2) {
+        obj1->removeConnection(MappedConnectableObject::Out, mId2);
+        obj2->removeConnection(MappedConnectableObject::In, mId1);
+        rv = true;
+    }
+
+    RELEASE_LOCK;
+
+    return rv;
+}
+
+bool
+MappedStudio::disconnectObject(MappedObjectId mId)
+{
+    GET_LOCK;
+
+    bool rv = false;
+
+    MappedConnectableObject *obj =
+        dynamic_cast<MappedConnectableObject *>(getObjectById(mId));
+
+    if (obj) {
+        while (1) {
+            MappedObjectValueList list =
+                obj->getConnections(MappedConnectableObject::In);
+            if (list.empty())
+                break;
+            MappedObjectId otherId = MappedObjectId(*list.begin());
+            disconnectObjects(otherId, mId);
+        }
+        while (1) {
+            MappedObjectValueList list =
+                obj->getConnections(MappedConnectableObject::Out);
+            if (list.empty())
+                break;
+            MappedObjectId otherId = MappedObjectId(*list.begin());
+            disconnectObjects(mId, otherId);
+        }
+    }
+
+    rv = true;
+
+    RELEASE_LOCK;
+
+    return rv;
+}
+
+
+
+// Clear down the whole studio
+//
+void
+MappedStudio::clear()
+{
+    GET_LOCK;
+
+    for (MappedObjectMap::iterator i = m_objects.begin();
+            i != m_objects.end(); ++i) {
+
+        for (MappedObjectCategory::iterator j = i->second.begin();
+                j != i->second.end(); ++j) {
+
+            delete j->second;
+        }
+    }
+
+    m_objects.clear();
+
+    // reset running object id
+    m_runningObjectId = 1;
+
+    RELEASE_LOCK;
+}
+
+bool
+MappedStudio::clearObject(MappedObjectId id)
+{
+    bool rv = false;
+
+    GET_LOCK;
+
+    for (MappedObjectMap::iterator i = m_objects.begin();
+            i != m_objects.end(); ++i) {
+
+        MappedObjectCategory::iterator j = i->second.find(id);
+        if (j != i->second.end()) {
+            // if the object has a parent other than the studio,
+            // persuade that parent to abandon it
+            MappedObject *parent = j->second->getParent();
+            if (parent && !dynamic_cast<MappedStudio *>(parent)) {
+                parent->removeChild(j->second);
+            }
+
+            i->second.erase(j);
+            rv = true;
+            break;
+        }
+    }
+
+    RELEASE_LOCK;
+
+    return rv;
+}
+
+
+MappedObjectPropertyList
+MappedStudio::getPropertyList(const MappedObjectProperty &property)
+{
+    MappedObjectPropertyList list;
+
+    if (property == "") {
+        // something
+    }
+
+    return list;
+}
+
+bool
+MappedStudio::getProperty(const MappedObjectProperty &,
+                          MappedObjectValue &)
+{
+    return false;
+}
+
+MappedObject*
+MappedStudio::getObjectById(MappedObjectId id)
+{
+    GET_LOCK;
+    MappedObject *rv = 0;
+
+    for (MappedObjectMap::iterator i = m_objects.begin();
+            i != m_objects.end(); ++i) {
+
+        MappedObjectCategory::iterator j = i->second.find(id);
+        if (j != i->second.end()) {
+            rv = j->second;
+            break;
+        }
+    }
+
+    RELEASE_LOCK;
+    return rv;
+}
+
+MappedObject*
+MappedStudio::getObjectByIdAndType(MappedObjectId id, MappedObjectType type)
+{
+    GET_LOCK;
+    MappedObject *rv = 0;
+
+    MappedObjectCategory &category = m_objects[type];
+    MappedObjectCategory::iterator i = category.find(id);
+    if (i != category.end()) {
+        rv = i->second;
+    }
+
+    RELEASE_LOCK;
+    return rv;
+}
+
+MappedObject*
+MappedStudio::getFirst(MappedObjectType type)
+{
+    return getObjectOfType(type);
+}
+
+MappedObject*
+MappedStudio::getNext(MappedObject *object)
+{
+    GET_LOCK;
+
+    MappedObjectCategory &category = m_objects[object->getType()];
+
+    bool next = false;
+    MappedObject *rv = 0;
+
+    for (MappedObjectCategory::iterator i = category.begin();
+            i != category.end(); ++i) {
+        if (i->second->getId() == object->getId())
+            next = true;
+        else if (next) {
+            rv = i->second;
+            break;
+        }
+    }
+
+    RELEASE_LOCK;
+    return rv;
+}
+
+void
+MappedStudio::setProperty(const MappedObjectProperty &property,
+                          MappedObjectValue /*value*/)
+{
+    if (property == "") {}
+
+}
+
+MappedAudioFader *
+MappedStudio::getAudioFader(InstrumentId id)
+{
+    GET_LOCK;
+
+    MappedObjectCategory &category = m_objects[AudioFader];
+    MappedAudioFader *rv = 0;
+
+    for (MappedObjectCategory::iterator i = category.begin();
+            i != category.end(); ++i) {
+        MappedAudioFader *fader = dynamic_cast<MappedAudioFader *>(i->second);
+        if (fader && (fader->getInstrument() == id)) {
+            rv = fader;
+            break;
+        }
+    }
+
+    RELEASE_LOCK;
+    return rv;
+}
+
+MappedAudioBuss *
+MappedStudio::getAudioBuss(int bussNumber)
+{
+    GET_LOCK;
+
+    MappedObjectCategory &category = m_objects[AudioBuss];
+    MappedAudioBuss *rv = 0;
+
+    for (MappedObjectCategory::iterator i = category.begin();
+            i != category.end(); ++i) {
+        MappedAudioBuss *buss = dynamic_cast<MappedAudioBuss *>(i->second);
+        if (buss && (buss->getBussId() == bussNumber)) {
+            rv = buss;
+            break;
+        }
+    }
+
+    RELEASE_LOCK;
+    return rv;
+}
+
+MappedAudioInput *
+MappedStudio::getAudioInput(int inputNumber)
+{
+    GET_LOCK;
+
+    MappedObjectCategory &category = m_objects[AudioInput];
+    MappedAudioInput *rv = 0;
+
+    for (MappedObjectCategory::iterator i = category.begin();
+            i != category.end(); ++i) {
+        MappedAudioInput *input = dynamic_cast<MappedAudioInput *>(i->second);
+        if (input && (input->getInputNumber() == inputNumber)) {
+            rv = input;
+            break;
+        }
+    }
+
+    RELEASE_LOCK;
+    return rv;
+}
+
+
+// -------------- MappedConnectableObject -----------------
+//
+//
+MappedConnectableObject::MappedConnectableObject(MappedObject *parent,
+        const std::string &name,
+        MappedObjectType type,
+        MappedObjectId id):
+        MappedObject(parent,
+                     name,
+                     type,
+                     id)
+{}
+
+MappedConnectableObject::~MappedConnectableObject()
+{}
+
+void
+MappedConnectableObject::setConnections(ConnectionDirection dir,
+                                        MappedObjectValueList conns)
+{
+    if (dir == In)
+        m_connectionsIn = conns;
+    else
+        m_connectionsOut = conns;
+}
+
+void
+MappedConnectableObject::addConnection(ConnectionDirection dir,
+                                       MappedObjectId id)
+{
+    MappedObjectValueList &list =
+        (dir == In ? m_connectionsIn : m_connectionsOut);
+
+    for (MappedObjectValueList::iterator i = list.begin(); i != list.end(); ++i) {
+        if (*i == id) {
+            return ;
+        }
+    }
+
+    list.push_back(MappedObjectValue(id));
+}
+
+void
+MappedConnectableObject::removeConnection(ConnectionDirection dir,
+        MappedObjectId id)
+{
+    MappedObjectValueList &list =
+        (dir == In ? m_connectionsIn : m_connectionsOut);
+
+    for (MappedObjectValueList::iterator i = list.begin(); i != list.end(); ++i) {
+        if (*i == id) {
+            list.erase(i);
+            return ;
+        }
+    }
+}
+
+MappedObjectValueList
+MappedConnectableObject::getConnections(ConnectionDirection dir)
+{
+    if (dir == In)
+        return m_connectionsIn;
+    else
+        return m_connectionsOut;
+}
+
+
+// ------------ MappedAudioFader ----------------
+//
+MappedAudioFader::MappedAudioFader(MappedObject *parent,
+                                   MappedObjectId id,
+                                   MappedObjectValue channels):
+        MappedConnectableObject(parent,
+                                "MappedAudioFader",
+                                AudioFader,
+                                id),
+        m_level(0.0),  // dB
+        m_recordLevel(0.0),
+        m_instrumentId(0),
+        m_pan(0),
+        m_channels(channels),
+        m_inputChannel(0)
+{}
+
+MappedAudioFader::~MappedAudioFader()
+{}
+
+
+MappedObjectPropertyList
+MappedAudioFader::getPropertyList(const MappedObjectProperty &property)
+{
+    MappedObjectPropertyList list;
+
+    if (property == "") {
+        list.push_back(MappedAudioFader::FaderLevel);
+        list.push_back(MappedAudioFader::FaderRecordLevel);
+        list.push_back(MappedObject::Instrument);
+        list.push_back(MappedAudioFader::Pan);
+        list.push_back(MappedAudioFader::Channels);
+        list.push_back(MappedConnectableObject::ConnectionsIn);
+        list.push_back(MappedConnectableObject::ConnectionsOut);
+    } else if (property == MappedObject::Instrument) {
+        list.push_back(MappedObjectProperty("%1").arg(m_instrumentId));
+    } else if (property == MappedAudioFader::FaderLevel) {
+        list.push_back(MappedObjectProperty("%1").arg(m_level));
+    } else if (property == MappedAudioFader::FaderRecordLevel) {
+        list.push_back(MappedObjectProperty("%1").arg(m_recordLevel));
+    } else if (property == MappedAudioFader::Channels) {
+        list.push_back(MappedObjectProperty("%1").arg(m_channels));
+    } else if (property == MappedAudioFader::InputChannel) {
+        list.push_back(MappedObjectProperty("%1").arg(m_inputChannel));
+    } else if (property == MappedAudioFader::Pan) {
+        list.push_back(MappedObjectProperty("%1").arg(m_pan));
+    } else if (property == MappedConnectableObject::ConnectionsIn) {
+        MappedObjectValueList::const_iterator
+        it = m_connectionsIn.begin();
+
+        for ( ; it != m_connectionsIn.end(); ++it) {
+            list.push_back(QString("%1").arg(*it));
+        }
+    } else if (property == MappedConnectableObject::ConnectionsOut) {
+        MappedObjectValueList::const_iterator
+        it = m_connectionsOut.begin();
+
+        for ( ; it != m_connectionsOut.end(); ++it) {
+            list.push_back(QString("%1").arg(*it));
+        }
+    }
+
+    return list;
+}
+
+bool
+MappedAudioFader::getProperty(const MappedObjectProperty &property,
+                              MappedObjectValue &value)
+{
+    if (property == FaderLevel) {
+        value = m_level;
+    } else if (property == Instrument) {
+        value = m_instrumentId;
+    } else if (property == FaderRecordLevel) {
+        value = m_recordLevel;
+    } else if (property == Channels) {
+        value = m_channels;
+    } else if (property == InputChannel) {
+        value = m_inputChannel;
+    } else if (property == Pan) {
+        value = m_pan;
+    } else {
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "MappedAudioFader::getProperty - "
+        << "unsupported or non-scalar property" << std::endl;
+#endif
+
+        return false;
+    }
+    return true;
+}
+
+void
+MappedAudioFader::setProperty(const MappedObjectProperty &property,
+                              MappedObjectValue value)
+{
+    bool updateLevels = false;
+
+    if (property == MappedAudioFader::FaderLevel) {
+        m_level = value;
+        updateLevels = true;
+    } else if (property == MappedObject::Instrument) {
+        m_instrumentId = InstrumentId(value);
+        updateLevels = true;
+    } else if (property == MappedAudioFader::FaderRecordLevel) {
+        m_recordLevel = value;
+    } else if (property == MappedAudioFader::Channels) {
+        m_channels = value;
+    } else if (property == MappedAudioFader::InputChannel) {
+        m_inputChannel = value;
+    } else if (property == MappedAudioFader::Pan) {
+        m_pan = value;
+        updateLevels = true;
+    } else if (property == MappedConnectableObject::ConnectionsIn) {
+        m_connectionsIn.clear();
+        m_connectionsIn.push_back(value);
+    } else if (property == MappedConnectableObject::ConnectionsOut) {
+        m_connectionsOut.clear();
+        m_connectionsOut.push_back(value);
+    } else {
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "MappedAudioFader::setProperty - "
+        << "unsupported property" << std::endl;
+#endif
+
+        return ;
+    }
+
+    /*
+    std::cout << "MappedAudioFader::setProperty - "
+              << property << " = " << value << std::endl;
+              */
+
+    if (updateLevels) {
+        MappedStudio *studio =
+            dynamic_cast<MappedStudio*>(getParent());
+
+        if (studio) {
+            studio->getSoundDriver()->setAudioInstrumentLevels
+            (m_instrumentId, m_level, m_pan);
+        }
+    }
+}
+
+// ---------------- MappedAudioBuss -------------------
+//
+//
+MappedAudioBuss::MappedAudioBuss(MappedObject *parent,
+                                 MappedObjectId id) :
+        MappedConnectableObject(parent,
+                                "MappedAudioBuss",
+                                AudioBuss,
+                                id),
+        m_bussId(0),
+        m_level(0),
+        m_pan(0)
+{}
+
+MappedAudioBuss::~MappedAudioBuss()
+{}
+
+MappedObjectPropertyList
+MappedAudioBuss::getPropertyList(const MappedObjectProperty &property)
+{
+    MappedObjectPropertyList list;
+
+    if (property == "") {
+        list.push_back(MappedAudioBuss::BussId);
+        list.push_back(MappedAudioBuss::Level);
+        list.push_back(MappedAudioBuss::Pan);
+        list.push_back(MappedConnectableObject::ConnectionsIn);
+        list.push_back(MappedConnectableObject::ConnectionsOut);
+    } else if (property == BussId) {
+        list.push_back(MappedObjectProperty("%1").arg(m_bussId));
+    } else if (property == Level) {
+        list.push_back(MappedObjectProperty("%1").arg(m_level));
+    } else if (property == MappedConnectableObject::ConnectionsIn) {
+        MappedObjectValueList::const_iterator
+        it = m_connectionsIn.begin();
+
+        for ( ; it != m_connectionsIn.end(); ++it) {
+            list.push_back(QString("%1").arg(*it));
+        }
+    } else if (property == MappedConnectableObject::ConnectionsOut) {
+        MappedObjectValueList::const_iterator
+        it = m_connectionsOut.begin();
+
+        for ( ; it != m_connectionsOut.end(); ++it) {
+            list.push_back(QString("%1").arg(*it));
+        }
+    }
+
+    return list;
+}
+
+bool
+MappedAudioBuss::getProperty(const MappedObjectProperty &property,
+                             MappedObjectValue &value)
+{
+    if (property == BussId) {
+        value = m_bussId;
+    } else if (property == Level) {
+        value = m_level;
+    } else if (property == Pan) {
+        value = m_pan;
+    } else {
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "MappedAudioBuss::getProperty - "
+        << "unsupported or non-scalar property" << std::endl;
+#endif
+
+        return false;
+    }
+    return true;
+}
+
+void
+MappedAudioBuss::setProperty(const MappedObjectProperty &property,
+                             MappedObjectValue value)
+{
+    bool updateLevels = false;
+
+    if (property == MappedAudioBuss::BussId) {
+        m_bussId = (int)value;
+        updateLevels = true;
+    } else if (property == MappedAudioBuss::Level) {
+        m_level = value;
+        updateLevels = true;
+    } else if (property == MappedAudioBuss::Pan) {
+        m_pan = value;
+        updateLevels = true;
+    } else if (property == MappedConnectableObject::ConnectionsIn) {
+        m_connectionsIn.clear();
+        m_connectionsIn.push_back(value);
+    } else if (property == MappedConnectableObject::ConnectionsOut) {
+        m_connectionsOut.clear();
+        m_connectionsOut.push_back(value);
+    } else {
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "MappedAudioBuss::setProperty - "
+        << "unsupported property" << std::endl;
+#endif
+
+        return ;
+    }
+
+    if (updateLevels) {
+        MappedStudio *studio =
+            dynamic_cast<MappedStudio*>(getParent());
+
+        if (studio) {
+            studio->getSoundDriver()->setAudioBussLevels
+            (m_bussId, m_level, m_pan);
+        }
+    }
+}
+
+std::vector<InstrumentId>
+MappedAudioBuss::getInstruments()
+{
+    std::vector<InstrumentId> rv;
+
+    GET_LOCK;
+
+    MappedObject *studioObject = getParent();
+    while (!dynamic_cast<MappedStudio *>(studioObject))
+        studioObject = studioObject->getParent();
+
+    std::vector<MappedObject *> objects =
+        static_cast<MappedStudio *>(studioObject)->
+        getObjectsOfType(MappedObject::AudioFader);
+
+    for (std::vector<MappedObject *>::iterator i = objects.begin();
+            i != objects.end(); ++i) {
+        MappedAudioFader *fader = dynamic_cast<MappedAudioFader *>(*i);
+        if (fader) {
+            MappedObjectValueList connections = fader->getConnections
+                                                (MappedConnectableObject::Out);
+            if (!connections.empty() && (*connections.begin() == getId())) {
+                rv.push_back(fader->getInstrument());
+            }
+        }
+    }
+
+    RELEASE_LOCK;
+
+    return rv;
+}
+
+
+// ---------------- MappedAudioInput -------------------
+//
+//
+MappedAudioInput::MappedAudioInput(MappedObject *parent,
+                                   MappedObjectId id) :
+        MappedConnectableObject(parent,
+                                "MappedAudioInput",
+                                AudioInput,
+                                id)
+{}
+
+MappedAudioInput::~MappedAudioInput()
+{}
+
+MappedObjectPropertyList
+MappedAudioInput::getPropertyList(const MappedObjectProperty &property)
+{
+    MappedObjectPropertyList list;
+
+    if (property == "") {
+        list.push_back(MappedAudioInput::InputNumber);
+    } else if (property == InputNumber) {
+        list.push_back(MappedObjectProperty("%1").arg(m_inputNumber));
+    }
+
+    return list;
+}
+
+bool
+MappedAudioInput::getProperty(const MappedObjectProperty &property,
+                              MappedObjectValue &value)
+{
+    if (property == InputNumber) {
+        value = m_inputNumber;
+    } else {
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "MappedAudioInput::getProperty - "
+        << "no properties available" << std::endl;
+#endif
+
+    }
+    return false;
+}
+
+void
+MappedAudioInput::setProperty(const MappedObjectProperty &property,
+                              MappedObjectValue value)
+{
+    if (property == InputNumber) {
+        m_inputNumber = value;
+    } else {
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "MappedAudioInput::setProperty - "
+        << "no properties available" << std::endl;
+#endif
+
+    }
+    return ;
+}
+
+
+MappedPluginSlot::MappedPluginSlot(MappedObject *parent, MappedObjectId id) :
+        MappedObject(parent, "MappedPluginSlot", PluginSlot, id)
+{
+#ifdef DEBUG_MAPPEDSTUDIO
+    std::cerr << "MappedPluginSlot::MappedPluginSlot: id = " << id << std::endl;
+#endif
+}
+
+MappedPluginSlot::~MappedPluginSlot()
+{
+#ifdef DEBUG_MAPPEDSTUDIO
+    std::cerr << "MappedPluginSlot::~MappedPluginSlot: id = " << getId() << ", identifier = " << m_identifier << std::endl;
+#endif
+
+    if (m_identifier != "") {
+
+        // shut down and remove the plugin instance we have running
+
+        MappedStudio *studio =
+            dynamic_cast<MappedStudio*>(getParent());
+
+        if (studio) {
+            SoundDriver *drv = studio->getSoundDriver();
+
+            if (drv) {
+                drv->removePluginInstance(m_instrument, m_position);
+            }
+        }
+    }
+}
+
+MappedObjectPropertyList
+MappedPluginSlot::getPropertyList(const MappedObjectProperty &property)
+{
+    MappedObjectPropertyList list;
+
+    if (property == "") {
+        list.push_back(PortCount);
+        list.push_back(Instrument);
+        list.push_back(Bypassed);
+        list.push_back(PluginName);
+        list.push_back(Label);
+        list.push_back(Author);
+        list.push_back(Copyright);
+        list.push_back(Category);
+    } else if (property == Programs) {
+
+        // The set of available programs is dynamic -- it can change
+        // while a plugin is instantiated.  So we query it on demand
+        // each time.
+
+        MappedStudio *studio =
+            dynamic_cast<MappedStudio*>(getParent());
+
+        if (studio) {
+            QStringList programs =
+                studio->getSoundDriver()->getPluginInstancePrograms(m_instrument,
+                        m_position);
+
+            for (int i = 0; i < int(programs.count()); ++i) {
+                list.push_back(programs[i]);
+            }
+        }
+
+    } else {
+        std::cerr << "MappedPluginSlot::getPropertyList: not a list property"
+        << std::endl;
+    }
+
+    return list;
+}
+
+bool
+MappedPluginSlot::getProperty(const MappedObjectProperty &property,
+                              MappedObjectValue &value)
+{
+    if (property == PortCount) {
+        value = m_portCount;
+    } else if (property == Instrument) {
+        value = m_instrument;
+    } else if (property == Position) {
+        value = m_position;
+    } else if (property == Bypassed) {
+        value = m_bypassed;
+    } else {
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "MappedPluginSlot::getProperty - "
+        << "unsupported or non-scalar property" << std::endl;
+#endif
+
+        return false;
+    }
+    return true;
+}
+
+bool
+MappedPluginSlot::getProperty(const MappedObjectProperty &property,
+                              QString &value)
+{
+    if (property == Identifier) {
+        value = m_identifier;
+    } else if (property == PluginName) {
+        value = m_name;
+    } else if (property == Label) {
+        value = m_label;
+    } else if (property == Author) {
+        value = m_author;
+    } else if (property == Copyright) {
+        value = m_copyright;
+    } else if (property == Category) {
+        value = m_category;
+    } else if (property == Program) {
+
+        MappedStudio *studio =
+            dynamic_cast<MappedStudio*>(getParent());
+
+        if (studio) {
+            value = studio->getSoundDriver()->getPluginInstanceProgram(m_instrument,
+                    m_position);
+        }
+    } else {
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "MappedPluginSlot::getProperty - "
+        << "unsupported or non-scalar property" << std::endl;
+#endif
+
+        return false;
+    }
+    return true;
+}
+
+QString
+MappedPluginSlot::getProgram(int bank, int program)
+{
+    MappedStudio *studio =
+        dynamic_cast<MappedStudio*>(getParent());
+
+    if (studio) {
+        return
+            studio->getSoundDriver()->getPluginInstanceProgram(m_instrument,
+                    m_position,
+                    bank,
+                    program);
+    }
+
+    return QString();
+}
+
+unsigned long
+MappedPluginSlot::getProgram(QString name)
+{
+    MappedStudio *studio =
+        dynamic_cast<MappedStudio*>(getParent());
+
+    if (studio) {
+        return
+            studio->getSoundDriver()->getPluginInstanceProgram(m_instrument,
+                    m_position,
+                    name);
+    }
+
+    return 0;
+}
+
+void
+MappedPluginSlot::setProperty(const MappedObjectProperty &property,
+                              MappedObjectValue value)
+{
+    if (property == Instrument) {
+        m_instrument = InstrumentId(value);
+    } else if (property == PortCount) {
+        m_portCount = int(value);
+    } else if (property == Position) {
+        m_position = int(value);
+    } else if (property == Bypassed) {
+        m_bypassed = bool(value);
+
+        MappedStudio *studio =
+            dynamic_cast<MappedStudio*>(getParent());
+
+        if (studio) {
+            studio->getSoundDriver()->setPluginInstanceBypass(m_instrument,
+                    m_position,
+                    m_bypassed);
+        }
+    }
+}
+
+void
+MappedPluginSlot::setProperty(const MappedObjectProperty &property,
+                              QString value)
+{
+    if (property == Identifier) {
+
+        if (m_identifier == value)
+            return ;
+
+        // shut down and remove the plugin instance we have running
+
+        MappedStudio *studio =
+            dynamic_cast<MappedStudio*>(getParent());
+
+        if (studio) {
+            SoundDriver *drv = studio->getSoundDriver();
+
+            if (drv) {
+
+                // We don't call drv->removePluginInstance at this
+                // point: the sequencer will deal with that when we
+                // call setPluginInstance below.  If we removed the
+                // instance here, we might cause the library we want
+                // for the new plugin instance to be unloaded and then
+                // loaded again, which is hardly the most efficient.
+
+                m_identifier = value;
+
+                // populate myself and my ports
+                PluginFactory *factory = PluginFactory::instanceFor(m_identifier);
+                if (!factory) {
+                    std::cerr << "WARNING: MappedPluginSlot::setProperty(identifier): No plugin factory for identifier " << m_identifier << "!" << std::endl;
+                    m_identifier = "";
+                    return ;
+                }
+
+                factory->populatePluginSlot(m_identifier, *this);
+
+                // now create the new instance
+                drv->setPluginInstance(m_instrument,
+                                       m_identifier,
+                                       m_position);
+            }
+        }
+
+        m_configuration.clear();
+
+    } else if (property == PluginName) {
+        m_name = value;
+    } else if (property == Label) {
+        m_label = value;
+    } else if (property == Author) {
+        m_author = value;
+    } else if (property == Copyright) {
+        m_copyright = value;
+    } else if (property == Category) {
+        m_category = value;
+    } else if (property == Program) {
+
+        MappedStudio *studio =
+            dynamic_cast<MappedStudio*>(getParent());
+
+        if (studio) {
+            studio->getSoundDriver()->setPluginInstanceProgram(m_instrument,
+                    m_position,
+                    value);
+        }
+    } else {
+
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "MappedPluginSlot::setProperty - "
+        << "unsupported or non-scalar property" << std::endl;
+#endif
+
+    }
+}
+
+void
+MappedPluginSlot::setPropertyList(const MappedObjectProperty &property,
+                                  const MappedObjectPropertyList &values)
+{
+    if (property == Configuration) {
+
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "MappedPluginSlot::setPropertyList(configuration): configuration is:" << std::endl;
+#endif
+
+        MappedStudio *studio =
+            dynamic_cast<MappedStudio*>(getParent());
+
+        for (MappedObjectPropertyList::const_iterator i = values.begin();
+                i != values.end(); ++i) {
+
+            QString key = *i;
+            QString value = *++i;
+
+#ifdef DEBUG_MAPPEDSTUDIO
+
+            std::cerr << key << " = " << value << std::endl;
+#endif
+
+            if (m_configuration.find(key) != m_configuration.end() &&
+                    m_configuration[key] == value)
+                continue;
+
+            if (studio) {
+                QString rv =
+                    studio->getSoundDriver()->configurePlugin(m_instrument,
+                            m_position,
+                            key, value);
+                if (rv && rv != "") {
+                    throw(rv);
+                }
+            }
+        }
+
+        m_configuration.clear();
+
+        for (MappedObjectPropertyList::const_iterator i = values.begin();
+                i != values.end(); ++i) {
+
+            QString key = *i;
+            QString value = *++i;
+
+            m_configuration[key] = value;
+        }
+    } else {
+
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "MappedPluginSlot::setPropertyList - "
+        << "not a list property" << std::endl;
+#endif
+
+    }
+}
+
+void
+MappedPluginSlot::setPort(unsigned long portNumber, float value)
+{
+    std::vector<MappedObject*> ports = getChildObjects();
+    std::vector<MappedObject*>::iterator it = ports.begin();
+    MappedPluginPort *port = 0;
+
+    for (; it != ports.end(); it++) {
+        port = dynamic_cast<MappedPluginPort *>(*it);
+        if (port && (unsigned long)port->getPortNumber() == portNumber) {
+            port->setValue(value);
+        }
+    }
+}
+
+float
+MappedPluginSlot::getPort(unsigned long portNumber)
+{
+    std::vector<MappedObject*> ports = getChildObjects();
+    std::vector<MappedObject*>::iterator it = ports.begin();
+    MappedPluginPort *port = 0;
+
+    for (; it != ports.end(); it++) {
+        port = dynamic_cast<MappedPluginPort *>(*it);
+        if (port && (unsigned long)port->getPortNumber() == portNumber) {
+            return port->getValue();
+        }
+    }
+
+    return 0;
+}
+
+
+MappedPluginPort::MappedPluginPort(MappedObject *parent, MappedObjectId id) :
+        MappedObject(parent, "MappedPluginPort", PluginPort, id)
+{}
+
+MappedPluginPort::~MappedPluginPort()
+{}
+
+MappedObjectPropertyList
+MappedPluginPort::getPropertyList(const MappedObjectProperty &property)
+{
+    MappedObjectPropertyList list;
+
+    if (property == "") {
+        list.push_back(PortNumber);
+        list.push_back(Minimum);
+        list.push_back(Maximum);
+        list.push_back(Default);
+        list.push_back(DisplayHint);
+        list.push_back(Value);
+        list.push_back(Name);
+    } else {
+        std::cerr << "MappedPluginSlot::getPropertyList: not a list property"
+        << std::endl;
+    }
+
+    return list;
+}
+
+bool
+MappedPluginPort::getProperty(const MappedObjectProperty &property,
+                              MappedObjectValue &value)
+{
+    if (property == PortNumber) {
+        value = m_portNumber;
+    } else if (property == Minimum) {
+        value = m_minimum;
+    } else if (property == Maximum) {
+        value = m_maximum;
+    } else if (property == Default) {
+        value = m_default;
+    } else if (property == DisplayHint) {
+        value = m_displayHint;
+    } else if (property == Value) {
+        return getValue();
+    } else {
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "MappedPluginPort::getProperty - "
+        << "unsupported or non-scalar property" << std::endl;
+#endif
+
+        return false;
+    }
+    return true;
+}
+
+bool
+MappedPluginPort::getProperty(const MappedObjectProperty &property,
+                              QString &value)
+{
+    if (property == Name) {
+        value = m_name;
+    } else {
+
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "MappedPluginPort::getProperty - "
+        << "unsupported or non-scalar property" << std::endl;
+#endif
+
+        return false;
+    }
+    return true;
+}
+
+void
+MappedPluginPort::setValue(MappedObjectValue value)
+{
+    MappedPluginSlot *slot =
+        dynamic_cast<MappedPluginSlot *>(getParent());
+
+    if (slot) {
+
+        MappedStudio *studio =
+            dynamic_cast<MappedStudio *>(slot->getParent());
+
+        if (studio) {
+            SoundDriver *drv = studio->getSoundDriver();
+
+            if (drv) {
+                drv->setPluginInstancePortValue(slot->getInstrument(),
+                                                slot->getPosition(),
+                                                m_portNumber, value);
+            }
+        }
+    }
+}
+
+float
+MappedPluginPort::getValue() const
+{
+    const MappedPluginSlot *slot =
+        dynamic_cast<const MappedPluginSlot *>(getParent());
+
+    if (slot) {
+
+        const MappedStudio *studio =
+            dynamic_cast<const MappedStudio *>(slot->getParent());
+
+        if (studio) {
+            SoundDriver *drv =
+                const_cast<SoundDriver *>(studio->getSoundDriver());
+
+            if (drv) {
+                return drv->getPluginInstancePortValue(slot->getInstrument(),
+                                                       slot->getPosition(),
+                                                       m_portNumber);
+            }
+        }
+    }
+
+    return 0;
+}
+
+void
+MappedPluginPort::setProperty(const MappedObjectProperty &property,
+                              MappedObjectValue value)
+{
+    if (property == PortNumber) {
+        m_portNumber = int(value);
+    } else if (property == Minimum) {
+        m_minimum = value;
+    } else if (property == Maximum) {
+        m_maximum = value;
+    } else if (property == Default) {
+        m_default = value;
+    } else if (property == DisplayHint) {
+        m_displayHint = PluginPort::PortDisplayHint(value);
+    } else if (property == Value) {
+        setValue(value);
+    } else {
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "MappedPluginPort::setProperty - "
+        << "unsupported or non-scalar property" << std::endl;
+#endif
+
+    }
+}
+
+
+void
+MappedPluginPort::setProperty(const MappedObjectProperty &property,
+                              QString value)
+{
+    if (property == Name) {
+        m_name = value;
+    } else {
+
+#ifdef DEBUG_MAPPEDSTUDIO
+        std::cerr << "MappedPluginPort::setProperty - "
+        << "unsupported or non-scalar property" << std::endl;
+#endif
+
+    }
+}
+
+
+}
+
+
+
diff --git a/src/sound/MappedStudio.h b/src/sound/MappedStudio.h
new file mode 100644
index 0000000..0896e6b
--- /dev/null
+++ b/src/sound/MappedStudio.h
@@ -0,0 +1,552 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include <map>
+#include <string>
+#include <vector>
+#include <qdatastream.h>
+#include <qstring.h>
+
+#include "MappedCommon.h"
+#include "Instrument.h"
+#include "Device.h"
+
+#include "AudioPluginInstance.h" // for PluginPort::PortDisplayHint //!!!???
+
+#ifndef _MAPPEDSTUDIO_H_
+#define _MAPPEDSTUDIO_H_
+
+
+// A sequencer-side representation of certain elements in the
+// gui that enables us to control outgoing or incoming audio
+// and MIDI with run-time only persistence.  Placeholders for
+// our Studio elements on the sequencer.
+
+namespace Rosegarden
+{
+
+class SoundDriver;
+
+
+// Types are in MappedCommon.h
+//
+class MappedObject
+{
+public:
+
+    // Some common properties
+    //
+    static const MappedObjectProperty Name;
+    static const MappedObjectProperty Instrument;
+    static const MappedObjectProperty Position;
+
+    // The object we can create
+    //
+    typedef enum
+    {
+        Studio,
+        AudioFader,          // connectable fader - interfaces with devices
+        AudioBuss,           // connectable buss - inferfaces with faders
+        AudioInput,          // connectable record input
+        PluginSlot,
+        PluginPort
+
+    } MappedObjectType;
+
+    MappedObject(MappedObject *parent,
+                 const std::string &name,
+                 MappedObjectType type,
+                 MappedObjectId id):
+        m_type(type),
+        m_id(id), 
+        m_name(name),
+        m_parent(parent) {;}
+
+    virtual ~MappedObject() {;}
+
+    MappedObjectId getId() { return m_id; }
+    MappedObjectType getType() { return m_type; }
+
+    std::string getName() { return m_name; }
+    void setName(const std::string &name) { m_name= name; }
+
+    // Get and set properties
+    //
+    virtual MappedObjectPropertyList
+        getPropertyList(const MappedObjectProperty &property) = 0;
+
+    virtual bool getProperty(const MappedObjectProperty &property,
+                             MappedObjectValue &value) = 0;
+
+    // Only relevant to objects that have string properties
+    // 
+    virtual bool getProperty(const MappedObjectProperty &/* property */,
+                             QString &/* value */) { return false; }
+
+    virtual void setProperty(const MappedObjectProperty &property,
+                             MappedObjectValue value) = 0;
+
+    // Only relevant to objects that have string properties
+    // 
+    virtual void setProperty(const MappedObjectProperty &/* property */,
+                             QString /* value */) { }
+
+    // Only relevant to objects that have list properties
+    //
+    virtual void setPropertyList(const MappedObjectProperty &/* property */,
+                                 const MappedObjectPropertyList &/* values */) { }
+
+    // Ownership
+    //
+    MappedObject* getParent() { return m_parent; }
+    const MappedObject* getParent() const { return m_parent; }
+    void setParent(MappedObject *parent) { m_parent = parent; }
+
+    // Get a list of child ids - get a list of a certain type
+    //
+    MappedObjectPropertyList getChildren();
+    MappedObjectPropertyList getChildren(MappedObjectType type);
+
+    // Child management
+    //
+    void addChild(MappedObject *mO);
+    void removeChild(MappedObject *mO);
+
+    // Destruction
+    //
+    void destroy();
+    void destroyChildren();
+
+    std::vector<MappedObject*> getChildObjects() { return m_children; }
+
+protected:
+
+    MappedObjectType m_type;
+    MappedObjectId   m_id;
+    std::string      m_name;
+
+    MappedObject                *m_parent;
+    std::vector<MappedObject*>   m_children;
+};
+
+
+class MappedAudioFader;
+class MappedAudioBuss;
+class MappedAudioInput;
+
+// Works as a factory and virtual plug-board for all our other
+// objects whether they be MIDI or audio.
+//
+//
+//
+class MappedStudio : public MappedObject
+{
+public:
+    MappedStudio();
+    ~MappedStudio();
+
+    // Create a new slider of a certain type for a certain
+    // type of device.
+    //
+    MappedObject* createObject(MappedObjectType type);
+
+    // And create an object with a specified id
+    //
+    MappedObject* createObject(MappedObjectType type, 
+                               MappedObjectId id);
+
+    bool connectObjects(MappedObjectId mId1, MappedObjectId mId2);
+    bool disconnectObjects(MappedObjectId mId1, MappedObjectId mId2);
+    bool disconnectObject(MappedObjectId mId);
+
+    // Destroy a MappedObject by ID
+    //
+    bool destroyObject(MappedObjectId id);
+
+    // Get an object by ID only
+    //
+    MappedObject* getObjectById(MappedObjectId);
+
+    // Get an object by ID and type.  (Returns 0 if the ID does not
+    // exist or exists but is not of the correct type.)  This is
+    // faster than getObjectById if you know the type already.
+    //
+    MappedObject* getObjectByIdAndType(MappedObjectId, MappedObjectType);
+
+    // Get an arbitrary object of a given type - to see if any exist
+    //
+    MappedObject* getObjectOfType(MappedObjectType type);
+
+    // Find out how many objects there are of a certain type
+    //
+    unsigned int getObjectCount(MappedObjectType type);
+
+    // iterators
+    MappedObject* getFirst(MappedObjectType type);
+    MappedObject* getNext(MappedObject *object);
+
+    std::vector<MappedObject *> getObjectsOfType(MappedObjectType type);
+
+    // Empty the studio of everything
+    //
+    void clear();
+
+    // Clear a MappedObject reference from the Studio
+    //
+    bool clearObject(MappedObjectId id);
+
+    // Property list
+    //
+    virtual MappedObjectPropertyList getPropertyList(
+            const MappedObjectProperty &property);
+
+    virtual bool getProperty(const MappedObjectProperty &property,
+                             MappedObjectValue &value);
+
+    virtual void setProperty(const MappedObjectProperty &property,
+                             MappedObjectValue value);
+
+    // Get an audio fader for an InstrumentId.  Convenience function.
+    //
+    MappedAudioFader *getAudioFader(InstrumentId id);
+    MappedAudioBuss *getAudioBuss(int bussNumber); // not buss no., not object id
+    MappedAudioInput *getAudioInput(int inputNumber); // likewise
+
+    // Return the object vector
+    //
+    //std::vector<MappedObject*>* getObjects() const { return &m_objects; }
+
+    // DCOP streaming
+    //
+    /* dunno if we need this
+    friend QDataStream& operator>>(QDataStream &dS, MappedStudio *mS);
+    friend QDataStream& operator<<(QDataStream &dS, MappedStudio *mS);
+    friend QDataStream& operator>>(QDataStream &dS, MappedStudio &mS);
+    friend QDataStream& operator<<(QDataStream &dS, const MappedStudio &mS);
+    */
+
+
+    // Set the driver object so that we can do things like
+    // initialise plugins etc.
+    //
+    SoundDriver* getSoundDriver() { return m_soundDriver; }
+    const SoundDriver* getSoundDriver() const { return m_soundDriver; }
+    void setSoundDriver(SoundDriver *driver) { m_soundDriver = driver; }
+
+protected:
+
+private:
+
+    // We give everything we create a unique MappedObjectId for
+    // this session.  So store the running total in here.
+    //
+    MappedObjectId             m_runningObjectId;
+
+    // All of our mapped (virtual) studio resides in this container as
+    // well as having all their parent/child relationships.  Because
+    // some things are just blobs with no connections we need to
+    // maintain both - don't forget about this.
+    //
+    // Note that object IDs are globally unique, not just unique within
+    // a category.
+    //
+    typedef std::map<MappedObjectId, MappedObject *> MappedObjectCategory;
+    typedef std::map<MappedObjectType, MappedObjectCategory> MappedObjectMap;
+    MappedObjectMap m_objects;
+    
+    // Driver object
+    //
+    SoundDriver *m_soundDriver;
+};
+
+
+// A connectable AudioObject that provides a connection framework
+// for MappedAudioFader and MappedAudioBuss (for example).  An
+// abstract base class.
+//
+// n input connections and m output connections - subclasses
+// can do the cleverness if n != m
+//
+
+class MappedConnectableObject : public MappedObject
+{
+public:
+    static const MappedObjectProperty ConnectionsIn;
+    static const MappedObjectProperty ConnectionsOut;
+
+    typedef enum
+    {
+        In,
+        Out
+    } ConnectionDirection;
+
+    MappedConnectableObject(MappedObject *parent,
+                            const std::string &name,
+                            MappedObjectType type,
+                            MappedObjectId id);
+
+    ~MappedConnectableObject();
+
+    void setConnections(ConnectionDirection dir,
+                        MappedObjectValueList conns);
+
+    void addConnection(ConnectionDirection dir, MappedObjectId id);
+    void removeConnection(ConnectionDirection dir, MappedObjectId id);
+
+    MappedObjectValueList getConnections (ConnectionDirection dir);
+
+protected:
+
+    // Which audio connections we have
+    //
+    MappedObjectValueList      m_connectionsIn;
+    MappedObjectValueList      m_connectionsOut;
+};
+
+// Audio fader
+//
+class MappedAudioFader : public MappedConnectableObject
+{
+public:
+    static const MappedObjectProperty Channels;
+
+    // properties
+    //
+    static const MappedObjectProperty FaderLevel;
+    static const MappedObjectProperty FaderRecordLevel;
+    static const MappedObjectProperty Pan;
+    static const MappedObjectProperty InputChannel;
+
+    MappedAudioFader(MappedObject *parent,
+                     MappedObjectId id,
+                     MappedObjectValue channels = 2); // stereo default
+    ~MappedAudioFader();
+
+    virtual MappedObjectPropertyList getPropertyList(
+                        const MappedObjectProperty &property);
+
+    virtual bool getProperty(const MappedObjectProperty &property,
+                             MappedObjectValue &value);
+
+    virtual void setProperty(const MappedObjectProperty &property,
+                             MappedObjectValue value);
+
+    InstrumentId getInstrument() const { return m_instrumentId; }
+
+protected:
+
+    MappedObjectValue             m_level;
+    MappedObjectValue             m_recordLevel;
+    InstrumentId      m_instrumentId;
+
+    // Stereo pan (-1.0 to +1.0)
+    //
+    MappedObjectValue             m_pan;
+
+    // How many channels we carry
+    //
+    MappedObjectValue             m_channels;
+
+    // If we have an input, which channel we take from it (if we are
+    // a mono fader at least)
+    //
+    MappedObjectValue             m_inputChannel;
+};
+
+class MappedAudioBuss : public MappedConnectableObject
+{
+public:
+    // A buss is much simpler than an instrument fader.  It's always
+    // stereo, and just has a level and pan associated with it.  The
+    // level may be a submaster fader level or a send mix level, it
+    // depends on what the purpose of the buss is.  At the moment we
+    // just have a 1-1 relationship between busses and submasters, and
+    // no send channels.
+
+    static const MappedObjectProperty BussId;
+    static const MappedObjectProperty Pan;
+    static const MappedObjectProperty Level;
+
+    MappedAudioBuss(MappedObject *parent,
+                    MappedObjectId id);
+    ~MappedAudioBuss();
+
+    virtual MappedObjectPropertyList getPropertyList(
+                        const MappedObjectProperty &property);
+
+    virtual bool getProperty(const MappedObjectProperty &property,
+                             MappedObjectValue &value);
+
+    virtual void setProperty(const MappedObjectProperty &property,
+                             MappedObjectValue value);
+
+    MappedObjectValue getBussId() { return m_bussId; }
+
+    // super-convenience function: retrieve the ids of the instruments
+    // connected to this buss
+    std::vector<InstrumentId> getInstruments();
+
+protected:
+    int m_bussId;
+    MappedObjectValue m_level;
+    MappedObjectValue m_pan;
+};
+
+class MappedAudioInput : public MappedConnectableObject
+{
+public:
+    // An input is simpler still -- no properties at all, apart from
+    // the input number, otherwise just the connections
+
+    static const MappedObjectProperty InputNumber;
+
+    MappedAudioInput(MappedObject *parent,
+                     MappedObjectId id);
+    ~MappedAudioInput();
+
+    virtual MappedObjectPropertyList getPropertyList(
+                        const MappedObjectProperty &property);
+
+    virtual bool getProperty(const MappedObjectProperty &property,
+                             MappedObjectValue &value);
+
+    virtual void setProperty(const MappedObjectProperty &property,
+                             MappedObjectValue value);
+
+    MappedObjectValue getInputNumber() { return m_inputNumber; }
+
+protected:
+    MappedObjectValue m_inputNumber;
+};
+
+class MappedPluginSlot : public MappedObject
+{
+public:
+    static const MappedObjectProperty Identifier;
+    static const MappedObjectProperty PluginName;
+    static const MappedObjectProperty Label;
+    static const MappedObjectProperty Author;
+    static const MappedObjectProperty Copyright;
+    static const MappedObjectProperty Category;
+    static const MappedObjectProperty PortCount;
+    static const MappedObjectProperty Ports;
+    static const MappedObjectProperty Program;
+    static const MappedObjectProperty Programs; // list property
+    static const MappedObjectProperty Instrument;
+    static const MappedObjectProperty Position;
+    static const MappedObjectProperty Bypassed;
+    static const MappedObjectProperty Configuration; // list property
+
+    MappedPluginSlot(MappedObject *parent, MappedObjectId id);
+    ~MappedPluginSlot();
+
+    virtual MappedObjectPropertyList getPropertyList(
+                        const MappedObjectProperty &property);
+
+    virtual bool getProperty(const MappedObjectProperty &property,
+                             MappedObjectValue &value);
+
+    virtual bool getProperty(const MappedObjectProperty &property,
+                             QString &value);
+
+    virtual void setProperty(const MappedObjectProperty &property,
+                             MappedObjectValue value);
+
+    virtual void setProperty(const MappedObjectProperty &property,
+                             QString value);
+
+    virtual void setPropertyList(const MappedObjectProperty &,
+                                 const MappedObjectPropertyList &);
+
+    void  setPort(unsigned long portNumber, float value);
+    float getPort(unsigned long portNumber);
+
+    InstrumentId getInstrument() const { return m_instrument; }
+    int getPosition() const { return m_position; }
+
+    QString getProgram(int bank, int program);
+    unsigned long getProgram(QString name); // rv is bank << 16 + program
+
+protected:
+    QString                   m_identifier;
+
+    QString                   m_name;
+    QString                   m_label;
+    QString                   m_author;
+    QString                   m_copyright;
+    QString                   m_category;
+    unsigned long             m_portCount;
+
+    InstrumentId              m_instrument;
+    int                       m_position;
+    bool                      m_bypassed;
+
+    std::map<QString, QString> m_configuration;
+};
+
+class MappedPluginPort : public MappedObject
+{
+public:
+    static const MappedObjectProperty PortNumber;
+    static const MappedObjectProperty Name;
+    static const MappedObjectProperty Minimum;
+    static const MappedObjectProperty Maximum;
+    static const MappedObjectProperty Default;
+    static const MappedObjectProperty DisplayHint;
+    static const MappedObjectProperty Value;
+
+    MappedPluginPort(MappedObject *parent, MappedObjectId id);
+    ~MappedPluginPort();
+
+    virtual MappedObjectPropertyList getPropertyList(
+                        const MappedObjectProperty &property);
+
+    virtual bool getProperty(const MappedObjectProperty &property,
+                             MappedObjectValue &value);
+
+    virtual bool getProperty(const MappedObjectProperty &property,
+                             QString &value);
+
+    virtual void setProperty(const MappedObjectProperty &property,
+                             MappedObjectValue value);
+
+    virtual void setProperty(const MappedObjectProperty &property,
+                             QString value);
+
+    void setValue(MappedObjectValue value);
+    MappedObjectValue getValue() const;
+
+    int getPortNumber() const { return m_portNumber; }
+
+protected:
+    int                     m_portNumber;
+    QString                 m_name;
+    MappedObjectValue       m_minimum;
+    MappedObjectValue       m_maximum;
+    MappedObjectValue       m_default;
+    PluginPort::PortDisplayHint m_displayHint;
+
+};
+
+    
+}
+
+#endif // _MAPPEDSTUDIO_H_
diff --git a/src/sound/Midi.h b/src/sound/Midi.h
new file mode 100644
index 0000000..65bfe93
--- /dev/null
+++ b/src/sound/Midi.h
@@ -0,0 +1,184 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+
+#ifndef _ROSEGARDEN_MIDI_H_
+#define _ROSEGARDEN_MIDI_H_
+
+#include "Instrument.h" // for MidiByte
+#include <string>
+
+// Yes we use the STL here.  Don't worry, it's fine.
+//
+//
+
+namespace Rosegarden
+{
+// Within the namespace we define our static MIDI messages 
+// that'll help us create and understand MIDI files.
+//
+// CreateMessageByte(MSG, CHANNEL)  = (MSG) | (CHANNEL)
+//
+//
+
+const std::string MIDI_FILE_HEADER              = "MThd";
+const std::string MIDI_TRACK_HEADER             = "MTrk";
+
+const MidiByte MIDI_STATUS_BYTE_MASK       = 0x80;
+const MidiByte MIDI_MESSAGE_TYPE_MASK      = 0xF0;
+const MidiByte MIDI_CHANNEL_NUM_MASK       = 0x0F;
+
+// our basic MIDI messages
+//
+const MidiByte MIDI_NOTE_OFF               = 0x80;
+const MidiByte MIDI_NOTE_ON                = 0x90;
+const MidiByte MIDI_POLY_AFTERTOUCH        = 0xA0;
+const MidiByte MIDI_CTRL_CHANGE            = 0xB0;
+const MidiByte MIDI_PROG_CHANGE            = 0xC0;
+const MidiByte MIDI_CHNL_AFTERTOUCH        = 0xD0;
+const MidiByte MIDI_PITCH_BEND             = 0xE0;
+
+// channel mode
+//
+const MidiByte MIDI_SELECT_CHNL_MODE       = 0xB0;
+
+// system messages
+const MidiByte MIDI_SYSTEM_EXCLUSIVE       = 0xF0;
+const MidiByte MIDI_TC_QUARTER_FRAME       = 0xF1;
+const MidiByte MIDI_SONG_POSITION_PTR      = 0xF2;
+const MidiByte MIDI_SONG_SELECT            = 0xF3;
+const MidiByte MIDI_TUNE_REQUEST           = 0xF6;
+const MidiByte MIDI_END_OF_EXCLUSIVE       = 0xF7;
+
+const MidiByte MIDI_TIMING_CLOCK           = 0xF8;
+const MidiByte MIDI_START                  = 0xFA;
+const MidiByte MIDI_CONTINUE               = 0xFB;
+const MidiByte MIDI_STOP                   = 0xFC;
+const MidiByte MIDI_ACTIVE_SENSING         = 0xFE;
+const MidiByte MIDI_SYSTEM_RESET           = 0xFF;
+
+// System Exclusive Extensions
+//
+
+// Non-commercial use
+//
+const MidiByte MIDI_SYSEX_NONCOMMERCIAL    = 0x7D;
+
+// Universal non-real time use
+// Format:
+//
+//   0xF0 0x7E <device id> <sub id #1> <sub id #2> <data> 0xF7
+//
+const MidiByte MIDI_SYSEX_NON_RT           = 0x7E;
+
+// RealTime e.g Midi Machine Control (MMC)
+//
+//   0xF0 0x7F <device id> <sub id #1> <sub id #2> <data> 0xF7
+//
+const MidiByte MIDI_SYSEX_RT               = 0x7F;
+
+// Sub IDs for RealTime SysExs
+//
+const MidiByte MIDI_SYSEX_RT_COMMAND       = 0x06;
+const MidiByte MIDI_SYSEX_RT_RESPONSE      = 0x07;
+
+// MMC commands
+//
+const MidiByte MIDI_MMC_STOP               = 0x01;
+const MidiByte MIDI_MMC_PLAY               = 0x02;
+const MidiByte MIDI_MMC_DEFERRED_PLAY      = 0x03;
+const MidiByte MIDI_MMC_FAST_FORWARD       = 0x04;
+const MidiByte MIDI_MMC_REWIND             = 0x05;
+const MidiByte MIDI_MMC_RECORD_STROBE      = 0x06; // punch in
+const MidiByte MIDI_MMC_RECORD_EXIT        = 0x07; // punch out
+const MidiByte MIDI_MMC_RECORD_PAUSE       = 0x08;
+const MidiByte MIDI_MMC_PAUSE              = 0x08;
+const MidiByte MIDI_MMC_EJECT              = 0x0A;
+const MidiByte MIDI_MMC_LOCATE             = 0x44; // jump to
+
+
+// Midi Event Code for META Event
+//
+const MidiByte MIDI_FILE_META_EVENT        = 0xFF;
+
+// META Event Codes
+//
+const MidiByte MIDI_SEQUENCE_NUMBER        = 0x00;
+const MidiByte MIDI_TEXT_EVENT             = 0x01;
+const MidiByte MIDI_COPYRIGHT_NOTICE       = 0x02;
+const MidiByte MIDI_TRACK_NAME             = 0x03;
+const MidiByte MIDI_INSTRUMENT_NAME        = 0x04;
+const MidiByte MIDI_LYRIC                  = 0x05;
+const MidiByte MIDI_TEXT_MARKER            = 0x06;
+const MidiByte MIDI_CUE_POINT              = 0x07;
+const MidiByte MIDI_CHANNEL_PREFIX         = 0x20;
+
+// There is contention over what 0x21 really means.
+// It's either a miswritten CHANNEL PREFIX or it's
+// a non-standard PORT MAPPING used by a sequencer.
+// Either way we include it (and generally ignore it)
+// as it's a part of many MIDI files that already 
+// exist.
+const MidiByte MIDI_CHANNEL_PREFIX_OR_PORT = 0x21;
+
+const MidiByte MIDI_END_OF_TRACK           = 0x2F;
+const MidiByte MIDI_SET_TEMPO              = 0x51;
+const MidiByte MIDI_SMPTE_OFFSET           = 0x54;
+const MidiByte MIDI_TIME_SIGNATURE         = 0x58;
+const MidiByte MIDI_KEY_SIGNATURE          = 0x59;
+const MidiByte MIDI_SEQUENCER_SPECIFIC     = 0x7F;
+
+// Some controllers
+//
+const MidiByte MIDI_CONTROLLER_BANK_MSB      = 0x00;
+const MidiByte MIDI_CONTROLLER_VOLUME        = 0x07;
+const MidiByte MIDI_CONTROLLER_BANK_LSB      = 0x20;
+const MidiByte MIDI_CONTROLLER_MODULATION    = 0x01;
+const MidiByte MIDI_CONTROLLER_PAN           = 0x0A;
+const MidiByte MIDI_CONTROLLER_SUSTAIN       = 0x40;
+const MidiByte MIDI_CONTROLLER_RESONANCE     = 0x47;
+const MidiByte MIDI_CONTROLLER_RELEASE       = 0x48;
+const MidiByte MIDI_CONTROLLER_ATTACK        = 0x49;
+const MidiByte MIDI_CONTROLLER_FILTER        = 0x4A;
+const MidiByte MIDI_CONTROLLER_REVERB        = 0x5B;
+const MidiByte MIDI_CONTROLLER_CHORUS        = 0x5D;
+
+// Registered and Non-Registered Parameter Controllers
+//
+const MidiByte MIDI_CONTROLLER_NRPN_1        = 0x62;
+const MidiByte MIDI_CONTROLLER_NRPN_2        = 0x63;
+const MidiByte MIDI_CONTROLLER_RPN_1         = 0x64;
+const MidiByte MIDI_CONTROLLER_RPN_2         = 0x65;
+
+const MidiByte MIDI_CONTROLLER_SOUNDS_OFF    = 0x78;
+const MidiByte MIDI_CONTROLLER_RESET         = 0x79; // reset all controllers
+const MidiByte MIDI_CONTROLLER_LOCAL         = 0x7A; // 0 = off, 127 = on
+const MidiByte MIDI_CONTROLLER_ALL_NOTES_OFF = 0x7B;
+
+
+// MIDI percussion channel
+const MidiByte MIDI_PERCUSSION_CHANNEL     = 9;
+
+
+
+}
+
+
+#endif // _ROSEGARDEN_MIDI_H_
diff --git a/src/sound/MidiEvent.cpp b/src/sound/MidiEvent.cpp
new file mode 100644
index 0000000..975b7aa
--- /dev/null
+++ b/src/sound/MidiEvent.cpp
@@ -0,0 +1,289 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+ 
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+ 
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+ 
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include "Midi.h"
+#include "MidiEvent.h"
+#include <iostream>
+
+// MidiEvent is a representation of MIDI which we use
+// for the import/export of MidiFiles.  It uses std::strings for
+// meta event messages which makes them nice and easy to handle.
+//
+//
+namespace Rosegarden
+{
+
+using std::string;
+using std::cout;
+using std::endl;
+
+MidiEvent::MidiEvent()
+{}
+
+MidiEvent::MidiEvent(timeT deltaTime,
+                     MidiByte eventCode):
+        m_deltaTime(deltaTime),
+        m_duration(0),
+        m_eventCode(eventCode),
+        m_data1(0),
+        m_data2(0),
+        m_metaEventCode(0),
+        m_metaMessage("")
+{}
+
+MidiEvent::MidiEvent(timeT deltaTime,
+                     MidiByte eventCode,
+                     MidiByte data1):
+        m_deltaTime(deltaTime),
+        m_duration(0),
+        m_eventCode(eventCode),
+        m_data1(data1),
+        m_data2(0),
+        m_metaEventCode(0),
+        m_metaMessage("")
+{}
+
+MidiEvent::MidiEvent(timeT deltaTime,
+                     MidiByte eventCode,
+                     MidiByte data1,
+                     MidiByte data2):
+        m_deltaTime(deltaTime),
+        m_duration(0),
+        m_eventCode(eventCode),
+        m_data1(data1),
+        m_data2(data2),
+        m_metaEventCode(0),
+        m_metaMessage("")
+
+{}
+
+MidiEvent::MidiEvent(timeT deltaTime,
+                     MidiByte eventCode,
+                     MidiByte metaEventCode,
+                     const string &metaMessage):
+        m_deltaTime(deltaTime),
+        m_duration(0),
+        m_eventCode(eventCode),
+        m_data1(0),
+        m_data2(0),
+        m_metaEventCode(metaEventCode),
+        m_metaMessage(metaMessage)
+{}
+
+MidiEvent::MidiEvent(timeT deltaTime,
+                     MidiByte eventCode,
+                     const string &sysEx):
+        m_deltaTime(deltaTime),
+        m_duration(0),
+        m_eventCode(eventCode),
+        m_data1(0),
+        m_data2(0),
+        m_metaEventCode(0),
+        m_metaMessage(sysEx)
+{}
+
+MidiEvent::~MidiEvent()
+{}
+
+// Show a representation of our MidiEvent purely for information
+// purposes (also demos how we decode them)
+//
+//
+void
+MidiEvent::print()
+{
+    timeT tempo;
+    int tonality;
+    string sharpflat;
+
+    if (m_metaEventCode) {
+        switch (m_metaEventCode) {
+        case MIDI_SEQUENCE_NUMBER:
+            cout << "MIDI SEQUENCE NUMBER" << endl;
+            break;
+
+        case MIDI_TEXT_EVENT:
+            cout << "MIDI TEXT:\t\"" << m_metaMessage << "\"" << endl;
+            break;
+
+        case MIDI_COPYRIGHT_NOTICE:
+            cout << "COPYRIGHT:\t\"" << m_metaMessage << "\"" << endl;
+
+        case MIDI_TRACK_NAME:
+            cout << "TRACK NAME:\t\"" << m_metaMessage << "\"" << endl;
+            break;
+
+        case MIDI_INSTRUMENT_NAME:
+            cout << "INSTRUMENT NAME:\t\"" << m_metaMessage << "\"" << endl;
+            break;
+
+        case MIDI_LYRIC:
+            cout << "LYRIC:\t\"" << m_metaMessage << "\"" << endl;
+            break;
+
+        case MIDI_TEXT_MARKER:
+            cout << "MARKER:\t\"" << m_metaMessage << "\"" << endl;
+            break;
+
+        case MIDI_CUE_POINT:
+            cout << "CUE POINT:\t\"" << m_metaMessage << "\"" << endl;
+            break;
+
+            // Sets a Channel number for a TRACK before it starts
+        case MIDI_CHANNEL_PREFIX:
+            cout << "CHANNEL PREFIX:\t"
+            << (timeT)m_metaMessage[0]
+            << endl;
+            break;
+
+            // These are actually the same case but this is not an
+            // official META event - it just crops up a lot.  We
+            // assume it's a MIDI_CHANNEL_PREFIX though
+            //
+        case MIDI_CHANNEL_PREFIX_OR_PORT:
+            cout << "FIXED CHANNEL PREFIX:\t"
+            << (timeT)m_metaMessage[0] << endl;
+            break;
+
+        case MIDI_END_OF_TRACK:
+            cout << "END OF TRACK" << endl;
+            break;
+
+        case MIDI_SET_TEMPO:
+            tempo =
+                ((timeT)(((MidiByte)m_metaMessage[0]) << 16)) +
+                ((timeT)(((MidiByte)m_metaMessage[1]) << 8)) +
+                (short)(MidiByte)m_metaMessage[2];
+
+            tempo = 60000000 / tempo;
+            cout << "SET TEMPO:\t" << tempo << endl;
+            break;
+
+        case MIDI_SMPTE_OFFSET:
+            cout << "SMPTE TIME CODE:\t"
+            << (timeT)m_metaMessage[0]
+            << ":" << (timeT)m_metaMessage[1]
+            << ":" << (timeT)m_metaMessage[2]
+            << "  -  fps = " << (timeT)m_metaMessage[3]
+            << "  - subdivsperframe = "
+            << (timeT)m_metaMessage[4]
+            << endl;
+            break;
+
+        case MIDI_TIME_SIGNATURE:
+            cout << "TIME SIGNATURE:\t"
+            << (timeT)m_metaMessage[0]
+            << "/"
+            << (1 << (timeT)m_metaMessage[1]) << endl;
+            break;
+
+        case MIDI_KEY_SIGNATURE:
+            tonality = (int)m_metaMessage[0];
+
+            if (tonality < 0) {
+                sharpflat = -tonality + " flat";
+            } else {
+                sharpflat = tonality;
+                sharpflat += " sharp";
+            }
+
+            cout << "KEY SIGNATURE:\t" << sharpflat << " "
+            << (((int)m_metaMessage[1]) == 0 ? "major" : "minor")
+            << endl;
+
+            break;
+
+        case MIDI_SEQUENCER_SPECIFIC:
+            cout << "SEQUENCER SPECIFIC:\t\"" << m_metaMessage << endl;
+            break;
+
+
+        default:
+            cout << "Undefined MIDI META event - "
+            << (timeT)m_metaEventCode << endl;
+            break;
+        }
+    } else {
+        switch (m_eventCode & MIDI_MESSAGE_TYPE_MASK) {
+        case MIDI_NOTE_ON:
+            cout << "NOTE ON:\t" << (int)m_data1 << " - "
+            << (int)m_data2 << endl;
+            break;
+
+        case MIDI_NOTE_OFF:
+            cout << "NOTE OFF:\t" << (int)m_data1 << " - "
+            << (int)m_data2 << endl;
+            break;
+
+        case MIDI_POLY_AFTERTOUCH:
+            cout << "POLY AFTERTOUCH:\t" << (int)m_data1
+            << " - " << (int)m_data2 << endl;
+            break;
+
+        case MIDI_CTRL_CHANGE:
+            cout << "CTRL CHANGE:\t" << (int)m_data1
+            << " - " << (int)m_data2 << endl;
+            break;
+
+        case MIDI_PITCH_BEND:
+            cout << "PITCH BEND:\t" << (int)m_data1
+            << " - " << (int)m_data2 << endl;
+            break;
+
+        case MIDI_PROG_CHANGE:
+            cout << "PROG CHANGE:\t" << (int)m_data1 << endl;
+            break;
+
+        case MIDI_CHNL_AFTERTOUCH:
+            cout << "CHNL AFTERTOUCH\t" << (int)m_data1 << endl;
+            break;
+
+        default:
+            cout << "Undefined MIDI event" << endl;
+            break;
+        }
+    }
+
+
+    return ;
+}
+
+// Adds the argument to _deltaTime and returns the result
+// thus aggregating the times as we go aint
+timeT
+MidiEvent::addTime(const timeT &time)
+{
+    m_deltaTime += time;
+    return m_deltaTime;
+}
+
+
+// Compare based on time
+//
+bool
+operator<(const MidiEvent &a, const MidiEvent &b)
+{
+    return a.getTime() < b.getTime();
+}
+
+
+}
+
+
diff --git a/src/sound/MidiEvent.h b/src/sound/MidiEvent.h
new file mode 100644
index 0000000..b2192b4
--- /dev/null
+++ b/src/sound/MidiEvent.h
@@ -0,0 +1,141 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+
+#ifndef _ROSEGARDEN_MIDI_EVENT_H_
+#define _ROSEGARDEN_MIDI_EVENT_H_
+
+#include "Midi.h"
+#include "Event.h"
+
+// MidiEvent holds MIDI and Event data during MIDI file I/O.
+// We don't use this class at all for playback or recording of MIDI -
+// for that look at MappedEvent and MappedComposition.
+//
+// Rosegarden doesn't have any internal concept of MIDI events, only
+// Events which are a superset of MIDI functionality.
+//
+// Check out Event in base/ for more information.
+//
+//
+//
+
+namespace Rosegarden
+{
+class MidiEvent
+{
+
+public:
+    MidiEvent();
+
+    // No data event
+    //
+    MidiEvent(timeT deltaTime,
+              MidiByte eventCode);
+
+    // single data byte case
+    //
+    MidiEvent(timeT deltaTime,
+              MidiByte eventCode,
+              MidiByte data1);
+
+    // double data byte
+    //
+    MidiEvent(timeT deltaTime,
+              MidiByte eventCode,
+              MidiByte data1,
+              MidiByte data2);
+
+    // Meta event
+    //
+    MidiEvent(timeT deltaTime,
+              MidiByte eventCode,
+              MidiByte metaEventCode,
+              const std::string &metaMessage);
+
+    // Sysex style constructor
+    //
+    MidiEvent(timeT deltaTime,
+              MidiByte eventCode,
+              const std::string &sysEx);
+
+
+    ~MidiEvent();
+
+    // View our event as text
+    //
+    void print();
+
+
+    void setTime(const timeT &time) { m_deltaTime = time; }
+    void setDuration(const timeT& duration) {m_duration = duration;}
+    timeT addTime(const timeT &time);
+
+    MidiByte getMessageType() const
+        { return ( m_eventCode & MIDI_MESSAGE_TYPE_MASK ); }
+
+    MidiByte getChannelNumber() const
+        { return ( m_eventCode & MIDI_CHANNEL_NUM_MASK ); }
+
+    timeT getTime() const { return m_deltaTime; }
+    timeT getDuration() const { return m_duration; }
+
+    MidiByte getPitch() const { return m_data1; }
+    MidiByte getVelocity() const { return m_data2; }
+    MidiByte getData1() const { return m_data1; }
+    MidiByte getData2() const { return m_data2; }
+    MidiByte getEventCode() const { return m_eventCode; }
+
+    bool isMeta() const { return(m_eventCode == MIDI_FILE_META_EVENT); }
+
+    MidiByte getMetaEventCode() const { return m_metaEventCode; }
+    std::string getMetaMessage() const { return m_metaMessage; }
+    void setMetaMessage(const std::string &meta) { m_metaMessage = meta; }
+
+    friend bool operator<(const MidiEvent &a, const MidiEvent &b);
+
+private:
+
+    MidiEvent& operator=(const MidiEvent);
+
+    timeT m_deltaTime;
+    timeT m_duration;
+    MidiByte          m_eventCode;
+    MidiByte          m_data1;         // or Note
+    MidiByte          m_data2;         // or Velocity
+
+    MidiByte          m_metaEventCode;
+    std::string       m_metaMessage;
+
+};
+
+// Comparator for sorting
+//
+struct MidiEventCmp
+{
+    bool operator()(const MidiEvent &mE1, const MidiEvent &mE2) const
+                    { return mE1.getTime() < mE2.getTime(); }
+    bool operator()(const MidiEvent *mE1, const MidiEvent *mE2) const
+                    { return mE1->getTime() < mE2->getTime(); }
+};
+
+}
+
+#endif // _ROSEGARDEN_MIDI_EVENT_H_
diff --git a/src/sound/MidiFile.cpp b/src/sound/MidiFile.cpp
new file mode 100644
index 0000000..76d5c85
--- /dev/null
+++ b/src/sound/MidiFile.cpp
@@ -0,0 +1,2261 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+ 
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+ 
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+ 
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+
+#include <iostream>
+#include "misc/Debug.h"
+#include <kapplication.h>
+#include <fstream>
+#include <string>
+#include <cstdio>
+#include <algorithm>
+
+#include "Midi.h"
+#include "MidiFile.h"
+#include "Segment.h"
+#include "NotationTypes.h"
+#include "BaseProperties.h"
+#include "SegmentNotationHelper.h"
+#include "SegmentPerformanceHelper.h"
+#include "CompositionTimeSliceAdapter.h"
+#include "AnalysisTypes.h"
+#include "Track.h"
+#include "Instrument.h"
+#include "Quantizer.h"
+#include "Studio.h"
+#include "MidiTypes.h"
+#include "Profiler.h"
+
+//#define MIDI_DEBUG 1
+
+#if (__GNUC__ < 3)
+#include <strstream>
+#define stringstream strstream
+#else
+#include <sstream>
+#endif
+
+#include <kapp.h>
+
+namespace Rosegarden
+{
+
+using std::string;
+using std::ifstream;
+using std::stringstream;
+using std::cerr;
+using std::endl;
+using std::ends;
+using std::ios;
+
+MidiFile::MidiFile(Studio *studio):
+        SoundFile(std::string("unnamed.mid")),
+        m_timingDivision(0),
+        m_format(MIDI_FILE_NOT_LOADED),
+        m_numberOfTracks(0),
+        m_containsTimeChanges(false),
+        m_trackByteCount(0),
+        m_decrementCount(false),
+        m_studio(studio)
+{}
+
+MidiFile::MidiFile(const std::string &fn,
+                   Studio *studio):
+        SoundFile(fn),
+        m_timingDivision(0),
+        m_format(MIDI_FILE_NOT_LOADED),
+        m_numberOfTracks(0),
+        m_containsTimeChanges(false),
+        m_trackByteCount(0),
+        m_decrementCount(false),
+        m_studio(studio)
+{}
+
+// Make sure we clear away the m_midiComposition
+//
+MidiFile::~MidiFile()
+{
+    clearMidiComposition();
+}
+
+
+// A couple of convenience functions.   Watch the byte conversions out
+// of the STL strings.
+//
+//
+long
+MidiFile::midiBytesToLong(const string& bytes)
+{
+    if (bytes.length() != 4) {
+#ifdef MIDI_DEBUG
+        std::cerr << "WARNING: Wrong length for long data (" << bytes.length()
+        << ", should be 4)" << endl;
+#endif
+
+        throw (Exception("Wrong length for long data in MIDI stream"));
+    }
+
+    long longRet = ((long)(((MidiByte)bytes[0]) << 24)) |
+                   ((long)(((MidiByte)bytes[1]) << 16)) |
+                   ((long)(((MidiByte)bytes[2]) << 8)) |
+                   ((long)((MidiByte)(bytes[3])));
+
+    std::cerr << "midiBytesToLong(" << int((MidiByte)bytes[0]) << "," << int((MidiByte)bytes[1]) << "," << int((MidiByte)bytes[2]) << "," << int((MidiByte)bytes[3]) << ") -> " << longRet << std::endl;
+
+    return longRet;
+}
+
+int
+MidiFile::midiBytesToInt(const string& bytes)
+{
+    if (bytes.length() != 2) {
+#ifdef MIDI_DEBUG
+        std::cerr << "WARNING: Wrong length for int data (" << bytes.length()
+        << ", should be 2)" << endl;
+#endif
+
+        throw (Exception("Wrong length for int data in MIDI stream"));
+    }
+
+    int intRet = ((int)(((MidiByte)bytes[0]) << 8)) |
+                 ((int)(((MidiByte)bytes[1])));
+    return (intRet);
+}
+
+
+
+// Gets a single byte from the MIDI byte stream.  For each track
+// section we can read only a specified number of bytes held in
+// m_trackByteCount.
+//
+MidiByte
+MidiFile::getMidiByte(ifstream* midiFile)
+{
+    static int bytesGot = 0; // purely for progress reporting purposes
+
+    if (midiFile->eof()) {
+        throw(Exception("End of MIDI file encountered while reading"));
+    }
+
+    if (m_decrementCount && m_trackByteCount <= 0) {
+        throw(Exception("Attempt to get more bytes than expected on Track"));
+    }
+
+    char byte;
+    if (midiFile->read(&byte, 1)) {
+
+        --m_trackByteCount;
+
+        // update a progress dialog if we have one
+        //
+        ++bytesGot;
+        if (bytesGot % 2000 == 0) {
+
+            emit setProgress((int)(double(midiFile->tellg()) /
+                                   double(m_fileSize) * 20.0));
+            kapp->processEvents(50);
+        }
+
+        return (MidiByte)byte;
+    }
+
+    throw(Exception("Attempt to read past MIDI file end"));
+}
+
+
+// Gets a specified number of bytes from the MIDI byte stream.  For
+// each track section we can read only a specified number of bytes
+// held in m_trackByteCount.
+//
+string
+MidiFile::getMidiBytes(ifstream* midiFile, unsigned long numberOfBytes)
+{
+    string stringRet;
+    char fileMidiByte;
+    static int bytesGot = 0; // purely for progress reporting purposes
+
+    if (midiFile->eof()) {
+#ifdef MIDI_DEBUG
+        std::cerr << "MIDI file EOF - got "
+        << stringRet.length() << " bytes out of "
+        << numberOfBytes << endl;
+#endif
+
+        throw(Exception("End of MIDI file encountered while reading"));
+
+    }
+
+    if (m_decrementCount && (numberOfBytes > (unsigned long)m_trackByteCount)) {
+#ifdef MIDI_DEBUG
+        std::cerr << "Attempt to get more bytes than allowed on Track ("
+        << numberOfBytes
+        << " > "
+        << m_trackByteCount << endl;
+#endif
+
+        //!!! Investigate -- I'm seeing this on new-notation-quantization
+        // branch: load glazunov.rg, run Interpret on first segment, export
+        // and attempt to import again
+
+        throw(Exception("Attempt to get more bytes than expected on Track"));
+    }
+
+    while (stringRet.length() < numberOfBytes &&
+            midiFile->read(&fileMidiByte, 1)) {
+        stringRet += fileMidiByte;
+    }
+
+    // if we've reached the end of file without fulfilling the
+    // quota then panic as our parsing has performed incorrectly
+    //
+    if (stringRet.length() < numberOfBytes) {
+        stringRet = "";
+#ifdef MIDI_DEBUG
+
+        cerr << "Attempt to read past file end - got "
+        << stringRet.length() << " bytes out of "
+        << numberOfBytes << endl;
+#endif
+
+        throw(Exception("Attempt to read past MIDI file end"));
+
+    }
+
+    // decrement the byte count
+    if (m_decrementCount)
+        m_trackByteCount -= stringRet.length();
+
+    // update a progress dialog if we have one
+    //
+    bytesGot += numberOfBytes;
+    if (bytesGot % 2000 == 0) {
+        emit setProgress((int)(double(midiFile->tellg()) /
+                               double(m_fileSize) * 20.0));
+        kapp->processEvents(50);
+    }
+
+    return stringRet;
+}
+
+
+// Get a long number of variable length from the MIDI byte stream.
+//
+//
+long
+MidiFile::getNumberFromMidiBytes(ifstream* midiFile, int firstByte)
+{
+    long longRet = 0;
+    MidiByte midiByte;
+
+    if (firstByte >= 0) {
+        midiByte = (MidiByte)firstByte;
+    } else if (midiFile->eof()) {
+        return longRet;
+    } else {
+        midiByte = getMidiByte(midiFile);
+    }
+
+    longRet = midiByte;
+    if (midiByte & 0x80 ) {
+        longRet &= 0x7F;
+        do {
+            midiByte = getMidiByte(midiFile);
+            longRet = (longRet << 7) + (midiByte & 0x7F);
+        } while (!midiFile->eof() && (midiByte & 0x80));
+    }
+
+    return longRet;
+}
+
+
+
+// Seeks to the next track in the midi file and sets the number
+// of bytes to be read in the counter m_trackByteCount.
+//
+bool
+MidiFile::skipToNextTrack(ifstream *midiFile)
+{
+    string buffer, buffer2;
+    m_trackByteCount = -1;
+    m_decrementCount = false;
+
+    while (!midiFile->eof() && (m_decrementCount == false )) {
+        buffer = getMidiBytes(midiFile, 4);
+
+#if (__GNUC__ < 3)
+
+        if (buffer.compare(MIDI_TRACK_HEADER, 0, 4) == 0)
+#else
+
+        if (buffer.compare(0, 4, MIDI_TRACK_HEADER) == 0)
+#endif
+
+        {
+            m_trackByteCount = midiBytesToLong(getMidiBytes(midiFile, 4));
+            m_decrementCount = true;
+        }
+
+    }
+
+    if ( m_trackByteCount == -1 ) // we haven't found a track
+        return (false);
+    else
+        return (true);
+}
+
+
+// Read in a MIDI file.  The parsing process throws string
+// exceptions back up here if we run into trouble which we
+// can then pass back out to whoever called us using a nice
+// bool.
+//
+//
+bool
+MidiFile::open()
+{
+    bool retOK = true;
+    m_error = "";
+
+#ifdef MIDI_DEBUG
+
+    std::cerr << "MidiFile::open() : fileName = " << m_fileName.c_str() << endl;
+#endif
+
+    // Open the file
+    ifstream *midiFile = new ifstream(m_fileName.c_str(), ios::in | ios::binary);
+
+    try {
+        if (*midiFile) {
+
+            // Set file size so we can count it off
+            //
+            midiFile->seekg(0, std::ios::end);
+            m_fileSize = midiFile->tellg();
+            midiFile->seekg(0, std::ios::beg);
+
+            // Parse the MIDI header first.  The first 14 bytes of the file.
+            if (!parseHeader(getMidiBytes(midiFile, 14))) {
+                m_format = MIDI_FILE_NOT_LOADED;
+                m_error = "Not a MIDI file.";
+                return (false);
+            }
+
+            m_containsTimeChanges = false;
+
+            TrackId i = 0;
+
+            for (unsigned int j = 0; j < m_numberOfTracks; ++j) {
+
+//#ifdef MIDI_DEBUG
+                std::cerr << "Parsing Track " << j << endl;
+//#endif
+
+                if (!skipToNextTrack(midiFile)) {
+#ifdef MIDI_DEBUG
+                    cerr << "Couldn't find Track " << j << endl;
+#endif
+
+                    m_error = "File corrupted or in non-standard format?";
+                    m_format = MIDI_FILE_NOT_LOADED;
+                    return (false);
+                }
+
+#ifdef MIDI_DEBUG
+                std::cerr << "Track has " << m_trackByteCount << " bytes" << std::endl;
+#endif
+
+                // Run through the events taking them into our internal
+                // representation.
+                if (!parseTrack(midiFile, i)) {
+//#ifdef MIDI_DEBUG
+                    std::cerr << "Track " << j << " parsing failed" << endl;
+//#endif
+
+                    m_error = "File corrupted or in non-standard format?";
+                    m_format = MIDI_FILE_NOT_LOADED;
+                    return (false);
+                }
+
+                ++i; // j is the source track number, i the destination
+            }
+
+            m_numberOfTracks = i;
+        } else {
+            m_error = "File not found or not readable.";
+            m_format = MIDI_FILE_NOT_LOADED;
+            return (false);
+        }
+
+        // Close the file now
+        midiFile->close();
+    } catch (Exception e) {
+#ifdef MIDI_DEBUG
+        std::cerr << "MidiFile::open() - caught exception - "
+        << e.getMessage() << endl;
+#endif
+
+        m_error = e.getMessage();
+        retOK = false;
+    }
+
+    return (retOK);
+}
+
+// Parse and ensure the MIDI Header is legitimate
+//
+//
+bool
+MidiFile::parseHeader(const string &midiHeader)
+{
+    if (midiHeader.size() < 14) {
+#ifdef MIDI_DEBUG
+        std::cerr << "MidiFile::parseHeader() - file header undersized" << endl;
+#endif
+
+        return (false);
+    }
+
+#if (__GNUC__ < 3)
+    if (midiHeader.compare(MIDI_FILE_HEADER, 0, 4) != 0)
+#else
+
+    if (midiHeader.compare(0, 4, MIDI_FILE_HEADER) != 0)
+#endif
+
+    {
+#ifdef MIDI_DEBUG
+        std::cerr << "MidiFile::parseHeader()"
+        << "- file header not found or malformed"
+        << endl;
+#endif
+
+        return (false);
+    }
+
+    if (midiBytesToLong(midiHeader.substr(4, 4)) != 6L) {
+#ifdef MIDI_DEBUG
+        std::cerr << "MidiFile::parseHeader()"
+        << " - header length incorrect"
+        << endl;
+#endif
+
+        return (false);
+    }
+
+    m_format = (MIDIFileFormatType) midiBytesToInt(midiHeader.substr(8, 2));
+    m_numberOfTracks = midiBytesToInt(midiHeader.substr(10, 2));
+    m_timingDivision = midiBytesToInt(midiHeader.substr(12, 2));
+
+    if ( m_format == MIDI_SEQUENTIAL_TRACK_FILE ) {
+#ifdef MIDI_DEBUG
+        std::cerr << "MidiFile::parseHeader()"
+        << "- can't load sequential track file"
+        << endl;
+#endif
+
+        return (false);
+    }
+
+
+#ifdef MIDI_DEBUG
+    if ( m_timingDivision < 0 ) {
+        std::cerr << "MidiFile::parseHeader()"
+        << " - file uses SMPTE timing"
+        << endl;
+    }
+#endif
+
+    return (true);
+}
+
+
+
+// Extract the contents from a MIDI file track and places it into
+// our local map of MIDI events.
+//
+//
+bool
+MidiFile::parseTrack(ifstream* midiFile, TrackId &lastTrackNum)
+{
+    MidiByte midiByte, metaEventCode, data1, data2;
+    MidiByte eventCode = 0x80;
+    std::string metaMessage;
+    unsigned int messageLength;
+    unsigned long deltaTime;
+    unsigned long accumulatedTime = 0;
+
+    // The trackNum passed in to this method is the default track for
+    // all events provided they're all on the same channel.  If we find
+    // events on more than one channel, we increment trackNum and record
+    // the mapping from channel to trackNum in this channelTrackMap.
+    // We then return the new trackNum by reference so the calling
+    // method knows we've got more tracks than expected.
+
+    // This would be a vector<TrackId> but TrackId is unsigned
+    // and we need -1 to indicate "not yet used"
+    std::vector<int> channelTrackMap(16, -1);
+
+    // This is used to store the last absolute time found on each track,
+    // allowing us to modify delta-times correctly when separating events
+    // out from one to multiple tracks
+    //
+    std::map<int, unsigned long> trackTimeMap;
+
+    // Meta-events don't have a channel, so we place them in a fixed
+    // track number instead
+    TrackId metaTrack = lastTrackNum;
+
+    // Remember the last non-meta status byte (-1 if we haven't seen one)
+    int runningStatus = -1;
+
+    bool firstTrack = true;
+
+    std::cerr << "Parse track: last track number is " << lastTrackNum << std::endl;
+
+    while (!midiFile->eof() && ( m_trackByteCount > 0 ) ) {
+        if (eventCode < 0x80) {
+#ifdef MIDI_DEBUG
+            cerr << "WARNING: Invalid event code " << eventCode
+            << " in MIDI file" << endl;
+#endif
+
+            throw (Exception("Invalid event code found"));
+        }
+
+        deltaTime = getNumberFromMidiBytes(midiFile);
+
+#ifdef MIDI_DEBUG
+        cerr << "read delta time " << deltaTime << endl;
+#endif
+
+        // Get a single byte
+        midiByte = getMidiByte(midiFile);
+
+        if (!(midiByte & MIDI_STATUS_BYTE_MASK)) {
+            if (runningStatus < 0) {
+                throw (Exception("Running status used for first event in track"));
+            }
+
+            eventCode = (MidiByte)runningStatus;
+            data1 = midiByte;
+
+#ifdef MIDI_DEBUG
+            std::cerr << "using running status (byte " << int(midiByte) << " found)" << std::endl;
+#endif
+
+        } else {
+#ifdef MIDI_DEBUG
+            std::cerr << "have new event code " << int(midiByte) << std::endl;
+#endif
+
+            eventCode = midiByte;
+            data1 = getMidiByte(midiFile);
+        }
+
+        if (eventCode == MIDI_FILE_META_EVENT) // meta events
+        {
+            //            metaEventCode = getMidiByte(midiFile);
+            metaEventCode = data1;
+            messageLength = getNumberFromMidiBytes(midiFile);
+
+#ifdef MIDI_DEBUG
+
+            std::cerr << "Meta event of type " << int(metaEventCode) << " and " << messageLength << " bytes found" << std::endl;
+#endif
+
+            metaMessage = getMidiBytes(midiFile, messageLength);
+
+            if (metaEventCode == MIDI_TIME_SIGNATURE ||
+                    metaEventCode == MIDI_SET_TEMPO)
+            {
+                m_containsTimeChanges = true;
+            }
+
+            long gap = accumulatedTime - trackTimeMap[metaTrack];
+            accumulatedTime += deltaTime;
+            deltaTime += gap;
+            trackTimeMap[metaTrack] = accumulatedTime;
+
+            MidiEvent *e = new MidiEvent(deltaTime,
+                                         MIDI_FILE_META_EVENT,
+                                         metaEventCode,
+                                         metaMessage);
+
+            m_midiComposition[metaTrack].push_back(e);
+
+        } else // the rest
+        {
+            runningStatus = eventCode;
+
+            MidiEvent *midiEvent;
+
+            int channel = (eventCode & MIDI_CHANNEL_NUM_MASK);
+            if (channelTrackMap[channel] == -1) {
+                if (!firstTrack) {
+                    ++lastTrackNum;
+		} else {
+                    firstTrack = false;
+		}
+		std::cerr << "MidiFile: new channel map entry: channel " << channel << " -> track " << lastTrackNum << std::endl;
+                channelTrackMap[channel] = lastTrackNum;
+		m_trackChannelMap[lastTrackNum] = channel;
+            }
+
+            TrackId trackNum = channelTrackMap[channel];
+
+	    {
+		static int prevTrackNum = -1, prevChannel = -1;
+		if (prevTrackNum != (int) trackNum ||
+		    prevChannel != (int) channel) {
+		    std::cerr << "MidiFile: track number for channel " << channel << " is " << trackNum << std::endl;
+		    prevTrackNum = trackNum;
+		    prevChannel = channel;
+		}
+	    }
+
+            // accumulatedTime is abs time of last event on any track;
+            // trackTimeMap[trackNum] is that of last event on this track
+
+            long gap = accumulatedTime - trackTimeMap[trackNum];
+            accumulatedTime += deltaTime;
+            deltaTime += gap;
+            trackTimeMap[trackNum] = accumulatedTime;
+
+            switch (eventCode & MIDI_MESSAGE_TYPE_MASK) {
+            case MIDI_NOTE_ON:
+            case MIDI_NOTE_OFF:
+            case MIDI_POLY_AFTERTOUCH:
+            case MIDI_CTRL_CHANGE:
+                data2 = getMidiByte(midiFile);
+
+                // create and store our event
+                midiEvent = new MidiEvent(deltaTime, eventCode, data1, data2);
+
+                /*
+                std::cerr << "MIDI event for channel " << channel << " (track "
+                << trackNum << ")" << std::endl;
+                midiEvent->print();
+                          */
+
+
+                m_midiComposition[trackNum].push_back(midiEvent);
+                break;
+
+            case MIDI_PITCH_BEND:
+                data2 = getMidiByte(midiFile);
+
+                // create and store our event
+                midiEvent = new MidiEvent(deltaTime, eventCode, data1, data2);
+                m_midiComposition[trackNum].push_back(midiEvent);
+                break;
+
+            case MIDI_PROG_CHANGE:
+            case MIDI_CHNL_AFTERTOUCH:
+                // create and store our event
+		std::cerr << "Program change or channel aftertouch: time " << deltaTime << ", code " << (int)eventCode << ", data " << (int) data1  << " going to track " << trackNum << std::endl;
+                midiEvent = new MidiEvent(deltaTime, eventCode, data1);
+                m_midiComposition[trackNum].push_back(midiEvent);
+                break;
+
+            case MIDI_SYSTEM_EXCLUSIVE:
+                messageLength = getNumberFromMidiBytes(midiFile, data1);
+
+#ifdef MIDI_DEBUG
+
+                std::cerr << "SysEx of " << messageLength << " bytes found" << std::endl;
+#endif
+
+                metaMessage = getMidiBytes(midiFile, messageLength);
+
+                if (MidiByte(metaMessage[metaMessage.length() - 1]) !=
+                        MIDI_END_OF_EXCLUSIVE) {
+#ifdef MIDI_DEBUG
+                    std::cerr << "MidiFile::parseTrack() - "
+                    << "malformed or unsupported SysEx type"
+                    << std::endl;
+#endif
+
+                    continue;
+                }
+
+                // chop off the EOX
+                // length fixed by Pedro Lopez-Cabanillas (20030523)
+                //
+                metaMessage = metaMessage.substr(0, metaMessage.length() - 1);
+
+                midiEvent = new MidiEvent(deltaTime,
+                                          MIDI_SYSTEM_EXCLUSIVE,
+                                          metaMessage);
+                m_midiComposition[trackNum].push_back(midiEvent);
+                break;
+
+            case MIDI_END_OF_EXCLUSIVE:
+#ifdef MIDI_DEBUG
+
+                std::cerr << "MidiFile::parseTrack() - "
+                << "Found a stray MIDI_END_OF_EXCLUSIVE" << std::endl;
+#endif
+
+                break;
+
+            default:
+#ifdef MIDI_DEBUG
+
+                std::cerr << "MidiFile::parseTrack()"
+                << " - Unsupported MIDI Event Code:  "
+                << (int)eventCode << endl;
+#endif
+
+                break;
+            }
+        }
+    }
+
+    return (true);
+}
+
+// borrowed from ALSA pcm_timer.c
+//
+static unsigned long gcd(unsigned long a, unsigned long b)
+{
+    unsigned long r;
+    if (a < b) {
+        r = a;
+        a = b;
+        b = r;
+    }
+    while ((r = a % b) != 0) {
+        a = b;
+        b = r;
+    }
+    return b;
+}
+
+// If we wanted to abstract the MidiFile class to make it more useful to
+// other applications (and formats) we'd make this method and its twin
+// pure virtual.
+//
+bool
+MidiFile::convertToRosegarden(Composition &composition, ConversionType type)
+{
+    Profiler profiler("MidiFile::convertToRosegarden");
+
+    MidiTrack::iterator midiEvent;
+    Segment *rosegardenSegment;
+    Segment *conductorSegment = 0;
+    Event *rosegardenEvent;
+    string trackName;
+
+    // Time conversions
+    //
+    timeT rosegardenTime = 0;
+    timeT rosegardenDuration = 0;
+    timeT maxTime = 0;
+
+    // To create rests
+    //
+    timeT endOfLastNote;
+
+    // Event specific vars
+    //
+    int numerator = 4;
+    int denominator = 4;
+    timeT segmentTime;
+
+    // keys
+    int accidentals;
+    bool isMinor;
+    bool isSharp;
+
+    if (type == CONVERT_REPLACE)
+        composition.clear();
+
+    timeT origin = 0;
+    if (type == CONVERT_APPEND && composition.getDuration() > 0) {
+        origin = composition.getBarEndForTime(composition.getDuration());
+    }
+
+    TrackId compTrack = 0;
+    for (Composition::iterator ci = composition.begin();
+            ci != composition.end(); ++ci) {
+        if ((*ci)->getTrack() >= compTrack)
+            compTrack = (*ci)->getTrack() + 1;
+    }
+
+    Track *track = 0;
+
+    // precalculate the timing factor
+    //
+    // [cc] -- attempt to avoid floating-point rounding errors
+    timeT crotchetTime = Note(Note::Crotchet).getDuration();
+    int divisor = m_timingDivision ? m_timingDivision : 96;
+
+    unsigned long multiplier = crotchetTime;
+    int g = (int)gcd(crotchetTime, divisor);
+    multiplier /= g;
+    divisor /= g;
+
+    timeT maxRawTime = LONG_MAX;
+    if (multiplier > divisor)
+        maxRawTime = (maxRawTime / multiplier) * divisor;
+
+    bool haveTimeSignatures = false;
+    InstrumentId compInstrument = MidiInstrumentBase;
+
+    // Clear down the assigned Instruments we already have
+    //
+    if (type == CONVERT_REPLACE) {
+	m_studio->unassignAllInstruments();
+    }
+
+    std::vector<Segment *> addedSegments;
+
+#ifdef MIDI_DEBUG
+
+    std::cerr << "NUMBER OF TRACKS = " << m_numberOfTracks << endl;
+    std::cerr << "MIDI COMP SIZE = " << m_midiComposition.size() << endl;
+#endif
+
+    for (TrackId i = 0; i < m_numberOfTracks; i++ ) {
+        segmentTime = 0;
+        trackName = string("Imported MIDI");
+
+        // progress - 20% total in file import itself and then 80%
+        // split over these tracks
+        emit setProgress(20 +
+                         (int)((80.0 * double(i) / double(m_numberOfTracks))));
+        kapp->processEvents(50);
+
+        // Convert the deltaTime to an absolute time since
+        // the start of the segment.  The addTime method
+        // returns the sum of the current Midi Event delta
+        // time plus the argument.
+        //
+        for (midiEvent = m_midiComposition[i].begin();
+                midiEvent != m_midiComposition[i].end();
+                ++midiEvent) {
+            segmentTime = (*midiEvent)->addTime(segmentTime);
+        }
+
+        // Consolidate NOTE ON and NOTE OFF events into a NOTE ON with
+        // a duration.
+        //
+	consolidateNoteOffEvents(i);
+
+	if (m_trackChannelMap.find(i) != m_trackChannelMap.end()) {
+	    compInstrument = MidiInstrumentBase + m_trackChannelMap[i];
+	} else {
+	    compInstrument = MidiInstrumentBase;
+	}
+
+        rosegardenSegment = new Segment;
+        rosegardenSegment->setTrack(compTrack);
+        rosegardenSegment->setStartTime(0);
+
+        track = new Track(compTrack,         // id
+                          compInstrument,    // instrument
+                          compTrack,         // position
+                          trackName,         // name
+                          false);           // muted
+
+	std::cerr << "New Rosegarden track: id = " << compTrack << ", instrument = " << compInstrument << ", name = " << trackName << std::endl;
+
+        // rest creation token needs to be reset here
+        //
+        endOfLastNote = 0;
+
+        int msb = -1, lsb = -1; // for bank selects
+        Instrument *instrument = 0;
+
+        for (midiEvent = m_midiComposition[i].begin();
+	     midiEvent != m_midiComposition[i].end();
+	     midiEvent++) {
+            rosegardenEvent = 0;
+
+            // [cc] -- avoid floating-point where possible
+
+            timeT rawTime = (*midiEvent)->getTime();
+
+            if (rawTime < maxRawTime) {
+                rosegardenTime = origin +
+                                 timeT((rawTime * multiplier) / divisor);
+            } else {
+                rosegardenTime = origin +
+                                 timeT((double(rawTime) * multiplier) / double(divisor) + 0.01);
+            }
+
+            rosegardenDuration =
+                timeT(((*midiEvent)->getDuration() * multiplier) / divisor);
+
+#ifdef MIDI_DEBUG
+
+            std::cerr << "MIDI file import: origin " << origin
+            << ", event time " << rosegardenTime
+            << ", duration " << rosegardenDuration
+            << ", event type " << (int)(*midiEvent)->getMessageType()
+            << ", previous max time " << maxTime
+            << ", potential max time " << (rosegardenTime + rosegardenDuration)
+            << ", ev raw time " << (*midiEvent)->getTime()
+            << ", crotchet " << crotchetTime
+            << ", multiplier " << multiplier
+            << ", divisor " << divisor
+            << std::endl;
+#endif
+
+            if (rosegardenTime + rosegardenDuration > maxTime) {
+                maxTime = rosegardenTime + rosegardenDuration;
+            }
+
+            //	    timeT fillFromTime = rosegardenTime;
+            if (rosegardenSegment->empty()) {
+                //		fillFromTime = composition.getBarStartForTime(rosegardenTime);
+                endOfLastNote = composition.getBarStartForTime(rosegardenTime);
+            }
+
+            if ((*midiEvent)->isMeta()) {
+
+                switch ((*midiEvent)->getMetaEventCode()) {
+
+                case MIDI_TEXT_EVENT: {
+                        std::string text = (*midiEvent)->getMetaMessage();
+                        rosegardenEvent =
+                            Text(text).getAsEvent(rosegardenTime);
+                    }
+                    break;
+
+                case MIDI_LYRIC: {
+                        std::string text = (*midiEvent)->getMetaMessage();
+//		    std::cerr << "lyric event: text=\""
+//			      << text << "\", time=" << rosegardenTime << std::endl;
+                        rosegardenEvent =
+                            Text(text, Text::Lyric).
+                            getAsEvent(rosegardenTime);
+                    }
+                    break;
+
+                case MIDI_TEXT_MARKER: {
+                        std::string text = (*midiEvent)->getMetaMessage();
+                        composition.addMarker(new Marker
+                                              (rosegardenTime, text, ""));
+                    }
+                    break;
+
+                case MIDI_COPYRIGHT_NOTICE:
+                    if (type == CONVERT_REPLACE) {
+                        composition.setCopyrightNote((*midiEvent)->
+                                                     getMetaMessage());
+                    }
+                    break;
+
+                case MIDI_TRACK_NAME:
+                    track->setLabel((*midiEvent)->getMetaMessage());
+                    break;
+
+		case MIDI_INSTRUMENT_NAME:
+		    rosegardenSegment->setLabel((*midiEvent)->getMetaMessage());
+		    break;
+
+                case MIDI_END_OF_TRACK: {
+                        timeT trackEndTime = rosegardenTime;
+                        if (trackEndTime <= 0) {
+                            trackEndTime = crotchetTime * 4 * numerator / denominator;
+                        }
+                        if (endOfLastNote < trackEndTime) {
+                            //If there's nothing in the segment yet, then we
+                            //shouldn't fill with rests because we don't want
+                            //to cause the otherwise empty segment to be created
+                            if (rosegardenSegment->size() > 0) {
+                                rosegardenSegment->fillWithRests(trackEndTime);
+                            }
+                        }
+                    }
+                    break;
+
+                case MIDI_SET_TEMPO: {
+                        MidiByte m0 = (*midiEvent)->getMetaMessage()[0];
+                        MidiByte m1 = (*midiEvent)->getMetaMessage()[1];
+                        MidiByte m2 = (*midiEvent)->getMetaMessage()[2];
+
+                        long tempo = (((m0 << 8) + m1) << 8) + m2;
+
+                        if (tempo != 0) {
+                            double qpm = 60000000.0 / double(tempo);
+                            tempoT rgt(Composition::getTempoForQpm(qpm));
+                            std::cout << "MidiFile: converted MIDI tempo " << tempo << " to Rosegarden tempo " << rgt << std::endl;
+                            composition.addTempoAtTime(rosegardenTime, rgt);
+                        }
+                    }
+                    break;
+
+                case MIDI_TIME_SIGNATURE:
+                    numerator = (int) (*midiEvent)->getMetaMessage()[0];
+                    denominator = 1 << ((int)(*midiEvent)->getMetaMessage()[1]);
+
+                    // NB. a MIDI time signature also has
+                    // metamessage[2] and [3], containing some timing data
+
+                    if (numerator == 0)
+                        numerator = 4;
+                    if (denominator == 0)
+                        denominator = 4;
+
+                    composition.addTimeSignature
+                    (rosegardenTime,
+                     TimeSignature(numerator, denominator));
+                    haveTimeSignatures = true;
+                    break;
+
+                case MIDI_KEY_SIGNATURE:
+                    // get the details
+                    accidentals = (int) (*midiEvent)->getMetaMessage()[0];
+                    isMinor = (int) (*midiEvent)->getMetaMessage()[1];
+                    isSharp = accidentals < 0 ? false : true;
+                    accidentals = accidentals < 0 ? -accidentals : accidentals;
+                    // create the key event
+                    //
+                    try {
+                        rosegardenEvent = Rosegarden::Key
+                                          (accidentals, isSharp, isMinor).
+                                          getAsEvent(rosegardenTime);
+                    }
+                    catch (...) {
+#ifdef MIDI_DEBUG
+                        std::cerr << "MidiFile::convertToRosegarden - "
+                        << " badly formed key signature"
+                        << std::endl;
+#endif
+
+                        break;
+                    }
+                    break;
+
+                case MIDI_SEQUENCE_NUMBER:
+                case MIDI_CHANNEL_PREFIX_OR_PORT:
+                case MIDI_CUE_POINT:
+                case MIDI_CHANNEL_PREFIX:
+                case MIDI_SEQUENCER_SPECIFIC:
+                case MIDI_SMPTE_OFFSET:
+                default:
+#ifdef MIDI_DEBUG
+
+                    std::cerr << "MidiFile::convertToRosegarden - "
+                    << "unsupported META event code "
+                    << (int)((*midiEvent)->getMetaEventCode()) << endl;
+#endif
+
+                    break;
+                }
+
+            } else
+                switch ((*midiEvent)->getMessageType()) {
+                case MIDI_NOTE_ON:
+
+                    // A zero velocity here is a virtual "NOTE OFF"
+                    // so we ignore this event
+                    //
+                    if ((*midiEvent)->getVelocity() == 0)
+                        break;
+
+                    endOfLastNote = rosegardenTime + rosegardenDuration;
+
+                    //std::cerr << "MidiFile::convertToRosegarden: note at " << rosegardenTime << ", midi time " << (*midiEvent)->getTime() << std::endl;
+
+                    // create and populate event
+                    rosegardenEvent = new Event(Note::EventType,
+                                                rosegardenTime,
+                                                rosegardenDuration);
+                    rosegardenEvent->set
+			<Int>(BaseProperties::PITCH,
+			      (*midiEvent)->getPitch());
+                    rosegardenEvent->set
+			<Int>(BaseProperties::VELOCITY,
+			      (*midiEvent)->getVelocity());
+                    break;
+
+                    // We ignore any NOTE OFFs here as we've already
+                    // converted NOTE ONs to have duration
+                    //
+                case MIDI_NOTE_OFF:
+                    continue;
+                    break;
+
+                case MIDI_PROG_CHANGE:
+                    // Attempt to turn the prog change we've found into an
+                    // Instrument.  Send the program number and whether or
+                    // not we're on the percussion channel.
+                    //
+                    // Note that we make no attempt to do the right
+                    // thing with program changes during a track -- we
+                    // just save them as events.  Only the first is
+                    // used to select the instrument.  If it's at time
+                    // zero, it's not saved as an event.
+                    //
+//		    std::cerr << "Program change found" << std::endl;
+
+                    if (!instrument) {
+
+			bool percussion = (*midiEvent)->getChannelNumber() ==
+			    MIDI_PERCUSSION_CHANNEL;
+			int program = (*midiEvent)->getData1();
+
+			if (type == CONVERT_REPLACE) {
+
+			    instrument = m_studio->getInstrumentById(compInstrument);
+			    if (instrument) {
+				instrument->setPercussion(percussion);
+				instrument->setSendProgramChange(true);
+				instrument->setProgramChange(program);
+				instrument->setSendBankSelect(msb >= 0 || lsb >= 0);
+				if (instrument->sendsBankSelect()) {
+				    instrument->setMSB(msb >= 0 ? msb : 0);
+				    instrument->setLSB(lsb >= 0 ? lsb : 0);
+				}
+			    }
+			} else { // not CONVERT_REPLACE
+			    instrument =
+				m_studio->assignMidiProgramToInstrument
+				(program, msb, lsb, percussion);
+			}
+		    }
+
+		    // assign it here
+		    if (instrument) {
+			track->setInstrument(instrument->getId());
+			// We used to set the segment name from the instrument
+			// here, but now we do them all at the end only if the
+			// segment has no other name set (e.g. from instrument
+			// meta event)
+			if ((*midiEvent)->getTime() == 0) break; // no insert
+		    }
+
+                    // did we have a bank select? if so, insert that too
+
+                    if (msb >= 0) {
+                        rosegardenSegment->insert
+                        (Controller(MIDI_CONTROLLER_BANK_MSB, msb).
+                         getAsEvent(rosegardenTime));
+                    }
+                    if (lsb >= 0) {
+                        rosegardenSegment->insert
+                        (Controller(MIDI_CONTROLLER_BANK_LSB, msb).
+                         getAsEvent(rosegardenTime));
+                    }
+
+                    rosegardenEvent =
+                        ProgramChange((*midiEvent)->getData1()).
+                        getAsEvent(rosegardenTime);
+                    break;
+
+                case MIDI_CTRL_CHANGE:
+
+                    // If it's a bank select, interpret it (or remember
+                    // for later insertion) instead of just inserting it
+                    // as a Rosegarden event
+
+                    if ((*midiEvent)->getData1() == MIDI_CONTROLLER_BANK_MSB) {
+                        msb = (*midiEvent)->getData2();
+                        break;
+                    }
+
+                    if ((*midiEvent)->getData1() == MIDI_CONTROLLER_BANK_LSB) {
+                        lsb = (*midiEvent)->getData2();
+                        break;
+                    }
+
+                    // If it's something we can use as an instrument
+                    // parameter, and it's at time zero, and we already
+                    // have an instrument, then apply it to the instrument
+                    // instead of inserting
+
+                    if (instrument && (*midiEvent)->getTime() == 0) {
+                        if ((*midiEvent)->getData1() == MIDI_CONTROLLER_VOLUME) {
+                            instrument->setVolume((*midiEvent)->getData2());
+                            break;
+                        }
+                        if ((*midiEvent)->getData1() == MIDI_CONTROLLER_PAN) {
+                            instrument->setPan((*midiEvent)->getData2());
+                            break;
+                        }
+                        if ((*midiEvent)->getData1() == MIDI_CONTROLLER_ATTACK) {
+                            instrument->setControllerValue(MIDI_CONTROLLER_ATTACK, (*midiEvent)->getData2());
+                            break;
+                        }
+                        if ((*midiEvent)->getData1() == MIDI_CONTROLLER_RELEASE) {
+                            instrument->setControllerValue(MIDI_CONTROLLER_RELEASE, (*midiEvent)->getData2());
+                            break;
+                        }
+                        if ((*midiEvent)->getData1() == MIDI_CONTROLLER_FILTER) {
+                            instrument->setControllerValue(MIDI_CONTROLLER_FILTER, (*midiEvent)->getData2());
+                            break;
+                        }
+                        if ((*midiEvent)->getData1() == MIDI_CONTROLLER_RESONANCE) {
+                            instrument->setControllerValue(MIDI_CONTROLLER_RESONANCE, (*midiEvent)->getData2());
+                            break;
+                        }
+                        if ((*midiEvent)->getData1() == MIDI_CONTROLLER_CHORUS) {
+                            instrument->setControllerValue(MIDI_CONTROLLER_CHORUS, (*midiEvent)->getData2());
+                            break;
+                        }
+                        if ((*midiEvent)->getData1() == MIDI_CONTROLLER_REVERB) {
+                            instrument->setControllerValue(MIDI_CONTROLLER_REVERB, (*midiEvent)->getData2());
+                            break;
+                        }
+                    }
+
+                    rosegardenEvent =
+                        Controller((*midiEvent)->getData1(),
+                                   (*midiEvent)->getData2()).
+                        getAsEvent(rosegardenTime);
+                    break;
+
+                case MIDI_PITCH_BEND:
+                    rosegardenEvent =
+                        PitchBend((*midiEvent)->getData2(),
+                                  (*midiEvent)->getData1()).
+                        getAsEvent(rosegardenTime);
+                    break;
+
+                case MIDI_SYSTEM_EXCLUSIVE:
+                    rosegardenEvent =
+                        SystemExclusive((*midiEvent)->getMetaMessage()).
+                        getAsEvent(rosegardenTime);
+                    break;
+
+                case MIDI_POLY_AFTERTOUCH:
+                    rosegardenEvent =
+                        KeyPressure((*midiEvent)->getData1(),
+                                    (*midiEvent)->getData2()).
+                        getAsEvent(rosegardenTime);
+                    break;
+
+                case MIDI_CHNL_AFTERTOUCH:
+                    rosegardenEvent =
+                        ChannelPressure((*midiEvent)->getData1()).
+                        getAsEvent(rosegardenTime);
+                    break;
+
+                default:
+#ifdef MIDI_DEBUG
+
+                    std::cerr << "MidiFile::convertToRosegarden - "
+                    << "Unsupported event code = "
+                    << (int)(*midiEvent)->getMessageType() << std::endl;
+#endif
+
+                    break;
+                }
+
+            if (rosegardenEvent) {
+                //		if (fillFromTime < rosegardenTime) {
+                //		    rosegardenSegment->fillWithRests(fillFromTime, rosegardenTime);
+                //		}
+                if (endOfLastNote < rosegardenTime) {
+                    rosegardenSegment->fillWithRests(endOfLastNote, rosegardenTime);
+                }
+                rosegardenSegment->insert(rosegardenEvent);
+            }
+        }
+
+        if (rosegardenSegment->size() > 0) {
+
+            // if all we have is key signatures and rests, take this
+            // to be a conductor segment and don't insert it
+            //
+            bool keySigsOnly = true;
+            bool haveKeySig = false;
+            for (Segment::iterator i = rosegardenSegment->begin();
+                    i != rosegardenSegment->end(); ++i) {
+                if (!(*i)->isa(Rosegarden::Key::EventType) &&
+                        !(*i)->isa(Note::EventRestType)) {
+                    keySigsOnly = false;
+                    break;
+                } else if ((*i)->isa(Rosegarden::Key::EventType)) {
+                    haveKeySig = true;
+                }
+            }
+
+            if (keySigsOnly) {
+                conductorSegment = rosegardenSegment;
+                continue;
+            } else if (!haveKeySig && conductorSegment) {
+                // copy across any key sigs from the conductor segment
+
+                timeT segmentStartTime = rosegardenSegment->getStartTime();
+                timeT earliestEventEndTime = segmentStartTime;
+
+                for (Segment::iterator i = conductorSegment->begin();
+                        i != conductorSegment->end(); ++i) {
+                    if ((*i)->getAbsoluteTime() + (*i)->getDuration() <
+                            earliestEventEndTime) {
+                        earliestEventEndTime =
+                            (*i)->getAbsoluteTime() + (*i)->getDuration();
+                    }
+                    rosegardenSegment->insert(new Event(**i));
+                }
+
+                if (earliestEventEndTime < segmentStartTime) {
+                    rosegardenSegment->fillWithRests(earliestEventEndTime,
+                                                     segmentStartTime);
+                }
+            }
+
+#ifdef MIDI_DEBUG
+            std::cerr << "MIDI import: adding segment with start time " << rosegardenSegment->getStartTime() << " and end time " << rosegardenSegment->getEndTime() << std::endl;
+	    if (rosegardenSegment->getEndTime() == 2880) {
+		std::cerr << "events:" << std::endl;
+		for (Segment::iterator i = rosegardenSegment->begin();
+		     i != rosegardenSegment->end(); ++i) {
+		    std::cerr << "type = " << (*i)->getType() << std::endl;
+		    std::cerr << "time = " << (*i)->getAbsoluteTime() << std::endl;
+		    std::cerr << "duration = " << (*i)->getDuration() << std::endl;
+		}
+	    }
+#endif
+
+            // add the Segment to the Composition and increment the
+            // Rosegarden segment number
+            //
+            composition.addTrack(track);
+            composition.addSegment(rosegardenSegment);
+	    addedSegments.push_back(rosegardenSegment);
+            compTrack++;
+
+        } else {
+            delete rosegardenSegment;
+            rosegardenSegment = 0;
+            delete track;
+            track = 0;
+        }
+    }
+
+    if (type == CONVERT_REPLACE || maxTime > composition.getEndMarker()) {
+        composition.setEndMarker(composition.getBarEndForTime(maxTime));
+    }
+
+    for (std::vector<Segment *>::iterator i = addedSegments.begin();
+	 i != addedSegments.end(); ++i) {
+	Segment *s = *i;
+	if (s) {
+	    timeT duration = s->getEndMarkerTime() - s->getStartTime();
+/*
+	    std::cerr << "duration = " << duration << " (start "
+		      << s->getStartTime() << ", end " << s->getEndTime()
+		      << ", marker " << s->getEndMarkerTime() << ")" << std::endl;
+*/
+	    if (duration == 0) {
+		s->setEndMarkerTime(s->getStartTime() +
+				    Note(Note::Crotchet).getDuration());
+	    }
+	    Instrument *instr = m_studio->getInstrumentFor(s);
+	    if (instr) {
+		if (s->getLabel() == "") {
+		    s->setLabel(m_studio->getSegmentName(instr->getId()));
+		}
+	    }
+	}
+    }
+
+    return true;
+}
+
+// Takes a Composition and turns it into internal MIDI representation
+// that can then be written out to file.
+//
+// For the moment we should watch to make sure that multiple Segment
+// (parts) don't equate to multiple segments in the MIDI Composition.
+//
+// This is a two pass operation - firstly convert the RG Composition
+// into MIDI events and insert anything extra we need (i.e. NOTE OFFs)
+// with absolute times before then processing all timings into delta
+// times.
+//
+//
+void
+MidiFile::convertToMidi(Composition &comp)
+{
+    MidiEvent *midiEvent;
+    int conductorTrack = 0;
+
+    timeT midiEventAbsoluteTime;
+    MidiByte midiVelocity;
+    MidiByte midiChannel = 0;
+
+    // [cc] int rather than floating point
+    //
+    m_timingDivision = 480; //!!! make this configurable
+    timeT crotchetDuration = Note(Note::Crotchet).getDuration();
+
+    // Export as this format only
+    //
+    m_format = MIDI_SIMULTANEOUS_TRACK_FILE;
+
+    // Clear out the MidiComposition internal store
+    //
+    clearMidiComposition();
+
+    // Insert the Rosegarden Signature Track here and any relevant
+    // file META information - this will get written out just like
+    // any other MIDI track.
+    //
+    midiEvent = new MidiEvent(0, MIDI_FILE_META_EVENT, MIDI_COPYRIGHT_NOTICE,
+                              comp.getCopyrightNote());
+
+    m_midiComposition[conductorTrack].push_back(midiEvent);
+
+    midiEvent = new MidiEvent(0, MIDI_FILE_META_EVENT, MIDI_CUE_POINT,
+                              "Created by Rosegarden");
+
+    m_midiComposition[conductorTrack].push_back(midiEvent);
+
+    midiEvent = new MidiEvent(0, MIDI_FILE_META_EVENT, MIDI_CUE_POINT,
+                              "http://www.rosegardenmusic.com/");
+
+    m_midiComposition[conductorTrack].push_back(midiEvent);
+
+    // Insert tempo events
+    //
+    for (int i = 0; i < comp.getTempoChangeCount(); i++) // i=0 should be comp.getStart-something
+    {
+        std::pair<timeT, tempoT> tempo = comp.getTempoChange(i);
+
+        midiEventAbsoluteTime = tempo.first * m_timingDivision
+                                / crotchetDuration;
+
+        double qpm = Composition::getTempoQpm(tempo.second);
+        long tempoValue = long(60000000.0 / qpm + 0.01);
+
+        string tempoString;
+        tempoString += (MidiByte) ( tempoValue >> 16 & 0xFF );
+        tempoString += (MidiByte) ( tempoValue >> 8 & 0xFF );
+        tempoString += (MidiByte) ( tempoValue & 0xFF );
+
+        midiEvent = new MidiEvent(midiEventAbsoluteTime,
+                                  MIDI_FILE_META_EVENT,
+                                  MIDI_SET_TEMPO,
+                                  tempoString);
+
+        m_midiComposition[conductorTrack].push_back(midiEvent);
+    }
+
+    // Insert time signatures (don't worry that the times might be out
+    // of order with those of the tempo events -- we sort the track later)
+    //
+    for (int i = 0; i < comp.getTimeSignatureCount(); i++) {
+        std::pair<timeT, TimeSignature> timeSig =
+            comp.getTimeSignatureChange(i);
+
+        midiEventAbsoluteTime = timeSig.first * m_timingDivision
+                                / crotchetDuration;
+
+        string timeSigString;
+        timeSigString += (MidiByte) (timeSig.second.getNumerator());
+        int denominator = timeSig.second.getDenominator();
+        int denPowerOf2 = 0;
+
+        // Work out how many powers of two are in the denominator
+        //
+        while (denominator >>= 1)
+            denPowerOf2++;
+
+        timeSigString += (MidiByte) denPowerOf2;
+
+        // The third byte is the number of MIDI clocks per beat.
+        // There are 24 clocks per quarter-note (the MIDI clock
+        // is tempo-independent and is not related to the timebase).
+        //
+        int cpb = 24 * timeSig.second.getBeatDuration() / crotchetDuration;
+        timeSigString += (MidiByte) cpb;
+
+        // And the fourth byte is always 8, for us (it expresses
+        // the number of notated 32nd-notes in a MIDI quarter-note,
+        // for applications that may want to notate and perform
+        // in different units)
+        //
+        timeSigString += (MidiByte) 8;
+
+        midiEvent = new MidiEvent(midiEventAbsoluteTime,
+                                  MIDI_FILE_META_EVENT,
+                                  MIDI_TIME_SIGNATURE,
+                                  timeSigString);
+
+        m_midiComposition[conductorTrack].push_back(midiEvent);
+    }
+
+    // Insert markers
+    // fix for bug#
+    Composition::markercontainer marks = comp.getMarkers();
+
+    for (unsigned int i = 0; i < marks.size(); i++) {
+        midiEventAbsoluteTime = marks[i]->getTime() * m_timingDivision
+                                / crotchetDuration;
+
+        midiEvent = new MidiEvent( midiEventAbsoluteTime,
+                                   MIDI_FILE_META_EVENT,
+                                   MIDI_TEXT_MARKER,
+                                   marks[i]->getName() );
+
+        m_midiComposition[conductorTrack].push_back(midiEvent);
+    }
+
+    m_numberOfTracks = 1;
+    std::map<int, int> trackPosMap; // RG track pos -> MIDI track no
+
+    // In pass one just insert all events including new NOTE OFFs at the right
+    // absolute times.
+    //
+    for (Composition::const_iterator segment = comp.begin();
+            segment != comp.end(); ++segment) {
+
+        // We use this later to get NOTE durations
+        //
+        SegmentPerformanceHelper helper(**segment);
+
+        Track *track = comp.getTrackById((*segment)->getTrack());
+
+        if (track->isMuted()) continue;
+
+	// Fix #1602023, map Rosegarden tracks to MIDI tracks, instead of
+	// putting each segment out on a new track
+
+	int trackPosition = track->getPosition();
+	bool firstSegmentThisTrack = false;
+
+	if (trackPosMap.find(trackPosition) == trackPosMap.end()) {
+	    firstSegmentThisTrack = true;
+	    trackPosMap[trackPosition] = m_numberOfTracks++;
+	}
+
+	int trackNumber = trackPosMap[trackPosition];
+
+	MidiTrack &mtrack = m_midiComposition[trackNumber];
+
+	midiEvent = new MidiEvent(0,
+				  MIDI_FILE_META_EVENT,
+				  MIDI_TRACK_NAME,
+				  track->getLabel());
+
+	mtrack.push_back(midiEvent);
+
+        // Get the Instrument
+        //
+        Instrument *instr =
+            m_studio->getInstrumentById(track->getInstrument());
+
+	if (firstSegmentThisTrack) {
+
+	    MidiByte program = 0;
+	    midiChannel = 0;
+
+	    bool useBank = false;
+	    MidiByte lsb = 0;
+	    MidiByte msb = 0;
+
+	    if (instr) {
+		midiChannel = instr->getMidiChannel();
+		program = instr->getProgramChange();
+		if (instr->sendsBankSelect()) {
+		    lsb = instr->getLSB();
+		    msb = instr->getMSB();
+		    useBank = true;
+		}
+	    }
+	    
+	    if (useBank) {
+		
+		// insert a bank select
+		
+		if (msb != 0) {
+		    midiEvent = new MidiEvent(0,
+					      MIDI_CTRL_CHANGE | midiChannel,
+					      MIDI_CONTROLLER_BANK_MSB,
+					      msb);
+		    mtrack.push_back(midiEvent);
+		}
+
+		if (lsb != 0) {
+		    midiEvent = new MidiEvent(0,
+					      MIDI_CTRL_CHANGE | midiChannel,
+					      MIDI_CONTROLLER_BANK_LSB,
+					      lsb);
+		    mtrack.push_back(midiEvent);
+		}
+	    }
+
+	    // insert a program change
+	    midiEvent = new MidiEvent(0,  // time
+				      MIDI_PROG_CHANGE | midiChannel,
+				      program);
+	    mtrack.push_back(midiEvent);
+
+	    if (instr) {
+
+		// MidiInstrument parameters: volume, pan, attack,
+		// release, filter, resonance, chorus, reverb.  Always
+		// write these: the Instrument has an additional parameter
+		// to record whether they should be sent, but it isn't
+		// actually set anywhere so we have to ignore it.
+
+		static int controllers[] = {
+		    MIDI_CONTROLLER_ATTACK,
+		    MIDI_CONTROLLER_RELEASE,
+		    MIDI_CONTROLLER_FILTER,
+		    MIDI_CONTROLLER_RESONANCE,
+		    MIDI_CONTROLLER_CHORUS,
+		    MIDI_CONTROLLER_REVERB
+		};
+
+		mtrack.push_back
+		    (new MidiEvent(0, MIDI_CTRL_CHANGE | midiChannel,
+				   MIDI_CONTROLLER_VOLUME, instr->getVolume()));
+
+		mtrack.push_back
+		    (new MidiEvent(0, MIDI_CTRL_CHANGE | midiChannel,
+				   MIDI_CONTROLLER_PAN, instr->getPan()));
+
+		for (int i = 0; i < sizeof(controllers)/sizeof(controllers[0]); ++i) {
+		    try {
+			mtrack.push_back
+			    (new MidiEvent
+			     (0, MIDI_CTRL_CHANGE | midiChannel, controllers[i],
+			      instr->getControllerValue(controllers[i])));
+		    } catch (...) {
+			/* do nothing */
+		    }
+		}
+	    } // if (instr)
+	} // if (firstSegmentThisTrack)
+
+        timeT segmentMidiDuration =
+            ((*segment)->getEndMarkerTime() -
+             (*segment)->getStartTime()) * m_timingDivision /
+            crotchetDuration;
+
+        for (Segment::iterator el = (*segment)->begin();
+	     (*segment)->isBeforeEndMarker(el); ++el) {
+            midiEventAbsoluteTime =
+                (*el)->getAbsoluteTime() + (*segment)->getDelay();
+
+            timeT absoluteTimeLimit = midiEventAbsoluteTime;
+            if ((*segment)->isRepeating()) {
+                absoluteTimeLimit = ((*segment)->getRepeatEndTime() - 1) +
+		    (*segment)->getDelay();
+            }
+
+            if ((*segment)->getRealTimeDelay() != RealTime::zeroTime) {
+                RealTime evRT = comp.getElapsedRealTime(midiEventAbsoluteTime);
+                timeT timeBeforeDelay = midiEventAbsoluteTime;
+                midiEventAbsoluteTime = comp.getElapsedTimeForRealTime
+		    (evRT + (*segment)->getRealTimeDelay());
+                absoluteTimeLimit += (midiEventAbsoluteTime - timeBeforeDelay);
+            }
+
+            midiEventAbsoluteTime =
+                midiEventAbsoluteTime * m_timingDivision / crotchetDuration;
+            absoluteTimeLimit =
+                absoluteTimeLimit * m_timingDivision / crotchetDuration;
+
+            while (midiEventAbsoluteTime <= absoluteTimeLimit) {
+
+                try {
+
+                    if ((*el)->isa(Note::EventType)) {
+                        if ((*el)->has(BaseProperties::VELOCITY))
+                            midiVelocity = (*el)->get
+				<Int>(BaseProperties::VELOCITY);
+                        else
+                            midiVelocity = 127;
+
+                        // Get the sounding time for the matching NOTE_OFF.
+                        // We use SegmentPerformanceHelper::getSoundingDuration()
+                        // to work out the tied duration of the NOTE.
+                        timeT soundingDuration = helper.getSoundingDuration(el);
+                        if (soundingDuration > 0) {
+
+                            timeT midiEventEndTime = midiEventAbsoluteTime +
+				soundingDuration * m_timingDivision /
+				crotchetDuration;
+
+                            long pitch = 60;
+                            (*el)->get
+				<Int>(BaseProperties::PITCH, pitch);
+                            pitch += (*segment)->getTranspose();
+
+                            // insert the NOTE_ON at the appropriate channel
+                            //
+                            midiEvent =
+                                new MidiEvent(midiEventAbsoluteTime,
+                                              MIDI_NOTE_ON | midiChannel,
+                                              pitch,
+                                              midiVelocity);
+
+                            mtrack.push_back(midiEvent);
+
+                            // insert the matching NOTE OFF
+                            //
+                            midiEvent =
+                                new MidiEvent(midiEventEndTime,
+                                              MIDI_NOTE_OFF | midiChannel,
+                                              pitch,
+                                              127); // full volume silence
+
+                            mtrack.push_back(midiEvent);
+                        }
+                    } else if ((*el)->isa(PitchBend::EventType)) {
+                        PitchBend pb(**el);
+                        midiEvent =
+                            new MidiEvent(midiEventAbsoluteTime,
+                                          MIDI_PITCH_BEND | midiChannel,
+                                          pb.getLSB(), pb.getMSB());
+
+                        mtrack.push_back(midiEvent);
+                    } else if ((*el)->isa(Rosegarden::Key::EventType)) {
+                        Rosegarden::Key key(**el);
+
+                        int accidentals = key.getAccidentalCount();
+                        if (!key.isSharp())
+                            accidentals = -accidentals;
+
+                        // stack out onto the meta string
+                        //
+                        std::string metaMessage;
+                        metaMessage += MidiByte(accidentals);
+                        metaMessage += MidiByte(key.isMinor());
+
+                        midiEvent =
+                            new MidiEvent(midiEventAbsoluteTime,
+                                          MIDI_FILE_META_EVENT,
+                                          MIDI_KEY_SIGNATURE,
+                                          metaMessage);
+
+                        //mtrack.push_back(midiEvent);
+
+                    } else if ((*el)->isa(Controller::EventType)) {
+                        Controller c(**el);
+                        midiEvent =
+                            new MidiEvent(midiEventAbsoluteTime,
+                                          MIDI_CTRL_CHANGE | midiChannel,
+                                          c.getNumber(), c.getValue());
+
+                        mtrack.push_back(midiEvent);
+                    } else if ((*el)->isa(ProgramChange::EventType)) {
+                        ProgramChange pc(**el);
+                        midiEvent =
+                            new MidiEvent(midiEventAbsoluteTime,
+                                          MIDI_PROG_CHANGE | midiChannel,
+                                          pc.getProgram());
+
+                        mtrack.push_back(midiEvent);
+                    } else if ((*el)->isa(SystemExclusive::EventType)) {
+                        SystemExclusive s(**el);
+                        std::string data = s.getRawData();
+
+                        // check for closing EOX and add one if none found
+                        //
+                        if (MidiByte(data[data.length() - 1]) != MIDI_END_OF_EXCLUSIVE) {
+                            data += MIDI_END_OF_EXCLUSIVE;
+                        }
+
+                        // construct plain SYSEX event
+                        //
+                        midiEvent = new MidiEvent(midiEventAbsoluteTime,
+                                                  MIDI_SYSTEM_EXCLUSIVE,
+                                                  data);
+
+                        mtrack.push_back(midiEvent);
+
+                    } else if ((*el)->isa(ChannelPressure::EventType)) {
+                        ChannelPressure cp(**el);
+                        midiEvent =
+                            new MidiEvent(midiEventAbsoluteTime,
+                                          MIDI_CHNL_AFTERTOUCH | midiChannel,
+                                          cp.getPressure());
+
+                        mtrack.push_back(midiEvent);
+                    } else if ((*el)->isa(KeyPressure::EventType)) {
+                        KeyPressure kp(**el);
+                        midiEvent =
+                            new MidiEvent(midiEventAbsoluteTime,
+                                          MIDI_POLY_AFTERTOUCH | midiChannel,
+                                          kp.getPitch(), kp.getPressure());
+
+                        mtrack.push_back(midiEvent);
+                    } else if ((*el)->isa(Text::EventType)) {
+                        Text text(**el);
+                        std::string metaMessage = text.getText();
+
+                        MidiByte midiTextType = MIDI_TEXT_EVENT;
+
+                        if (text.getTextType() == Text::Lyric) {
+                            midiTextType = MIDI_LYRIC;
+                        }
+
+                        if (text.getTextType() != Text::Annotation) {
+                            // (we don't write annotations)
+
+                            midiEvent =
+                                new MidiEvent(midiEventAbsoluteTime,
+                                              MIDI_FILE_META_EVENT,
+                                              midiTextType,
+                                              metaMessage);
+
+                            mtrack.push_back(midiEvent);
+                        }
+                    } else if ((*el)->isa(Note::EventRestType)) {
+                        // skip legitimately
+                    } else {
+                        /*
+                          cerr << "MidiFile::convertToMidi - "
+                          << "unsupported MidiType \""
+                          << (*el)->getType()
+                          << "\" at export"
+                          << std::endl;
+                        */
+                    }
+
+                } catch (MIDIValueOutOfRange r) {
+#ifdef MIDI_DEBUG
+                    std::cerr << "MIDI value out of range at "
+			      << (*el)->getAbsoluteTime() << std::endl;
+#endif
+
+                } catch (Event::NoData d) {
+#ifdef MIDI_DEBUG
+                    std::cerr << "Caught Event::NoData at "
+			      << (*el)->getAbsoluteTime() << ", message is:"
+			      << std::endl << d.getMessage() << std::endl;
+#endif
+
+                } catch (Event::BadType b) {
+#ifdef MIDI_DEBUG
+                    std::cerr << "Caught Event::BadType at "
+			      << (*el)->getAbsoluteTime() << ", message is:"
+			      << std::endl << b.getMessage() << std::endl;
+#endif
+
+                } catch (SystemExclusive::BadEncoding e) {
+#ifdef MIDI_DEBUG
+                    std::cerr << "Caught bad SysEx encoding at "
+			      << (*el)->getAbsoluteTime() << std::endl;
+#endif
+
+                }
+
+                if (segmentMidiDuration > 0) {
+                    midiEventAbsoluteTime += segmentMidiDuration;
+                } else
+                    break;
+            }
+        }
+    }
+
+    // Now gnash through the MIDI events and turn the absolute times
+    // into delta times.
+    //
+    //
+    MidiTrack::iterator it;
+    timeT deltaTime, lastMidiTime;
+
+    for (TrackId i = 0; i < m_numberOfTracks; i++) {
+        lastMidiTime = 0;
+
+        // First sort the track with the MidiEvent comparator.  Use
+        // stable_sort so that events with equal times are maintained
+        // in their current order (important for e.g. bank-program
+        // pairs, or the controllers at the start of the track which
+        // should follow the program so we can treat them correctly
+        // when re-reading).
+        //
+        std::stable_sort(m_midiComposition[i].begin(),
+                         m_midiComposition[i].end(),
+                         MidiEventCmp());
+
+        for (it = m_midiComposition[i].begin();
+                it != m_midiComposition[i].end();
+                it++) {
+            deltaTime = (*it)->getTime() - lastMidiTime;
+            lastMidiTime = (*it)->getTime();
+            (*it)->setTime(deltaTime);
+        }
+
+        // Insert end of track event (delta time = 0)
+        //
+        midiEvent = new MidiEvent(0, MIDI_FILE_META_EVENT,
+                                  MIDI_END_OF_TRACK, "");
+
+        m_midiComposition[i].push_back(midiEvent);
+
+    }
+
+    return ;
+}
+
+
+
+// Convert an integer into a two byte representation and
+// write out to the MidiFile.
+//
+void
+MidiFile::intToMidiBytes(std::ofstream* midiFile, int number)
+{
+    MidiByte upper;
+    MidiByte lower;
+
+    upper = (number & 0xFF00) >> 8;
+    lower = (number & 0x00FF);
+
+    *midiFile << (MidiByte) upper;
+    *midiFile << (MidiByte) lower;
+
+}
+
+void
+MidiFile::longToMidiBytes(std::ofstream* midiFile, unsigned long number)
+{
+    MidiByte upper1;
+    MidiByte lower1;
+    MidiByte upper2;
+    MidiByte lower2;
+
+    upper1 = (number & 0xff000000) >> 24;
+    lower1 = (number & 0x00ff0000) >> 16;
+    upper2 = (number & 0x0000ff00) >> 8;
+    lower2 = (number & 0x000000ff);
+
+    *midiFile << (MidiByte) upper1;
+    *midiFile << (MidiByte) lower1;
+    *midiFile << (MidiByte) upper2;
+    *midiFile << (MidiByte) lower2;
+
+}
+
+// Turn a delta time into a MIDI time - overlapping into
+// a maximum of four bytes using the MSB as the carry on
+// flag.
+//
+std::string
+MidiFile::longToVarBuffer(unsigned long number)
+{
+    std::string rS;
+
+    long inNumber = number;
+    long outNumber;
+
+    // get the lowest 7 bits of the number
+    outNumber = number & 0x7f;
+
+    // Shift and test and move the numbers
+    // on if we need them - setting the MSB
+    // as we go.
+    //
+    while ((inNumber >>= 7 ) > 0) {
+        outNumber <<= 8;
+        outNumber |= 0x80;
+        outNumber += (inNumber & 0x7f);
+    }
+
+    // Now move the converted number out onto the buffer
+    //
+    while (true) {
+        rS += (MidiByte)(outNumber & 0xff);
+        if (outNumber & 0x80)
+            outNumber >>= 8;
+        else
+            break;
+    }
+
+    return rS;
+}
+
+
+
+// Write out the MIDI file header
+//
+bool
+MidiFile::writeHeader(std::ofstream* midiFile)
+{
+    // Our identifying Header string
+    //
+    *midiFile << MIDI_FILE_HEADER.c_str();
+
+    // Write number of Bytes to follow
+    //
+    *midiFile << (MidiByte) 0x00;
+    *midiFile << (MidiByte) 0x00;
+    *midiFile << (MidiByte) 0x00;
+    *midiFile << (MidiByte) 0x06;
+
+    // Write File Format
+    //
+    *midiFile << (MidiByte) 0x00;
+    *midiFile << (MidiByte) m_format;
+
+    // Number of Tracks we're writing out
+    //
+    intToMidiBytes(midiFile, m_numberOfTracks);
+
+    // Timing Division
+    //
+    intToMidiBytes(midiFile, m_timingDivision);
+
+    return (true);
+}
+
+// Write a MIDI track to file
+//
+bool
+MidiFile::writeTrack(std::ofstream* midiFile, TrackId trackNumber)
+{
+    bool retOK = true;
+    MidiByte eventCode = 0;
+    MidiTrack::iterator midiEvent;
+
+    // First we write into the trackBuffer, then write it out to the
+    // file with it's accompanying length.
+    //
+    string trackBuffer;
+
+    long progressTotal = m_midiComposition[trackNumber].size();
+    long progressCount = 0;
+
+    for (midiEvent = m_midiComposition[trackNumber].begin();
+            midiEvent != m_midiComposition[trackNumber].end();
+            midiEvent++) {
+        // Write the time to the buffer in MIDI format
+        //
+        //
+        trackBuffer += longToVarBuffer((*midiEvent)->getTime());
+
+        if ((*midiEvent)->isMeta()) {
+            trackBuffer += MIDI_FILE_META_EVENT;
+            trackBuffer += (*midiEvent)->getMetaEventCode();
+
+            // Variable length number field
+            trackBuffer += longToVarBuffer((*midiEvent)->
+                                           getMetaMessage().length());
+
+            trackBuffer += (*midiEvent)->getMetaMessage();
+        } else {
+            // Send the normal event code (with encoded channel information)
+            //
+            // Fix for 674731 by Pedro Lopez-Cabanillas (20030531)
+            if (((*midiEvent)->getEventCode() != eventCode) ||
+                    ((*midiEvent)->getEventCode() == MIDI_SYSTEM_EXCLUSIVE)) {
+                trackBuffer += (*midiEvent)->getEventCode();
+                eventCode = (*midiEvent)->getEventCode();
+            }
+
+            // Send the relevant data
+            //
+            switch ((*midiEvent)->getMessageType()) {
+            case MIDI_NOTE_ON:
+            case MIDI_NOTE_OFF:
+            case MIDI_POLY_AFTERTOUCH:
+                trackBuffer += (*midiEvent)->getData1();
+                trackBuffer += (*midiEvent)->getData2();
+                break;
+
+            case MIDI_CTRL_CHANGE:
+                trackBuffer += (*midiEvent)->getData1();
+                trackBuffer += (*midiEvent)->getData2();
+                break;
+
+            case MIDI_PROG_CHANGE:
+                trackBuffer += (*midiEvent)->getData1();
+                break;
+
+            case MIDI_CHNL_AFTERTOUCH:
+                trackBuffer += (*midiEvent)->getData1();
+                break;
+
+            case MIDI_PITCH_BEND:
+                trackBuffer += (*midiEvent)->getData1();
+                trackBuffer += (*midiEvent)->getData2();
+                break;
+
+            case MIDI_SYSTEM_EXCLUSIVE:
+
+                // write out message length
+                trackBuffer +=
+                    longToVarBuffer((*midiEvent)->getMetaMessage().length());
+
+                // now the message
+                trackBuffer += (*midiEvent)->getMetaMessage();
+
+                break;
+
+            default:
+#ifdef MIDI_DEBUG
+
+                std::cerr << "MidiFile::writeTrack()"
+                << " - cannot write unsupported MIDI event"
+                << endl;
+#endif
+
+                break;
+            }
+        }
+
+        // For the moment just keep the app updating until we work
+        // out a good way of accounting for this write.
+        //
+        ++progressCount;
+
+        if (progressCount % 500 == 0) {
+            emit setProgress(progressCount * 100 / progressTotal);
+            kapp->processEvents(500);
+        }
+    }
+
+    // Now we write the track - First the standard header..
+    //
+    *midiFile << MIDI_TRACK_HEADER.c_str();
+
+    // ..now the length of the buffer..
+    //
+    longToMidiBytes(midiFile, (long)trackBuffer.length());
+
+    // ..then the buffer itself..
+    //
+    *midiFile << trackBuffer;
+
+    return (retOK);
+}
+
+// Writes out a MIDI file from the internal Midi representation
+//
+bool
+MidiFile::write()
+{
+    bool retOK = true;
+
+    std::ofstream *midiFile =
+        new std::ofstream(m_fileName.c_str(), ios::out | ios::binary);
+
+
+    if (!(*midiFile)) {
+#ifdef MIDI_DEBUG
+        std::cerr << "MidiFile::write() - can't write file" << endl;
+#endif
+
+        m_format = MIDI_FILE_NOT_LOADED;
+        return false;
+    }
+
+    // Write out the Header
+    //
+    writeHeader(midiFile);
+
+    // And now the tracks
+    //
+    for (TrackId i = 0; i < m_numberOfTracks; i++ )
+        if (!writeTrack(midiFile, i))
+            retOK = false;
+
+    midiFile->close();
+
+    if (!retOK)
+        m_format = MIDI_FILE_NOT_LOADED;
+
+    return (retOK);
+}
+
+// Delete dead NOTE OFF and NOTE ON/Zero Velocty Events after
+// reading them and modifying their relevant NOTE ONs
+//
+bool
+MidiFile::consolidateNoteOffEvents(TrackId track)
+{
+    MidiTrack::iterator nOE, mE = m_midiComposition[track].begin();
+    bool notesOnTrack = false;
+    bool noteOffFound;
+
+    for (;mE != m_midiComposition[track].end(); mE++) {
+        if ((*mE)->getMessageType() == MIDI_NOTE_ON && (*mE)->getVelocity() > 0) {
+            // We've found a note - flag it
+            //
+            if (!notesOnTrack)
+                notesOnTrack = true;
+
+            noteOffFound = false;
+
+            for (nOE = mE; nOE != m_midiComposition[track].end(); nOE++) {
+                if (((*nOE)->getChannelNumber() == (*mE)->getChannelNumber()) &&
+                        ((*nOE)->getPitch() == (*mE)->getPitch()) &&
+                        ((*nOE)->getMessageType() == MIDI_NOTE_OFF ||
+                         ((*nOE)->getMessageType() == MIDI_NOTE_ON &&
+                          (*nOE)->getVelocity() == 0x00))) {
+                    (*mE)->setDuration((*nOE)->getTime() - (*mE)->getTime());
+
+                    delete *nOE;
+                    m_midiComposition[track].erase(nOE);
+
+                    noteOffFound = true;
+                    break;
+                }
+            }
+
+            // If no matching NOTE OFF has been found then set
+            // Event duration to length of Segment
+            //
+            if (noteOffFound == false) {
+                --nOE; // avoid crash due to nOE == track.end()
+                (*mE)->setDuration((*nOE)->getTime() - (*mE)->getTime());
+            }
+        }
+    }
+
+    return notesOnTrack;
+}
+
+// Clear down the MidiFile Composition
+//
+void
+MidiFile::clearMidiComposition()
+{
+    for (MidiComposition::iterator ci = m_midiComposition.begin();
+            ci != m_midiComposition.end(); ++ci) {
+
+        //std::cerr << "MidiFile::clearMidiComposition: track " << ci->first << std::endl;
+
+        for (MidiTrack::iterator ti = ci->second.begin();
+                ti != ci->second.end(); ++ti) {
+            delete *ti;
+        }
+
+        ci->second.clear();
+    }
+
+    m_midiComposition.clear();
+    m_trackChannelMap.clear();
+}
+
+// Doesn't do anything yet - doesn't need to.  We need to satisfy
+// the pure virtual function in the base class.
+//
+void
+MidiFile::close()
+{}
+
+
+
+}
+
+#include "MidiFile.moc"
diff --git a/src/sound/MidiFile.h b/src/sound/MidiFile.h
new file mode 100644
index 0000000..da97374
--- /dev/null
+++ b/src/sound/MidiFile.h
@@ -0,0 +1,173 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+
+#ifndef _ROSEGARDEN_MIDI_FILE_H_
+#define _ROSEGARDEN_MIDI_FILE_H_
+
+#include <fstream>
+#include <string>
+#include <list>
+#include <map>
+
+#include <qobject.h>
+
+#include "Midi.h"
+#include "MidiEvent.h"
+#include "Composition.h"
+#include "SoundFile.h"
+
+// Conversion class for Composition to and
+// from MIDI Files.  Despite the fact you can reuse this
+// object it's probably safer just to create it for a
+// single way conversion and then throw it away (MIDI
+// to Composition conversion invalidates the internal
+// MIDI model).
+//
+// Derived from SoundFile but still had some features
+// in common with it which could theoretically be moved
+// up into the base for use in other derived classes.
+//
+// [rwb]
+//
+//
+
+namespace Rosegarden
+{
+
+// Our internal MIDI structure is just a list of MidiEvents.
+// We use a list and not a set because we want the order of
+// the events to be arbitrary until we explicitly sort them
+// (necessary when converting Composition absolute times to
+// MIDI delta times).
+//
+typedef std::vector<MidiEvent *> MidiTrack;
+typedef std::map<unsigned int, MidiTrack> MidiComposition;
+
+class Studio;
+
+class MidiFile : public QObject, public SoundFile
+{
+    Q_OBJECT
+public:
+
+    typedef enum
+    {
+        MIDI_SINGLE_TRACK_FILE          = 0x00,
+        MIDI_SIMULTANEOUS_TRACK_FILE    = 0x01,
+        MIDI_SEQUENTIAL_TRACK_FILE      = 0x02,
+        MIDI_CONVERTED_TO_APPLICATION   = 0xFE,
+        MIDI_FILE_NOT_LOADED            = 0xFF
+    } MIDIFileFormatType;
+
+    typedef enum
+    {
+        CONVERT_REPLACE,
+        CONVERT_AUGMENT,
+        CONVERT_APPEND
+    } ConversionType;
+
+    MidiFile(Studio *studio);
+    MidiFile (const std::string &fn, Studio *studio);
+    ~MidiFile();
+
+    // Declare our virtuals
+    //
+    virtual bool open();
+    virtual bool write();
+    virtual void close();
+
+    int timingDivision() { return m_timingDivision; }
+    MIDIFileFormatType format() { return m_format; }
+    unsigned int numberOfTracks() { return m_numberOfTracks; }
+    bool hasTimeChanges() { return m_containsTimeChanges; }
+
+    // If a file open or save failed
+    std::string getError() { return m_error; }
+
+    /**
+     * Convert a MIDI file to a Rosegarden composition.  Return true
+     * for success.
+     */
+    bool convertToRosegarden(Composition &c, ConversionType type);
+
+    /**
+     * Convert a Rosegarden composition to MIDI format, storing the
+     * result internally for later writing.
+     */
+    void convertToMidi(Composition &comp);
+
+signals:
+    void setProgress(int);
+    void incrementProgress(int);
+    
+private:
+
+    int                    m_timingDivision;   // pulses per quarter note
+    MIDIFileFormatType     m_format;
+    unsigned int           m_numberOfTracks;
+    bool                   m_containsTimeChanges;
+
+    // Internal counters
+    //
+    long                  m_trackByteCount;
+    bool                  m_decrementCount;
+
+    // Internal MidiComposition
+    //
+    MidiComposition       m_midiComposition;
+    std::map<int, int>    m_trackChannelMap;
+
+    // Clear the m_midiComposition
+    //
+    void clearMidiComposition();
+
+    // Split the tasks up with these top level private methods
+    //
+    bool parseHeader(const std::string& midiHeader);
+    bool parseTrack(std::ifstream* midiFile, unsigned int &trackNum);
+    bool writeHeader(std::ofstream* midiFile);
+    bool writeTrack(std::ofstream* midiFile, unsigned int trackNum);
+
+    bool consolidateNoteOffEvents(TrackId track);
+
+    // Internal convenience functions
+    //
+    int midiBytesToInt(const std::string& bytes);
+    long midiBytesToLong(const std::string& bytes);
+    long getNumberFromMidiBytes(std::ifstream* midiFile, int firstByte = -1);
+    MidiByte getMidiByte(std::ifstream* midiFile);
+    std::string getMidiBytes(std::ifstream* midiFile,
+                                   unsigned long bytes);
+    bool skipToNextTrack(std::ifstream *midiFile);
+    void intToMidiBytes(std::ofstream* midiFile, int number);
+    void longToMidiBytes(std::ofstream* midiFile, unsigned long number);
+    std::string longToVarBuffer(unsigned long number);
+
+    // The pointer to the Studio for Instrument stuff
+    //
+    Studio   *m_studio;
+
+    std::string m_error;
+};
+
+}
+
+#endif // _ROSEGARDEN_MIDI_FILE_H_
diff --git a/src/sound/MidiMapping.xml b/src/sound/MidiMapping.xml
new file mode 100644
index 0000000..13b7138
--- /dev/null
+++ b/src/sound/MidiMapping.xml
@@ -0,0 +1,133 @@
+
+<midi>
+    <bank id="0" name="General MIDI">
+        <program id="0" name="Acoustic Grand Piano"/>
+        <program id="1" name="Bright Acoustic Piano"/>
+        <program id="2" name="Electric Grand Piano"/>
+        <program id="3" name="Honky-tonk Piano"/>
+        <program id="4" name="Electric Piano 1"/>
+        <program id="5" name="Electric Piano 2"/>
+        <program id="6" name="Harpsichord"/>
+        <program id="7" name="Clavi"/>
+        <program id="8" name="Celesta"/>
+        <program id="9" name="Glockenspiel"/>
+        <program id="10" name="Music Box"/>
+        <program id="11" name="Vibraphone"/>
+        <program id="12" name="Marimba"/>
+        <program id="13" name="Xylophone"/>
+        <program id="14" name="Tubular Bells"/>
+        <program id="15" name="Dulcimer"/>
+        <program id="16" name="Drawbar Organ"/>
+        <program id="17" name="Percussive Organ"/>
+        <program id="18" name="Rock Organ"/>
+        <program id="19" name="Church Organ"/>
+        <program id="20" name="Reed Organ"/>
+        <program id="21" name="Accordion"/>
+        <program id="22" name="Harmonica"/>
+        <program id="23" name="Tango Accordion"/>
+        <program id="24" name="Acoustic Guitar (nylon)"/>
+        <program id="25" name="Acoustic Guitar (steel)"/>
+        <program id="26" name="Electric Guitar (jazz)"/>
+        <program id="27" name="Electric Guitar (clean)"/>
+        <program id="28" name="Electric Guitar (muted)"/>
+        <program id="29" name="Overdriven Guitar"/>
+        <program id="30" name="Distortion Guitar"/>
+        <program id="31" name="Guitar harmonics"/>
+        <program id="32" name="Acoustic Bass"/>
+        <program id="33" name="Fingered Bass"/>
+        <program id="34" name="Picked Bass"/>
+        <program id="35" name="Fretless Bass"/>
+        <program id="36" name="Slap Bass 1"/>
+        <program id="37" name="Slap Bass 2"/>
+        <program id="38" name="Synth Bass 1"/>
+        <program id="39" name="Synth Bass 2"/>
+        <program id="40" name="Violin"/>
+        <program id="41" name="Viola"/>
+        <program id="42" name="Cello"/>
+        <program id="43" name="Contrabass"/>
+        <program id="44" name="Tremolo Strings"/>
+        <program id="45" name="Pizzicato Strings"/>
+        <program id="46" name="Orchestral Harp"/>
+        <program id="47" name="Timpani"/>
+        <program id="48" name="String Ensemble 1"/>
+        <program id="49" name="String Ensemble 2"/>
+        <program id="50" name="SynthStrings 1"/>
+        <program id="51" name="SynthStrings 2"/>
+        <program id="52" name="Choir Aahs"/>
+        <program id="53" name="Voice Oohs"/>
+        <program id="54" name="Synth Voice"/>
+        <program id="55" name="Orchestra Hit"/>
+        <program id="56" name="Trumpet"/>
+        <program id="57" name="Trombone"/>
+        <program id="58" name="Tuba"/>
+        <program id="59" name="Muted Trumpet"/>
+        <program id="60" name="French Horn"/>
+        <program id="61" name="Brass Section"/>
+        <program id="62" name="SynthBrass 1"/>
+        <program id="63" name="SynthBrass 2"/>
+        <program id="64" name="Soprano Sax"/>
+        <program id="65" name="Alto Sax"/>
+        <program id="66" name="Tenor Sax"/>
+        <program id="67" name="Baritone Sax"/>
+        <program id="68" name="Oboe"/>
+        <program id="69" name="English Horn"/>
+        <program id="70" name="Bassoon"/>
+        <program id="71" name="Clarinet"/>
+        <program id="72" name="Piccolo"/>
+        <program id="73" name="Flute"/>
+        <program id="74" name="Recorder"/>
+        <program id="75" name="Pan Flute"/>
+        <program id="76" name="Blown Bottle"/>
+        <program id="77" name="Shakuhachi"/>
+        <program id="78" name="Whistle"/>
+        <program id="79" name="Ocarina"/>
+        <program id="80" name="Lead 1 (square)"/>
+        <program id="81" name="Lead 2 (sawtooth)"/>
+        <program id="82" name="Lead 3 (calliope)"/>
+        <program id="83" name="Lead 4 (chiff)"/>
+        <program id="84" name="Lead 5 (charang)"/>
+        <program id="85" name="Lead 6 (voice)"/>
+        <program id="86" name="Lead 7 (fifths)"/>
+        <program id="87" name="Lead 8 (bass + lead)"/>
+        <program id="88" name="Pad 1 (new age)"/>
+        <program id="89" name="Pad 2 (warm)"/>
+        <program id="90" name="Pad 3 (polysynth)"/>
+        <program id="91" name="Pad 4 (choir)"/>
+        <program id="92" name="Pad 5 (bowed)"/>
+        <program id="93" name="Pad 6 (metallic)"/>
+        <program id="94" name="Pad 7 (halo)"/>
+        <program id="95" name="Pad 8 (sweep)"/>
+        <program id="96" name="FX 1 (rain)"/>
+        <program id="97" name="FX 2 (soundtrack)"/>
+        <program id="98" name="FX 3 (crystal)"/>
+        <program id="99" name="FX 4 (atmosphere)"/>
+        <program id="100" name="FX 5 (brightness)"/>
+        <program id="101" name="FX 6 (goblins)"/>
+        <program id="102" name="FX 7 (echoes)"/>
+        <program id="103" name="FX 8 (sci-fi)"/>
+        <program id="104" name="Sitar"/>
+        <program id="105" name="Banjo"/>
+        <program id="106" name="Shamisen"/>
+        <program id="107" name="Koto"/>
+        <program id="108" name="Kalimba"/>
+        <program id="109" name="Bag pipe"/>
+        <program id="110" name="Fiddle"/>
+        <program id="111" name="Shanai"/>
+        <program id="112" name="Tinkle Bell"/>
+        <program id="113" name="Agogo"/>
+        <program id="114" name="Steel Drums"/>
+        <program id="115" name="Woodblock"/>
+        <program id="116" name="Taiko Drum"/>
+        <program id="117" name="Melodic Tom"/>
+        <program id="118" name="Synth Drum"/>
+        <program id="119" name="Reverse Cymbal"/>
+        <program id="120" name="Guitar Fret Noise"/>
+        <program id="121" name="Breath Noise"/>
+        <program id="122" name="Seashore"/>
+        <program id="123" name="Bird Tweet"/>
+        <program id="124" name="Telephone Ring"/>
+        <program id="125" name="Helicopter"/>
+        <program id="126" name="Applause"/>
+        <program id="127" name="Gunshot"/>
+    </bank>
+</midi>
diff --git a/src/sound/PeakFile.cpp b/src/sound/PeakFile.cpp
new file mode 100644
index 0000000..8881114
--- /dev/null
+++ b/src/sound/PeakFile.cpp
@@ -0,0 +1,1033 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*- /*
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+ 
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+ 
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+ 
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include <cmath>
+#include <cstdlib>
+#include <kapplication.h>
+
+#include <qdatetime.h>
+#include <qstringlist.h>
+#include <qpalette.h>
+#include <kapp.h>
+
+#include "PeakFile.h"
+#include "AudioFile.h"
+#include "Profiler.h"
+
+using std::cout;
+using std::cerr;
+using std::endl;
+
+//#define DEBUG_PEAKFILE 1
+//#define DEBUG_PEAKFILE_BRIEF 1
+//#define DEBUG_PEAKFILE_CACHE 1
+
+#ifdef DEBUG_PEAKFILE
+#define DEBUG_PEAKFILE_BRIEF 1
+#endif
+
+namespace Rosegarden
+{
+
+PeakFile::PeakFile(AudioFile *audioFile):
+        SoundFile(audioFile->getPeakFilename()),
+        m_audioFile(audioFile),
+        m_version( -1),                          // -1 defines new file - start at 0
+        m_format(1),                            // default is 8-bit peak format
+        m_pointsPerValue(0),
+        m_blockSize(256),                       // default block size is 256 samples
+        m_channels(0),
+        m_numberOfPeaks(0),
+        m_positionPeakOfPeaks(0),
+        m_offsetToPeaks(0),
+        m_modificationTime(QDate(1970, 1, 1), QTime(0, 0, 0)),
+        m_chunkStartPosition(0),
+        m_lastPreviewStartTime(0, 0),
+        m_lastPreviewEndTime(0, 0),
+        m_lastPreviewWidth( -1),
+        m_lastPreviewShowMinima(false)
+{}
+
+PeakFile::~PeakFile()
+{}
+
+bool
+PeakFile::open()
+{
+    // Set the file size
+    //
+    QFileInfo info(QString(m_fileName.c_str()));
+    m_fileSize = info.size();
+
+    // If we're already open then don't open again
+    //
+    if (m_inFile && m_inFile->is_open())
+        return true;
+
+    // Open
+    //
+    m_inFile = new std::ifstream(m_fileName.c_str(),
+                                 std::ios::in | std::ios::binary);
+    // Check we're open
+    //
+    if (!(*m_inFile))
+        return false;
+
+    try {
+        parseHeader();
+    } catch (BadSoundFileException s) {
+
+#ifdef DEBUG_PEAKFILE
+        cerr << "PeakFile::open - EXCEPTION \"" << s.getMessage() << "\""
+        << endl;
+#endif
+
+        return false;
+    }
+
+    return true;
+}
+
+void
+PeakFile::parseHeader()
+{
+    if (!(*m_inFile))
+        return ;
+
+    m_inFile->seekg(0, std::ios::beg);
+
+    // get full header length
+    //
+    std::string header = getBytes(128);
+
+#if (__GNUC__ < 3)
+
+    if (header.compare(AUDIO_BWF_PEAK_ID, 0, 4) != 0)
+#else
+
+    if (header.compare(0, 4, AUDIO_BWF_PEAK_ID) != 0)
+#endif
+
+    {
+        throw(BadSoundFileException(m_fileName, "PeakFile::parseHeader - can't find LEVL identifier"));
+    }
+
+    int length = getIntegerFromLittleEndian(header.substr(4, 4));
+
+    // Get the length of the header minus the first 8 bytes
+    //
+    if (length == 0)
+        throw(BadSoundFileException(m_fileName, "PeakFile::parseHeader - can't get header length"));
+
+    // Get the file information
+    //
+    m_version = getIntegerFromLittleEndian(header.substr(8, 4));
+    m_format = getIntegerFromLittleEndian(header.substr(12, 4));
+    m_pointsPerValue = getIntegerFromLittleEndian(header.substr(16, 4));
+    m_blockSize = getIntegerFromLittleEndian(header.substr(20, 4));
+    m_channels = getIntegerFromLittleEndian(header.substr(24, 4));
+    m_numberOfPeaks = getIntegerFromLittleEndian(header.substr(28, 4));
+    m_positionPeakOfPeaks = getIntegerFromLittleEndian(header.substr(32, 4));
+
+    // Read in date string and convert it up to QDateTime
+    //
+    QString dateString = QString(header.substr(40, 28).c_str());
+
+    QStringList dateTime = QStringList::split(":", dateString);
+
+    m_modificationTime.setDate(QDate(dateTime[0].toInt(),
+                                     dateTime[1].toInt(),
+                                     dateTime[2].toInt()));
+
+    m_modificationTime.setTime(QTime(dateTime[3].toInt(),
+                                     dateTime[4].toInt(),
+                                     dateTime[5].toInt(),
+                                     dateTime[6].toInt()));
+
+    //printStats();
+
+}
+
+void
+PeakFile::printStats()
+{
+    cout << endl;
+    cout << "STATS for PeakFile \"" << m_fileName << "\"" << endl
+    << "-----" << endl << endl;
+
+    cout << "  VERSION = " << m_version << endl
+    << "  FORMAT  = " << m_format << endl
+    << "  BYTES/VALUE = " << m_pointsPerValue << endl
+    << "  BLOCKSIZE   = " << m_blockSize << endl
+    << "  CHANNELS    = " << m_channels << endl
+    << "  PEAK FRAMES = " << m_numberOfPeaks << endl
+    << "  PEAK OF PKS = " << m_positionPeakOfPeaks << endl
+    << endl;
+
+    cout << "DATE" << endl
+    << "----" << endl << endl
+    << "  YEAR    = " << m_modificationTime.date().year() << endl
+    << "  MONTH   = " << m_modificationTime.date().month() << endl
+    << "  DAY     = " << m_modificationTime.date().day() << endl
+    << "  HOUR    = " << m_modificationTime.time().hour() << endl
+    << "  MINUTE  = " << m_modificationTime.time().minute()
+    << endl
+    << "  SECOND  = " << m_modificationTime.time().second()
+    << endl
+    << "  MSEC    = " << m_modificationTime.time().msec()
+    << endl << endl;
+}
+
+bool
+PeakFile::write()
+{
+    return write(5); // default update every 5%
+}
+
+bool
+PeakFile::write(unsigned short updatePercentage)
+{
+    if (m_outFile) {
+        m_outFile->close();
+        delete m_outFile;
+    }
+
+    // Attempt to open AudioFile so that we can extract sample data
+    // for preview file generation
+    //
+    try {
+        if (!m_audioFile->open())
+            return false;
+    } catch (BadSoundFileException e) {
+#ifdef DEBUG_PEAKFILE
+        std::cerr << "PeakFile::write - \"" << e.getMessage() << "\"" << std::endl;
+#endif
+
+        return false;
+    }
+
+    // create and test that we've made it
+    m_outFile = new std::ofstream(m_fileName.c_str(),
+                                  std::ios::out | std::ios::binary);
+    if (!(*m_outFile))
+        return false;
+
+    // write out the header
+    writeHeader(m_outFile);
+
+    // and now the peak values
+    writePeaks(updatePercentage, m_outFile);
+
+    return true;
+}
+
+// Close the peak file and tidy up
+//
+void
+PeakFile::close()
+{
+    // Close any input file handle
+    //
+    if (m_inFile && m_inFile->is_open()) {
+        m_inFile->close();
+        delete m_inFile;
+        m_inFile = 0;
+    }
+
+    if (m_outFile == 0)
+        return ;
+
+    // Seek to start of chunk
+    //
+    m_outFile->seekp(m_chunkStartPosition, std::ios::beg);
+
+    // Seek to size field at set it
+    //
+    m_outFile->seekp(4, std::ios::cur);
+    putBytes(m_outFile, getLittleEndianFromInteger(m_bodyBytes + 120, 4));
+
+    // Seek to format and set it (m_format is only set at the
+    // end of writePeaks()
+    //
+    m_outFile->seekp(4, std::ios::cur);
+    putBytes(m_outFile, getLittleEndianFromInteger(m_format, 4));
+
+    // Seek to number of peak frames and write value
+    //
+    m_outFile->seekp(12, std::ios::cur);
+    putBytes(m_outFile,
+             getLittleEndianFromInteger(m_numberOfPeaks, 4));
+
+    // Peak of peaks
+    //
+    putBytes(m_outFile,
+             getLittleEndianFromInteger(m_positionPeakOfPeaks, 4));
+
+    // Seek to date field
+    //
+    m_outFile->seekp(4, std::ios::cur);
+
+    // Set modification time to now
+    //
+    m_modificationTime = m_modificationTime.currentDateTime();
+
+    QString fDate;
+    fDate.sprintf("%04d:%02d:%02d:%02d:%02d:%02d:%03d",
+                  m_modificationTime.date().year(),
+                  m_modificationTime.date().month(),
+                  m_modificationTime.date().day(),
+                  m_modificationTime.time().hour(),
+                  m_modificationTime.time().minute(),
+                  m_modificationTime.time().second(),
+                  m_modificationTime.time().msec());
+
+    std::string dateString(fDate.data());
+
+    // Pad with spaces to make up to 28 bytes long and output
+    //
+    dateString += "     ";
+    putBytes(m_outFile, dateString);
+
+    // Ok, now close and tidy up
+    //
+    m_outFile->close();
+    delete m_outFile;
+    m_outFile = 0;
+}
+
+// If the audio file is more recently modified that the modification time
+// on this peak file then we're invalid.  The action to rectify this is
+// usually to regenerate the peak data.
+//
+bool
+PeakFile::isValid()
+{
+    if (m_audioFile->getModificationDateTime() > m_modificationTime)
+        return false;
+
+    return true;
+}
+
+bool
+PeakFile::writeToHandle(std::ofstream *file,
+                        unsigned short /*updatePercentage*/)
+{
+    // Remember the position where we pass in the ofstream pointer
+    // so we can return there to write close() information.
+    //
+    m_chunkStartPosition = file->tellp();
+
+    return false;
+}
+
+// Build up a header string and then pump it out to the file handle
+//
+void
+PeakFile::writeHeader(std::ofstream *file)
+{
+    if (!file || !(*file))
+        return ;
+
+    std::string header;
+
+    // The "levl" identifer for this chunk
+    //
+    header += AUDIO_BWF_PEAK_ID;
+
+    // Add a four byte version of the size of the header chunk (120
+    // bytes from this point onwards)
+    //
+    header += getLittleEndianFromInteger(120, 4);
+
+    // A four byte version number (incremented every time)
+    //
+    header += getLittleEndianFromInteger(++m_version, 4);
+
+    // Format of the peak points - 1 = unsigned char
+    //                             2 = unsigned short
+    //
+    header += getLittleEndianFromInteger(m_format, 4);
+
+    // Points per value          - 1 = 1 peak and has vertical about x-axis
+    //                             2 = 2 peaks so differs above and below x-axis
+    //
+    // .. hardcode to 2 for the mo
+    m_pointsPerValue = 2;
+    header += getLittleEndianFromInteger(m_pointsPerValue, 4);
+
+    // Block size - default and recommended is 256
+    //
+    header += getLittleEndianFromInteger(m_blockSize, 4);
+
+    // Set channels up if they're currently empty
+    //
+    if (m_channels == 0 && m_audioFile)
+        m_channels = m_audioFile->getChannels();
+
+    // Peak channels - same as AudioFile channels
+    //
+    header += getLittleEndianFromInteger(m_channels, 4);
+
+    // Number of peak frames - we write this at close() and so
+    // for the moment put spacing 0's in.
+    header += getLittleEndianFromInteger(0, 4);
+
+    // Position of peak of peaks - written at close()
+    //
+    header += getLittleEndianFromInteger(0, 4);
+
+    // Offset to start of peaks - usually the total size of this header
+    //
+    header += getLittleEndianFromInteger(128, 4);
+
+    // Creation timestamp - fill in on close() so just use spacing
+    // of 28 bytes for the moment.
+    //
+    header += getLittleEndianFromInteger(0, 28);
+
+    // reserved space - 60 bytes
+    header += getLittleEndianFromInteger(0, 60);
+
+    //cout << "HEADER LENGTH = " << header.length() << endl;
+
+    // write out the header
+    //
+    putBytes(file, header);
+}
+
+bool
+PeakFile::scanToPeak(int peak)
+{
+    if (!m_inFile)
+        return false;
+
+    if (!m_inFile->is_open())
+        return false;
+
+    // Scan to start of chunk and then seek to peak number
+    //
+    ssize_t pos = (ssize_t)m_chunkStartPosition + 128 +
+                  peak * m_format * m_channels * m_pointsPerValue;
+
+    ssize_t off = pos - m_inFile->tellg();
+
+    if (off == 0) {
+        return true;
+    } else if (off < 0) {
+        //	std::cerr << "PeakFile::scanToPeak: warning: seeking backwards for peak " << peak << " (" << m_inFile->tellg() << " -> " << pos << ")" << std::endl;
+        m_inFile->seekg(pos);
+    } else {
+        m_inFile->seekg(off, std::ios::cur);
+    }
+
+    // Ensure we re-read the input buffer if we're
+    // doing buffered reads as it's now meaningless
+    //
+    m_loseBuffer = true;
+
+    if (m_inFile->eof()) {
+        m_inFile->clear();
+        return false;
+    }
+
+    return true;
+}
+
+bool
+PeakFile::scanForward(int numberOfPeaks)
+{
+    if (!m_inFile)
+        return false;
+
+    if (!m_inFile->is_open())
+        return false;
+
+    // Seek forward and number of peaks
+    //
+    m_inFile->seekg(numberOfPeaks * m_format * m_channels * m_pointsPerValue,
+                    std::ios::cur);
+
+    // Ensure we re-read the input buffer
+    m_loseBuffer = true;
+
+    if (m_inFile->eof()) {
+        m_inFile->clear();
+        return false;
+    }
+
+    return true;
+}
+
+
+void
+PeakFile::writePeaks(unsigned short /*updatePercentage*/,
+                     std::ofstream *file)
+{
+    if (!file || !(*file))
+        return ;
+    m_keepProcessing = true;
+
+#ifdef DEBUG_PEAKFILE
+
+    cout << "PeakFile::writePeaks - calculating peaks" << endl;
+#endif
+
+    // Scan to beginning of audio data
+    m_audioFile->scanTo(RealTime(0, 0));
+
+    // Store our samples
+    //
+    std::vector<std::pair<int, int> > channelPeaks;
+    std::string samples;
+    unsigned char *samplePtr;
+
+    int sampleValue;
+    int sampleMax = 0 ;
+    int sampleFrameCount = 0;
+
+    int channels = m_audioFile->getChannels();
+    int bytes = m_audioFile->getBitsPerSample() / 8;
+
+    m_format = bytes;
+    if (bytes == 3 || bytes == 4) // 24-bit PCM or 32-bit float
+        m_format = 2; // write 16-bit PCM instead
+
+    // for the progress dialog
+    unsigned int apprxTotalBytes = m_audioFile->getSize();
+    unsigned int byteCount = 0;
+
+    for (int i = 0; i < channels; i++)
+        channelPeaks.push_back(std::pair<int, int>());
+
+    // clear down info
+    m_numberOfPeaks = 0;
+    m_bodyBytes = 0;
+    m_positionPeakOfPeaks = 0;
+
+    while (m_keepProcessing) {
+        try {
+            samples = m_audioFile->
+                      getBytes(m_blockSize * channels * bytes);
+        } catch (BadSoundFileException e) {
+            std::cerr << "PeakFile::writePeaks: " << e.getMessage()
+            << std::endl;
+            break;
+        }
+
+        // If no bytes or less than the total number of bytes are returned
+        // then break out
+        //
+        if (samples.length() == 0 ||
+                samples.length() < (m_blockSize * m_audioFile->getChannels()
+                                    * bytes))
+            break;
+
+        byteCount += samples.length();
+
+        emit setProgress((int)(double(byteCount) /
+                               double(apprxTotalBytes) * 100.0));
+        kapp->processEvents();
+
+        samplePtr = (unsigned char *)samples.c_str();
+
+        for (int i = 0; i < m_blockSize; i++) {
+            for (unsigned int ch = 0; ch < m_audioFile->getChannels(); ch++) {
+                // Single byte format values range from 0-255 and then
+                // shifted down about the x-axis.  Double byte and above
+                // are already centred about x-axis.
+                //
+                if (bytes == 1) {
+                    // get value
+                    sampleValue = int(*samplePtr) - 128;
+                    samplePtr++;
+                } else if (bytes == 2) {
+                    unsigned char b2 = samplePtr[0];
+                    unsigned char b1 = samplePtr[1];
+                    unsigned int bits = (b1 << 8) + b2;
+                    sampleValue = (short)bits;
+                    samplePtr += 2;
+                } else if (bytes == 3) {
+                    unsigned char b3 = samplePtr[0];
+                    unsigned char b2 = samplePtr[1];
+                    unsigned char b1 = samplePtr[2];
+                    unsigned int bits = (b1 << 24) + (b2 << 16) + (b3 << 8);
+
+                    // write out as 16-bit (m_format == 2)
+                    sampleValue = int(bits) / 65536;
+
+                    samplePtr += 3;
+                } else if (bytes == 4)  // IEEE float (enforced by RIFFAudioFile)
+                {
+                    // write out as 16-bit (m_format == 2)
+                    float val = *(float *)samplePtr;
+                    sampleValue = (int)(32767.0 * val);
+                    samplePtr += 4;
+                } else {
+                    throw(BadSoundFileException(m_fileName, "PeakFile::writePeaks - unsupported bit depth"));
+                }
+
+                // First time for each channel
+                //
+                if (i == 0) {
+                    channelPeaks[ch].first = sampleValue;
+                    channelPeaks[ch].second = sampleValue;
+                } else {
+                    // Compare and store
+                    //
+                    if (sampleValue > channelPeaks[ch].first)
+                        channelPeaks[ch].first = sampleValue;
+
+                    if (sampleValue < channelPeaks[ch].second)
+                        channelPeaks[ch].second = sampleValue;
+                }
+
+                // Store peak of peaks if it fits
+                //
+                if (abs(sampleValue) > sampleMax) {
+                    sampleMax = abs(sampleValue);
+                    m_positionPeakOfPeaks = sampleFrameCount;
+                }
+            }
+
+            // for peak of peaks as well as frame count
+            sampleFrameCount++;
+        }
+
+        // Write absolute peak data in channel order
+        //
+        for (unsigned int i = 0; i < m_audioFile->getChannels(); i++) {
+            putBytes(file, getLittleEndianFromInteger(channelPeaks[i].first,
+                     m_format));
+            putBytes(file, getLittleEndianFromInteger(channelPeaks[i].second,
+                     m_format));
+            m_bodyBytes += m_format * 2;
+        }
+
+        // increment number of peak frames
+        m_numberOfPeaks++;
+    }
+
+#ifdef DEBUG_PEAKFILE
+    cout << "PeakFile::writePeaks - "
+    << "completed peaks" << endl;
+#endif
+
+}
+
+// Get a normalised vector for the preview at a given horizontal resolution.
+// We return a value for each channel and if returnLow is set we also return
+// an interleaved low value for each channel.
+//
+//
+std::vector<float>
+PeakFile::getPreview(const RealTime &startTime,
+                     const RealTime &endTime,
+                     int width,
+                     bool showMinima)
+{
+#ifdef DEBUG_PEAKFILE_BRIEF
+    std::cout << "PeakFile::getPreview - "
+    << "startTime = " << startTime
+    << ", endTime = " << endTime
+    << ", width = " << width
+    << ", showMinima = " << showMinima << std::endl;
+#endif
+
+    if (getSize() == 0) {
+        std::cout << "PeakFile::getPreview - PeakFile size == 0" << std::endl;
+        return std::vector<float>();
+    }
+
+    // Regenerate cache on these conditions
+    //
+    if (!m_peakCache.length()) {
+#ifdef DEBUG_PEAKFILE_CACHE
+        std::cerr << "PeakFile::getPreview - no peak cache" << std::endl;
+#endif
+
+        if (getSize() < (256 *1024)) // if less than 256K PeakFile
+        {
+            // Scan to start of peak data
+            scanToPeak(0);
+            try
+            {
+                m_peakCache = getBytes(m_inFile, getSize() - 128);
+            } catch (BadSoundFileException e)
+            {
+                std::cerr << "PeakFile::getPreview: " << e.getMessage()
+                << std::endl;
+            }
+
+#ifdef DEBUG_PEAKFILE_CACHE
+            std::cout << "PeakFile::getPreview - generated peak cache - "
+            << "size = " << m_peakCache.length() << std::endl;
+#endif
+
+        } else {
+#ifdef DEBUG_PEAKFILE_CACHE
+            std::cout << "PeakFile::getPreview - file size = " << getSize()
+            << ", not generating cache" << std::endl;
+#endif
+
+        }
+    }
+
+    // Check to see if we hit the "lastPreview" cache by comparing the last
+    // query parameters we used.
+    //
+    if (startTime == m_lastPreviewStartTime && endTime == m_lastPreviewEndTime
+            && width == m_lastPreviewWidth && showMinima == m_lastPreviewShowMinima) {
+#ifdef DEBUG_PEAKFILE_CACHE
+        std::cout << "PeakFile::getPreview - hit last preview cache" << std::endl;
+#endif
+
+        return m_lastPreviewCache;
+    } else {
+#ifdef DEBUG_PEAKFILE_CACHE
+        std::cout << "PeakFile::getPreview - last preview " << m_lastPreviewStartTime
+        << " -> " << m_lastPreviewEndTime << ", w " << m_lastPreviewWidth << "; this " << startTime << " -> " << endTime << ", w " << width << std::endl;
+#endif
+
+    }
+
+    // Clear the cache - we need to regenerate it
+    //
+    m_lastPreviewCache.clear();
+
+    int startPeak = getPeak(startTime);
+    int endPeak = getPeak(endTime);
+
+    // Sanity check
+    if (startPeak > endPeak)
+        return m_lastPreviewCache;
+
+    // Actual possible sample length in RealTime
+    //
+    double step = double(endPeak - startPeak) / double(width);
+    std::string peakData;
+    int peakNumber;
+
+#ifdef DEBUG_PEAKFILE_BRIEF
+
+    std::cout << "PeakFile::getPreview - getting preview for \""
+    << m_audioFile->getFilename() << "\"" << endl;
+#endif
+
+    // Get a divisor
+    //
+    float divisor = 0.0f;
+    switch (m_format) {
+    case 1:
+        divisor = SAMPLE_MAX_8BIT;
+        break;
+
+    case 2:
+        divisor = SAMPLE_MAX_16BIT;
+        break;
+
+    default:
+#ifdef DEBUG_PEAKFILE_BRIEF
+
+        std::cout << "PeakFile::getPreview - "
+        << "unsupported peak length format (" << m_format << ")"
+        << endl;
+#endif
+
+        return m_lastPreviewCache;
+    }
+
+    float *hiValues = new float[m_channels];
+    float *loValues = new float[m_channels];
+
+    for (int i = 0; i < width; i++) {
+
+        peakNumber = startPeak + int(double(i) * step);
+        int nextPeakNumber = startPeak + int(double(i + 1) * step);
+
+        // Seek to value
+        //
+        if (!m_peakCache.length()) {
+
+            if (scanToPeak(peakNumber) == false) {
+#ifdef DEBUG_PEAKFILE
+                std::cout << "PeakFile::getPreview: scanToPeak(" << peakNumber << ") failed" << std::endl;
+#endif
+
+                m_lastPreviewCache.push_back(0.0f);
+            }
+        }
+#ifdef DEBUG_PEAKFILE
+        std::cout << "PeakFile::getPreview: step is " << step << ", format * pointsPerValue * chans is " << (m_format * m_pointsPerValue * m_channels) << std::endl;
+        std::cout << "i = " << i << ", peakNumber = " << peakNumber << ", nextPeakNumber = " << nextPeakNumber << std::endl;
+#endif
+
+        for (int ch = 0; ch < m_channels; ch++) {
+            hiValues[ch] = 0.0f;
+            loValues[ch] = 0.0f;
+        }
+
+        // Get peak value over channels
+        //
+        for (int k = 0; peakNumber < nextPeakNumber; ++k) {
+
+            for (int ch = 0; ch < m_channels; ch++) {
+
+                if (!m_peakCache.length()) {
+
+                    try {
+                        peakData = getBytes(m_inFile, m_format * m_pointsPerValue);
+                    } catch (BadSoundFileException e) {
+                        // Problem with the get - probably an EOF
+                        // return the results so far.
+                        //
+#ifdef DEBUG_PEAKFILE
+                        std::cout << "PeakFile::getPreview - \"" << e.getMessage() << "\"\n"
+                        << endl;
+#endif
+
+                        goto done;
+                    }
+#ifdef DEBUG_PEAKFILE
+                    std::cout << "PeakFile::getPreview - "
+                    << "read from file" << std::endl;
+#endif
+
+                } else {
+
+                    int valueNum = peakNumber * m_channels + ch;
+                    int charNum = valueNum * m_format * m_pointsPerValue;
+                    int charLength = m_format * m_pointsPerValue;
+
+                    // Get peak value from the cached string if
+                    // the value is valid.
+                    //
+                    if (charNum + charLength <= m_peakCache.length()) {
+                        peakData = m_peakCache.substr(charNum, charLength);
+#ifdef DEBUG_PEAKFILE
+
+                        std::cout << "PeakFile::getPreview - "
+                        << "hit peakCache" << std::endl;
+#endif
+
+                    }
+                }
+
+
+                if (peakData.length() != (unsigned int)(m_format *
+                                                        m_pointsPerValue)) {
+                    // We didn't get the whole peak block - return what
+                    // we've got so far
+                    //
+#ifdef DEBUG_PEAKFILE
+                    std::cout << "PeakFile::getPreview - "
+                    << "failed to get complete peak block"
+                    << endl;
+#endif
+
+                    goto done;
+                }
+
+                int intDivisor = int(divisor);
+                int inValue =
+                    getIntegerFromLittleEndian(peakData.substr(0, m_format));
+
+                while (inValue > intDivisor) {
+                    inValue -= (1 << (m_format * 8));
+                }
+
+#ifdef DEBUG_PEAKFILE
+                std::cout << "found potential hivalue " << inValue << std::endl;
+#endif
+
+                if (k == 0 || inValue > hiValues[ch]) {
+                    hiValues[ch] = float(inValue);
+                }
+
+                if (m_pointsPerValue == 2) {
+
+                    inValue =
+                        getIntegerFromLittleEndian(
+                            peakData.substr(m_format, m_format));
+
+                    while (inValue > intDivisor) {
+                        inValue -= (1 << (m_format * 8));
+                    }
+
+                    if (k == 0 || inValue < loValues[ch]) {
+                        loValues[ch] = inValue;
+                    }
+                }
+            }
+
+            ++peakNumber;
+        }
+
+        for (int ch = 0; ch < m_channels; ++ch) {
+
+            float value = hiValues[ch] / divisor;
+
+#ifdef DEBUG_PEAKFILE_BRIEF
+
+            std::cout << "VALUE = " << hiValues[ch] / divisor << std::endl;
+#endif
+
+            if (showMinima) {
+                m_lastPreviewCache.push_back(loValues[ch] / divisor);
+            } else {
+                value = fabs(value);
+                if (m_pointsPerValue == 2) {
+                    value = std::max(value, fabsf(loValues[ch] / divisor));
+                }
+                m_lastPreviewCache.push_back(value);
+            }
+        }
+    }
+
+done:
+    resetStream();
+    delete[] hiValues;
+    delete[] loValues;
+
+    // We have a good preview in the cache so store our parameters
+    //
+    m_lastPreviewStartTime = startTime;
+    m_lastPreviewEndTime = endTime;
+    m_lastPreviewWidth = width;
+    m_lastPreviewShowMinima = showMinima;
+
+#ifdef DEBUG_PEAKFILE_BRIEF
+
+    std::cout << "Returning " << m_lastPreviewCache.size() << " items" << std::endl;
+#endif
+
+    return m_lastPreviewCache;
+}
+
+int
+PeakFile::getPeak(const RealTime &time)
+{
+    double frames = ((time.sec * 1000000.0) + time.usec()) *
+                    m_audioFile->getSampleRate() / 1000000.0;
+    return int(frames / double(m_blockSize));
+}
+
+RealTime
+PeakFile::getTime(int peak)
+{
+    int usecs = int((double)peak * (double)m_blockSize *
+                    double(1000000.0) / double(m_audioFile->getSampleRate()));
+    return RealTime(usecs / 1000000, (usecs % 1000000) * 1000);
+}
+
+// Get pairs of split points for areas that exceed a percentage
+// threshold
+//
+std::vector<SplitPointPair>
+PeakFile::getSplitPoints(const RealTime &startTime,
+                         const RealTime &endTime,
+                         int threshold,
+                         const RealTime &minLength)
+{
+    std::vector<SplitPointPair> points;
+    std::string peakData;
+
+    int startPeak = getPeak(startTime);
+    int endPeak = getPeak(endTime);
+
+    if (endPeak < startPeak)
+        return std::vector<SplitPointPair>();
+
+    scanToPeak(startPeak);
+
+    float divisor = 0.0f;
+    switch (m_format) {
+    case 1:
+        divisor = SAMPLE_MAX_8BIT;
+        break;
+
+    case 2:
+        divisor = SAMPLE_MAX_16BIT;
+        break;
+
+    default:
+        return points;
+    }
+
+    float value;
+    float fThreshold = float(threshold) / 100.0;
+    bool belowThreshold = true;
+    RealTime startSplit = RealTime::zeroTime;
+    bool inSplit = false;
+
+    for (int i = startPeak; i < endPeak; i++) {
+        value = 0.0;
+
+        for (int ch = 0; ch < m_channels; ch++) {
+            try {
+                peakData = getBytes(m_inFile, m_format * m_pointsPerValue);
+            } catch (BadSoundFileException e) {
+                std::cerr << "PeakFile::getSplitPoints: "
+                << e.getMessage() << std::endl;
+                break;
+            }
+
+            if (peakData.length() == (unsigned int)(m_format *
+                                                    m_pointsPerValue)) {
+                int peakValue =
+                    getIntegerFromLittleEndian(peakData.substr(0, m_format));
+
+                value += fabs(float(peakValue) / divisor);
+            }
+        }
+
+        value /= float(m_channels);
+
+        if (belowThreshold) {
+            if (value > fThreshold) {
+                startSplit = getTime(i);
+                inSplit = true;
+                belowThreshold = false;
+            }
+        } else {
+            if (value < fThreshold && getTime(i) - startSplit > minLength) {
+                // insert values
+                if (inSplit) {
+                    points.push_back(SplitPointPair(startSplit, getTime(i)));
+                }
+                inSplit = false;
+                belowThreshold = true;
+            }
+        }
+    }
+
+    // if we've got a split point open the close it
+    if (inSplit) {
+        points.push_back(SplitPointPair(startSplit,
+                                        getTime(endPeak)));
+    }
+
+    return points;
+}
+
+
+}
+
+
+#include "PeakFile.moc"
diff --git a/src/sound/PeakFile.h b/src/sound/PeakFile.h
new file mode 100644
index 0000000..26ef71c
--- /dev/null
+++ b/src/sound/PeakFile.h
@@ -0,0 +1,196 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*- /*
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+#include <vector>
+
+#include <qobject.h>
+#include <qdatetime.h>
+
+#include "SoundFile.h"
+#include "RealTime.h"
+
+#ifndef _PEAKFILE_H_
+#define _PEAKFILE_H_
+
+// A PeakFile is generated to the BWF Supplement 3 Peak Envelope Chunk
+// format as defined here:
+//
+// http://www.ebu.ch/pmc_bwf.html
+//
+// To comply with BWF format files this chunk can be embedded into
+// the sample file itself (writeToHandle()) or used to generate an
+// external peak file (write()).  At the moment the only type of file
+// with an embedded peak chunk is the BWF file itself.
+//
+//
+
+
+
+namespace Rosegarden
+{
+
+class AudioFile;
+
+
+typedef std::pair<RealTime, RealTime> SplitPointPair;
+
+class PeakFile : public QObject, public SoundFile
+{
+    Q_OBJECT
+
+public:
+    PeakFile(AudioFile *audioFile);
+    virtual ~PeakFile();
+
+    // Copy constructor
+    //
+    PeakFile(const PeakFile &);
+
+    // Standard file methods
+    //
+    virtual bool open();
+    virtual void close();
+
+    // Write to standard peak file
+    //
+    virtual bool write();
+
+    // Write the file, emit progress signal and process app events
+    //
+    virtual bool write(unsigned short updatePercentage);
+
+    // Write peak chunk to file handle (BWF)
+    //
+    bool writeToHandle(std::ofstream *file, unsigned short updatePercentage);
+
+    // Is the peak file valid and up to date?
+    //
+    bool isValid();
+
+    // Vital file stats
+    //
+    void printStats();
+
+    // Get a preview of a section of the audio file where that section
+    // is "width" pixels.
+    //
+    std::vector<float> getPreview(const RealTime &startTime,
+                                  const RealTime &endTime,
+                                  int width,
+                                  bool showMinima);
+
+    AudioFile* getAudioFile() { return m_audioFile; }
+    const AudioFile* getAudioFile() const { return m_audioFile; }
+
+    // Scan to a peak and scan forward a number of peaks
+    //
+    bool scanToPeak(int peak);
+    bool scanForward(int numberOfPeaks);
+
+    // Find threshold crossing points
+    //
+    std::vector<SplitPointPair> getSplitPoints(const RealTime &startTime,
+                                               const RealTime &endTime,
+                                               int threshold,
+                                               const RealTime &minLength);
+    // Accessors
+    //
+    int getVersion() const { return m_version; }
+    int getFormat() const { return m_format; }
+    int getPointsPerValue() const { return m_pointsPerValue; }
+    int getBlockSize() const { return m_blockSize; }
+    int getChannels() const { return m_channels; }
+    int getNumberOfPeaks() const { return m_numberOfPeaks; }
+    int getPositionPeakOfPeaks() const { return m_positionPeakOfPeaks; }
+    int getOffsetToPeaks() const { return m_offsetToPeaks; }
+    int getBodyBytes() const { return m_bodyBytes; }
+    QDateTime getModificationTime() const { return m_modificationTime; }
+    std::streampos getChunkStartPosition() const
+        { return m_chunkStartPosition; }
+
+    bool isProcessingPeaks() const { return m_keepProcessing; }
+    void setProcessingPeaks(bool value) { m_keepProcessing = value; }
+
+signals:
+    void setProgress(int);
+    
+protected:
+    // Write the peak header and the peaks themselves
+    //
+    void writeHeader(std::ofstream *file);
+    void writePeaks(unsigned short updatePercentage,
+                    std::ofstream *file);
+
+    // Get the position of a peak for a given time
+    //
+    int getPeak(const RealTime &time);
+
+    // And the time of a peak
+    //
+    RealTime getTime(int peak);
+
+    // Parse the header
+    //
+    void parseHeader();
+
+    AudioFile *m_audioFile;
+
+    // Some Peak Envelope Chunk parameters
+    //
+    int m_version;
+    int m_format;  // bytes in peak value (1 or 2)
+    int m_pointsPerValue;
+    int m_blockSize;
+    int m_channels;
+    int m_numberOfPeaks;
+    int m_positionPeakOfPeaks;
+    int m_offsetToPeaks;
+    int m_bodyBytes;
+
+    // Peak timestamp
+    //
+    QDateTime m_modificationTime;
+
+    std::streampos m_chunkStartPosition;
+
+    // For cacheing of peak information in memory we use the last query 
+    // parameters as our key to the cached data.
+    //
+    RealTime           m_lastPreviewStartTime;
+    RealTime           m_lastPreviewEndTime;
+    int                m_lastPreviewWidth;
+    bool               m_lastPreviewShowMinima;
+    std::vector<float> m_lastPreviewCache;
+
+    // Do we actually want to keep processing this peakfile?
+    // In case we get a cancel.
+    //
+    bool               m_keepProcessing;
+
+    std::string        m_peakCache;
+    
+};
+
+}
+
+
+#endif // _PEAKFILE_H_
+
+
diff --git a/src/sound/PeakFileManager.cpp b/src/sound/PeakFileManager.cpp
new file mode 100644
index 0000000..10293e6
--- /dev/null
+++ b/src/sound/PeakFileManager.cpp
@@ -0,0 +1,327 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+
+// Accepts a file handle positioned somewhere in sample data (could
+// be at the start) along with the necessary meta information for
+// decoding (channels, bits per sample) and turns the sample data
+// into peak data and generates a BWF format peak chunk file.  This
+// file can exist by itself (in the case this is being generated
+// by a WAV) or be accomodated inside a BWF format file.
+//
+//
+
+#include <string>
+#include <vector>
+
+#include <qobject.h>
+
+#include "PeakFileManager.h"
+#include "AudioFile.h"
+#include "RealTime.h"
+#include "PeakFile.h"
+
+namespace Rosegarden
+{
+
+
+PeakFileManager::PeakFileManager():
+        m_updatePercentage(0),
+        m_currentPeakFile(0)
+{}
+
+PeakFileManager::~PeakFileManager()
+{}
+
+// Inserts PeakFile based on AudioFile if it doesn't already exist
+bool
+PeakFileManager::insertAudioFile(AudioFile *audioFile)
+{
+    std::vector<PeakFile*>::iterator it;
+
+    for (it = m_peakFiles.begin(); it != m_peakFiles.end(); it++) {
+        if ((*it)->getAudioFile()->getId() == audioFile->getId())
+            return false;
+    }
+
+    /*
+    std::cout << "PeakFileManager::insertAudioFile - creating peak file "
+              << m_peakFiles.size() + 1
+              << " for \"" << audioFile->getFilename()
+              << "\"" << std::endl;
+    */
+
+    // Insert
+    m_peakFiles.push_back(new PeakFile(audioFile));
+
+    return true;
+}
+
+// Removes peak file from PeakFileManager - doesn't affect audioFile
+//
+bool
+PeakFileManager::removeAudioFile(AudioFile *audioFile)
+{
+    std::vector<PeakFile*>::iterator it;
+
+    for (it = m_peakFiles.begin(); it != m_peakFiles.end(); it++) {
+        if ((*it)->getAudioFile()->getId() == audioFile->getId()) {
+            if (m_currentPeakFile == *it)
+                m_currentPeakFile = 0;
+            delete *it;
+            m_peakFiles.erase(it);
+            return true;
+        }
+    }
+
+    return false;
+}
+
+// Auto-insert PeakFile into manager if it doesn't already exist
+//
+PeakFile*
+PeakFileManager::getPeakFile(AudioFile *audioFile)
+{
+    std::vector<PeakFile*>::iterator it;
+    PeakFile *ptr = 0;
+
+    while (ptr == 0) {
+        for (it = m_peakFiles.begin(); it != m_peakFiles.end(); it++)
+            if ((*it)->getAudioFile()->getId() == audioFile->getId())
+                ptr = *it;
+
+        // If nothing is found then insert and retry
+        //
+        if (ptr == 0) {
+            // Insert - if we fail we return as empty
+            //
+            if (insertAudioFile(audioFile) == false)
+                return 0;
+        }
+    }
+
+    return ptr;
+}
+
+
+// Does a given AudioFile have a valid peak file or peak chunk?
+//
+bool
+PeakFileManager::hasValidPeaks(AudioFile *audioFile)
+{
+    if (audioFile->getType() == WAV) {
+        // Check external peak file
+        PeakFile *peakFile = getPeakFile(audioFile);
+
+        if (peakFile == 0) {
+#ifdef DEBUG_PEAKFILEMANAGER
+            std::cerr << "PeakFileManager::hasValidPeaks - no peak file found"
+            << std::endl;
+#endif
+
+            return false;
+        }
+        // If it doesn't open and parse correctly
+        if (peakFile->open() == false)
+            return false;
+
+        // or if the data is old or invalid
+        if (peakFile->isValid() == false)
+            return false;
+
+    } else if (audioFile->getType() == BWF) {
+        // check internal peak chunk
+    } else {
+#ifdef DEBUG_PEAKFILEMANAGER
+        std::cout << "PeakFileManager::hasValidPeaks - unsupported file type"
+        << std::endl;
+#endif
+
+        return false;
+    }
+
+    return true;
+
+}
+
+// Generate the peak file.  Checks to see if peak file exists
+// already and if so if it's up to date.  If it isn't then we
+// regenerate.
+//
+void
+PeakFileManager::generatePeaks(AudioFile *audioFile,
+                               unsigned short updatePercentage)
+{
+#ifdef DEBUG_PEAKFILEMANAGER
+    std::cout << "PeakFileManager::generatePeaks - generating peaks for \""
+    << audioFile->getFilename() << "\"" << std::endl;
+#endif
+
+    if (audioFile->getType() == WAV) {
+        m_currentPeakFile = getPeakFile(audioFile);
+
+        QObject::connect(m_currentPeakFile, SIGNAL(setProgress(int)),
+                         this, SIGNAL(setProgress(int)));
+
+        // Just write out a peak file
+        //
+        if (m_currentPeakFile->write(updatePercentage) == false) {
+            std::cerr << "Can't write peak file for " << audioFile->getFilename() << " - no preview generated" << std::endl;
+            throw BadPeakFileException
+            (audioFile->getFilename(), __FILE__, __LINE__);
+        }
+
+        // The m_currentPeakFile might have been cancelled (see stopPreview())
+        //
+        if (m_currentPeakFile) {
+            // close writes out important things
+            m_currentPeakFile->close();
+            m_currentPeakFile->disconnect();
+        }
+    } else if (audioFile->getType() == BWF) {
+        // write the file out and incorporate the peak chunk
+    } else {
+#ifdef DEBUG_PEAKFILEMANAGER
+        std::cerr << "PeakFileManager::generatePeaks - unsupported file type"
+        << std::endl;
+#endif
+
+        return ;
+    }
+
+    m_currentPeakFile = 0;
+
+}
+
+std::vector<float>
+PeakFileManager::getPreview(AudioFile *audioFile,
+                            const RealTime &startTime,
+                            const RealTime &endTime,
+                            int width,
+                            bool showMinima)
+{
+    std::vector<float> rV;
+
+    // If we've got no channels then the audio file hasn't
+    // completed (recording) - so don't generate a preview
+    //
+    if (audioFile->getChannels() == 0)
+        return rV;
+
+    if (audioFile->getType() == WAV) {
+        PeakFile *peakFile = getPeakFile(audioFile);
+
+        // just write out a peak file
+        try {
+            peakFile->open();
+            rV = peakFile->getPreview(startTime,
+                                      endTime,
+                                      width,
+                                      showMinima);
+        } catch (SoundFile::BadSoundFileException e) {
+#ifdef DEBUG_PEAKFILEMANAGER
+            std::cout << "PeakFileManager::getPreview "
+            << "\"" << e << "\"" << std::endl;
+#else
+
+            ;
+#endif
+
+            throw BadPeakFileException(e);
+        }
+    } else if (audioFile->getType() == BWF) {
+        // write the file out and incorporate the peak chunk
+    }
+#ifdef DEBUG_PEAKFILEMANAGER
+    else {
+        std::cerr << "PeakFileManager::getPreview - unsupported file type"
+        << std::endl;
+    }
+#endif
+
+    return rV;
+}
+
+void
+PeakFileManager::clear()
+{
+    std::vector<PeakFile*>::iterator it;
+
+    for (it = m_peakFiles.begin(); it != m_peakFiles.end(); it++)
+        delete (*it);
+
+    m_peakFiles.erase(m_peakFiles.begin(), m_peakFiles.end());
+
+    m_currentPeakFile = 0;
+}
+
+
+std::vector<SplitPointPair>
+PeakFileManager::getSplitPoints(AudioFile *audioFile,
+                                const RealTime &startTime,
+                                const RealTime &endTime,
+                                int threshold,
+                                const RealTime &minTime)
+{
+    PeakFile *peakFile = getPeakFile(audioFile);
+
+    if (peakFile == 0)
+        return std::vector<SplitPointPair>();
+
+    return peakFile->getSplitPoints(startTime,
+                                    endTime,
+                                    threshold,
+                                    minTime);
+
+}
+
+void
+PeakFileManager::stopPreview()
+{
+    if (m_currentPeakFile) {
+        // Stop processing
+        //
+        QString fileName = QString(m_currentPeakFile->getFilename().data());
+        m_currentPeakFile->setProcessingPeaks(false);
+        m_currentPeakFile->disconnect();
+
+        QFile file(fileName);
+        bool removed = file.remove();
+
+#ifdef DEBUG_PEAKFILEMANAGER
+
+        if (removed) {
+            std::cout << "PeakFileManager::stopPreview() - removed preview"
+            << std::endl;
+        }
+#endif
+        //delete m_currentPeakFile;
+        m_currentPeakFile = 0;
+    }
+}
+
+
+
+
+}
+
+
+#include "PeakFileManager.moc"
diff --git a/src/sound/PeakFileManager.h b/src/sound/PeakFileManager.h
new file mode 100644
index 0000000..07ff704
--- /dev/null
+++ b/src/sound/PeakFileManager.h
@@ -0,0 +1,162 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+
+// Accepts an AudioFIle and turns the sample data into peak data for
+// storage in a peak file or a BWF format peak chunk.  Pixmaps or
+// sample data is returned to callers on demand using these cached
+// values.
+//
+//
+
+#ifndef _PEAKFILEMANAGER_H_
+#define _PEAKFILEMANAGER_H_
+
+#include <string>
+#include <iostream>
+#include <fstream>
+#include <vector>
+
+#include <qobject.h>
+
+
+#include "PeakFile.h"
+
+namespace Rosegarden
+{
+
+class AudioFile;
+class RealTime;
+
+class PeakFileManager : public QObject
+{
+    Q_OBJECT
+public:
+    // updatePercentage tells this object how often to throw a
+    // percentage complete message - active between 0-100 only
+    // if it's set to 5 then we send an update exception every
+    // five percent.  The percentage complete is sent with 
+    // each exception.
+    //
+    PeakFileManager();
+    virtual ~PeakFileManager();
+    
+    class BadPeakFileException : public Exception
+    {
+    public:
+        BadPeakFileException(std::string path) :
+            Exception("Bad peak file " + path), m_path(path) { }
+        BadPeakFileException(std::string path, std::string file, int line) :
+            Exception("Bad peak file " + path, file, line), m_path(path) { }
+        BadPeakFileException(const SoundFile::BadSoundFileException &e) :
+            Exception("Bad peak file (malformed audio?) " + e.getPath()), m_path(e.getPath()) { }
+
+        ~BadPeakFileException() throw() { }
+
+        std::string getPath() const { return m_path; }
+
+    private:
+        std::string m_path;
+    };
+
+private:
+    PeakFileManager(const PeakFileManager &pFM);
+    PeakFileManager& operator=(const PeakFileManager &);
+
+public:
+    // Check that a given audio file has a valid and up to date
+    // peak file or peak chunk.
+    //
+    bool hasValidPeaks(AudioFile *audioFile);
+    // throw BadSoundFileException, BadPeakFileException
+
+    // Generate a peak file from file details - if the peak file already
+    // exists _and_ it's up to date then we don't do anything.  For BWF
+    // files we generate an internal peak chunk.
+    //
+    //
+    void generatePeaks(AudioFile *audioFile,
+                       unsigned short updatePercentage);
+    // throw BadSoundFileException, BadPeakFileException
+
+    // Get a vector of floats as the preview
+    //
+    std::vector<float> getPreview(AudioFile *audioFile,
+                                  const RealTime &startTime,
+                                  const RealTime &endTime,
+                                  int   width,
+                                  bool  showMinima);
+    // throw BadSoundFileException, BadPeakFileException
+    
+    // Remove cache for a single audio file (if audio file to be deleted etc)
+    // 
+    bool removeAudioFile(AudioFile *audioFile);
+
+    // Clear down
+    //
+    void clear();
+                    
+    // Get split points for a peak file
+    //
+    std::vector<SplitPointPair> 
+        getSplitPoints(AudioFile *audioFile,
+                       const RealTime &startTime,
+                       const RealTime &endTime,
+                       int threshold,
+                       const RealTime &minTime);
+
+    std::vector<PeakFile*>::const_iterator begin() const
+                { return m_peakFiles.begin(); }
+
+    std::vector<PeakFile*>::const_iterator end() const
+                { return m_peakFiles.end(); }
+
+    // Stop a preview during its build
+    //
+    void stopPreview();
+
+signals:
+    void setProgress(int);
+
+protected:
+
+    // Add and remove from our PeakFile cache
+    //
+    bool insertAudioFile(AudioFile *audioFile);
+    PeakFile* getPeakFile(AudioFile *audioFile);
+
+    std::vector<PeakFile*> m_peakFiles;
+    unsigned short m_updatePercentage;  // how often we send updates 
+
+    // Whilst processing - the current PeakFile
+    //
+    PeakFile              *m_currentPeakFile;
+
+
+};
+
+
+}
+
+
+#endif // _PEAKFILEMANAGER_H_
+
+
diff --git a/src/sound/PlayableAudioFile.cpp b/src/sound/PlayableAudioFile.cpp
new file mode 100644
index 0000000..b5ddcf7
--- /dev/null
+++ b/src/sound/PlayableAudioFile.cpp
@@ -0,0 +1,1086 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "PlayableAudioFile.h"
+#include <cassert>
+
+namespace Rosegarden
+{
+
+//#define DEBUG_RING_BUFFER_POOL 1
+//#define DEBUG_PLAYABLE 1
+//#define DEBUG_PLAYABLE_READ 1
+
+class RingBufferPool
+{
+public:
+    typedef float sample_t;
+
+    RingBufferPool(size_t bufferSize);
+    virtual ~RingBufferPool();
+
+    /**
+     * Set the default size for buffers.  Buffers currently allocated
+     * will not be resized until they are returned.
+     */
+    void setBufferSize(size_t n);
+
+    size_t getBufferSize() const
+    {
+        return m_bufferSize;
+    }
+
+    /**
+     * Discard or create buffers as necessary so as to have n buffers
+     * in the pool.  This will not discard any buffers that are
+     * currently allocated, so if more than n are allocated, more than
+     * n will remain.
+     */
+    void setPoolSize(size_t n);
+
+    size_t getPoolSize() const
+    {
+        return m_buffers.size();
+    }
+
+    /**
+     * Return true if n buffers available, false otherwise.
+     */
+    bool getBuffers(size_t n, RingBuffer<sample_t> **buffers);
+
+    /**
+     * Return a buffer to the pool.
+     */
+    void returnBuffer(RingBuffer<sample_t> *buffer);
+
+protected:
+    // Want to avoid memory allocation if possible when marking a buffer
+    // unallocated or allocated, so we use a single container for all
+
+    typedef std::pair<RingBuffer<sample_t> *, bool> AllocPair;
+    typedef std::vector<AllocPair> AllocList;
+    AllocList m_buffers;
+
+    size_t m_bufferSize;
+    size_t m_available;
+
+    pthread_mutex_t m_lock;
+};
+
+
+RingBufferPool::RingBufferPool(size_t bufferSize) :
+    m_bufferSize(bufferSize),
+    m_available(0)
+{
+    pthread_mutex_t initialisingMutex = PTHREAD_MUTEX_INITIALIZER;
+    memcpy(&m_lock, &initialisingMutex, sizeof(pthread_mutex_t));
+}
+
+RingBufferPool::~RingBufferPool()
+{
+    size_t allocatedCount = 0;
+    for (AllocList::iterator i = m_buffers.begin(); i != m_buffers.end(); ++i) {
+        if (i->second)
+            ++allocatedCount;
+    }
+
+    if (allocatedCount > 0) {
+        std::cerr << "WARNING: RingBufferPool::~RingBufferPool: deleting pool with " << allocatedCount << " allocated buffers" << std::endl;
+    }
+
+    for (AllocList::iterator i = m_buffers.begin(); i != m_buffers.end(); ++i) {
+        delete i->first;
+    }
+
+    m_buffers.clear();
+
+    pthread_mutex_destroy(&m_lock);
+}
+
+void
+RingBufferPool::setBufferSize(size_t n)
+{
+    if (m_bufferSize == n)
+        return ;
+
+    pthread_mutex_lock(&m_lock);
+
+#ifdef DEBUG_RING_BUFFER_POOL
+
+    std::cerr << "RingBufferPool::setBufferSize: from " << m_bufferSize
+	      << " to " << n << std::endl;
+    int c = 0;
+#endif
+
+    for (AllocList::iterator i = m_buffers.begin(); i != m_buffers.end(); ++i) {
+        if (!i->second) {
+            delete i->first;
+            i->first = new RingBuffer<sample_t>(n);
+#ifdef DEBUG_RING_BUFFER_POOL
+
+            std::cerr << "Resized buffer " << c++ << std::endl;
+#endif
+
+        } else {
+#ifdef DEBUG_RING_BUFFER_POOL
+            std::cerr << "Buffer " << c++ << " is already in use, resizing in place" << std::endl;
+#endif
+
+            i->first->resize(n);
+        }
+    }
+
+    m_bufferSize = n;
+    pthread_mutex_unlock(&m_lock);
+}
+
+void
+RingBufferPool::setPoolSize(size_t n)
+{
+    pthread_mutex_lock(&m_lock);
+
+#ifdef DEBUG_RING_BUFFER_POOL
+
+    std::cerr << "RingBufferPool::setPoolSize: from " << m_buffers.size()
+	      << " to " << n << std::endl;
+#endif
+
+    size_t allocatedCount = 0, count = 0;
+
+    for (AllocList::iterator i = m_buffers.begin(); i != m_buffers.end(); ++i) {
+        if (i->second)
+            ++allocatedCount;
+        ++count;
+    }
+
+    if (count > n) {
+        for (AllocList::iterator i = m_buffers.begin(); i != m_buffers.end(); ) {
+            if (!i->second) {
+                delete i->first;
+                m_buffers.erase(i);
+                if (--count == n)
+                    break;
+            } else {
+                ++i;
+            }
+        }
+    }
+
+    while (count < n) {
+        m_buffers.push_back(AllocPair(new RingBuffer<sample_t>(m_bufferSize),
+                                      false));
+        ++count;
+    }
+
+    m_available = std::max(allocatedCount, n) - allocatedCount;
+
+#ifdef DEBUG_RING_BUFFER_POOL
+
+    std::cerr << "RingBufferPool::setPoolSize: have " << m_buffers.size()
+	      << " buffers (" << allocatedCount << " allocated, " << m_available << " available)" << std::endl;
+#endif
+
+    pthread_mutex_unlock(&m_lock);
+}
+
+bool
+RingBufferPool::getBuffers(size_t n, RingBuffer<sample_t> **buffers)
+{
+    pthread_mutex_lock(&m_lock);
+
+    size_t count = 0;
+
+    for (AllocList::iterator i = m_buffers.begin(); i != m_buffers.end(); ++i) {
+        if (!i->second && ++count == n)
+            break;
+    }
+
+    if (count < n) {
+#ifdef DEBUG_RING_BUFFER_POOL
+        std::cerr << "RingBufferPool::getBuffers(" << n << "): not available (in pool of " << m_buffers.size() << "), resizing" << std::endl;
+#endif
+
+        AllocList newBuffers;
+
+        while (count < n) {
+            for (size_t i = 0; i < m_buffers.size(); ++i) {
+                newBuffers.push_back(m_buffers[i]);
+            }
+            for (size_t i = 0; i < m_buffers.size(); ++i) {
+                newBuffers.push_back(AllocPair(new RingBuffer<sample_t>(m_bufferSize),
+                                               false));
+            }
+            count += m_buffers.size();
+            m_available += m_buffers.size();
+        }
+
+        m_buffers = newBuffers;
+    }
+
+    count = 0;
+
+#ifdef DEBUG_RING_BUFFER_POOL
+
+    std::cerr << "RingBufferPool::getBuffers(" << n << "): available" << std::endl;
+#endif
+
+    for (AllocList::iterator i = m_buffers.begin(); i != m_buffers.end(); ++i) {
+        if (!i->second) {
+            i->second = true;
+            i->first->reset();
+            i->first->mlock();
+            buffers[count] = i->first;
+            --m_available;
+            if (++count == n)
+                break;
+        }
+    }
+
+#ifdef DEBUG_RING_BUFFER_POOL
+    std::cerr << "RingBufferPool::getBuffers: " << m_available << " remain in pool of " << m_buffers.size() << std::endl;
+#endif
+
+    pthread_mutex_unlock(&m_lock);
+    return true;
+}
+
+void
+RingBufferPool::returnBuffer(RingBuffer<sample_t> *buffer)
+{
+    pthread_mutex_lock(&m_lock);
+
+#ifdef DEBUG_RING_BUFFER_POOL
+
+    std::cerr << "RingBufferPool::returnBuffer" << std::endl;
+#endif
+
+    buffer->munlock();
+
+    for (AllocList::iterator i = m_buffers.begin(); i != m_buffers.end(); ++i) {
+	if (i->first == buffer) {
+	    i->second = false;
+	    ++m_available;
+	    if (buffer->getSize() != m_bufferSize) {
+		delete buffer;
+		i->first = new RingBuffer<sample_t>(m_bufferSize);
+	    }
+	}
+    }
+
+#ifdef DEBUG_RING_BUFFER_POOL
+    std::cerr << "RingBufferPool::returnBuffer: " << m_available << " remain in pool of " << m_buffers.size() << std::endl;
+#endif
+
+    pthread_mutex_unlock(&m_lock);
+}
+
+
+AudioCache PlayableAudioFile::m_smallFileCache;
+
+std::vector<PlayableAudioFile::sample_t *> PlayableAudioFile::m_workBuffers;
+size_t PlayableAudioFile::m_workBufferSize = 0;
+
+char *PlayableAudioFile::m_rawFileBuffer;
+size_t PlayableAudioFile::m_rawFileBufferSize = 0;
+
+RingBufferPool *PlayableAudioFile::m_ringBufferPool = 0;
+
+size_t PlayableAudioFile::m_xfadeFrames = 30;
+
+PlayableAudioFile::PlayableAudioFile(InstrumentId instrumentId,
+                                     AudioFile *audioFile,
+                                     const RealTime &startTime,
+                                     const RealTime &startIndex,
+                                     const RealTime &duration,
+                                     size_t bufferSize,
+                                     size_t smallFileSize,
+                                     int targetChannels,
+                                     int targetSampleRate) :
+    m_startTime(startTime),
+    m_startIndex(startIndex),
+    m_duration(duration),
+    m_file(0),
+    m_audioFile(audioFile),
+    m_instrumentId(instrumentId),
+    m_targetChannels(targetChannels),
+    m_targetSampleRate(targetSampleRate),
+    m_fileEnded(false),
+    m_firstRead(true),
+    m_runtimeSegmentId( -1),
+    m_isSmallFile(false),
+    m_currentScanPoint(RealTime::zeroTime),
+    m_smallFileScanFrame(0),
+    m_autoFade(false),
+    m_fadeInTime(RealTime::zeroTime),
+    m_fadeOutTime(RealTime::zeroTime)
+{
+#ifdef DEBUG_PLAYABLE
+    std::cerr << "PlayableAudioFile::PlayableAudioFile - creating " << this << " for instrument " << instrumentId << " with file " << (m_audioFile ? m_audioFile->getShortFilename() : "(none)") << std::endl;
+#endif
+
+    if (!m_ringBufferPool) {
+        //!!! Problematic -- how do we deal with different playable audio
+        // files requiring different buffer sizes?  That shouldn't be the
+        // usual case, but it's not unthinkable.
+        m_ringBufferPool = new RingBufferPool(bufferSize);
+    } else {
+        m_ringBufferPool->setBufferSize
+	    (std::max(bufferSize, m_ringBufferPool->getBufferSize()));
+    }
+
+    initialise(bufferSize, smallFileSize);
+}
+
+
+void
+PlayableAudioFile::setRingBufferPoolSizes(size_t n, size_t nframes)
+{
+    if (!m_ringBufferPool) {
+        m_ringBufferPool = new RingBufferPool(nframes);
+    } else {
+        m_ringBufferPool->setBufferSize
+	    (std::max(nframes, m_ringBufferPool->getBufferSize()));
+    }
+    m_ringBufferPool->setPoolSize(n);
+}
+
+
+void
+PlayableAudioFile::initialise(size_t bufferSize, size_t smallFileSize)
+{
+#ifdef DEBUG_PLAYABLE
+    std::cerr << "PlayableAudioFile::initialise() " << this << std::endl;
+#endif
+
+    checkSmallFileCache(smallFileSize);
+
+    if (!m_isSmallFile) {
+
+        m_file = new std::ifstream(m_audioFile->getFilename().c_str(),
+                                   std::ios::in | std::ios::binary);
+
+        if (!*m_file) {
+            std::cerr << "ERROR: PlayableAudioFile::initialise: Failed to open audio file " << m_audioFile->getFilename() << std::endl;
+            delete m_file;
+            m_file = 0;
+        }
+    }
+
+    // Scan to the beginning of the data chunk we need
+    //
+#ifdef DEBUG_PLAYABLE
+    std::cerr << "PlayableAudioFile::initialise - scanning to " << m_startIndex << std::endl;
+#endif
+
+    if (m_file) {
+        scanTo(m_startIndex);
+    } else {
+        m_fileEnded = false;
+        m_currentScanPoint = m_startIndex;
+	m_smallFileScanFrame = RealTime::realTime2Frame
+	    (m_currentScanPoint, m_audioFile->getSampleRate());
+    }
+
+#ifdef DEBUG_PLAYABLE
+    std::cerr << "PlayableAudioFile::initialise: buffer size is " << bufferSize << " frames, file size is " << m_audioFile->getSize() << std::endl;
+#endif
+
+    if (m_targetChannels <= 0)
+        m_targetChannels = m_audioFile->getChannels();
+    if (m_targetSampleRate <= 0)
+        m_targetSampleRate = m_audioFile->getSampleRate();
+
+    m_ringBuffers = new RingBuffer<sample_t> *[m_targetChannels];
+    for (int ch = 0; ch < m_targetChannels; ++ch) {
+        m_ringBuffers[ch] = 0;
+    }
+}
+
+PlayableAudioFile::~PlayableAudioFile()
+{
+    if (m_file) {
+        m_file->close();
+        delete m_file;
+    }
+
+    returnRingBuffers();
+    delete[] m_ringBuffers;
+    m_ringBuffers = 0;
+
+    if (m_isSmallFile) {
+        m_smallFileCache.decrementReference(m_audioFile);
+    }
+
+#ifdef DEBUG_PLAYABLE 
+    //    std::cerr << "PlayableAudioFile::~PlayableAudioFile - destroying - " << this << std::endl;
+#endif
+}
+
+void
+PlayableAudioFile::returnRingBuffers()
+{
+    for (int i = 0; i < m_targetChannels; ++i) {
+	if (m_ringBuffers[i]) {
+	    m_ringBufferPool->returnBuffer(m_ringBuffers[i]);
+	    m_ringBuffers[i] = 0;
+	}
+    }
+}
+
+bool
+PlayableAudioFile::scanTo(const RealTime &time)
+{
+#ifdef DEBUG_PLAYABLE_READ
+    std::cerr << "PlayableAudioFile::scanTo(" << time << ")" << std::endl;
+#endif
+
+    m_fileEnded = false; // until we know otherwise -- this flag is an
+    // optimisation, not a reliable record
+
+    bool ok = false;
+
+    if (m_isSmallFile) {
+
+	m_currentScanPoint = time;
+	m_smallFileScanFrame = RealTime::realTime2Frame
+	    (time, m_audioFile->getSampleRate());
+#ifdef DEBUG_PLAYABLE_READ
+	std::cerr << "... maps to frame " << m_smallFileScanFrame << std::endl;
+#endif
+	ok = true;
+
+    } else {
+
+	ok = m_audioFile->scanTo(m_file, time);
+	if (ok) {
+	    m_currentScanPoint = time;
+	}
+    }
+
+#ifdef DEBUG_PLAYABLE_READ
+    std::cerr << "PlayableAudioFile::scanTo(" << time << "): set m_currentScanPoint to " << m_currentScanPoint << std::endl;
+#endif
+
+    m_firstRead = true; // so we know to xfade in
+
+    return ok;
+}
+
+
+size_t
+PlayableAudioFile::getSampleFramesAvailable()
+{
+    size_t actual = 0;
+
+    if (m_isSmallFile) {
+	size_t cchannels;
+	size_t cframes;
+	(void)m_smallFileCache.getData(m_audioFile, cchannels, cframes);
+	if (cframes > m_smallFileScanFrame)
+	    return cframes - m_smallFileScanFrame;
+	else
+	    return 0;
+    }
+
+    for (int ch = 0; ch < m_targetChannels; ++ch) {
+	if (!m_ringBuffers[ch])
+	    return 0;
+	size_t thisChannel = m_ringBuffers[ch]->getReadSpace();
+	if (ch == 0 || thisChannel < actual)
+	    actual = thisChannel;
+    }
+
+#ifdef DEBUG_PLAYABLE
+    std::cerr << "PlayableAudioFile(" << (m_audioFile ? m_audioFile->getShortFilename() : "(none)") << " " << this << ")::getSampleFramesAvailable: have " << actual << std::endl;
+#endif
+
+    return actual;
+}
+
+size_t
+PlayableAudioFile::addSamples(std::vector<sample_t *> &destination,
+			      size_t channels, size_t nframes, size_t offset)
+{
+#ifdef DEBUG_PLAYABLE_READ
+    std::cerr << "PlayableAudioFile::addSamples(" << nframes << "): channels " << channels << ", my target channels " << m_targetChannels << std::endl;
+#endif
+
+    if (!m_isSmallFile) {
+
+	size_t qty = 0;
+	bool done = m_fileEnded;
+
+	for (int ch = 0; ch < int(channels) && ch < m_targetChannels; ++ch) {
+	    if (!m_ringBuffers[ch])
+		return 0; //!!! fatal
+	    size_t here = m_ringBuffers[ch]->readAdding(destination[ch] + offset, nframes);
+	    if (ch == 0 || here < qty)
+		qty = here;
+	    if (done && (m_ringBuffers[ch]->getReadSpace() > 0))
+		done = false;
+	}
+
+	for (int ch = channels; ch < m_targetChannels; ++ch) {
+	    m_ringBuffers[ch]->skip(nframes);
+	}
+
+	if (done) {
+#ifdef DEBUG_PLAYABLE_READ
+	    std::cerr << "PlayableAudioFile::addSamples(" << nframes << "): reached end, returning buffers" << std::endl;
+#endif
+
+	    returnRingBuffers();
+	}
+
+#ifdef DEBUG_PLAYABLE_READ
+	std::cerr << "PlayableAudioFile::addSamples(" << nframes << "): returning " << qty << " frames (at least " << (m_ringBuffers[0] ? m_ringBuffers[0]->getReadSpace() : 0) << " remaining)" << std::endl;
+#endif
+
+	return qty;
+
+    } else {
+
+	size_t cchannels;
+	size_t cframes;
+	float **cached = m_smallFileCache.getData(m_audioFile, cchannels, cframes);
+
+	if (!cached) {
+	    std::cerr << "WARNING: PlayableAudioFile::addSamples: Failed to find small file in cache" << std::endl;
+	    m_isSmallFile = false;
+	} else {
+
+	    size_t scanFrame = m_smallFileScanFrame;
+
+	    if (scanFrame >= cframes) {
+		m_fileEnded = true;
+		return 0;
+	    }
+
+	    size_t endFrame = scanFrame + nframes;
+	    size_t n = nframes;
+
+	    if (endFrame >= cframes) {
+		m_fileEnded = true;
+		n = cframes - scanFrame;
+	    }
+
+#ifdef DEBUG_PLAYABLE_READ
+	    std::cerr << "PlayableAudioFile::addSamples: it's a small file: want frames " << scanFrame << " to " << endFrame << " of " << cframes << std::endl;
+#endif
+
+	    size_t xfadeIn = (m_firstRead ? m_xfadeFrames : 0);
+	    size_t xfadeOut = (m_fileEnded ? m_xfadeFrames : 0);
+
+	    // all this could be neater!
+
+	    if (channels == 1 && cchannels == 2) { // mix
+		for (size_t i = 0; i < n; ++i) {
+		    sample_t v =
+			cached[0][scanFrame + i] +
+			cached[1][scanFrame + i];
+		    //		    if ((i + 1) < xfadeIn)
+		    //	v = (v * (i + 1)) / xfadeIn;
+		    //if ((n - i) < xfadeOut)
+		    //	v = (v * (n - i)) / xfadeOut;
+		    destination[0][i + offset] += v;
+		}
+	    } else {
+		for (size_t ch = 0; ch < channels; ++ch) {
+		    int sch = ch;
+		    if (ch >= cchannels) {
+			if (channels == 2 && cchannels == 1)
+			    sch = 0;
+			else
+			    break;
+		    } else {
+			for (size_t i = 0; i < n; ++i) {
+			    sample_t v = cached[sch][scanFrame + i];
+			    //	    if ((i + 1) < xfadeIn)
+			    //	v = (v * (i + 1)) / xfadeIn;
+			    //if ((n - i) < xfadeOut)
+			    //	v = (v * (n - i)) / xfadeOut;
+			    destination[ch][i + offset] += v;
+			}
+		    }
+		}
+	    }
+
+	    m_smallFileScanFrame += nframes;
+	    m_currentScanPoint = m_currentScanPoint +
+		RealTime::frame2RealTime(nframes, m_targetSampleRate);
+	    return nframes;
+	}
+    }
+
+    return 0;
+}
+
+void
+PlayableAudioFile::checkSmallFileCache(size_t smallFileSize)
+{
+    if (m_smallFileCache.has(m_audioFile)) {
+
+#ifdef DEBUG_PLAYABLE
+	std::cerr << "PlayableAudioFile::checkSmallFileCache: Found file in small file cache" << std::endl;
+#endif
+
+	m_smallFileCache.incrementReference(m_audioFile);
+	m_isSmallFile = true;
+
+    } else if (m_audioFile->getSize() <= smallFileSize) {
+
+	std::ifstream file(m_audioFile->getFilename().c_str(),
+			   std::ios::in | std::ios::binary);
+
+	if (!file) {
+	    std::cerr << "ERROR: PlayableAudioFile::checkSmallFileCache: Failed to open audio file " << m_audioFile->getFilename() << std::endl;
+	    return ;
+	}
+
+#ifdef DEBUG_PLAYABLE
+	std::cerr << "PlayableAudioFile::checkSmallFileCache: Adding file to small file cache" << std::endl;
+#endif
+
+	// We always encache files with their original number of
+	// channels (because they might be called for in any channel
+	// configuration subsequently) but with the current sample
+	// rate, not their original one.
+
+	m_audioFile->scanTo(&file, RealTime::zeroTime);
+
+	size_t reqd = m_audioFile->getSize() / m_audioFile->getBytesPerFrame();
+	unsigned char *buffer = new unsigned char[m_audioFile->getSize()];
+	size_t obtained = m_audioFile->getSampleFrames(&file, (char *)buffer, reqd);
+
+//	std::cerr <<"obtained=" << obtained << std::endl;
+
+	size_t nch = getSourceChannels();
+	size_t nframes = obtained;
+	if (int(getSourceSampleRate()) != m_targetSampleRate) {
+#ifdef DEBUG_PLAYABLE
+	    std::cerr << "PlayableAudioFile::checkSmallFileCache: Resampling badly from " << getSourceSampleRate() << " to " << m_targetSampleRate << std::endl;
+#endif
+	    nframes = size_t(float(nframes) * float(m_targetSampleRate) /
+			     float(getSourceSampleRate()));
+	}
+
+	std::vector<sample_t *> samples;
+	for (size_t ch = 0; ch < nch; ++ch) {
+	    samples.push_back(new sample_t[nframes]);
+	}
+
+	if (!m_audioFile->decode(buffer,
+				 obtained * m_audioFile->getBytesPerFrame(),
+				 m_targetSampleRate,
+				 nch,
+				 nframes,
+				 samples)) {
+	    std::cerr << "PlayableAudioFile::checkSmallFileCache: failed to decode file" << std::endl;
+	} else {
+	    sample_t **toCache = new sample_t * [nch];
+	    for (size_t ch = 0; ch < nch; ++ch) {
+		toCache[ch] = samples[ch];
+	    }
+	    m_smallFileCache.addData(m_audioFile, nch, nframes, toCache);
+	    m_isSmallFile = true;
+	}
+
+	delete[] buffer;
+
+	file.close();
+    }
+
+    if (m_isSmallFile) {
+	if (m_file) {
+	    m_file->close();
+	    delete m_file;
+	    m_file = 0;
+	}
+    }
+}
+
+
+void
+PlayableAudioFile::fillBuffers()
+{
+#ifdef DEBUG_PLAYABLE
+    if (m_audioFile) {
+	std::cerr << "PlayableAudioFile(" << m_audioFile->getShortFilename() << ")::fillBuffers() [async] -- scanning to " << m_startIndex << std::endl;
+    } else {
+	std::cerr << "PlayableAudioFile::fillBuffers() [async] -- scanning to " << m_startIndex << std::endl;
+    }
+#endif
+
+    if (!m_isSmallFile && (!m_file || !*m_file)) {
+	m_file = new std::ifstream(m_audioFile->getFilename().c_str(),
+				   std::ios::in | std::ios::binary);
+	if (!*m_file) {
+	    std::cerr << "ERROR: PlayableAudioFile::fillBuffers: Failed to open audio file " << m_audioFile->getFilename() << std::endl;
+	    delete m_file;
+	    m_file = 0;
+	    return ;
+	}
+    }
+
+    scanTo(m_startIndex);
+    updateBuffers();
+}
+
+void
+PlayableAudioFile::clearBuffers()
+{
+    returnRingBuffers();
+}
+
+bool
+PlayableAudioFile::fillBuffers(const RealTime &currentTime)
+{
+#ifdef DEBUG_PLAYABLE
+    if (!m_isSmallFile) {
+	if (m_audioFile) {
+	    std::cerr << "PlayableAudioFile(" << m_audioFile->getShortFilename() << " " << this << ")::fillBuffers(" << currentTime << "):\n my start time " << m_startTime << ", start index " << m_startIndex << ", duration " << m_duration << std::endl;
+	} else {
+	    std::cerr << "PlayableAudioFile::fillBuffers(" << currentTime << "): my start time " << m_startTime << ", start index " << m_startIndex << ", duration " << m_duration << std::endl;
+	}
+    }
+#endif
+
+    if (currentTime > m_startTime + m_duration) {
+
+#ifdef DEBUG_PLAYABLE
+	std::cerr << "PlayableAudioFile::fillBuffers: seeking past end, returning buffers" << std::endl;
+#endif
+
+	returnRingBuffers();
+	return true;
+    }
+
+    if (!m_isSmallFile && (!m_file || !*m_file)) {
+	m_file = new std::ifstream(m_audioFile->getFilename().c_str(),
+				   std::ios::in | std::ios::binary);
+	if (!*m_file) {
+	    std::cerr << "ERROR: PlayableAudioFile::fillBuffers: Failed to open audio file " << m_audioFile->getFilename() << std::endl;
+	    delete m_file;
+	    m_file = 0;
+	    return false;
+	}
+	scanTo(m_startIndex);
+    }
+
+    RealTime scanTime = m_startIndex;
+
+    if (currentTime > m_startTime) {
+	scanTime = m_startIndex + currentTime - m_startTime;
+    }
+
+    //    size_t scanFrames = RealTime::realTime2Frame
+    //	(scanTime,
+    //	 m_isSmallFile ? m_targetSampleRate : m_audioFile->getSampleRate());
+
+    if (scanTime != m_currentScanPoint) {
+	scanTo(scanTime);
+    }
+
+    if (!m_isSmallFile) {
+	for (int i = 0; i < m_targetChannels; ++i) {
+	    if (m_ringBuffers[i])
+		m_ringBuffers[i]->reset();
+	}
+	updateBuffers();
+    }
+
+    return true;
+}
+
+bool
+PlayableAudioFile::updateBuffers()
+{
+    if (m_isSmallFile)
+	return false;
+    if (!m_file)
+	return false;
+
+    if (m_fileEnded) {
+#ifdef DEBUG_PLAYABLE_READ
+	std::cerr << "PlayableAudioFile::updateBuffers: at end of file already" << std::endl;
+#endif
+
+	return false;
+    }
+
+    if (!m_ringBuffers[0]) {
+
+	if (m_targetChannels < 0) {
+	    std::cerr << "WARNING: PlayableAudioFile::updateBuffers: m_targetChannels < 0, can't allocate ring buffers" << std::endl;
+	    return false;
+	}
+
+	// need a buffer: can we get one?
+	if (!m_ringBufferPool->getBuffers(m_targetChannels, m_ringBuffers)) {
+	    std::cerr << "WARNING: PlayableAudioFile::updateBuffers: no ring buffers available" << std::endl;
+	    return false;
+	}
+    }
+
+    size_t nframes = 0;
+
+    for (int ch = 0; ch < m_targetChannels; ++ch) {
+	if (!m_ringBuffers[ch])
+	    continue;
+	size_t writeSpace = m_ringBuffers[ch]->getWriteSpace();
+	if (ch == 0 || writeSpace < nframes)
+	    nframes = writeSpace;
+    }
+
+    if (nframes == 0) {
+#ifdef DEBUG_PLAYABLE_READ
+	std::cerr << "PlayableAudioFile::updateBuffers: frames == 0, ignoring" << std::endl;
+#endif
+
+	return false;
+    }
+
+#ifdef DEBUG_PLAYABLE_READ
+    std::cerr << "PlayableAudioFile::updateBuffers: want " << nframes << " frames" << std::endl;
+#endif
+
+
+    RealTime block = RealTime::frame2RealTime(nframes, m_targetSampleRate);
+    if (m_currentScanPoint + block >= m_startIndex + m_duration) {
+	block = m_startIndex + m_duration - m_currentScanPoint;
+	if (block <= RealTime::zeroTime)
+	    nframes = 0;
+	else
+	    nframes = RealTime::realTime2Frame(block, m_targetSampleRate);
+	m_fileEnded = true;
+    }
+
+    size_t fileFrames = nframes;
+    if (m_targetSampleRate != int(getSourceSampleRate())) {
+	fileFrames = size_t(float(fileFrames) * float(getSourceSampleRate()) /
+			    float(m_targetSampleRate));
+    }
+
+#ifdef DEBUG_PLAYABLE_READ
+    std::cerr << "Want " << fileFrames << " (" << block << ") from file (" << (m_duration + m_startIndex - m_currentScanPoint - block) << " to go)" << std::endl;
+#endif
+
+    //!!! need to be doing this in initialise, want to avoid allocations here
+    if ((getBytesPerFrame() * fileFrames) > m_rawFileBufferSize) {
+	delete[] m_rawFileBuffer;
+	m_rawFileBufferSize = getBytesPerFrame() * fileFrames;
+#ifdef DEBUG_PLAYABLE_READ
+
+	std::cerr << "Expanding raw file buffer to " << m_rawFileBufferSize << " chars" << std::endl;
+#endif
+
+	m_rawFileBuffer = new char[m_rawFileBufferSize];
+    }
+
+    size_t obtained =
+	m_audioFile->getSampleFrames(m_file, m_rawFileBuffer, fileFrames);
+
+    if (obtained < fileFrames || m_file->eof()) {
+	m_fileEnded = true;
+    }
+
+#ifdef DEBUG_PLAYABLE
+    std::cerr << "requested " << fileFrames << " frames from file for " << nframes << " frames, got " << obtained << " frames" << std::endl;
+#endif
+
+    if (nframes > m_workBufferSize) {
+
+	for (size_t i = 0; i < m_workBuffers.size(); ++i) {
+	    delete[] m_workBuffers[i];
+	}
+
+	m_workBuffers.clear();
+	m_workBufferSize = nframes;
+#ifdef DEBUG_PLAYABLE_READ
+
+	std::cerr << "Expanding work buffer to " << m_workBufferSize << " frames" << std::endl;
+#endif
+
+	for (int i = 0; i < m_targetChannels; ++i) {
+	    m_workBuffers.push_back(new sample_t[m_workBufferSize]);
+	}
+
+    } else {
+
+	while (m_targetChannels > m_workBuffers.size()) {
+	    m_workBuffers.push_back(new sample_t[m_workBufferSize]);
+	}
+    }
+
+    if (m_audioFile->decode((const unsigned char *)m_rawFileBuffer,
+			    obtained * getBytesPerFrame(),
+			    m_targetSampleRate,
+			    m_targetChannels,
+			    nframes,
+			    m_workBuffers,
+			    false)) {
+
+	/*!!! No -- GUI and notification side of things isn't up to this yet,
+	  so comment it out just in case
+                                  
+	  if (m_autoFade) {
+                                  
+	  if (m_currentScanPoint < m_startIndex + m_fadeInTime) {
+                                  
+	  size_t fadeSamples =
+	  RealTime::realTime2Frame(m_fadeInTime, getTargetSampleRate());
+	  size_t originSamples =
+	  RealTime::realTime2Frame(m_currentScanPoint - m_startIndex,
+	  getTargetSampleRate());
+                                  
+	  for (size_t i = 0; i < nframes; ++i) {
+	  if (i + originSamples > fadeSamples) {
+	  break;
+	  }
+	  float gain = float(i + originSamples) / float(fadeSamples);
+	  for (int ch = 0; ch < m_targetChannels; ++ch) {
+	  m_workBuffers[ch][i] *= gain;
+	  }
+	  }
+	  }
+                                  
+	  if (m_currentScanPoint + block > 
+	  m_startIndex + m_duration - m_fadeOutTime) {
+                                 		
+	  size_t fadeSamples =
+	  RealTime::realTime2Frame(m_fadeOutTime, getTargetSampleRate());
+	  size_t originSamples = // counting from end
+	  RealTime::realTime2Frame
+	  (m_startIndex + m_duration - m_currentScanPoint,
+	  getTargetSampleRate());
+                                  
+	  for (size_t i = 0; i < nframes; ++i) {
+	  float gain = 1.0;
+	  if (originSamples < i) gain = 0.0;
+	  else {
+	  size_t fromEnd = originSamples - i;
+	  if (fromEnd < fadeSamples) {
+	  gain = float(fromEnd) / float(fadeSamples);
+	  }
+	  }
+	  for (int ch = 0; ch < m_targetChannels; ++ch) {
+	  m_workBuffers[ch][i] *= gain;
+	  }
+	  }
+	  }
+	  }
+	*/
+
+	m_currentScanPoint = m_currentScanPoint + block;
+
+	for (int ch = 0; ch < m_targetChannels; ++ch) {
+
+	    if (m_firstRead || m_fileEnded) {
+		float xfade = std::min(m_xfadeFrames, nframes);
+		if (m_firstRead) {
+		    for (size_t i = 0; i < xfade; ++i) {
+			m_workBuffers[ch][i] *= float(i + 1) / xfade;
+		    }
+		}
+		if (m_fileEnded) {
+		    for (size_t i = 0; i < xfade; ++i) {
+			m_workBuffers[ch][nframes - i - 1] *=
+			    float(i + 1) / xfade;
+		    }
+		}
+	    }
+
+	    if (m_ringBuffers[ch]) {
+		m_ringBuffers[ch]->write(m_workBuffers[ch], nframes);
+	    }
+	}
+    }
+
+    m_firstRead = false;
+
+    if (obtained < fileFrames) {
+	if (m_file) {
+	    m_file->close();
+	    delete m_file;
+	    m_file = 0;
+	}
+    }
+
+    return true;
+}
+
+
+// How many channels in the base AudioFile?
+//
+unsigned int
+PlayableAudioFile::getSourceChannels()
+{
+    if (m_audioFile) {
+	return m_audioFile->getChannels();
+    }
+    return 0;
+}
+
+unsigned int
+PlayableAudioFile::getTargetChannels()
+{
+    return m_targetChannels;
+}
+
+unsigned int
+PlayableAudioFile::getBytesPerFrame()
+{
+    if (m_audioFile) {
+	return m_audioFile->getBytesPerFrame();
+    }
+    return 0;
+}
+
+unsigned int
+PlayableAudioFile::getSourceSampleRate()
+{
+    if (m_audioFile) {
+	return m_audioFile->getSampleRate();
+    }
+    return 0;
+}
+
+unsigned int
+PlayableAudioFile::getTargetSampleRate()
+{
+    return m_targetSampleRate;
+}
+
+
+// How many bits per sample in the base AudioFile?
+//
+unsigned int
+PlayableAudioFile::getBitsPerSample()
+{
+    if (m_audioFile) {
+	return m_audioFile->getBitsPerSample();
+    }
+    return 0;
+}
+
+
+}
+
diff --git a/src/sound/PlayableAudioFile.h b/src/sound/PlayableAudioFile.h
new file mode 100644
index 0000000..648ad4c
--- /dev/null
+++ b/src/sound/PlayableAudioFile.h
@@ -0,0 +1,219 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _PLAYABLE_AUDIO_FILE_H_
+#define _PLAYABLE_AUDIO_FILE_H_
+
+#include "Instrument.h"
+#include "RingBuffer.h"
+#include "AudioFile.h"
+#include "AudioCache.h"
+
+#include <string>
+#include <map>
+
+namespace Rosegarden
+{
+
+class RingBufferPool;
+
+
+class PlayableAudioFile
+{
+public:
+    typedef float sample_t;
+
+    PlayableAudioFile(InstrumentId instrumentId,
+                      AudioFile *audioFile,
+                      const RealTime &startTime,
+                      const RealTime &startIndex,
+                      const RealTime &duration,
+                      size_t bufferSize = 4096,
+                      size_t smallFileSize = 131072,
+                      int targetChannels = -1, // default same as file
+                      int targetSampleRate = -1); // default same as file
+    ~PlayableAudioFile();
+
+    static void setRingBufferPoolSizes(size_t n, size_t nframes);
+
+    void setStartTime(const RealTime &time) { m_startTime = time; }
+    RealTime getStartTime() const { return m_startTime; }
+
+    void setDuration(const RealTime &time) { m_duration = time; }
+    RealTime getDuration() const { return m_duration; }
+    RealTime getEndTime() const { return m_startTime + m_duration; }
+
+    void setStartIndex(const RealTime &time) { m_startIndex = time; }
+    RealTime getStartIndex() const { return m_startIndex; }
+
+    bool isSmallFile() const { return m_isSmallFile; }
+
+    // Get audio file for interrogation
+    //
+    AudioFile* getAudioFile() const { return m_audioFile; }
+
+    // Get instrument ID - we need to be able to map back
+    // at the GUI.
+    //
+    InstrumentId getInstrument() const { return m_instrumentId; }
+
+    // Return the number of frames currently buffered.  The next call
+    // to getSamples on any channel is guaranteed to return at least
+    // this many samples.
+    //
+    size_t getSampleFramesAvailable();
+
+    // Read samples from the given channel on the file and add them
+    // into the destination.
+    //
+    // If insufficient frames are available, this will leave the
+    // excess samples unchanged.
+    //
+    // Returns the actual number of samples written.
+    //
+    // If offset is non-zero, the samples will be written starting at
+    // offset frames from the start of the target block.
+    //
+    size_t addSamples(std::vector<sample_t *> &target,
+                      size_t channels, size_t nframes, size_t offset = 0);
+
+    unsigned int getSourceChannels();
+    unsigned int getTargetChannels();
+    unsigned int getSourceSampleRate();
+    unsigned int getTargetSampleRate();
+
+    unsigned int getBitsPerSample();
+    unsigned int getBytesPerFrame();
+
+    // Clear out and refill the ring buffer for immediate
+    // (asynchronous) play.
+    //
+    void fillBuffers();
+
+    // Clear out and refill the ring buffer (in preparation for
+    // playback) according to the proposed play time.
+    //
+    // This call and updateBuffers are not thread-safe (for
+    // performance reasons).  They should be called for all files
+    // sequentially within a single thread.
+    //
+    bool fillBuffers(const RealTime &currentTime);
+
+    void clearBuffers();
+
+    // Update the buffer during playback.
+    //
+    // This call and fillBuffers are not thread-safe (for performance
+    // reasons).  They should be called for all files sequentially
+    // within a single thread.
+    //
+    bool updateBuffers();
+
+    // Has fillBuffers been called and completed yet?
+    //
+    bool isBuffered() const { return m_currentScanPoint > m_startIndex; }
+
+    // Has all the data in this file now been read into the buffers?
+    //
+    bool isFullyBuffered() const { return m_isSmallFile || m_fileEnded; }
+
+    // Stop playing this file.
+    // 
+    void cancel() { m_fileEnded = true; }
+
+    // Segment id that allows us to crosscheck against playing audio
+    // segments.
+    //
+    int getRuntimeSegmentId() const { return m_runtimeSegmentId; }
+    void setRuntimeSegmentId(int id) { m_runtimeSegmentId = id; }
+
+    // Auto fading of a playable audio file
+    //
+    bool isAutoFading() const { return m_autoFade; }
+    void setAutoFade(bool value) { m_autoFade = value; }
+
+    RealTime getFadeInTime() const { return m_fadeInTime; }
+    void setFadeInTime(const RealTime &time) 
+        { m_fadeInTime = time; }
+
+    RealTime getFadeOutTime() const { return m_fadeOutTime; }
+    void setFadeOutTime(const RealTime &time) 
+        { m_fadeOutTime = time; }
+
+
+protected: 
+    void initialise(size_t bufferSize, size_t smallFileSize);
+    void checkSmallFileCache(size_t smallFileSize);
+    bool scanTo(const RealTime &time);
+    void returnRingBuffers();
+
+    RealTime              m_startTime;
+    RealTime              m_startIndex;
+    RealTime              m_duration;
+
+    // Performance file handle - must open non-blocking to
+    // allow other potential PlayableAudioFiles access to
+    // the same file.
+    //
+    std::ifstream        *m_file;
+
+    // AudioFile handle
+    //
+    AudioFile            *m_audioFile;
+
+    // Originating Instrument Id
+    //
+    InstrumentId          m_instrumentId;
+
+    int                   m_targetChannels;
+    int                   m_targetSampleRate;
+
+    bool                  m_fileEnded;
+    bool                  m_firstRead;
+    static size_t         m_xfadeFrames;
+    int                   m_runtimeSegmentId;
+
+    static AudioCache     m_smallFileCache;
+    bool                  m_isSmallFile;
+
+    static std::vector<sample_t *> m_workBuffers;
+    static size_t         m_workBufferSize;
+    
+    static char          *m_rawFileBuffer;
+    static size_t         m_rawFileBufferSize;
+
+    RingBuffer<sample_t>  **m_ringBuffers;
+    static RingBufferPool  *m_ringBufferPool;
+
+    RealTime              m_currentScanPoint;
+    size_t                m_smallFileScanFrame;
+
+    bool                  m_autoFade;
+    RealTime  m_fadeInTime;
+    RealTime  m_fadeOutTime;
+
+private:
+    PlayableAudioFile(const PlayableAudioFile &pAF); // not provided
+};
+
+}
+
+#endif
diff --git a/src/sound/PluginFactory.cpp b/src/sound/PluginFactory.cpp
new file mode 100644
index 0000000..49c1014
--- /dev/null
+++ b/src/sound/PluginFactory.cpp
@@ -0,0 +1,120 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "PluginFactory.h"
+#include "PluginIdentifier.h"
+
+#ifdef HAVE_LADSPA
+#include "LADSPAPluginFactory.h"
+#endif
+
+#ifdef HAVE_DSSI
+#include "DSSIPluginFactory.h"
+#endif
+
+#include <iostream>
+
+namespace Rosegarden
+{
+
+int PluginFactory::m_sampleRate = 48000;
+
+#ifdef HAVE_LADSPA
+static LADSPAPluginFactory *_ladspaInstance = 0;
+#endif
+
+#ifdef HAVE_DSSI
+static LADSPAPluginFactory *_dssiInstance = 0;
+#endif
+
+PluginFactory *
+PluginFactory::instance(QString pluginType)
+{
+    if (pluginType == "ladspa") {
+#ifdef HAVE_LADSPA
+        if (!_ladspaInstance) {
+            std::cerr << "PluginFactory::instance(" << pluginType
+            << "): creating new LADSPAPluginFactory" << std::endl;
+            _ladspaInstance = new LADSPAPluginFactory();
+            _ladspaInstance->discoverPlugins();
+        }
+        return _ladspaInstance;
+#else
+
+        return 0;
+#endif
+
+    } else if (pluginType == "dssi") {
+#ifdef HAVE_DSSI
+        if (!_dssiInstance) {
+            std::cerr << "PluginFactory::instance(" << pluginType
+            << "): creating new DSSIPluginFactory" << std::endl;
+            _dssiInstance = new DSSIPluginFactory();
+            _dssiInstance->discoverPlugins();
+        }
+        return _dssiInstance;
+#else
+
+        return 0;
+#endif
+
+    }
+    else
+        return 0;
+}
+
+PluginFactory *
+PluginFactory::instanceFor(QString identifier)
+{
+    QString type, soName, label;
+    PluginIdentifier::parseIdentifier(identifier, type, soName, label);
+    return instance(type);
+}
+
+void
+PluginFactory::enumerateAllPlugins(MappedObjectPropertyList &list)
+{
+    PluginFactory *factory;
+
+    // Plugins can change the locale, store it for reverting afterwards
+    char *loc = setlocale(LC_ALL, 0);
+
+    // Query DSSI plugins before LADSPA ones.
+    // This is to provide for the interesting possibility of plugins
+    // providing either DSSI or LADSPA versions of themselves,
+    // returning both versions if the LADSPA identifiers are queried
+    // first but only the DSSI version if the DSSI identifiers are
+    // queried first.
+
+    factory = instance("dssi");
+    if (factory)
+        factory->enumeratePlugins(list);
+
+    factory = instance("ladspa");
+    if (factory)
+        factory->enumeratePlugins(list);
+
+    setlocale(LC_ALL, loc);
+}
+
+
+}
+
diff --git a/src/sound/PluginFactory.h b/src/sound/PluginFactory.h
new file mode 100644
index 0000000..820b233
--- /dev/null
+++ b/src/sound/PluginFactory.h
@@ -0,0 +1,97 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _PLUGIN_FACTORY_H_
+#define _PLUGIN_FACTORY_H_
+
+#include <qstring.h>
+#include <vector>
+
+#include "MappedCommon.h"
+
+namespace Rosegarden
+{
+
+class RunnablePluginInstance;
+class MappedPluginSlot;
+
+class PluginFactory
+{
+public:
+    static PluginFactory *instance(QString pluginType);
+    static PluginFactory *instanceFor(QString identifier);
+    static void enumerateAllPlugins(MappedObjectPropertyList &);
+
+    static void setSampleRate(int sampleRate) { m_sampleRate = sampleRate; }
+
+    /**
+     * Look up the plugin path and find the plugins in it.  Called 
+     * automatically after construction of a factory.
+     */
+    virtual void discoverPlugins() = 0;
+
+    /**
+     * Return a reference to a list of all plugin identifiers that can
+     * be created by this factory.
+     */
+    virtual const std::vector<QString> &getPluginIdentifiers() const = 0;
+
+    /**
+     * Append to the given list descriptions of all the available
+     * plugins and their ports.  This is in a standard format, see
+     * the LADSPA implementation for details.
+     */
+    virtual void enumeratePlugins(MappedObjectPropertyList &list) = 0;
+
+    /**
+     * Populate the given plugin slot with information about its
+     * plugin.  This is called from the plugin slot's set method
+     * when it's been asked to set its plugin identifier.  This
+     * method should also destroy and recreate the plugin slot's
+     * port child objects.
+     */
+    virtual void populatePluginSlot(QString identifier,
+                                    MappedPluginSlot &slot) = 0;
+
+    /**
+     * Instantiate a plugin.
+     */
+    virtual RunnablePluginInstance *instantiatePlugin(QString identifier,
+                                                      int instrumentId,
+                                                      int position,
+                                                      unsigned int sampleRate,
+                                                      unsigned int blockSize,
+                                                      unsigned int channels) = 0;
+
+protected:
+    PluginFactory() { }
+
+    // for call by RunnablePluginInstance dtor
+    virtual void releasePlugin(RunnablePluginInstance *, QString identifier) = 0;
+    friend class RunnablePluginInstance;
+
+    static int m_sampleRate;
+};
+
+
+}
+
+#endif
diff --git a/src/sound/PluginIdentifier.cpp b/src/sound/PluginIdentifier.cpp
new file mode 100644
index 0000000..25b3317
--- /dev/null
+++ b/src/sound/PluginIdentifier.cpp
@@ -0,0 +1,72 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "PluginIdentifier.h"
+#include <iostream>
+
+namespace Rosegarden
+{
+
+QString
+PluginIdentifier::createIdentifier(QString type,
+                                   QString soName,
+                                   QString label)
+{
+    QString identifier = type + ":" + soName + ":" + label;
+    return identifier;
+}
+
+void
+PluginIdentifier::parseIdentifier(QString identifier,
+                                  QString &type,
+                                  QString &soName,
+                                  QString &label)
+{
+    type = identifier.section(':', 0, 0);
+    soName = identifier.section(':', 1, 1);
+    label = identifier.section(':', 2);
+}
+
+bool
+PluginIdentifier::areIdentifiersSimilar(QString id1, QString id2)
+{
+    QString type1, type2, soName1, soName2, label1, label2;
+
+    parseIdentifier(id1, type1, soName1, label1);
+    parseIdentifier(id2, type2, soName2, label2);
+
+    if (type1 != type2 || label1 != label2)
+        return false;
+
+    bool similar = (soName1.section('/', -1).section('.', 0, 0) ==
+                    soName2.section('/', -1).section('.', 0, 0));
+
+    return similar;
+}
+
+// The prefix of this key is also used as a literal in base/AudioPluginInstance.C.
+// If you change one, change the other.
+// Better still, don't change one.
+QString
+PluginIdentifier::RESERVED_PROJECT_DIRECTORY_KEY = "__ROSEGARDEN__:__RESERVED__:ProjectDirectoryKey";
+
+}
+
diff --git a/src/sound/PluginIdentifier.h b/src/sound/PluginIdentifier.h
new file mode 100644
index 0000000..e8519ad
--- /dev/null
+++ b/src/sound/PluginIdentifier.h
@@ -0,0 +1,50 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _PLUGIN_IDENTIFIER_H_
+#define _PLUGIN_IDENTIFIER_H_
+
+#include <qstring.h>
+
+
+// A plugin identifier is simply a string; this class provides methods
+// to parse it into its constituent bits (plugin type, DLL path and label).
+
+namespace Rosegarden {
+
+class PluginIdentifier {
+
+public:
+ 
+    static QString createIdentifier(QString type, QString soName, QString label);
+
+    static void parseIdentifier(QString identifier,
+                                QString &type, QString &soName, QString &label);
+
+    static bool areIdentifiersSimilar(QString id1, QString id2);
+
+    // Not strictly related to identifiers
+    static QString RESERVED_PROJECT_DIRECTORY_KEY;
+};
+
+}
+
+#endif
diff --git a/src/sound/RIFFAudioFile.cpp b/src/sound/RIFFAudioFile.cpp
new file mode 100644
index 0000000..c34435f
--- /dev/null
+++ b/src/sound/RIFFAudioFile.cpp
@@ -0,0 +1,686 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "RIFFAudioFile.h"
+#include "RealTime.h"
+#include "Profiler.h"
+
+using std::cout;
+using std::cerr;
+using std::endl;
+
+//#define DEBUG_RIFF
+
+
+namespace Rosegarden
+{
+
+RIFFAudioFile::RIFFAudioFile(unsigned int id,
+                             const std::string &name,
+                             const std::string &fileName):
+    AudioFile(id, name, fileName),
+    m_subFormat(PCM),
+    m_bytesPerSecond(0),
+    m_bytesPerFrame(0)
+{}
+
+RIFFAudioFile::RIFFAudioFile(const std::string &fileName,
+                             unsigned int channels = 1,
+                             unsigned int sampleRate = 48000,
+                             unsigned int bytesPerSecond = 6000,
+                             unsigned int bytesPerFrame = 2,
+                             unsigned int bitsPerSample = 16):
+        AudioFile(0, "", fileName)
+{
+    m_bitsPerSample = bitsPerSample;
+    m_sampleRate = sampleRate;
+    m_bytesPerSecond = bytesPerSecond;
+    m_bytesPerFrame = bytesPerFrame;
+    m_channels = channels;
+
+    if (bitsPerSample == 16)
+        m_subFormat = PCM;
+    else if (bitsPerSample == 32)
+        m_subFormat = FLOAT;
+    else
+        throw(BadSoundFileException(m_fileName, "Rosegarden currently only supports 16 or 32-bit PCM or IEEE floating-point RIFF files for writing"));
+
+}
+
+RIFFAudioFile::~RIFFAudioFile()
+{}
+
+
+// Show some stats on this file
+//
+void
+RIFFAudioFile::printStats()
+{
+    cout << "filename         : " << m_fileName << endl
+    << "channels         : " << m_channels << endl
+    << "sample rate      : " << m_sampleRate << endl
+    << "bytes per second : " << m_bytesPerSecond << endl
+    << "bits per sample  : " << m_bitsPerSample << endl
+    << "bytes per frame  : " << m_bytesPerFrame << endl
+    << "file length      : " << m_fileSize << " bytes" << endl
+    << endl;
+}
+
+bool
+RIFFAudioFile::appendSamples(const std::string &buffer)
+{
+    /*
+    if (m_outFile == 0 || m_type != WAV)
+        return false;
+        */
+
+    // write out
+    putBytes(m_outFile, buffer);
+
+    return true;
+}
+
+bool
+RIFFAudioFile::appendSamples(const char *buf, unsigned int frames)
+{
+    putBytes(m_outFile, buf, frames * m_bytesPerFrame);
+    return true;
+}
+
+// scan on from a descriptor position
+bool
+RIFFAudioFile::scanForward(std::ifstream *file, const RealTime &time)
+{
+    // sanity
+    if (file == 0)
+        return false;
+
+    unsigned int totalSamples = m_sampleRate * time.sec +
+                                ( ( m_sampleRate * time.usec() ) / 1000000 );
+    unsigned int totalBytes = totalSamples * m_bytesPerFrame;
+
+    m_loseBuffer = true;
+
+    // do the seek
+    file->seekg(totalBytes, std::ios::cur);
+
+    if (file->eof())
+        return false;
+
+    return true;
+}
+
+bool
+RIFFAudioFile::scanForward(const RealTime &time)
+{
+    if (*m_inFile)
+        return scanForward(m_inFile, time);
+    else
+        return false;
+}
+
+bool
+RIFFAudioFile::scanTo(const RealTime &time)
+{
+    if (*m_inFile)
+        return scanTo(m_inFile, time);
+    else
+        return false;
+
+}
+
+bool
+RIFFAudioFile::scanTo(std::ifstream *file, const RealTime &time)
+{
+    // sanity
+    if (file == 0)
+        return false;
+
+    // whatever we do here we invalidate the read buffer
+    //
+    m_loseBuffer = true;
+
+    file->clear();
+
+    // seek past header - don't hardcode this - use the file format
+    // spec to get header length and then scoot to that.
+    //
+    file->seekg(16, std::ios::beg);
+
+    unsigned int lengthOfFormat = 0;
+
+    try {
+        lengthOfFormat = getIntegerFromLittleEndian(getBytes(file, 4));
+        file->seekg(lengthOfFormat, std::ios::cur);
+
+        // check we've got data chunk start
+	std::string chunkName;
+	int chunkLength = 0;
+
+        while ((chunkName = getBytes(file, 4)) != "data") {
+	    if (file->eof()) {
+		std::cerr << "RIFFAudioFile::scanTo(): failed to find data "
+			  << std::endl;
+		return false;
+	    }
+//#ifdef DEBUG_RIFF
+	    std::cerr << "RIFFAudioFile::scanTo(): skipping chunk: "
+		      << chunkName << std::endl;
+//#endif
+	    chunkLength = getIntegerFromLittleEndian(getBytes(file, 4));
+	    if (chunkLength < 0) {
+		std::cerr << "RIFFAudioFile::scanTo(): negative chunk length "
+			  << chunkLength << " for chunk " << chunkName << std::endl;
+		return false;
+	    }
+	    file->seekg(chunkLength, std::ios::cur);
+        }
+
+        // get the length of the data chunk, and scan past it as a side-effect
+	chunkLength = getIntegerFromLittleEndian(getBytes(file, 4));
+#ifdef DEBUG_RIFF
+
+        std::cout << "RIFFAudioFile::scanTo() - data chunk size = "
+        << chunkLength << std::endl;
+#endif
+
+    } catch (BadSoundFileException s) {
+#ifdef DEBUG_RIFF
+        std::cerr << "RIFFAudioFile::scanTo - EXCEPTION - \""
+        << s.getMessage() << "\"" << std::endl;
+#endif
+
+        return false;
+    }
+
+    // Ok, we're past all the header information in the data chunk.
+    // Now, how much do we scan forward?
+    //
+    size_t totalFrames = RealTime::realTime2Frame(time, m_sampleRate);
+
+    unsigned int totalBytes = totalFrames * m_bytesPerFrame;
+
+    // When using seekg we have to keep an eye on the boundaries ourselves
+    //
+    if (totalBytes > m_fileSize - (lengthOfFormat + 16 + 8)) {
+#ifdef DEBUG_RIFF
+        std::cerr << "RIFFAudioFile::scanTo() - attempting to move past end of "
+        << "data block" << std::endl;
+#endif
+
+        return false;
+    }
+
+#ifdef DEBUG_RIFF
+    std::cout << "RIFFAudioFile::scanTo - seeking to " << time
+    << " (" << totalBytes << " bytes from current " << file->tellg()
+    << ")" << std::endl;
+#endif
+
+    file->seekg(totalBytes, std::ios::cur);
+
+    return true;
+}
+
+// Get a certain number of sample frames - a frame is a set
+// of samples (all channels) for a given sample quanta.
+//
+// For example, getting one frame of 16-bit stereo will return
+// four bytes of data (two per channel).
+//
+//
+std::string
+RIFFAudioFile::getSampleFrames(std::ifstream *file, unsigned int frames)
+{
+    // sanity
+    if (file == 0)
+        return std::string("");
+
+    // Bytes per sample already takes into account the number
+    // of channels we're using
+    //
+    long totalBytes = frames * m_bytesPerFrame;
+
+    try {
+        return getBytes(file, totalBytes);
+    } catch (BadSoundFileException s) {
+        return "";
+    }
+}
+
+unsigned int
+RIFFAudioFile::getSampleFrames(std::ifstream *file, char *buf,
+                               unsigned int frames)
+{
+    if (file == 0)
+        return 0;
+    try {
+        return getBytes(file, buf, frames * m_bytesPerFrame) / m_bytesPerFrame;
+    } catch (BadSoundFileException s) {
+        return 0;
+    }
+}
+
+std::string
+RIFFAudioFile::getSampleFrames(unsigned int frames)
+{
+    if (*m_inFile) {
+        return getSampleFrames(m_inFile, frames);
+    } else {
+        return std::string("");
+    }
+}
+
+// Return a slice of frames over a time period
+//
+std::string
+RIFFAudioFile::getSampleFrameSlice(std::ifstream *file, const RealTime &time)
+{
+    // sanity
+    if (file == 0)
+        return std::string("");
+
+    long totalFrames = RealTime::realTime2Frame(time, m_sampleRate);
+    long totalBytes = totalFrames * m_bytesPerFrame;
+
+    try {
+        return getBytes(file, totalBytes);
+    } catch (BadSoundFileException s) {
+        return "";
+    }
+}
+
+std::string
+RIFFAudioFile::getSampleFrameSlice(const RealTime &time)
+{
+    if (*m_inFile) {
+        return getSampleFrameSlice(m_inFile, time);
+    } else {
+        return std::string("");
+    }
+}
+
+RealTime
+RIFFAudioFile::getLength()
+{
+    // Fixed header size = 44 but prove by getting it from the file too
+    //
+    unsigned int headerLength = 44;
+
+    if (m_inFile) {
+        m_inFile->seekg(16, std::ios::beg);
+        headerLength = getIntegerFromLittleEndian(getBytes(m_inFile, 4));
+        m_inFile->seekg(headerLength, std::ios::cur);
+        headerLength += (16 + 8);
+    }
+
+    if (!m_bytesPerFrame || !m_sampleRate) return RealTime::zeroTime;
+
+    double frames = (m_fileSize - headerLength) / m_bytesPerFrame;
+    double seconds = frames / ((double)m_sampleRate);
+
+    int secs = int(seconds);
+    int nsecs = int((seconds - secs) * 1000000000.0);
+
+    return RealTime(secs, nsecs);
+}
+
+
+// The RIFF file format chunk defines our internal meta data.
+//
+// Courtesy of:
+//   http://www.technology.niagarac.on.ca/courses/comp630/WavFileFormat.html
+//
+// 'The WAV file itself consists of three "chunks" of information:
+//  The RIFF chunk which identifies the file as a WAV file, The FORMAT
+//  chunk which identifies parameters such as sample rate and the DATA
+//  chunk which contains the actual data (samples).'
+//
+//
+void
+RIFFAudioFile::readFormatChunk()
+{
+    if (m_inFile == 0)
+        return ;
+
+    m_loseBuffer = true;
+
+    // seek to beginning
+    m_inFile->seekg(0, std::ios::beg);
+
+    // get the header string
+    //
+    std::string hS = getBytes(36);
+
+    // Look for the RIFF identifier and bomb out if we don't find it
+    //
+#if (__GNUC__ < 3)
+
+    if (hS.compare(AUDIO_RIFF_ID, 0, 4) != 0)
+#else
+
+    if (hS.compare(0, 4, AUDIO_RIFF_ID) != 0)
+#endif
+
+    {
+#ifdef DEBUG_RIFF
+        std::cerr << "RIFFAudioFile::readFormatChunk - "
+        << "can't find RIFF identifier\n";
+#endif
+
+        throw(BadSoundFileException(m_fileName, "RIFFAudioFile::readFormatChunk - can't find RIFF identifier"));
+    }
+
+    // Look for the WAV identifier
+    //
+#if (__GNUC__ < 3)
+    if (hS.compare(AUDIO_WAVE_ID, 8, 4) != 0)
+#else
+
+    if (hS.compare(8, 4, AUDIO_WAVE_ID) != 0)
+#endif
+
+    {
+#ifdef DEBUG_RIFF
+        std::cerr << "Can't find WAV identifier\n";
+#endif
+
+        throw(BadSoundFileException(m_fileName, "Can't find WAV identifier"));
+    }
+
+    // Look for the FORMAT identifier - note that this doesn't actually
+    // have to be in the first chunk we come across, but for the moment
+    // this is the only place we check for it because I'm lazy.
+    //
+    //
+#if (__GNUC__ < 3)
+    if (hS.compare(AUDIO_FORMAT_ID, 12, 4) != 0)
+#else
+
+    if (hS.compare(12, 4, AUDIO_FORMAT_ID) != 0)
+#endif
+
+    {
+#ifdef DEBUG_RIFF
+        std::cerr << "Can't find FORMAT identifier\n";
+#endif
+
+        throw(BadSoundFileException(m_fileName, "Can't find FORMAT identifier"));
+    }
+
+    // Little endian conversion of length bytes into file length
+    // (add on eight for RIFF id and length field and compare to
+    // real file size).
+    //
+    unsigned int length = getIntegerFromLittleEndian(hS.substr(4, 4)) + 8;
+
+    if (length != m_fileSize) {
+        std::cerr << "WARNING: RIFFAudioFile: incorrect length ("
+        << length << ", file size is " << m_fileSize << "), ignoring"
+        << std::endl;
+        length = m_fileSize;
+    }
+
+    // Check the format length
+    //
+    unsigned int lengthOfFormat = getIntegerFromLittleEndian(hS.substr(16, 4));
+
+    // Make sure we step to the end of the format chunk ignoring the
+    // tail if it exists
+    //
+    if (lengthOfFormat > 0x10) {
+#ifdef DEBUG_RIFF
+        std::cerr << "RIFFAudioFile::readFormatChunk - "
+        << "extended Format Chunk (" << lengthOfFormat << ")"
+        << std::endl;
+#endif
+
+        // ignore any overlapping bytes
+        m_inFile->seekg(lengthOfFormat - 0x10, std::ios::cur);
+    } else if (lengthOfFormat < 0x10) {
+#ifdef DEBUG_RIFF
+        std::cerr << "RIFFAudioFile::readFormatChunk - "
+        << "truncated Format Chunk (" << lengthOfFormat << ")"
+        << std::endl;
+#endif
+
+        m_inFile->seekg(lengthOfFormat - 0x10, std::ios::cur);
+        //throw(BadSoundFileException(m_fileName, "Format chunk too short"));
+    }
+
+
+    // Check sub format - we support PCM or IEEE floating point.
+    //
+    unsigned int subFormat = getIntegerFromLittleEndian(hS.substr(20, 2));
+
+    if (subFormat == 0x01) {
+        m_subFormat = PCM;
+    } else if (subFormat == 0x03) {
+        m_subFormat = FLOAT;
+    } else {
+        throw(BadSoundFileException(m_fileName, "Rosegarden currently only supports PCM or IEEE floating-point RIFF files"));
+    }
+
+    // We seem to have a good looking .WAV file - extract the
+    // sample information and populate this locally
+    //
+    unsigned int channelNumbers = getIntegerFromLittleEndian(hS.substr(22, 2));
+
+    switch (channelNumbers) {
+    case 0x01:
+    case 0x02:
+        m_channels = channelNumbers;
+        break;
+
+    default: {
+            throw(BadSoundFileException(m_fileName, "Unsupported number of channels"));
+        }
+        break;
+    }
+
+    // Now the rest of the information
+    //
+    m_sampleRate = getIntegerFromLittleEndian(hS.substr(24, 4));
+    m_bytesPerSecond = getIntegerFromLittleEndian(hS.substr(28, 4));
+    m_bytesPerFrame = getIntegerFromLittleEndian(hS.substr(32, 2));
+    m_bitsPerSample = getIntegerFromLittleEndian(hS.substr(34, 2));
+
+    if (m_subFormat == PCM) {
+        if (m_bitsPerSample != 8 && m_bitsPerSample != 16 && m_bitsPerSample != 24) {
+            throw BadSoundFileException("Rosegarden currently only supports 8-, 16- or 24-bit PCM in RIFF files");
+        }
+    } else if (m_subFormat == FLOAT) {
+        if (m_bitsPerSample != 32) {
+            throw BadSoundFileException("Rosegarden currently only supports 32-bit floating-point in RIFF files");
+        }
+    }
+
+   // printStats();
+
+}
+
+// Write out the format chunk from our internal data
+//
+void
+RIFFAudioFile::writeFormatChunk()
+{
+    if (m_outFile == 0 || m_type != WAV)
+        return ;
+
+    std::string outString;
+
+    // RIFF type is all we support for the moment
+    outString += AUDIO_RIFF_ID;
+
+    // Now write the total length of the file minus these first 8 bytes.
+    // We won't know this until we've finished recording the file.
+    //
+    outString += "0000";
+
+    // WAV file is all we support
+    //
+    outString += AUDIO_WAVE_ID;
+
+    // Begin the format chunk
+    outString += AUDIO_FORMAT_ID;
+
+    // length
+    //cout << "LENGTH = " << getLittleEndianFromInteger(0x10, 4) << endl;
+    outString += getLittleEndianFromInteger(0x10, 4);
+
+    // 1 for PCM, 3 for float
+    if (m_subFormat == PCM) {
+        outString += getLittleEndianFromInteger(0x01, 2);
+    } else {
+        outString += getLittleEndianFromInteger(0x03, 2);
+    }
+
+    // channel
+    outString += getLittleEndianFromInteger(m_channels, 2);
+
+    // sample rate
+    outString += getLittleEndianFromInteger(m_sampleRate, 4);
+
+    // bytes per second
+    outString += getLittleEndianFromInteger(m_bytesPerSecond, 4);
+
+    // bytes per sample
+    outString += getLittleEndianFromInteger(m_bytesPerFrame, 2);
+
+    // bits per sample
+    outString += getLittleEndianFromInteger(m_bitsPerSample, 2);
+
+    // Now mark the beginning of the "data" chunk and leave the file
+    // open for writing.
+    outString += "data";
+
+    // length of data to follow - again needs to be written after
+    // we've completed the file.
+    //
+    outString += "0000";
+
+    // write out
+    //
+    putBytes(m_outFile, outString);
+}
+
+
+AudioFileType
+RIFFAudioFile::identifySubType(const std::string &filename)
+{
+    std::ifstream *testFile =
+        new std::ifstream(filename.c_str(), std::ios::in | std::ios::binary);
+
+    if (!(*testFile))
+        return UNKNOWN;
+
+    std::string hS;
+    unsigned int numberOfBytes = 36;
+    char *bytes = new char[numberOfBytes];
+
+    testFile->read(bytes, numberOfBytes);
+    for (unsigned int i = 0; i < numberOfBytes; i++)
+        hS += (unsigned char)bytes[i];
+
+    AudioFileType type = UNKNOWN;
+
+    // Test for BWF first because it's an extension of a plain WAV
+    //
+#if (__GNUC__ < 3)
+
+    if (hS.compare(AUDIO_RIFF_ID, 0, 4) == 0 &&
+            hS.compare(AUDIO_WAVE_ID, 8, 4) == 0 &&
+            hS.compare(AUDIO_BWF_ID, 12, 4) == 0)
+#else
+
+    if (hS.compare(0, 4, AUDIO_RIFF_ID) == 0 &&
+            hS.compare(8, 4, AUDIO_WAVE_ID) == 0 &&
+            hS.compare(12, 4, AUDIO_BWF_ID) == 0)
+#endif
+
+    {
+        type = BWF;
+    }
+    // Now for a WAV
+#if (__GNUC__ < 3)
+    else if (hS.compare(AUDIO_RIFF_ID, 0, 4) == 0 &&
+             hS.compare(AUDIO_WAVE_ID, 8, 4) == 0)
+#else
+
+    else if (hS.compare(0, 4, AUDIO_RIFF_ID) == 0 &&
+             hS.compare(8, 4, AUDIO_WAVE_ID) == 0)
+#endif
+
+    {
+        type = WAV;
+    } else
+        type = UNKNOWN;
+
+    testFile->close();
+    delete [] bytes;
+
+    return type;
+}
+
+float
+RIFFAudioFile::convertBytesToSample(const unsigned char *ubuf)
+{
+    switch (getBitsPerSample()) {
+
+    case 8: {
+            // WAV stores 8-bit samples unsigned, other sizes signed.
+            return (float)(ubuf[0] - 128.0) / 128.0;
+        }
+
+    case 16: {
+            // Two's complement little-endian 16-bit integer.
+            // We convert endianness (if necessary) but assume 16-bit short.
+            unsigned char b2 = ubuf[0];
+            unsigned char b1 = ubuf[1];
+            unsigned int bits = (b1 << 8) + b2;
+            return (float)(short(bits)) / 32767.0;
+        }
+
+    case 24: {
+            // Two's complement little-endian 24-bit integer.
+            // Again, convert endianness but assume 32-bit int.
+            unsigned char b3 = ubuf[0];
+            unsigned char b2 = ubuf[1];
+            unsigned char b1 = ubuf[2];
+            // Rotate 8 bits too far in order to get the sign bit
+            // in the right place; this gives us a 32-bit value,
+            // hence the larger float divisor
+            unsigned int bits = (b1 << 24) + (b2 << 16) + (b3 << 8);
+            return (float)(int(bits)) / 2147483647.0;
+        }
+
+    case 32: {
+            // IEEE floating point
+            return *(float *)ubuf;
+        }
+
+    default:
+        return 0.0f;
+    }
+}
+
+}
+
diff --git a/src/sound/RIFFAudioFile.h b/src/sound/RIFFAudioFile.h
new file mode 100644
index 0000000..a846306
--- /dev/null
+++ b/src/sound/RIFFAudioFile.h
@@ -0,0 +1,168 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+
+// Resource Interchange File Formt - a chunk based audio 
+// file format.  Type of chunk varies with specialisation
+// of this class - WAV files are a specialisation with just
+// a format chunk, BWF has more chunks.
+//
+//
+
+#ifndef _RIFFAUDIOFILE_H_
+#define _RIFFAUDIOFILE_H_
+
+#include <string>
+#include <vector>
+
+#include "AudioFile.h"
+#include "RealTime.h"
+
+namespace Rosegarden
+{
+
+class RIFFAudioFile : public AudioFile
+{
+public:
+    RIFFAudioFile(unsigned int id,
+                  const std::string &name,
+                  const std::string &fileName);
+
+    RIFFAudioFile(const std::string &fileName,
+                  unsigned int channels,
+                  unsigned int sampleRate,
+                  unsigned int bytesPerSecond,
+                  unsigned int bytesPerFrame,
+                  unsigned int bitsPerSample);
+
+    ~RIFFAudioFile();
+
+    typedef enum {
+        PCM,
+        FLOAT
+    } SubFormat;
+
+    // Our main control methods - again keeping abstract at this level
+    //
+    //virtual bool open() = 0;
+    //virtual bool write() = 0;
+    //virtual void close() = 0;
+
+    // Show the information we have on this file
+    //
+    virtual void printStats();
+
+    // Slightly dodgy code here - we keep these functions here
+    // because I don't want to duplicate them in PlayableRIFFAudioFile
+    // and also don't want that class to inherit this one.
+    //
+    // Of course the file handle we use in might be pointing to
+    // any file - for the most part we just assume it's an audio
+    // file.
+    //
+    //
+    // Move file pointer to relative time in data chunk -
+    // shouldn't be less than zero.  Returns true if the
+    // scan time was valid and successful.
+    // 
+    virtual bool scanTo(const RealTime &time);
+    virtual bool scanTo(std::ifstream *file, const RealTime &time);
+
+    // Scan forward in a file by a certain amount of time
+    //
+    virtual bool scanForward(const RealTime &time);
+    virtual bool scanForward(std::ifstream *file, const RealTime &time);
+
+    // Return a number of samples - caller will have to
+    // de-interleave n-channel samples themselves.
+    //
+    virtual std::string getSampleFrames(std::ifstream *file,
+                                        unsigned int frames);
+    virtual unsigned int getSampleFrames(std::ifstream *file,
+                                         char *buf,
+                                         unsigned int frames);
+    virtual std::string getSampleFrames(unsigned int frames);
+
+    // Return a number of (possibly) interleaved samples
+    // over a time slice from current file pointer position.
+    //
+    virtual std::string getSampleFrameSlice(std::ifstream *file,
+                                            const RealTime &time);
+    virtual std::string getSampleFrameSlice(const RealTime &time);
+
+    // Append a string of samples to an already open (for writing)
+    // audio file.
+    //
+    virtual bool appendSamples(const std::string &buffer);
+    virtual bool appendSamples(const char *buf, unsigned int frames);
+
+    // Get the length of the sample in Seconds/Microseconds
+    //
+    virtual RealTime getLength();
+
+    // Accessors
+    //
+    virtual unsigned int getBytesPerFrame() { return m_bytesPerFrame; }
+    unsigned int getBytesPerSecond() { return m_bytesPerSecond; }
+
+    // Allow easy identification of wav file type
+    //
+    static AudioFileType identifySubType(const std::string &filename);
+
+    // Convert a single sample from byte format, given the right
+    // number of bytes for the sample width
+    float convertBytesToSample(const unsigned char *bytes);
+
+    // Decode and de-interleave the given samples that were retrieved
+    // from this file or another with the same format as it.  Place
+    // the results in the given float buffer.  Return true for
+    // success.  This function does crappy resampling if necessary.
+    // 
+    virtual bool decode(const unsigned char *sourceData,
+                        size_t sourceBytes,
+                        size_t targetSampleRate,
+                        size_t targetChannels,
+                        size_t targetFrames,
+                        std::vector<float *> &targetData,
+                        bool addToResultBuffers = false) = 0;
+
+protected:
+    //virtual void parseHeader(const std::string &header);
+    //virtual void parseBody();
+
+    // Find and read in the format chunk of a RIFF file - without
+    // this chunk we don't actually have a RIFF file.
+    //
+    void readFormatChunk();
+
+    // Write out the Format chunk from the internal data we have
+    //
+    void writeFormatChunk();
+
+    SubFormat m_subFormat;
+    unsigned int   m_bytesPerSecond;
+    unsigned int   m_bytesPerFrame;
+};
+
+}
+
+
+#endif // _RIFFAUDIOFILE_H_
diff --git a/src/sound/RecordableAudioFile.cpp b/src/sound/RecordableAudioFile.cpp
new file mode 100644
index 0000000..09420dd
--- /dev/null
+++ b/src/sound/RecordableAudioFile.cpp
@@ -0,0 +1,164 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "RecordableAudioFile.h"
+
+#include <cstdlib>
+
+//#define DEBUG_RECORDABLE 1
+
+namespace Rosegarden
+{
+
+RecordableAudioFile::RecordableAudioFile(AudioFile *audioFile,
+					 size_t bufferSize) :
+    m_audioFile(audioFile),
+    m_status(IDLE)
+{
+    for (unsigned int ch = 0; ch < audioFile->getChannels(); ++ch) {
+
+	m_ringBuffers.push_back(new RingBuffer<sample_t>(bufferSize));
+
+	if (!m_ringBuffers[ch]->mlock()) {
+	    std::cerr << "WARNING: RecordableAudioFile::initialise: couldn't lock buffer into real memory, performance may be impaired" << std::endl;
+	}
+    }
+}
+
+RecordableAudioFile::~RecordableAudioFile()
+{
+    write();
+    m_audioFile->close();
+    delete m_audioFile;
+
+    for (size_t i = 0; i < m_ringBuffers.size(); ++i) {
+	delete m_ringBuffers[i];
+    }
+}
+
+size_t
+RecordableAudioFile::buffer(const sample_t *data, int channel, size_t frames)
+{
+    if (channel >= int(m_ringBuffers.size())) {
+	std::cerr << "RecordableAudioFile::buffer: No such channel as "
+		  << channel << std::endl;
+	return 0;
+    }
+
+    size_t available = m_ringBuffers[channel]->getWriteSpace();
+
+    if (frames > available) {
+	std::cerr << "RecordableAudioFile::buffer: buffer maxed out!" << std::endl;
+	frames = available;
+    }
+
+#ifdef DEBUG_RECORDABLE
+    std::cerr << "RecordableAudioFile::buffer: buffering " << frames << " frames on channel " << channel << std::endl;
+#endif
+
+    m_ringBuffers[channel]->write(data, frames);
+    return frames;
+}
+
+void
+RecordableAudioFile::write()
+{
+    // Use a static buffer -- this obviously requires that write() is
+    // only called from a single thread
+    static size_t bufferSize = 0;
+    static sample_t *buffer = 0;
+    static char *encodeBuffer = 0;
+
+    unsigned int bits = m_audioFile->getBitsPerSample();
+
+    if (bits != 16 && bits != 32) {
+	std::cerr << "ERROR: RecordableAudioFile::write: file has " << bits
+		  << " bits per sample; only 16 or 32 are supported" << std::endl;
+	return ;
+    }
+
+    unsigned int channels = m_audioFile->getChannels();
+    unsigned char b1, b2;
+
+    // We need the same amount of available data on every channel
+    size_t s = 0;
+    for (unsigned int ch = 0; ch < channels; ++ch) {
+	size_t available = m_ringBuffers[ch]->getReadSpace();
+#ifdef DEBUG_RECORDABLE
+
+	std::cerr << "RecordableAudioFile::write: " << available << " frames available to write on channel " << ch << std::endl;
+#endif
+
+	if (ch == 0 || available < s)
+	    s = available;
+    }
+    if (s == 0)
+	return ;
+
+    size_t bufferReqd = channels * s;
+    if (bufferReqd > bufferSize) {
+	if (buffer) {
+	    buffer = (sample_t *)realloc(buffer, bufferReqd * sizeof(sample_t));
+	    encodeBuffer = (char *)realloc(encodeBuffer, bufferReqd * 4);
+	} else {
+	    buffer = (sample_t *) malloc(bufferReqd * sizeof(sample_t));
+	    encodeBuffer = (char *)malloc(bufferReqd * 4);
+	}
+	bufferSize = bufferReqd;
+    }
+
+    for (unsigned int ch = 0; ch < channels; ++ch) {
+	m_ringBuffers[ch]->read(buffer + ch * s, s);
+    }
+
+    // interleave and convert
+
+    if (bits == 16) {
+	size_t index = 0;
+	for (size_t i = 0; i < s; ++i) {
+	    for (unsigned int ch = 0; ch < channels; ++ch) {
+		float sample = buffer[i + ch * s];
+		b2 = (unsigned char)((long)(sample * 32767.0) & 0xff);
+		b1 = (unsigned char)((long)(sample * 32767.0) >> 8);
+		encodeBuffer[index++] = b2;
+		encodeBuffer[index++] = b1;
+	    }
+	}
+    } else {
+	char *encodePointer = encodeBuffer;
+	for (size_t i = 0; i < s; ++i) {
+	    for (unsigned int ch = 0; ch < channels; ++ch) {
+		float sample = buffer[i + ch * s];
+		*(float *)encodePointer = sample;
+		encodePointer += sizeof(float);
+	    }
+	}
+    }
+
+#ifdef DEBUG_RECORDABLE
+    std::cerr << "RecordableAudioFile::write: writing " << s << " frames at " << channels << " channels and " << bits << " bits to file" << std::endl;
+#endif
+
+    m_audioFile->appendSamples(encodeBuffer, s);
+}
+
+}
+
diff --git a/src/sound/RecordableAudioFile.h b/src/sound/RecordableAudioFile.h
new file mode 100644
index 0000000..06df6f0
--- /dev/null
+++ b/src/sound/RecordableAudioFile.h
@@ -0,0 +1,68 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RECORDABLE_AUDIO_FILE_H_
+#define _RECORDABLE_AUDIO_FILE_H_
+
+#include "RingBuffer.h"
+#include "AudioFile.h"
+
+#include <vector>
+
+namespace Rosegarden
+{
+
+// A wrapper class for writing out a recording file.  We assume the
+// data is provided by a process thread and the writes are requested
+// by a disk thread.
+//
+class RecordableAudioFile
+{
+public:
+    typedef float sample_t;
+
+    typedef enum
+    {
+        IDLE,
+        RECORDING,
+        DEFUNCT
+    } RecordStatus;
+
+    RecordableAudioFile(AudioFile *audioFile, // should be already open for writing
+                        size_t bufferSize);
+    ~RecordableAudioFile();
+
+    void setStatus(const RecordStatus &status) { m_status = status; }
+    RecordStatus getStatus() const { return m_status; }
+
+    size_t buffer(const sample_t *data, int channel, size_t frames);
+    void write();
+
+protected:
+    AudioFile            *m_audioFile;
+    RecordStatus          m_status;
+
+    std::vector<RingBuffer<sample_t> *> m_ringBuffers; // one per channel
+};
+
+}
+
+#endif
diff --git a/src/sound/RingBuffer.h b/src/sound/RingBuffer.h
new file mode 100644
index 0000000..0cc5dc6
--- /dev/null
+++ b/src/sound/RingBuffer.h
@@ -0,0 +1,572 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RINGBUFFER_H_
+#define _RINGBUFFER_H_
+
+#include <sys/types.h>
+#include <sys/mman.h>
+
+#include "Scavenger.h"
+
+//#define DEBUG_RINGBUFFER 1
+//#define DEBUG_RINGBUFFER_CREATE_DESTROY 1
+
+#ifdef DEBUG_RINGBUFFER
+#define DEBUG_RINGBUFFER_CREATE_DESTROY 1
+#endif
+
+#ifdef DEBUG_RINGBUFFER_CREATE_DESTROY
+#include <iostream>
+static int __extant_ringbuffers = 0;
+#endif
+
+namespace Rosegarden {
+
+/**
+ * RingBuffer implements a lock-free ring buffer for one writer and N
+ * readers, that is to be used to store a sample type T.
+ *
+ * For efficiency, RingBuffer frequently initialises samples by
+ * writing zeroes into their memory space, so T should normally be a
+ * simple type that can safely be set to zero using memset.
+ */
+
+template <typename T, int N = 1>
+class RingBuffer
+{
+public:
+    /**
+     * Create a ring buffer with room to write n samples.
+     *
+     * Note that the internal storage size will actually be n+1
+     * samples, as one element is unavailable for administrative
+     * reasons.  Since the ring buffer performs best if its size is a
+     * power of two, this means n should ideally be some power of two
+     * minus one.
+     */
+    RingBuffer(size_t n);
+
+    virtual ~RingBuffer();
+
+    /**
+     * Return the total capacity of the ring buffer in samples.
+     * (This is the argument n passed to the constructor.)
+     */
+    size_t getSize() const;
+
+    /**
+     * Resize the ring buffer.  This also empties it.  Actually swaps
+     * in a new, larger buffer; the old buffer is scavenged after a
+     * seemly delay.  Should be called from the write thread.
+     */
+    void resize(size_t newSize);
+
+    /**
+     * Lock the ring buffer into physical memory.  Returns true
+     * for success.
+     */
+    bool mlock();
+
+    /**
+     * Unlock the ring buffer from physical memory.  Returns true for
+     * success.
+     */
+    bool munlock();
+
+    /**
+     * Reset read and write pointers, thus emptying the buffer.
+     * Should be called from the write thread.
+     */
+    void reset();
+
+    /**
+     * Return the amount of data available for reading by reader R, in
+     * samples.
+     */
+    size_t getReadSpace(int R = 0) const;
+
+    /**
+     * Return the amount of space available for writing, in samples.
+     */
+    size_t getWriteSpace() const;
+
+    /**
+     * Read n samples from the buffer, for reader R.  If fewer than n
+     * are available, the remainder will be zeroed out.  Returns the
+     * number of samples actually read.
+     */
+    size_t read(T *destination, size_t n, int R = 0);
+
+    /**
+     * Read n samples from the buffer, for reader R, adding them to
+     * the destination.  If fewer than n are available, the remainder
+     * will be left alone.  Returns the number of samples actually
+     * read.
+     */
+    size_t readAdding(T *destination, size_t n, int R = 0);
+
+    /**
+     * Read one sample from the buffer, for reader R.  If no sample is
+     * available, this will silently return zero.  Calling this
+     * repeatedly is obviously slower than calling read once, but it
+     * may be good enough if you don't want to allocate a buffer to
+     * read into.
+     */
+    T readOne(int R = 0);
+
+    /**
+     * Read n samples from the buffer, if available, for reader R,
+     * without advancing the read pointer -- i.e. a subsequent read()
+     * or skip() will be necessary to empty the buffer.  If fewer than
+     * n are available, the remainder will be zeroed out.  Returns the
+     * number of samples actually read.
+     */
+    size_t peek(T *destination, size_t n, int R = 0) const;
+
+    /**
+     * Read one sample from the buffer, if available, without
+     * advancing the read pointer -- i.e. a subsequent read() or
+     * skip() will be necessary to empty the buffer.  Returns zero if
+     * no sample was available.
+     */
+    T peek(int R = 0) const;
+
+    /**
+     * Pretend to read n samples from the buffer, for reader R,
+     * without actually returning them (i.e. discard the next n
+     * samples).  Returns the number of samples actually available for
+     * discarding.
+     */
+    size_t skip(size_t n, int R = 0);
+
+    /**
+     * Write n samples to the buffer.  If insufficient space is
+     * available, not all samples may actually be written.  Returns
+     * the number of samples actually written.
+     */
+    size_t write(const T *source, size_t n);
+
+    /**
+     * Write n zero-value samples to the buffer.  If insufficient
+     * space is available, not all zeros may actually be written.
+     * Returns the number of zeroes actually written.
+     */
+    size_t zero(size_t n);
+
+protected:
+    T               *m_buffer;
+    volatile size_t  m_writer;
+    volatile size_t  m_readers[N];
+    size_t           m_size;
+    bool             m_mlocked;
+
+    static Scavenger<ScavengerArrayWrapper<T> > m_scavenger;
+
+private:
+    RingBuffer(const RingBuffer &); // not provided
+    RingBuffer &operator=(const RingBuffer &); // not provided
+};
+
+template <typename T, int N>
+Scavenger<ScavengerArrayWrapper<T> > RingBuffer<T, N>::m_scavenger;
+
+template <typename T, int N>
+RingBuffer<T, N>::RingBuffer(size_t n) :
+    m_buffer(new T[n + 1]),
+    m_writer(0),
+    m_size(n + 1),
+    m_mlocked(false)
+{
+#ifdef DEBUG_RINGBUFFER_CREATE_DESTROY
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::RingBuffer(" << n << ") [now have " << (++__extant_ringbuffers) << "]" << std::endl;
+#endif
+
+    for (int i = 0; i < N; ++i) m_readers[i] = 0;
+
+    m_scavenger.scavenge();
+}
+
+template <typename T, int N>
+RingBuffer<T, N>::~RingBuffer()
+{
+#ifdef DEBUG_RINGBUFFER_CREATE_DESTROY
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::~RingBuffer [now have " << (--__extant_ringbuffers) << "]" << std::endl;
+#endif
+
+    if (m_mlocked) {
+        ::munlock((void *)m_buffer, m_size * sizeof(T));
+    }
+    delete[] m_buffer;
+
+    m_scavenger.scavenge();
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::getSize() const
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::getSize(): " << m_size-1 << std::endl;
+#endif
+
+    return m_size - 1;
+}
+
+template <typename T, int N>
+void
+RingBuffer<T, N>::resize(size_t newSize)
+{
+#ifdef DEBUG_RINGBUFFER_CREATE_DESTROY
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::resize(" << newSize << ")" << std::endl;
+#endif
+
+    m_scavenger.scavenge();
+
+    if (m_mlocked) {
+        ::munlock((void *)m_buffer, m_size * sizeof(T));
+    }
+
+    m_scavenger.claim(new ScavengerArrayWrapper<T>(m_buffer));
+
+    reset();
+    m_buffer = new T[newSize + 1];
+    m_size = newSize + 1;
+
+    if (m_mlocked) {
+        if (::mlock((void *)m_buffer, m_size * sizeof(T))) {
+            m_mlocked = false;
+        }
+    }
+}
+
+template <typename T, int N>
+bool
+RingBuffer<T, N>::mlock()
+{
+    if (::mlock((void *)m_buffer, m_size * sizeof(T))) return false;
+    m_mlocked = true;
+    return true;
+}
+
+template <typename T, int N>
+bool
+RingBuffer<T, N>::munlock()
+{
+    if (::munlock((void *)m_buffer, m_size * sizeof(T))) return false;
+    m_mlocked = false;
+    return true;
+}
+
+template <typename T, int N>
+void
+RingBuffer<T, N>::reset()
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::reset" << std::endl;
+#endif
+
+    m_writer = 0;
+    for (int i = 0; i < N; ++i) m_readers[i] = 0;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::getReadSpace(int R) const
+{
+    size_t writer = m_writer;
+    size_t reader = m_readers[R];
+    size_t space = 0;
+
+    if (writer > reader) space = writer - reader;
+    else space = ((writer + m_size) - reader) % m_size;
+
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::getReadSpace(" << R << "): " << space << std::endl;
+#endif
+
+    return space;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::getWriteSpace() const
+{
+    size_t space = 0;
+    for (int i = 0; i < N; ++i) {
+        size_t here = (m_readers[i] + m_size - m_writer - 1) % m_size;
+        if (i == 0 || here < space) space = here;
+    }
+
+#ifdef DEBUG_RINGBUFFER
+    size_t rs(getReadSpace()), rp(m_readers[0]);
+
+    std::cerr << "RingBuffer: write space " << space << ", read space "
+              << rs << ", total " << (space + rs) << ", m_size " << m_size << std::endl;
+    std::cerr << "RingBuffer: reader " << rp << ", writer " << m_writer << std::endl;
+#endif
+
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::getWriteSpace(): " << space << std::endl;
+#endif
+
+    return space;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::read(T *destination, size_t n, int R)
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::read(dest, " << n << ", " << R << ")" << std::endl;
+#endif
+
+    size_t available = getReadSpace(R);
+    if (n > available) {
+#ifdef DEBUG_RINGBUFFER
+        std::cerr << "WARNING: Only " << available << " samples available"
+                  << std::endl;
+#endif
+        memset(destination + available, 0, (n - available) * sizeof(T));
+        n = available;
+    }
+    if (n == 0) return n;
+
+    size_t here = m_size - m_readers[R];
+    if (here >= n) {
+        memcpy(destination, m_buffer + m_readers[R], n * sizeof(T));
+    } else {
+        memcpy(destination, m_buffer + m_readers[R], here * sizeof(T));
+        memcpy(destination + here, m_buffer, (n - here) * sizeof(T));
+    }
+
+    m_readers[R] = (m_readers[R] + n) % m_size;
+
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::read: read " << n << ", reader now " << m_readers[R] << std::endl;
+#endif
+
+    return n;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::readAdding(T *destination, size_t n, int R)
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::readAdding(dest, " << n << ", " << R << ")" << std::endl;
+#endif
+
+    size_t available = getReadSpace(R);
+    if (n > available) {
+#ifdef DEBUG_RINGBUFFER
+        std::cerr << "WARNING: Only " << available << " samples available"
+                  << std::endl;
+#endif
+        n = available;
+    }
+    if (n == 0) return n;
+
+    size_t here = m_size - m_readers[R];
+
+    if (here >= n) {
+        for (size_t i = 0; i < n; ++i) {
+            destination[i] += (m_buffer + m_readers[R])[i];
+        }
+    } else {
+        for (size_t i = 0; i < here; ++i) {
+            destination[i] += (m_buffer + m_readers[R])[i];
+        }
+        for (size_t i = 0; i < (n - here); ++i) {
+            destination[i + here] += m_buffer[i];
+        }
+    }
+
+    m_readers[R] = (m_readers[R] + n) % m_size;
+    return n;
+}
+
+template <typename T, int N>
+T
+RingBuffer<T, N>::readOne(int R)
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::readOne(" << R << ")" << std::endl;
+#endif
+
+    if (m_writer == m_readers[R]) {
+#ifdef DEBUG_RINGBUFFER
+        std::cerr << "WARNING: No sample available"
+                  << std::endl;
+#endif
+        T t;
+        memset(&t, 0, sizeof(T));
+        return t;
+    }
+    T value = m_buffer[m_readers[R]];
+    if (++m_readers[R] == m_size) m_readers[R] = 0;
+    return value;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::peek(T *destination, size_t n, int R) const
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::peek(dest, " << n << ", " << R << ")" << std::endl;
+#endif
+
+    size_t available = getReadSpace(R);
+    if (n > available) {
+#ifdef DEBUG_RINGBUFFER
+	std::cerr << "WARNING: Only " << available << " samples available"
+		  << std::endl;
+#endif
+	memset(destination + available, 0, (n - available) * sizeof(T));
+	n = available;
+    }
+    if (n == 0) return n;
+
+    size_t here = m_size - m_readers[R];
+    if (here >= n) {
+	memcpy(destination, m_buffer + m_readers[R], n * sizeof(T));
+    } else {
+	memcpy(destination, m_buffer + m_readers[R], here * sizeof(T));
+	memcpy(destination + here, m_buffer, (n - here) * sizeof(T));
+    }
+
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::peek: read " << n << std::endl;
+#endif
+
+    return n;
+}
+
+template <typename T, int N>
+T
+RingBuffer<T, N>::peek(int R) const
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::peek(" << R << ")" << std::endl;
+#endif
+
+    if (m_writer == m_readers[R]) {
+#ifdef DEBUG_RINGBUFFER
+        std::cerr << "WARNING: No sample available"
+                  << std::endl;
+#endif
+        T t;
+        memset(&t, 0, sizeof(T));
+        return t;
+    }
+    T value = m_buffer[m_readers[R]];
+    return value;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::skip(size_t n, int R)
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::skip(" << n << ", " << R << ")" << std::endl;
+#endif
+
+    size_t available = getReadSpace(R);
+    if (n > available) {
+#ifdef DEBUG_RINGBUFFER
+        std::cerr << "WARNING: Only " << available << " samples available"
+                  << std::endl;
+#endif
+        n = available;
+    }
+    if (n == 0) return n;
+    m_readers[R] = (m_readers[R] + n) % m_size;
+    return n;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::write(const T *source, size_t n)
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::write(" << n << ")" << std::endl;
+#endif
+
+    size_t available = getWriteSpace();
+    if (n > available) {
+#ifdef DEBUG_RINGBUFFER
+        std::cerr << "WARNING: Only room for " << available << " samples"
+                  << std::endl;
+#endif
+        n = available;
+    }
+    if (n == 0) return n;
+
+    size_t here = m_size - m_writer;
+    if (here >= n) {
+        memcpy(m_buffer + m_writer, source, n * sizeof(T));
+    } else {
+        memcpy(m_buffer + m_writer, source, here * sizeof(T));
+        memcpy(m_buffer, source + here, (n - here) * sizeof(T));
+    }
+
+    m_writer = (m_writer + n) % m_size;
+
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::write: wrote " << n << ", writer now " << m_writer << std::endl;
+#endif
+
+    return n;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::zero(size_t n)
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::zero(" << n << ")" << std::endl;
+#endif
+
+    size_t available = getWriteSpace();
+    if (n > available) {
+#ifdef DEBUG_RINGBUFFER
+        std::cerr << "WARNING: Only room for " << available << " samples"
+                  << std::endl;
+#endif
+        n = available;
+    }
+    if (n == 0) return n;
+
+    size_t here = m_size - m_writer;
+    if (here >= n) {
+        memset(m_buffer + m_writer, 0, n * sizeof(T));
+    } else {
+        memset(m_buffer + m_writer, 0, here * sizeof(T));
+        memset(m_buffer, 0, (n - here) * sizeof(T));
+    }
+
+    m_writer = (m_writer + n) % m_size;
+    return n;
+}
+
+}
+
+#endif // _RINGBUFFER_H_
diff --git a/src/sound/RosegardenMidiRecord.mcopclass b/src/sound/RosegardenMidiRecord.mcopclass
new file mode 100644
index 0000000..41593f1
--- /dev/null
+++ b/src/sound/RosegardenMidiRecord.mcopclass
@@ -0,0 +1,5 @@
+Interface=RosegardenMidiRecord, Arts::MidiPort, Arts::Object
+Library=libRosegardenSequencer.la
+Language=C++
+Author="Richard Bown <bownie@bownie.com>", "Guillaume Laurent <glaurent@telegraph-road.org>", "Chris Cannam <cannam@all-day-breakfast.com>"
+URL="http://home"
diff --git a/src/sound/RunnablePluginInstance.cpp b/src/sound/RunnablePluginInstance.cpp
new file mode 100644
index 0000000..820aaf9
--- /dev/null
+++ b/src/sound/RunnablePluginInstance.cpp
@@ -0,0 +1,42 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "RunnablePluginInstance.h"
+#include "PluginFactory.h"
+
+#include <iostream>
+
+namespace Rosegarden
+{
+
+RunnablePluginInstance::~RunnablePluginInstance()
+{
+//    std::cerr << "RunnablePluginInstance::~RunnablePluginInstance" << std::endl;
+
+    if (m_factory) {
+//        std::cerr << "Asking factory to release " << m_identifier << std::endl;
+
+        m_factory->releasePlugin(this, m_identifier);
+    }
+}
+
+}
+
diff --git a/src/sound/RunnablePluginInstance.h b/src/sound/RunnablePluginInstance.h
new file mode 100644
index 0000000..f15f146
--- /dev/null
+++ b/src/sound/RunnablePluginInstance.h
@@ -0,0 +1,114 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RUNNABLE_PLUGIN_INSTANCE_H_
+#define _RUNNABLE_PLUGIN_INSTANCE_H_
+
+#include <qstring.h>
+#include <qstringlist.h>
+#include <vector>
+
+#include "RealTime.h"
+
+namespace Rosegarden
+{
+
+class PluginFactory;
+        
+/**
+ * RunnablePluginInstance is a very trivial interface that an audio
+ * process can use to refer to an instance of a plugin without needing
+ * to know what type of plugin it is.
+ *
+ * The audio code calls run() on an instance that has been passed to
+ * it, and assumes that the passing code has already initialised the
+ * plugin, connected its inputs and outputs and so on, and that there
+ * is an understanding in place about the sizes of the buffers in use
+ * by the plugin.  All of this depends on the subclass implementation.
+ */
+
+class RunnablePluginInstance
+{
+public:
+    typedef float sample_t;
+
+    virtual ~RunnablePluginInstance();
+
+    virtual bool isOK() const = 0;
+
+    virtual QString getIdentifier() const = 0;
+
+    /**
+     * Run for one block, starting at the given time.  The start time
+     * may be of interest to synths etc that may have queued events
+     * waiting.  Other plugins can ignore it.
+     */
+    virtual void run(const RealTime &blockStartTime) = 0;
+    
+    virtual size_t getBufferSize() = 0;
+
+    virtual size_t getAudioInputCount() = 0;
+    virtual size_t getAudioOutputCount() = 0;
+
+    virtual sample_t **getAudioInputBuffers() = 0;
+    virtual sample_t **getAudioOutputBuffers() = 0;
+
+    virtual QStringList getPrograms() { return QStringList(); }
+    virtual QString getCurrentProgram() { return QString(); }
+    virtual QString getProgram(int /* bank */, int /* program */) { return QString(); }
+    virtual unsigned long getProgram(QString /* name */) { return 0; } // bank << 16 + program
+    virtual void selectProgram(QString) { }
+
+    virtual void setPortValue(unsigned int port, float value) = 0;
+    virtual float getPortValue(unsigned int port) = 0;
+
+    virtual QString configure(QString /* key */, QString /* value */) { return QString(); }
+
+    virtual void sendEvent(const RealTime & /* eventTime */,
+                           const void * /* event */) { }
+
+    virtual bool isBypassed() const = 0;
+    virtual void setBypassed(bool value) = 0;
+
+    // This should be called after setup, but while not actually playing.
+    virtual size_t getLatency() = 0;
+
+    virtual void silence() = 0;
+    virtual void discardEvents() { }
+    virtual void setIdealChannelCount(size_t channels) = 0; // must also silence(); may also re-instantiate
+
+    void setFactory(PluginFactory *f) { m_factory = f; } // ew
+
+protected:
+    RunnablePluginInstance(PluginFactory *factory, QString identifier) :
+        m_factory(factory), m_identifier(identifier) { }
+
+    PluginFactory *m_factory;
+    QString m_identifier;
+
+    friend class PluginFactory;
+};
+
+typedef std::vector<RunnablePluginInstance *> RunnablePluginInstances;
+
+}
+
+#endif
diff --git a/src/sound/SF2PatchExtractor.cpp b/src/sound/SF2PatchExtractor.cpp
new file mode 100644
index 0000000..6ba8dc5
--- /dev/null
+++ b/src/sound/SF2PatchExtractor.cpp
@@ -0,0 +1,217 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "SF2PatchExtractor.h"
+
+#include <iostream>
+#include <fstream>
+#include <string>
+#include <map>
+#include <sys/types.h>
+
+namespace Rosegarden
+{
+
+using std::string;
+using std::cerr;
+using std::endl;
+using std::ifstream;
+using std::ios;
+
+
+struct Chunk
+{
+    char id[4];
+    u_int32_t size;
+
+    Chunk(ifstream *, bool idOnly = false);
+    bool isa(std::string s);
+};
+
+Chunk::Chunk(ifstream *file, bool idOnly)
+{
+    file->read((char *)this->id, 4);
+    size = 0;
+
+    if (idOnly)
+        return ;
+
+    unsigned char sz[4];
+    file->read((char *)sz, 4);
+    for (int i = 0; i < 4; ++i)
+        size += sz[i] << (i * 8);
+}
+
+bool
+Chunk::isa(string s)
+{
+    return string(id, 4) == s;
+}
+
+bool
+SF2PatchExtractor::isSF2File(string fileName)
+{
+    ifstream *file = new ifstream(fileName.c_str(), ios::in | ios::binary);
+    if (!file)
+        throw FileNotFoundException();
+
+    Chunk riffchunk(file);
+    if (!riffchunk.isa("RIFF")) {
+        file->close();
+        return false;
+    }
+
+    Chunk sfbkchunk(file, true);
+    if (!sfbkchunk.isa("sfbk")) {
+        file->close();
+        return false;
+    }
+
+    file->close();
+    return true;
+}
+
+SF2PatchExtractor::Device
+SF2PatchExtractor::read(string fileName)
+{
+    Device device;
+
+    ifstream *file = new ifstream(fileName.c_str(), ios::in | ios::binary);
+    if (!file)
+        throw FileNotFoundException();
+
+    Chunk riffchunk(file);
+    if (!riffchunk.isa("RIFF")) {
+        file->close();
+        throw WrongFileFormatException();
+    }
+
+    Chunk sfbkchunk(file, true);
+    if (!sfbkchunk.isa("sfbk")) {
+        file->close();
+        throw WrongFileFormatException();
+    }
+
+    while (!file->eof()) {
+
+        Chunk chunk(file);
+
+        if (!chunk.isa("LIST")) {
+            // cerr << "Skipping " << string(chunk.id, 4) << endl;
+            file->seekg(chunk.size, ios::cur);
+            continue;
+        }
+
+        Chunk listchunk(file, true);
+        if (!listchunk.isa("pdta")) {
+            // cerr << "Skipping " << string(id, 4) << endl;
+            file->seekg(chunk.size - 4, ios::cur);
+            continue;
+        }
+
+        int size = chunk.size - 4;
+        while (size > 0) {
+
+            Chunk pdtachunk(file);
+            size -= 8 + pdtachunk.size;
+            if (file->eof()) {
+                break;
+            }
+
+            if (!pdtachunk.isa("phdr")) { // preset header
+                // cerr << "Skipping " << string(pdtachunk.id, 4) << endl;
+                file->seekg(pdtachunk.size, ios::cur);
+                continue;
+            }
+
+            int presets = pdtachunk.size / 38;
+            for (int i = 0; i < presets; ++i) {
+
+                char name[21];
+                u_int16_t bank, program;
+
+                file->read((char *)name, 20);
+                name[20] = '\0';
+                file->read((char *)&program, 2);
+                file->read((char *)&bank, 2);
+
+                // cerr << "Read name as " << name << endl;
+
+                file->seekg(14, ios::cur);
+
+                if (i == presets - 1 &&
+                        bank == 255 &&
+                        program == 255 &&
+                        string(name) == "EOP")
+                    continue;
+
+                device[bank][program] = name;
+            }
+        }
+    }
+
+    file->close();
+    return device;
+}
+
+}
+
+
+#ifdef TEST_SF2_PATCH_EXTRACTOR
+
+int main(int argc, char **argv)
+{
+    using SF2PatchExtractor;
+
+    if (argc != 2) {
+        std::cerr << "Usage: " << argv[0] << " sf2filename" << std::endl;
+        return 2;
+    }
+
+    try {
+        SF2PatchExtractor::Device device =
+            SF2PatchExtractor::read(argv[1]);
+
+        std::cerr << "Done.  Presets are:" << std::endl;
+
+        for (SF2PatchExtractor::Device::iterator di = device.begin();
+                di != device.end(); ++di) {
+
+            std::cerr << "Bank " << di->first << ":" << std::endl;
+
+            for (SF2PatchExtractor::Bank::iterator bi = di->second.begin();
+                    bi != di->second.end();
+                    ++bi) {
+
+                std::cerr << "Program " << bi->first << ": \"" << bi->second
+                << "\"" << std::endl;
+            }
+        }
+    } catch (SF2PatchExtractor::WrongFileFormatException) {
+        std::cerr << "Wrong file format" << std::endl;
+    } catch (SF2PatchExtractor::FileNotFoundException) {
+        std::cerr << "File not found or couldn't be opened" << std::endl;
+    }
+
+    return 0;
+}
+
+#endif
diff --git a/src/sound/SF2PatchExtractor.h b/src/sound/SF2PatchExtractor.h
new file mode 100644
index 0000000..a9d5453
--- /dev/null
+++ b/src/sound/SF2PatchExtractor.h
@@ -0,0 +1,58 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _SF2_PATCH_EXTRACTOR_H_
+#define _SF2_PATCH_EXTRACTOR_H_
+
+#include <string>
+#include <map>
+
+namespace Rosegarden {
+
+/**
+ * Trivial class to suck the patch map out of a .sf2 SoundFont file.
+ * Inspired by (but not based on) sftovkb by Takashi Iwai.
+ * 
+ * SoundFont is a straightforward RIFF format so there's some
+ * redundancy between this and RIFFAudioFile -- we don't take any
+ * advantage of that, and this class is completely self-contained.
+ *
+ * Tolerates garbled files; will just suck all it can rather than
+ * throw an error, except if the file is not a SoundFont at all.
+ */
+
+class SF2PatchExtractor
+{
+public:
+    typedef std::map<int, std::string> Bank;
+    typedef std::map<int, Bank> Device;
+
+    struct FileNotFoundException { };
+    struct WrongFileFormatException { };
+
+    static bool isSF2File(std::string fileName);
+    static Device read(std::string fileName);
+};
+
+}
+
+#endif
+
diff --git a/src/sound/SampleWindow.h b/src/sound/SampleWindow.h
new file mode 100644
index 0000000..88400f7
--- /dev/null
+++ b/src/sound/SampleWindow.h
@@ -0,0 +1,192 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rosegarden
+    A MIDI and audio sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <richard.bown@ferventsoftware.com>
+
+    The moral rights of Guillaume Laurent, Chris Cannam, and Richard
+    Bown to claim authorship of this work have been asserted.
+
+    Other copyrights also apply to some parts of this work.  Please
+    see the AUTHORS file and individual file headers for details.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+/*
+    This file is derived from
+
+    Sonic Visualiser
+    An audio file viewer and annotation editor.
+    Centre for Digital Music, Queen Mary, University of London.
+    This file copyright 2006 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _SAMPLE_WINDOW_H_
+#define _SAMPLE_WINDOW_H_
+
+#include <cmath>
+#include <cstdlib>
+#include <iostream>
+#include <map>
+
+namespace Rosegarden 
+{
+
+template <typename T>
+class SampleWindow
+{
+public:
+    enum Type {
+        Rectangular,
+        Bartlett,
+        Hamming,
+        Hanning,
+        Blackman,
+        Gaussian,
+        Parzen,
+        Nuttall,
+        BlackmanHarris
+    };
+
+    /**
+     * Construct a windower of the given type.
+     */
+    SampleWindow(Type type, size_t size) : m_type(type), m_size(size) { encache(); }
+    SampleWindow(const SampleWindow &w) : m_type(w.m_type), m_size(w.m_size) { encache(); }
+    SampleWindow &operator=(const SampleWindow &w) {
+	if (&w == this) return *this;
+	m_type = w.m_type;
+	m_size = w.m_size;
+	encache();
+	return *this;
+    }
+    virtual ~SampleWindow() { delete[] m_cache; }
+    
+    void cut(T *src) const { cut(src, src); }
+    void cut(T *src, T *dst) const {
+	for (size_t i = 0; i < m_size; ++i) dst[i] = src[i] * m_cache[i];
+    }
+
+    T getArea() { return m_area; }
+    T getValue(size_t i) { return m_cache[i]; }
+
+    Type getType() const { return m_type; }
+    size_t getSize() const { return m_size; }
+
+protected:
+    Type m_type;
+    size_t m_size;
+    T *m_cache;
+    T m_area;
+    
+    void encache();
+    void cosinewin(T *, T, T, T, T);
+};
+
+template <typename T>
+void SampleWindow<T>::encache()
+{
+    int n = int(m_size);
+    T *mult = new T[n];
+    int i;
+    for (i = 0; i < n; ++i) mult[i] = 1.0;
+
+    switch (m_type) {
+		
+    case Rectangular:
+	for (i = 0; i < n; ++i) {
+	    mult[i] *= 0.5;
+	}
+	break;
+	    
+    case Bartlett:
+	for (i = 0; i < n/2; ++i) {
+	    mult[i] *= (i / T(n/2));
+	    mult[i + n/2] *= (1.0 - (i / T(n/2)));
+	}
+	break;
+	    
+    case Hamming:
+        cosinewin(mult, 0.54, 0.46, 0.0, 0.0);
+	break;
+	    
+    case Hanning:
+        cosinewin(mult, 0.50, 0.50, 0.0, 0.0);
+	break;
+	    
+    case Blackman:
+        cosinewin(mult, 0.42, 0.50, 0.08, 0.0);
+	break;
+	    
+    case Gaussian:
+	for (i = 0; i < n; ++i) {
+            mult[i] *= pow(2, - pow((i - (n-1)/2.0) / ((n-1)/2.0 / 3), 2));
+	}
+	break;
+	    
+    case Parzen:
+    {
+        int N = n-1;
+        for (i = 0; i < N/4; ++i) {
+            T m = 2 * pow(1.0 - (T(N)/2 - i) / (T(N)/2), 3);
+            mult[i] *= m;
+            mult[N-i] *= m;
+        }
+        for (i = N/4; i <= N/2; ++i) {
+            int wn = i - N/2;
+            T m = 1.0 - 6 * pow(wn / (T(N)/2), 2) * (1.0 - abs(wn) / (T(N)/2));
+            mult[i] *= m;
+            mult[N-i] *= m;
+        }            
+        break;
+    }
+
+    case Nuttall:
+        cosinewin(mult, 0.3635819, 0.4891775, 0.1365995, 0.0106411);
+	break;
+
+    case BlackmanHarris:
+        cosinewin(mult, 0.35875, 0.48829, 0.14128, 0.01168);
+        break;
+    }
+	
+    m_cache = mult;
+
+    m_area = 0;
+    for (int i = 0; i < n; ++i) {
+        m_area += m_cache[i];
+    }
+    m_area /= n;
+}
+
+template <typename T>
+void SampleWindow<T>::cosinewin(T *mult, T a0, T a1, T a2, T a3)
+{
+    int n = int(m_size);
+    for (int i = 0; i < n; ++i) {
+        mult[i] *= (a0
+                    - a1 * cos(2 * M_PI * i / n)
+                    + a2 * cos(4 * M_PI * i / n)
+                    - a3 * cos(6 * M_PI * i / n));
+    }
+}
+
+}
+
+#endif
diff --git a/src/sound/Scavenger.h b/src/sound/Scavenger.h
new file mode 100644
index 0000000..b27e848
--- /dev/null
+++ b/src/sound/Scavenger.h
@@ -0,0 +1,211 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _SCAVENGER_H_
+#define _SCAVENGER_H_
+
+#include <vector>
+#include <list>
+#include <sys/time.h>
+#include <pthread.h>
+#include <iostream>
+
+namespace Rosegarden
+{
+
+/**
+ * A very simple class that facilitates running things like plugins
+ * without locking, by collecting unwanted objects and deleting them
+ * after a delay so as to be sure nobody's in the middle of using
+ * them.  Requires scavenge() to be called regularly from a non-RT
+ * thread.
+ *
+ * This is currently not at all suitable for large numbers of objects
+ * -- it's just a quick hack for use with things like plugins.
+ */
+
+template <typename T>
+class Scavenger
+{
+public:
+    Scavenger(int sec = 2, int defaultObjectListSize = 200);
+    ~Scavenger();
+
+    /**
+     * Call from an RT thread etc., to pass ownership of t to us for
+     * later disposal.  Only one thread should be calling this on any
+     * given scavenger.
+     * 
+     * This is only lock-free so long as a slot is available in the
+     * object list; otherwise it takes a lock and allocates memory.
+     * Scavengers should always be used with an object list size
+     * sufficient to ensure that enough slots are always available in
+     * normal use.
+     */
+    void claim(T *t);
+
+    /**
+     * Call from a non-RT thread.
+     * Only one thread should be calling this on any given scavenger.
+     */
+    void scavenge();
+
+protected:
+    typedef std::pair<T *, int> ObjectTimePair;
+    typedef std::vector<ObjectTimePair> ObjectTimeList;
+    ObjectTimeList m_objects;
+    int m_sec;
+
+    typedef std::list<T *> ObjectList;
+    ObjectList m_excess;
+    int m_lastExcess;
+    pthread_mutex_t m_excessMutex;
+    void pushExcess(T *);
+    void clearExcess(int);
+
+    unsigned int m_claimed;
+    unsigned int m_scavenged;
+};
+
+/**
+ * A wrapper to permit arrays to be scavenged.
+ */
+
+template <typename T>
+class ScavengerArrayWrapper
+{
+public:
+    ScavengerArrayWrapper(T *array) : m_array(array) { }
+    ~ScavengerArrayWrapper() { delete[] m_array; }
+
+private:
+    T *m_array;
+};
+
+
+template <typename T>
+Scavenger<T>::Scavenger(int sec, int defaultObjectListSize) :
+    m_objects(ObjectTimeList(defaultObjectListSize)),
+    m_sec(sec),
+    m_lastExcess(0),
+    m_claimed(0),
+    m_scavenged(0)
+{
+    pthread_mutex_init(&m_excessMutex, NULL);
+}
+
+template <typename T>
+Scavenger<T>::~Scavenger()
+{
+    if (m_scavenged < m_claimed) {
+        for (size_t i = 0; i < m_objects.size(); ++i) {
+            ObjectTimePair &pair = m_objects[i];
+            if (pair.first != 0) {
+                T *ot = pair.first;
+                pair.first = 0;
+                delete ot;
+                ++m_scavenged;
+            }
+        }
+    }
+
+    clearExcess(0);
+
+    pthread_mutex_destroy(&m_excessMutex);
+}
+
+template <typename T>
+void
+Scavenger<T>::claim(T *t)
+{
+    struct timeval tv;
+    (void)gettimeofday(&tv, 0);
+    int sec = tv.tv_sec;
+
+    for (size_t i = 0; i < m_objects.size(); ++i) {
+        ObjectTimePair &pair = m_objects[i];
+        if (pair.first == 0) {
+            pair.second = sec;
+            pair.first = t;
+            ++m_claimed;
+            return;
+        }
+    }
+
+    std::cerr << "WARNING: Scavenger::claim(" << t << "): run out of slots, "
+              << "using non-RT-safe method" << std::endl;
+    pushExcess(t);
+}
+
+template <typename T>
+void
+Scavenger<T>::scavenge()
+{
+    if (m_scavenged >= m_claimed) return;
+    
+    struct timeval tv;
+    (void)gettimeofday(&tv, 0);
+    int sec = tv.tv_sec;
+
+    for (size_t i = 0; i < m_objects.size(); ++i) {
+        ObjectTimePair &pair = m_objects[i];
+        if (pair.first != 0 && pair.second + m_sec < sec) {
+            T *ot = pair.first;
+            pair.first = 0;
+            delete ot;
+            ++m_scavenged;
+        }
+    }
+
+    if (sec > m_lastExcess + m_sec) {
+        clearExcess(sec);
+    }
+}
+
+template <typename T>
+void
+Scavenger<T>::pushExcess(T *t)
+{
+    pthread_mutex_lock(&m_excessMutex);
+    m_excess.push_back(t);
+    struct timeval tv;
+    (void)gettimeofday(&tv, 0);
+    m_lastExcess = tv.tv_sec;
+    pthread_mutex_unlock(&m_excessMutex);
+}
+
+template <typename T>
+void
+Scavenger<T>::clearExcess(int sec)
+{
+    pthread_mutex_lock(&m_excessMutex);
+    for (typename ObjectList::iterator i = m_excess.begin();
+         i != m_excess.end(); ++i) {
+        delete *i;
+    }
+    m_excess.clear();
+    m_lastExcess = sec;
+    pthread_mutex_unlock(&m_excessMutex);
+}
+
+}
+
+#endif
diff --git a/src/sound/SequencerDataBlock.cpp b/src/sound/SequencerDataBlock.cpp
new file mode 100644
index 0000000..bc5e80b
--- /dev/null
+++ b/src/sound/SequencerDataBlock.cpp
@@ -0,0 +1,361 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "SequencerDataBlock.h"
+#include "MappedComposition.h"
+
+namespace Rosegarden
+{
+
+SequencerDataBlock::SequencerDataBlock(bool initialise)
+{
+    if (initialise)
+        clearTemporaries();
+}
+
+bool
+SequencerDataBlock::getVisual(MappedEvent &ev) const
+{
+    static int eventIndex = 0;
+
+    if (!m_haveVisualEvent) {
+        return false;
+    } else {
+        int thisEventIndex = m_visualEventIndex;
+        if (thisEventIndex == eventIndex)
+            return false;
+        ev = *((MappedEvent *) & m_visualEvent);
+        eventIndex = thisEventIndex;
+        return true;
+    }
+}
+
+void
+SequencerDataBlock::setVisual(const MappedEvent *ev)
+{
+    m_haveVisualEvent = false;
+    if (ev) {
+        *((MappedEvent *)&m_visualEvent) = *ev;
+        ++m_visualEventIndex;
+        m_haveVisualEvent = true;
+    }
+}
+
+int
+SequencerDataBlock::getRecordedEvents(MappedComposition &mC) const
+{
+    static int readIndex = -1;
+
+    if (readIndex == -1) {
+        readIndex = m_recordEventIndex;
+        return 0;
+    }
+
+    int currentIndex = m_recordEventIndex;
+    int count = 0;
+
+    MappedEvent *recordBuffer = (MappedEvent *)m_recordBuffer;
+
+    while (readIndex != currentIndex) {
+        mC.insert(new MappedEvent(recordBuffer[readIndex]));
+        if (++readIndex == SEQUENCER_DATABLOCK_RECORD_BUFFER_SIZE)
+            readIndex = 0;
+        ++count;
+    }
+
+    return count;
+}
+
+void
+SequencerDataBlock::addRecordedEvents(MappedComposition *mC)
+{
+    // ringbuffer
+    int index = m_recordEventIndex;
+    MappedEvent *recordBuffer = (MappedEvent *)m_recordBuffer;
+
+    for (MappedComposition::iterator i = mC->begin(); i != mC->end(); ++i) {
+        recordBuffer[index] = **i;
+        if (++index == SEQUENCER_DATABLOCK_RECORD_BUFFER_SIZE)
+            index = 0;
+    }
+
+    m_recordEventIndex = index;
+}
+
+int
+SequencerDataBlock::instrumentToIndex(InstrumentId id) const
+{
+    int i;
+
+    for (i = 0; i < m_knownInstrumentCount; ++i) {
+        if (m_knownInstruments[i] == id)
+            return i;
+    }
+
+    return -1;
+}
+
+int
+SequencerDataBlock::instrumentToIndexCreating(InstrumentId id)
+{
+    int i;
+
+    for (i = 0; i < m_knownInstrumentCount; ++i) {
+        if (m_knownInstruments[i] == id)
+            return i;
+    }
+
+    if (i == SEQUENCER_DATABLOCK_MAX_NB_INSTRUMENTS) {
+        std::cerr << "ERROR: SequencerDataBlock::instrumentToIndexCreating("
+        << id << "): out of instrument index space" << std::endl;
+        return -1;
+    }
+
+    m_knownInstruments[i] = id;
+    ++m_knownInstrumentCount;
+    return i;
+}
+
+bool
+SequencerDataBlock::getInstrumentLevel(InstrumentId id,
+                                       LevelInfo &info) const
+{
+    static int lastUpdateIndex[SEQUENCER_DATABLOCK_MAX_NB_INSTRUMENTS];
+
+    int index = instrumentToIndex(id);
+    if (index < 0) {
+        info.level = info.levelRight = 0;
+        return false;
+    }
+
+    int currentUpdateIndex = m_levelUpdateIndices[index];
+    info = m_levels[index];
+
+    /*
+    std::cout << "SequencerDataBlock::getInstrumentLevel - "
+              << "id = " << id
+              << ", level = " << info.level << std::endl;
+              */
+
+    if (lastUpdateIndex[index] != currentUpdateIndex) {
+        lastUpdateIndex[index] = currentUpdateIndex;
+        return true;
+    } else {
+        return false; // no change
+    }
+}
+
+bool
+SequencerDataBlock::getInstrumentLevelForMixer(InstrumentId id,
+        LevelInfo &info) const
+{
+    static int lastUpdateIndex[SEQUENCER_DATABLOCK_MAX_NB_INSTRUMENTS];
+
+    int index = instrumentToIndex(id);
+    if (index < 0) {
+        info.level = info.levelRight = 0;
+        return false;
+    }
+
+    int currentUpdateIndex = m_levelUpdateIndices[index];
+    info = m_levels[index];
+
+    if (lastUpdateIndex[index] != currentUpdateIndex) {
+        lastUpdateIndex[index] = currentUpdateIndex;
+        return true;
+    } else {
+        return false; // no change
+    }
+}
+
+void
+SequencerDataBlock::setInstrumentLevel(InstrumentId id, const LevelInfo &info)
+{
+    int index = instrumentToIndexCreating(id);
+    if (index < 0)
+        return ;
+
+    m_levels[index] = info;
+    ++m_levelUpdateIndices[index];
+}
+
+bool
+SequencerDataBlock::getInstrumentRecordLevel(InstrumentId id, LevelInfo &info) const
+{
+    static int lastUpdateIndex[SEQUENCER_DATABLOCK_MAX_NB_INSTRUMENTS];
+
+    int index = instrumentToIndex(id);
+    if (index < 0) {
+        info.level = info.levelRight = 0;
+        return false;
+    }
+
+    int currentUpdateIndex = m_recordLevelUpdateIndices[index];
+    info = m_recordLevels[index];
+
+    if (lastUpdateIndex[index] != currentUpdateIndex) {
+        lastUpdateIndex[index] = currentUpdateIndex;
+        return true;
+    } else {
+        return false; // no change
+    }
+}
+
+bool
+SequencerDataBlock::getInstrumentRecordLevelForMixer(InstrumentId id, LevelInfo &info) const
+{
+    static int lastUpdateIndex[SEQUENCER_DATABLOCK_MAX_NB_INSTRUMENTS];
+
+    int index = instrumentToIndex(id);
+    if (index < 0) {
+        info.level = info.levelRight = 0;
+        return false;
+    }
+
+    int currentUpdateIndex = m_recordLevelUpdateIndices[index];
+    info = m_recordLevels[index];
+
+    if (lastUpdateIndex[index] != currentUpdateIndex) {
+        lastUpdateIndex[index] = currentUpdateIndex;
+        return true;
+    } else {
+        return false; // no change
+    }
+}
+
+void
+SequencerDataBlock::setInstrumentRecordLevel(InstrumentId id, const LevelInfo &info)
+{
+    int index = instrumentToIndexCreating(id);
+    if (index < 0)
+        return ;
+
+    m_recordLevels[index] = info;
+    ++m_recordLevelUpdateIndices[index];
+}
+
+void
+SequencerDataBlock::setTrackLevel(TrackId id, const LevelInfo &info)
+{
+    if (m_controlBlock) {
+        setInstrumentLevel(m_controlBlock->getInstrumentForTrack(id), info);
+    }
+}
+
+bool
+SequencerDataBlock::getTrackLevel(TrackId id, LevelInfo &info) const
+{
+    info.level = info.levelRight = 0;
+
+    if (m_controlBlock) {
+        return getInstrumentLevel(m_controlBlock->getInstrumentForTrack(id),
+                                  info);
+    }
+
+    return false;
+}
+
+bool
+SequencerDataBlock::getSubmasterLevel(int submaster, LevelInfo &info) const
+{
+    static int lastUpdateIndex[SEQUENCER_DATABLOCK_MAX_NB_SUBMASTERS];
+
+    if (submaster < 0 || submaster > SEQUENCER_DATABLOCK_MAX_NB_SUBMASTERS) {
+        info.level = info.levelRight = 0;
+        return false;
+    }
+
+    int currentUpdateIndex = m_submasterLevelUpdateIndices[submaster];
+    info = m_submasterLevels[submaster];
+
+    if (lastUpdateIndex[submaster] != currentUpdateIndex) {
+        lastUpdateIndex[submaster] = currentUpdateIndex;
+        return true;
+    } else {
+        return false; // no change
+    }
+}
+
+void
+SequencerDataBlock::setSubmasterLevel(int submaster, const LevelInfo &info)
+{
+    if (submaster < 0 || submaster > SEQUENCER_DATABLOCK_MAX_NB_SUBMASTERS) {
+        return ;
+    }
+
+    m_submasterLevels[submaster] = info;
+    ++m_submasterLevelUpdateIndices[submaster];
+}
+
+bool
+SequencerDataBlock::getMasterLevel(LevelInfo &level) const
+{
+    static int lastUpdateIndex = 0;
+
+    int currentIndex = m_masterLevelUpdateIndex;
+    level = m_masterLevel;
+
+    if (lastUpdateIndex != currentIndex) {
+        lastUpdateIndex = currentIndex;
+        return true;
+    } else {
+        return false;
+    }
+}
+
+void
+SequencerDataBlock::setMasterLevel(const LevelInfo &info)
+{
+    m_masterLevel = info;
+    ++m_masterLevelUpdateIndex;
+}
+
+void
+SequencerDataBlock::clearTemporaries()
+{
+    m_controlBlock = 0;
+    m_positionSec = 0;
+    m_positionNsec = 0;
+    m_visualEventIndex = 0;
+    *((MappedEvent *)&m_visualEvent) = MappedEvent();
+    m_haveVisualEvent = false;
+    m_recordEventIndex = 0;
+    //!!!    m_recordLevel.level = 0;
+    //!!!    m_recordLevel.levelRight = 0;
+    memset(m_knownInstruments, 0,
+           SEQUENCER_DATABLOCK_MAX_NB_INSTRUMENTS * sizeof(InstrumentId));
+    m_knownInstrumentCount = 0;
+    memset(m_levelUpdateIndices, 0,
+           SEQUENCER_DATABLOCK_MAX_NB_INSTRUMENTS * sizeof(int));
+    memset(m_levels, 0,
+           SEQUENCER_DATABLOCK_MAX_NB_INSTRUMENTS * sizeof(LevelInfo));
+    memset(m_submasterLevelUpdateIndices, 0,
+           SEQUENCER_DATABLOCK_MAX_NB_SUBMASTERS * sizeof(int));
+    memset(m_submasterLevels, 0,
+           SEQUENCER_DATABLOCK_MAX_NB_SUBMASTERS * sizeof(LevelInfo));
+    m_masterLevelUpdateIndex = 0;
+    m_masterLevel.level = 0;
+    m_masterLevel.levelRight = 0;
+
+}
+
+}
+
diff --git a/src/sound/SequencerDataBlock.h b/src/sound/SequencerDataBlock.h
new file mode 100644
index 0000000..2cfdefe
--- /dev/null
+++ b/src/sound/SequencerDataBlock.h
@@ -0,0 +1,140 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _SEQUENCER_DATA_BLOCK_H_
+#define _SEQUENCER_DATA_BLOCK_H_
+
+#include "ControlBlock.h"
+#include "RealTime.h"
+#include "MappedEvent.h"
+
+namespace Rosegarden
+{
+        
+/**
+ * ONLY PUT PLAIN DATA HERE - NO POINTERS EVER
+ * (and this struct mustn't have a constructor)
+ */
+struct LevelInfo
+{
+    int level;
+    int levelRight; // if stereo audio
+};
+
+class MappedComposition;
+
+
+#define SEQUENCER_DATABLOCK_MAX_NB_INSTRUMENTS 512 // can't be a symbol
+#define SEQUENCER_DATABLOCK_MAX_NB_SUBMASTERS   64 // can't be a symbol
+#define SEQUENCER_DATABLOCK_RECORD_BUFFER_SIZE 1024 // MIDI events
+
+class SequencerDataBlock
+{
+public:
+    /**
+     * Constructor only initialises memory if initialise is true
+     */
+    SequencerDataBlock(bool initialise);
+
+    RealTime getPositionPointer() const {
+        return RealTime(m_positionSec, m_positionNsec);
+    }
+    void setPositionPointer(const RealTime &rt) {
+        m_positionSec = rt.sec;
+        m_positionNsec = rt.nsec;
+    }
+    
+    bool getVisual(MappedEvent &ev) const;
+    void setVisual(const MappedEvent *ev);
+
+    int getRecordedEvents(MappedComposition &) const;
+    void addRecordedEvents(MappedComposition *);
+
+    bool getTrackLevel(TrackId track, LevelInfo &) const;
+    void setTrackLevel(TrackId track, const LevelInfo &);
+
+    // Two of these to rather hamfistedly get around the fact
+    // we need to fetch this value twice - once from IPB, 
+    // and again for the Mixer.
+    //
+    bool getInstrumentLevel(InstrumentId id, LevelInfo &) const;
+    bool getInstrumentLevelForMixer(InstrumentId id, LevelInfo &) const;
+
+    void setInstrumentLevel(InstrumentId id, const LevelInfo &);
+
+    bool getInstrumentRecordLevel(InstrumentId id, LevelInfo &) const;
+    bool getInstrumentRecordLevelForMixer(InstrumentId id, LevelInfo &) const;
+
+    void setInstrumentRecordLevel(InstrumentId id, const LevelInfo &);
+
+    bool getSubmasterLevel(int submaster, LevelInfo &) const;
+    void setSubmasterLevel(int submaster, const LevelInfo &);
+
+    bool getMasterLevel(LevelInfo &) const;
+    void setMasterLevel(const LevelInfo &);
+
+    void setControlBlock(ControlBlock *cb) { m_controlBlock = cb; }
+    ControlBlock *getControlBlock() { return m_controlBlock; }
+
+    // Reset the temporaries on (for example) GUI restart
+    //
+    void clearTemporaries();
+    
+protected:
+    int instrumentToIndex(InstrumentId id) const;
+    int instrumentToIndexCreating(InstrumentId id);
+    ControlBlock *m_controlBlock;
+
+    // Two ints rather than a RealTime, as the RealTime default ctor
+    // initialises the space & so can't be used from the GUI's
+    // placement-new ctor (which has no write access and doesn't want
+    // it anyway).  Likewise we use char[] instead of MappedEvents
+
+    int m_positionSec;
+    int m_positionNsec;
+
+    int m_visualEventIndex;
+    bool m_haveVisualEvent;
+    char m_visualEvent[sizeof(MappedEvent)];
+    
+    int m_recordEventIndex;
+    char m_recordBuffer[sizeof(MappedEvent) *
+                        SEQUENCER_DATABLOCK_RECORD_BUFFER_SIZE];
+
+    InstrumentId m_knownInstruments[SEQUENCER_DATABLOCK_MAX_NB_INSTRUMENTS];
+    int m_knownInstrumentCount;
+
+    int m_levelUpdateIndices[SEQUENCER_DATABLOCK_MAX_NB_INSTRUMENTS];
+    LevelInfo m_levels[SEQUENCER_DATABLOCK_MAX_NB_INSTRUMENTS];
+
+    int m_recordLevelUpdateIndices[SEQUENCER_DATABLOCK_MAX_NB_INSTRUMENTS];
+    LevelInfo m_recordLevels[SEQUENCER_DATABLOCK_MAX_NB_INSTRUMENTS];
+
+    int m_submasterLevelUpdateIndices[SEQUENCER_DATABLOCK_MAX_NB_SUBMASTERS];
+    LevelInfo m_submasterLevels[SEQUENCER_DATABLOCK_MAX_NB_SUBMASTERS];
+
+    int m_masterLevelUpdateIndex;
+    LevelInfo m_masterLevel;
+};
+
+}
+
+#endif
diff --git a/src/sound/SoundDriver.cpp b/src/sound/SoundDriver.cpp
new file mode 100644
index 0000000..aab641c
--- /dev/null
+++ b/src/sound/SoundDriver.cpp
@@ -0,0 +1,391 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include <stdlib.h>
+
+#include "SoundDriver.h"
+#include "WAVAudioFile.h"
+#include "MappedStudio.h"
+#include "AudioPlayQueue.h"
+
+#include <unistd.h>
+#include <sys/time.h>
+#include <pthread.h> // for mutex
+
+//#define DEBUG_SOUND_DRIVER 1
+
+namespace Rosegarden
+{
+
+// ---------- SoundDriver -----------
+//
+
+
+SoundDriver::SoundDriver(MappedStudio *studio, const std::string &name):
+        m_name(name),
+        m_driverStatus(NO_DRIVER),
+        m_playStartPosition(0, 0),
+        m_startPlayback(false),
+        m_playing(false),
+        m_midiRecordDevice(0),
+        m_recordStatus(RECORD_OFF),
+        m_midiRunningId(MidiInstrumentBase),
+        m_audioRunningId(AudioInstrumentBase),
+        //    m_audioQueueScavenger(4, 50),
+        m_audioQueue(0),
+        m_lowLatencyMode(true),
+        m_audioRecFileFormat(RIFFAudioFile::FLOAT),
+        m_studio(studio),
+        m_sequencerDataBlock(0),
+        m_externalTransport(0),
+        m_mmcStatus(TRANSPORT_OFF),
+        m_mtcStatus(TRANSPORT_OFF),
+        m_mmcId(0),            // default MMC id of 0
+        m_midiClockEnabled(false),
+        m_midiClockInterval(0, 0),
+        m_midiClockSendTime(RealTime::zeroTime),
+        m_midiSongPositionPointer(0)
+{
+    m_audioQueue = new AudioPlayQueue();
+}
+
+
+SoundDriver::~SoundDriver()
+{
+    std::cout << "SoundDriver::~SoundDriver (exiting)" << std::endl;
+    delete m_audioQueue;
+}
+
+MappedInstrument*
+SoundDriver::getMappedInstrument(InstrumentId id)
+{
+    std::vector<MappedInstrument*>::const_iterator it;
+
+    for (it = m_instruments.begin(); it != m_instruments.end(); it++) {
+        if ((*it)->getId() == id)
+            return (*it);
+    }
+
+    return 0;
+}
+
+void
+SoundDriver::initialiseAudioQueue(const std::vector<MappedEvent> &events)
+{
+    AudioPlayQueue *newQueue = new AudioPlayQueue();
+
+    for (std::vector<MappedEvent>::const_iterator i = events.begin();
+            i != events.end(); ++i) {
+
+        // Check for existence of file - if the sequencer has died
+        // and been restarted then we're not always loaded up with
+        // the audio file references we should have.  In the future
+        // we could make this just get the gui to reload our files
+        // when (or before) this fails.
+        //
+        AudioFile *audioFile = getAudioFile(i->getAudioID());
+
+        if (audioFile) {
+            MappedAudioFader *fader =
+                dynamic_cast<MappedAudioFader*>
+                (getMappedStudio()->getAudioFader(i->getInstrument()));
+
+            if (!fader) {
+                std::cerr << "WARNING: SoundDriver::initialiseAudioQueue: no fader for audio instrument " << i->getInstrument() << std::endl;
+                continue;
+            }
+
+            unsigned int channels = fader->getPropertyList(
+                                        MappedAudioFader::Channels)[0].toInt();
+
+            //#define DEBUG_PLAYING_AUDIO
+#ifdef DEBUG_PLAYING_AUDIO
+
+            std::cout << "Creating playable audio file: id " << audioFile->getId() << ", event time " << i->getEventTime() << ", time now " << getSequencerTime() << ", start marker " << i->getAudioStartMarker() << ", duration " << i->getDuration() << ", instrument " << i->getInstrument() << " channels " << channels << std::endl;
+#endif
+
+            RealTime bufferLength = getAudioReadBufferLength();
+            int bufferFrames = RealTime::realTime2Frame
+                               (bufferLength, getSampleRate());
+
+            PlayableAudioFile *paf = 0;
+
+            try {
+                paf = new PlayableAudioFile(i->getInstrument(),
+                                            audioFile,
+                                            i->getEventTime(),
+                                            i->getAudioStartMarker(),
+                                            i->getDuration(),
+                                            bufferFrames,
+                                            getSmallFileSize() * 1024,
+                                            channels,
+                                            getSampleRate());
+            } catch (...) {
+                continue;
+            }
+
+            paf->setRuntimeSegmentId(i->getRuntimeSegmentId());
+
+            if (i->isAutoFading()) {
+                paf->setAutoFade(true);
+                paf->setFadeInTime(i->getFadeInTime());
+                paf->setFadeOutTime(i->getFadeInTime());
+
+                //#define DEBUG_AUTOFADING
+#ifdef DEBUG_AUTOFADING
+
+                std::cout << "SoundDriver::initialiseAudioQueue - "
+                << "PlayableAudioFile is AUTOFADING - "
+                << "in = " << i->getFadeInTime()
+                << ", out = " << i->getFadeOutTime()
+                << std::endl;
+#endif
+
+            }
+#ifdef DEBUG_AUTOFADING
+            else {
+                std::cout << "PlayableAudioFile has no AUTOFADE"
+                << std::endl;
+            }
+#endif
+
+            newQueue->addScheduled(paf);
+        } else {
+            std::cerr << "SoundDriver::initialiseAudioQueue - "
+            << "can't find audio file reference for id " << i->getAudioID()
+            << std::endl;
+
+            std::cerr << "SoundDriver::initialiseAudioQueue - "
+            << "try reloading the current Rosegarden file"
+            << std::endl;
+        }
+    }
+
+    std::cout << "SoundDriver::initialiseAudioQueue -- new queue has "
+    << newQueue->size() << " files"
+    << std::endl;
+
+    if (newQueue->empty()) {
+        if (m_audioQueue->empty()) {
+            delete newQueue;
+            return ;
+        }
+    }
+
+    AudioPlayQueue *oldQueue = m_audioQueue;
+    m_audioQueue = newQueue;
+    if (oldQueue)
+        m_audioQueueScavenger.claim(oldQueue);
+}
+
+void
+SoundDriver::clearAudioQueue()
+{
+    std::cout << "SoundDriver::clearAudioQueue" << std::endl;
+
+    if (m_audioQueue->empty())
+        return ;
+
+    AudioPlayQueue *newQueue = new AudioPlayQueue();
+    AudioPlayQueue *oldQueue = m_audioQueue;
+    m_audioQueue = newQueue;
+    if (oldQueue)
+        m_audioQueueScavenger.claim(oldQueue);
+}
+void
+SoundDriver::cancelAudioFile(MappedEvent *mE)
+{
+    std::cout << "SoundDriver::cancelAudioFile" << std::endl;
+
+    if (!m_audioQueue)
+        return ;
+
+    // For now we only permit cancelling unscheduled files.
+
+    const AudioPlayQueue::FileList &files = m_audioQueue->getAllUnscheduledFiles();
+    for (AudioPlayQueue::FileList::const_iterator fi = files.begin();
+            fi != files.end(); ++fi) {
+        PlayableAudioFile *file = *fi;
+        if (mE->getRuntimeSegmentId() == -1) {
+
+            // ERROR? The comparison between file->getAudioFile()->getId() of type unsigned int
+            //        and mE->getAudioID() of type int.
+            if (file->getInstrument() == mE->getInstrument() &&
+                    int(file->getAudioFile()->getId() == mE->getAudioID())) {
+                file->cancel();
+            }
+        } else {
+            if (file->getRuntimeSegmentId() == mE->getRuntimeSegmentId() &&
+                    file->getStartTime() == mE->getEventTime()) {
+                file->cancel();
+            }
+        }
+    }
+}
+
+const AudioPlayQueue *
+SoundDriver::getAudioQueue() const
+{
+    return m_audioQueue;
+}
+
+
+void
+SoundDriver::setMappedInstrument(MappedInstrument *mI)
+{
+    std::vector<MappedInstrument*>::iterator it;
+
+    // If we match then change existing entry
+    for (it = m_instruments.begin(); it != m_instruments.end(); it++) {
+        if ((*it)->getId() == mI->getId()) {
+            (*it)->setChannel(mI->getChannel());
+            (*it)->setType(mI->getType());
+            delete mI;
+            return ;
+        }
+    }
+
+    // else create a new one
+    m_instruments.push_back(mI);
+
+    std::cout << "SoundDriver: setMappedInstrument() : "
+    << "type = " << mI->getType() << " : "
+    << "channel = " << (int)(mI->getChannel()) << " : "
+    << "id = " << mI->getId() << std::endl;
+
+}
+
+unsigned int
+SoundDriver::getDevices()
+{
+    return m_devices.size();
+}
+
+MappedDevice
+SoundDriver::getMappedDevice(DeviceId id)
+{
+    MappedDevice retDevice;
+    std::vector<MappedInstrument*>::iterator it;
+
+    std::vector<MappedDevice*>::iterator dIt = m_devices.begin();
+    for (; dIt != m_devices.end(); dIt++) {
+        if ((*dIt)->getId() == id)
+            retDevice = **dIt;
+    }
+
+    // If we match then change existing entry
+    for (it = m_instruments.begin(); it != m_instruments.end(); it++) {
+        if ((*it)->getDevice() == id)
+            retDevice.push_back(*it);
+    }
+
+#ifdef DEBUG_SOUND_DRIVER
+    std::cout << "SoundDriver::getMappedDevice(" << id << ") - "
+    << "name = \"" << retDevice.getName()
+    << "\" type = " << retDevice.getType()
+    << " direction = " << retDevice.getDirection()
+    << " connection = \"" << retDevice.getConnection() << "\""
+    << " recording = " << retDevice.isRecording()
+    << std::endl;
+#endif
+
+    return retDevice;
+}
+
+
+
+bool
+SoundDriver::addAudioFile(const std::string &fileName, unsigned int id)
+{
+    AudioFile *ins = 0;
+
+    try {
+        ins = new WAVAudioFile(id, fileName, fileName);
+        ins->open();
+        m_audioFiles.push_back(ins);
+
+        //    std::cout << "Sequencer::addAudioFile() = \"" << fileName << "\"" << std::endl;
+
+        return true;
+
+    } catch (SoundFile::BadSoundFileException e) {
+        std::cerr << "SoundDriver::addAudioFile: Failed to add audio file " << fileName << ": " << e.getMessage() << std::endl;
+        delete ins;
+        return false;
+    }
+}
+
+bool
+SoundDriver::removeAudioFile(unsigned int id)
+{
+    std::vector<AudioFile*>::iterator it;
+    for (it = m_audioFiles.begin(); it != m_audioFiles.end(); it++) {
+        if ((*it)->getId() == id) {
+            std::cout << "Sequencer::removeAudioFile() = \"" <<
+            (*it)->getFilename() << "\"" << std::endl;
+
+            delete (*it);
+            m_audioFiles.erase(it);
+            return true;
+        }
+    }
+
+    return false;
+}
+
+AudioFile*
+SoundDriver::getAudioFile(unsigned int id)
+{
+    std::vector<AudioFile*>::iterator it;
+    for (it = m_audioFiles.begin(); it != m_audioFiles.end(); it++) {
+        if ((*it)->getId() == id)
+            return *it;
+    }
+
+    return 0;
+}
+
+void
+SoundDriver::clearAudioFiles()
+{
+    //    std::cout << "SoundDriver::clearAudioFiles() - clearing down audio files"
+    //              << std::endl;
+
+    std::vector<AudioFile*>::iterator it;
+    for (it = m_audioFiles.begin(); it != m_audioFiles.end(); it++)
+        delete(*it);
+
+    m_audioFiles.erase(m_audioFiles.begin(), m_audioFiles.end());
+}
+
+void
+SoundDriver::sleep(const RealTime &rt)
+{
+    // The usleep man page says it's deprecated and we should use
+    // nanosleep.  And that's what we did.  But it seems quite a few
+    // people don't have nanosleep, so we're reverting to usleep.
+
+    unsigned long usec = rt.sec * 1000000 + rt.usec();
+    usleep(usec);
+}
+
+
+}
+
diff --git a/src/sound/SoundDriver.h b/src/sound/SoundDriver.h
new file mode 100644
index 0000000..fabbaef
--- /dev/null
+++ b/src/sound/SoundDriver.h
@@ -0,0 +1,529 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include <string>
+#include <vector>
+#include <list>
+#include <qstringlist.h>
+
+#include "Device.h"
+#include "MappedComposition.h"
+#include "MappedInstrument.h"
+#include "MappedDevice.h"
+#include "SequencerDataBlock.h"
+#include "PlayableAudioFile.h"
+#include "Scavenger.h"
+#include "RIFFAudioFile.h" // for SubFormat enum
+
+// Abstract base to support SoundDrivers, such as ALSA.
+//
+// This base class provides the generic driver support for
+// these drivers with the Sequencer class owning an instance
+// of a sub class of this class and directing it and required
+// by the rosegardensequencer itself.
+//
+//
+
+#ifndef _SOUNDDRIVER_H_
+#define _SOUNDDRIVER_H_
+
+namespace Rosegarden
+{
+
+// Current recording status - whether we're monitoring anything
+// or recording.
+//
+typedef enum
+{
+    RECORD_OFF,
+    RECORD_ON,
+} RecordStatus;
+
+
+// Status of a SoundDriver - whether we're got an audio and
+// MIDI subsystem or not.  This is reported right up to the
+// gui.
+//
+typedef enum
+{
+    NO_DRIVER  = 0x00,          // Nothing's OK
+    AUDIO_OK   = 0x01,          // AUDIO's OK
+    MIDI_OK    = 0x02,          // MIDI's OK
+    VERSION_OK = 0x04           // GUI and sequencer versions match
+} SoundDriverStatus;
+
+
+// Used for MMC and MTC, not for JACK transport
+//
+typedef enum
+{
+    TRANSPORT_OFF,
+    TRANSPORT_MASTER,
+    TRANSPORT_SLAVE
+} TransportSyncStatus;
+
+
+// The NoteOffQueue holds a time ordered set of
+// pending MIDI NOTE OFF events.
+//
+class NoteOffEvent
+{
+public:
+    NoteOffEvent() {;}
+    NoteOffEvent(const RealTime &realTime,
+                 unsigned int pitch,
+                 MidiByte channel,
+                 InstrumentId instrument):
+        m_realTime(realTime),
+        m_pitch(pitch),
+        m_channel(channel),
+        m_instrument(instrument) {;}
+    ~NoteOffEvent() {;}
+
+    struct NoteOffEventCmp
+    {
+        bool operator()(NoteOffEvent *nO1, NoteOffEvent *nO2)
+        {
+            return nO1->getRealTime() < nO2->getRealTime();
+        }
+    };
+
+    void setRealTime(const RealTime &time) { m_realTime = time; }
+    RealTime getRealTime() const { return m_realTime; }
+
+    MidiByte getPitch() const { return m_pitch; }
+    MidiByte getChannel() const { return m_channel; }
+    InstrumentId getInstrument() const { return m_instrument; }
+
+private:
+    RealTime     m_realTime;
+    MidiByte     m_pitch;
+    MidiByte     m_channel;
+    InstrumentId m_instrument;
+
+};
+
+
+// The queue itself
+//
+class NoteOffQueue : public std::multiset<NoteOffEvent *,
+                     NoteOffEvent::NoteOffEventCmp>
+{
+public:
+    NoteOffQueue() {;}
+    ~NoteOffQueue() {;}
+private:
+};
+
+
+class MappedStudio;
+class ExternalTransport;
+class AudioPlayQueue;
+
+typedef std::vector<PlayableAudioFile *> PlayableAudioFileList;
+
+// The abstract SoundDriver
+//
+//
+class SoundDriver
+{
+public:
+    SoundDriver(MappedStudio *studio, const std::string &name);
+    virtual ~SoundDriver();
+
+    virtual bool initialise() = 0;
+    virtual void shutdown() { }
+
+    virtual void initialisePlayback(const RealTime &position) = 0;
+    virtual void stopPlayback() = 0;
+    virtual void punchOut() = 0; // stop recording, continue playing
+    virtual void resetPlayback(const RealTime &oldPosition, const RealTime &position) = 0;
+    virtual void allNotesOff() = 0;
+    
+    virtual RealTime getSequencerTime() = 0;
+
+    virtual MappedComposition *getMappedComposition() = 0;
+
+    virtual void startClocks() { }
+    virtual void stopClocks() { }
+
+    // Process some asynchronous events
+    //
+    virtual void processEventsOut(const MappedComposition &mC) = 0;
+
+    // Process some scheduled events on the output queue.  The
+    // slice times are here so that the driver can interleave
+    // note-off events as appropriate.
+    //
+    virtual void processEventsOut(const MappedComposition &mC,
+                                  const RealTime &sliceStart,
+                                  const RealTime &sliceEnd) = 0;
+
+    // Activate a recording state.  armedInstruments and audioFileNames
+    // can be NULL if no audio tracks recording.
+    //
+    virtual bool record(RecordStatus recordStatus,
+                        const std::vector<InstrumentId> *armedInstruments = 0,
+                        const std::vector<QString> *audioFileNames = 0) = 0;
+
+    // Process anything that's pending
+    //
+    virtual void processPending() = 0;
+
+    // Get the driver's operating sample rate
+    //
+    virtual unsigned int getSampleRate() const = 0;
+
+    // Plugin instance management
+    //
+    virtual void setPluginInstance(InstrumentId id,
+                                   QString identifier,
+                                   int position) = 0;
+
+    virtual void removePluginInstance(InstrumentId id,
+                                      int position) = 0;
+
+    // Clear down and remove all plugin instances
+    //
+    virtual void removePluginInstances() = 0;
+
+    virtual void setPluginInstancePortValue(InstrumentId id,
+                                            int position,
+                                            unsigned long portNumber,
+                                            float value) = 0;
+
+    virtual float getPluginInstancePortValue(InstrumentId id,
+                                             int position,
+                                             unsigned long portNumber) = 0;
+
+    virtual void setPluginInstanceBypass(InstrumentId id,
+                                         int position,
+                                         bool value) = 0;
+
+    virtual QStringList getPluginInstancePrograms(InstrumentId id,
+                                                  int position) = 0;
+
+    virtual QString getPluginInstanceProgram(InstrumentId id,
+                                             int position) = 0;
+
+    virtual QString getPluginInstanceProgram(InstrumentId id,
+                                             int position,
+                                             int bank,
+                                             int program) = 0;
+
+    virtual unsigned long getPluginInstanceProgram(InstrumentId id,
+                                                   int position,
+                                                   QString name) = 0;
+    
+    virtual void setPluginInstanceProgram(InstrumentId id,
+                                          int position,
+                                          QString program) = 0;
+
+    virtual QString configurePlugin(InstrumentId id,
+                                    int position,
+                                    QString key,
+                                    QString value) = 0;
+
+    virtual void setAudioBussLevels(int bussId,
+                                    float dB,
+                                    float pan) = 0;
+
+    virtual void setAudioInstrumentLevels(InstrumentId id,
+                                          float dB,
+                                          float pan) = 0;
+
+    // Poll for new clients (for new Devices/Instruments)
+    //
+    virtual bool checkForNewClients() = 0;
+
+    // Set a loop position at the driver (used for transport)
+    //
+    virtual void setLoop(const RealTime &loopStart, const RealTime &loopEnd)
+        = 0;
+
+    virtual void sleep(const RealTime &rt);
+
+    virtual QString getStatusLog() { return ""; }
+
+    // Mapped Instruments
+    //
+    void setMappedInstrument(MappedInstrument *mI);
+    MappedInstrument* getMappedInstrument(InstrumentId id);
+
+    // Return the current status of the driver
+    //
+    unsigned int getStatus() const { return m_driverStatus; }
+
+    // Are we playing?
+    //
+    bool isPlaying() const { return m_playing; }
+
+    // Are we counting?  By default a subclass probably wants to
+    // return true, if it doesn't know better.
+    //
+    virtual bool areClocksRunning() const = 0;
+
+    RealTime getStartPosition() const { return m_playStartPosition; }
+    RecordStatus getRecordStatus() const { return m_recordStatus; }
+
+    // Return a MappedDevice full of the Instrument mappings
+    // that the driver has discovered.  The gui can then use
+    // this list (complete with names) to generate its proper
+    // Instruments under the MidiDevice and AudioDevice.
+    //
+    MappedDevice getMappedDevice(DeviceId id);
+
+    // Return the number of devices we've found
+    //
+    unsigned int getDevices();
+
+    virtual bool canReconnect(Device::DeviceType) { return false; }
+
+    virtual DeviceId addDevice(Device::DeviceType,
+                               MidiDevice::DeviceDirection) {
+        return Device::NO_DEVICE;
+    }
+    virtual void removeDevice(DeviceId) { }
+    virtual void renameDevice(DeviceId, QString) { }
+
+    virtual unsigned int getConnections(Device::DeviceType,
+                                        MidiDevice::DeviceDirection) { return 0; }
+    virtual QString getConnection(Device::DeviceType,
+                                  MidiDevice::DeviceDirection,
+                                  unsigned int) { return ""; }
+    virtual void setConnection(DeviceId, QString) { }
+    virtual void setPlausibleConnection(DeviceId id, QString c) { setConnection(id, c); }
+
+    virtual unsigned int getTimers() { return 0; }
+    virtual QString getTimer(unsigned int) { return ""; }
+    virtual QString getCurrentTimer() { return ""; }
+    virtual void setCurrentTimer(QString) { }
+
+    virtual void getAudioInstrumentNumbers(InstrumentId &, int &) = 0;
+    virtual void getSoftSynthInstrumentNumbers(InstrumentId &, int &) = 0;
+
+    // Plugin management -- SoundDrivers should maintain a plugin
+    // scavenger which the audio process code can use for defunct
+    // plugins.  Ownership of plugin is passed to the SoundDriver.
+    //
+    virtual void claimUnwantedPlugin(void *plugin) = 0;
+
+    // This causes all scavenged plugins to be destroyed.  It
+    // should only be called in non-RT contexts.
+    //
+    virtual void scavengePlugins() = 0;
+
+    // Handle audio file references
+    //
+    void clearAudioFiles();
+    bool addAudioFile(const std::string &fileName, unsigned int id);
+    bool removeAudioFile(unsigned int id);
+                    
+    void initialiseAudioQueue(const std::vector<MappedEvent> &audioEvents);
+    void clearAudioQueue();
+    const AudioPlayQueue *getAudioQueue() const;
+
+    RIFFAudioFile::SubFormat getAudioRecFileFormat() const { return m_audioRecFileFormat; }
+
+
+    // Latencies
+    //
+    virtual RealTime getAudioPlayLatency() { return RealTime::zeroTime; }
+    virtual RealTime getAudioRecordLatency() { return RealTime::zeroTime; }
+    virtual RealTime getInstrumentPlayLatency(InstrumentId) { return RealTime::zeroTime; }
+    virtual RealTime getMaximumPlayLatency() { return RealTime::zeroTime; }
+
+    // Buffer sizes
+    //
+    void setAudioBufferSizes(RealTime mix, RealTime read, RealTime write,
+                             int smallFileSize) {
+        m_audioMixBufferLength = mix;
+        m_audioReadBufferLength = read;
+        m_audioWriteBufferLength = write;
+        m_smallFileSize = smallFileSize;
+    }
+
+    RealTime getAudioMixBufferLength() { return m_audioMixBufferLength; }
+    RealTime getAudioReadBufferLength() { return m_audioReadBufferLength; }
+    RealTime getAudioWriteBufferLength() { return m_audioWriteBufferLength; }
+    int getSmallFileSize() { return m_smallFileSize; }
+
+    void setLowLatencyMode(bool ll) { m_lowLatencyMode = ll; }
+    bool getLowLatencyMode() const { return m_lowLatencyMode; }
+
+    // Cancel the playback of an audio file - either by instrument and audio file id
+    // or by audio segment id.
+    //
+    void cancelAudioFile(MappedEvent *mE);
+
+    // Studio linkage
+    //
+    MappedStudio* getMappedStudio() { return m_studio; }
+    void setMappedStudio(MappedStudio *studio) { m_studio = studio; }
+
+    // Modify MIDI record device
+    //
+    void setMidiRecordDevice(DeviceId id) { m_midiRecordDevice = id; }
+    DeviceId getMIDIRecordDevice() const { return m_midiRecordDevice; }
+
+    // MIDI Realtime Sync setting
+    //
+    TransportSyncStatus getMIDISyncStatus() const { return m_midiSyncStatus; }
+    void setMIDISyncStatus(TransportSyncStatus status) { m_midiSyncStatus = status; }
+
+    // MMC master/slave setting
+    //
+    TransportSyncStatus getMMCStatus() const { return m_mmcStatus; }
+    void setMMCStatus(TransportSyncStatus status) { m_mmcStatus = status; }
+
+    // MTC master/slave setting
+    //
+    TransportSyncStatus getMTCStatus() const { return m_mtcStatus; }
+    void setMTCStatus(TransportSyncStatus status) { m_mtcStatus = status; }
+
+    // MMC Id
+    //
+    int getMMCId() const { return ((int)(m_mmcId)); }
+    void setMMCId(int id) { m_mmcId = (MidiByte)(id); }
+
+    // Set MIDI clock interval - allow redefinition above to ensure
+    // we handle this reset correctly.
+    //
+    virtual void setMIDIClockInterval(RealTime interval) 
+        { m_midiClockInterval = interval; }
+
+    // Get and set the mapper which may optionally be used to
+    // store recording levels etc for communication back to the GUI.
+    // (If a subclass wants this and finds it's not there, it should
+    // simply continue without.)
+    //
+    SequencerDataBlock *getSequencerDataBlock() { return m_sequencerDataBlock; }
+    void setSequencerDataBlock(SequencerDataBlock *d) { m_sequencerDataBlock = d; }
+
+    ExternalTransport *getExternalTransportControl() const {
+        return m_externalTransport;
+    }
+    void setExternalTransportControl(ExternalTransport *transport) {
+        m_externalTransport = transport;
+    }
+
+    // Do any bits and bobs of work that need to be done continuously
+    // (this is called repeatedly whether playing or not).
+    //
+    virtual void runTasks() { }
+
+    // Report a failure back to the GUI - ideally.  Default does nothing.
+    //
+    virtual void reportFailure(MappedEvent::FailureCode) { }
+
+protected:
+    // Helper functions to be implemented by subclasses
+    //
+    virtual void processMidiOut(const MappedComposition &mC,
+                                const RealTime &sliceStart,
+                                const RealTime &sliceEnd) = 0;
+    virtual void generateInstruments() = 0;
+
+    // Audio
+    //
+    AudioFile* getAudioFile(unsigned int id);
+
+    std::string                                 m_name;
+    unsigned int                                m_driverStatus;
+    RealTime                                    m_playStartPosition;
+    bool                                        m_startPlayback;
+    bool                                        m_playing;
+
+    // MIDI Note-off handling
+    //
+    NoteOffQueue                                m_noteOffQueue;
+
+    // This is our driver's own list of MappedInstruments and MappedDevices.  
+    // These are uncoupled at this level - the Instruments and Devices float
+    // free and only index each other - the Devices hold information only like
+    // name, id and if the device is duplex capable.
+    //
+    typedef std::vector<MappedInstrument*> MappedInstrumentList;
+    MappedInstrumentList m_instruments;
+
+    typedef std::vector<MappedDevice*> MappedDeviceList;
+    MappedDeviceList m_devices;
+
+    DeviceId                                    m_midiRecordDevice;
+
+    MappedComposition                           m_recordComposition;
+    MappedComposition                           m_returnComposition;
+    RecordStatus                                m_recordStatus;
+
+
+    InstrumentId                                m_midiRunningId;
+    InstrumentId                                m_audioRunningId;
+
+    // Subclass _MUST_ scavenge this regularly:
+    Scavenger<AudioPlayQueue>                   m_audioQueueScavenger;
+    AudioPlayQueue                             *m_audioQueue;
+
+    // A list of AudioFiles that we can play.
+    //
+    std::vector<AudioFile*>                     m_audioFiles;
+
+    RealTime m_audioMixBufferLength;
+    RealTime m_audioReadBufferLength;
+    RealTime m_audioWriteBufferLength;
+    int m_smallFileSize;
+    bool m_lowLatencyMode;
+
+    RIFFAudioFile::SubFormat m_audioRecFileFormat;
+    
+    // Virtual studio hook
+    //
+    MappedStudio                *m_studio;
+
+    // Sequencer data block for communication back to GUI
+    //
+    SequencerDataBlock          *m_sequencerDataBlock;
+    
+    // Controller to make externally originated transport requests on
+    //
+    ExternalTransport           *m_externalTransport;
+
+    // MMC and MTC status and ID
+    //
+    TransportSyncStatus          m_midiSyncStatus;
+    TransportSyncStatus          m_mmcStatus;
+    TransportSyncStatus          m_mtcStatus;
+    MidiByte                     m_mmcId;      // device id
+
+    // MIDI clock interval
+    //
+    bool                         m_midiClockEnabled;
+    RealTime                     m_midiClockInterval;
+    RealTime                     m_midiClockSendTime;
+
+    // MIDI Song Position pointer
+    //
+    long                         m_midiSongPositionPointer;
+
+};
+
+}
+
+#endif // _SOUNDDRIVER_H_
+
diff --git a/src/sound/SoundDriverFactory.cpp b/src/sound/SoundDriverFactory.cpp
new file mode 100644
index 0000000..d081a4e
--- /dev/null
+++ b/src/sound/SoundDriverFactory.cpp
@@ -0,0 +1,66 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "DummyDriver.h"
+
+#ifdef HAVE_ALSA
+#include "AlsaDriver.h"
+#endif
+
+#include "SoundDriverFactory.h"
+
+namespace Rosegarden
+{
+
+SoundDriver *
+SoundDriverFactory::createDriver(MappedStudio *studio)
+{
+    SoundDriver *driver = 0;
+    bool initialised = false;
+#ifdef NO_SOUND
+
+    driver = new DummyDriver(studio);
+#else
+#ifdef HAVE_ALSA
+
+    driver = new AlsaDriver(studio);
+#endif
+#endif
+
+    initialised = driver->initialise();
+
+    if ( ! initialised ) {
+        driver->shutdown();
+        delete driver;
+
+        // if the driver couldn't be initialised, then
+        // fall to the DummyDriver as a last chance,
+        // so GUI can still be used for notation.
+        //
+        driver = new DummyDriver(studio);
+        driver->initialise();
+    }
+    return driver;
+}
+
+
+}
+
+
diff --git a/src/sound/SoundDriverFactory.h b/src/sound/SoundDriverFactory.h
new file mode 100644
index 0000000..56bc889
--- /dev/null
+++ b/src/sound/SoundDriverFactory.h
@@ -0,0 +1,37 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef SOUND_DRIVER_FACTORY_H
+#define SOUND_DRIVER_FACTORY_H
+
+namespace Rosegarden {
+
+class SoundDriver;
+
+class SoundDriverFactory
+{
+public:
+    static SoundDriver *createDriver(MappedStudio *studio);
+};
+
+}
+
+#endif
+
diff --git a/src/sound/SoundFile.cpp b/src/sound/SoundFile.cpp
new file mode 100644
index 0000000..b87cef0
--- /dev/null
+++ b/src/sound/SoundFile.cpp
@@ -0,0 +1,295 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*-
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+ 
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+ 
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+ 
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+
+#include "SoundFile.h"
+#include "Profiler.h"
+
+
+//#define DEBUG_SOUNDFILE 1
+
+namespace Rosegarden
+
+{
+
+SoundFile::SoundFile(const std::string &fileName):
+        m_fileName(fileName),
+        m_readChunkPtr( -1),
+        m_readChunkSize(4096),  // 4k blocks
+        m_inFile(0),
+        m_outFile(0),
+        m_loseBuffer(false),
+        m_fileSize(0)
+{}
+
+// Tidies up for any dervied classes
+//
+SoundFile::~SoundFile()
+{
+    if (m_inFile) {
+        m_inFile->close();
+        delete m_inFile;
+    }
+
+    if (m_outFile) {
+        m_outFile->close();
+        delete m_outFile;
+    }
+
+}
+
+// Read in a specified number of bytes and return them
+// as a string.
+//
+std::string
+SoundFile::getBytes(std::ifstream *file, unsigned int numberOfBytes)
+{
+    if (file->eof()) {
+        // Reset the input stream so it's operational again
+        //
+        file->clear();
+
+        throw(BadSoundFileException(m_fileName, "SoundFile::getBytes() - EOF encountered"));
+    }
+
+    if (!(*file)) {
+        std::cerr << "SoundFile::getBytes() -  stream is not well";
+    }
+
+
+    std::string rS;
+    char *fileBytes = new char[numberOfBytes];
+
+    file->read(fileBytes, numberOfBytes);
+
+    for (int i = 0; i < file->gcount(); i++)
+        rS += (unsigned char)fileBytes[i];
+
+#ifdef DEBUG_SOUNDFILE
+    // complain but return
+    //
+    if (rS.length() < numberOfBytes)
+        std::cerr << "SoundFile::getBytes() - couldn't get all bytes ("
+        << rS.length() << " from " << numberOfBytes << ")"
+        << std::endl;
+#endif
+
+    // clear down
+    delete [] fileBytes;
+
+    return rS;
+}
+
+// Read a specified number of bytes into a buffer.
+//
+size_t
+SoundFile::getBytes(std::ifstream *file, char *buf, size_t n)
+{
+    if (!(*file)) {
+        std::cerr << "SoundFile::getBytes() -  stream is not well";
+        return 0;
+    }
+
+    if (file->eof()) {
+        file->clear();
+        return 0;
+    }
+
+    file->read(buf, n);
+    return file->gcount();
+}
+
+// A buffered read based on the current file handle.
+//
+std::string
+SoundFile::getBytes(unsigned int numberOfBytes)
+{
+    if (m_inFile == 0)
+        throw(BadSoundFileException(m_fileName, "SoundFile::getBytes - no open file handle"));
+
+    if (m_inFile->eof()) {
+        // Reset the input stream so it's operational again
+        //
+        m_inFile->clear();
+
+        throw(BadSoundFileException(m_fileName, "SoundFile::getBytes() - EOF encountered"));
+    }
+
+
+    // If this flag is set we dump the buffer and re-read it -
+    // should be set if specialised class is scanning about
+    // when we're doing buffered reads
+    //
+    if (m_loseBuffer) {
+        m_readChunkPtr = -1;
+        m_loseBuffer = false;
+    }
+
+    std::string rS;
+    char *fileBytes = new char[m_readChunkSize];
+    int oldLength;
+
+    while (rS.length() < numberOfBytes && !m_inFile->eof()) {
+        if (m_readChunkPtr == -1) {
+            // clear buffer
+            m_readBuffer = "";
+
+            // reset read pointer
+            m_readChunkPtr = 0;
+
+            // Try to read the whole chunk
+            //
+            m_inFile->read(fileBytes, m_readChunkSize);
+
+            // file->gcount holds the number of bytes we've actually read
+            // so copy them across into our string
+            //
+            for (int i = 0; i < m_inFile->gcount(); i++)
+                m_readBuffer += (unsigned char)fileBytes[i];
+        }
+
+        // Can we fulfill our request at this pass?  If so read the
+        // bytes across and we'll exit at the end of this loop.
+        // m_readChunkPtr keeps our position for next time.
+        //
+        if (numberOfBytes - rS.length() <= m_readBuffer.length() -
+                m_readChunkPtr) {
+            oldLength = rS.length();
+
+            rS += m_readBuffer.substr(m_readChunkPtr,
+                                      numberOfBytes - oldLength);
+
+            m_readChunkPtr += rS.length() - oldLength;
+        } else {
+            // Fill all we can this time and reset the m_readChunkPtr
+            // so that we fetch another chunk of bytes from the file.
+            //
+            rS += m_readBuffer.substr(m_readChunkPtr,
+                                      m_readChunkSize - m_readChunkPtr);
+            m_readChunkPtr = -1;
+        }
+
+        // If we're EOF here we must've read and copied across everything
+        // we can do.  Reset and break out.
+        //
+        if (m_inFile->eof()) {
+            m_inFile->clear();
+            break;
+        }
+
+    }
+
+#ifdef DEBUG_SOUNDFILE
+    // complain but return
+    //
+    if (rS.length() < numberOfBytes)
+        std::cerr << "SoundFile::getBytes() buffered - couldn't get all bytes ("
+        << rS.length() << " from " << numberOfBytes << ")"
+        << std::endl;
+#endif
+
+    delete [] fileBytes;
+
+    // Reset and return if EOF
+    //
+    if (m_inFile->eof())
+        m_inFile->clear();
+
+    return rS;
+}
+
+
+// Write out a sequence of FileBytes to the stream
+//
+void
+SoundFile::putBytes(std::ofstream *file,
+                    const std::string oS)
+{
+    for (unsigned int i = 0; i < oS.length(); i++)
+        *file << (FileByte) oS[i];
+}
+
+void
+SoundFile::putBytes(std::ofstream *file, const char *buffer, size_t n)
+{
+    file->write(buffer, n);
+}
+
+
+// Clip off any path from the filename
+std::string
+SoundFile::getShortFilename() const
+{
+    std::string rS = m_fileName;
+    unsigned int pos = rS.find_last_of("/");
+
+    if (pos > 0 && ( pos + 1 ) < rS.length())
+        rS = rS.substr(pos + 1, rS.length());
+
+    return rS;
+}
+
+
+// Turn a little endian binary std::string into an integer
+//
+int
+SoundFile::getIntegerFromLittleEndian(const std::string &s)
+{
+    int r = 0;
+
+    for (unsigned int i = 0; i < s.length(); i++) {
+        r += (int)(((FileByte)s[i]) << (i * 8));
+    }
+
+    return r;
+}
+
+
+// Turn a value into a little endian string of "length"
+//
+std::string
+SoundFile::getLittleEndianFromInteger(unsigned int value, unsigned int length)
+{
+    std::string r = "";
+
+    do {
+        r += (unsigned char)((long)((value >> (8 * r.length())) & 0xff));
+    } while (r.length() < length);
+
+    return r;
+}
+
+int
+SoundFile::getIntegerFromBigEndian(const std::string &s)
+{
+    return 0;
+}
+
+std::string
+SoundFile::getBigEndianFromInteger(unsigned int value, unsigned int length)
+{
+    std::string r;
+
+    return r;
+}
+
+
+}
+
diff --git a/src/sound/SoundFile.h b/src/sound/SoundFile.h
new file mode 100644
index 0000000..b048226
--- /dev/null
+++ b/src/sound/SoundFile.h
@@ -0,0 +1,155 @@
+// -*- c-indentation-style:"stroustrup" c-basic-offset: 4 -*- /*
+/*
+  Rosegarden
+  A sequencer and musical notation editor.
+
+  This program is Copyright 2000-2008
+  Guillaume Laurent   <glaurent@telegraph-road.org>,
+  Chris Cannam        <cannam@all-day-breakfast.com>,
+  Richard Bown        <bownie@bownie.com>
+
+  The moral right of the authors to claim authorship of this work
+  has been asserted.
+
+  This program is free software; you can redistribute it and/or
+  modify it under the terms of the GNU General Public License as
+  published by the Free Software Foundation; either version 2 of the
+  License, or (at your option) any later version.  See the file
+  COPYING included with this distribution for more information.
+*/
+
+
+#ifndef _SOUNDFILE_H_
+#define _SOUNDFILE_H_
+
+// SoundFile is an abstract base class defining behaviour for both
+// MidiFiles and AudioFiles.  The getBytes routine is buffered into
+// suitably sized chunks to prevent excessive file reads.
+// 
+//
+
+#include <iostream>
+#include <fstream>
+#include <string>
+
+#include "Exception.h"
+
+namespace Rosegarden
+{
+
+
+// Constants related to RIFF/WAV files
+//
+const std::string AUDIO_RIFF_ID = "RIFF";
+const std::string AUDIO_WAVE_ID = "WAVE";
+const std::string AUDIO_FORMAT_ID = "fmt ";   // Always four bytes
+
+const std::string AUDIO_BWF_ID = "bext";       // BWF chunk id
+const std::string AUDIO_BWF_PEAK_ID = "levl";  // BWF peak chunk id
+
+
+const float SAMPLE_MAX_8BIT  = (float)(0xff);
+const float SAMPLE_MAX_16BIT = (float)(0xffff/2);
+const float SAMPLE_MAX_24BIT = (float)(0xffffff/2);
+
+
+
+typedef unsigned char FileByte; 
+
+class SoundFile
+{
+public:
+    SoundFile(const std::string &fileName);
+    virtual ~SoundFile();
+
+    class BadSoundFileException : public Exception
+    {
+    public:
+        BadSoundFileException(std::string path) :
+            Exception("Bad sound file " + path), m_path(path) { }
+        BadSoundFileException(std::string path, std::string message) :
+            Exception("Bad sound file " + path + ": " + message), m_path(path) { }
+        BadSoundFileException(std::string path, std::string file, int line) :
+            Exception("Bad sound file " + path, file, line), m_path(path) { }
+
+        ~BadSoundFileException() throw() { }
+
+        std::string getPath() const { return m_path; }
+
+    private:
+        std::string m_path;
+    };
+
+    // All files should be able open, write and close
+    virtual bool open() = 0;
+    virtual bool write() = 0;
+    virtual void close() = 0;
+
+    std::string getShortFilename() const;
+    std::string getFilename() const { return m_fileName; }
+    void setFilename(const std::string &fileName) { m_fileName = fileName; }
+
+    // Useful methods that operate on our file data
+    //
+    int getIntegerFromLittleEndian(const std::string &s);
+    std::string getLittleEndianFromInteger(unsigned int value,
+                                           unsigned int length);
+
+    int getIntegerFromBigEndian(const std::string &s);
+    std::string getBigEndianFromInteger(unsigned int value,
+                                        unsigned int length);
+
+    // Buffered read - allow this to be public
+    //
+    std::string getBytes(unsigned int numberOfBytes);
+
+    // Return file size
+    //
+    unsigned int getSize() const { return m_fileSize; }
+
+    void resetStream() { m_inFile->seekg(0); m_inFile->clear(); }
+
+    // check EOF status
+    //
+    bool isEof() const 
+        { if (m_inFile) return m_inFile->eof(); else return true; }
+
+protected:
+    std::string m_fileName;
+
+    // get some bytes from an input stream - unbuffered as we can
+    // modify the file stream
+    std::string getBytes(std::ifstream *file, unsigned int numberOfBytes);
+
+    // Get n bytes from an input stream and write them into buffer.
+    // Return the actual number of bytes read.
+    size_t getBytes(std::ifstream *file, char *buffer, size_t n);
+
+    // write some bytes to an output stream
+    void putBytes(std::ofstream *file, const std::string outputString);
+
+    // write some bytes to an output stream
+    void putBytes(std::ofstream *file, const char *buffer, size_t n);
+
+    // Read buffering - define chunk size and buffer file reading
+    //
+    int            m_readChunkPtr;
+    int            m_readChunkSize;
+    std::string    m_readBuffer;
+
+    std::ifstream *m_inFile;
+    std::ofstream *m_outFile;
+    
+    bool           m_loseBuffer; // do we need to dump the read buffer
+                                 // and re-fill it?
+
+    unsigned int   m_fileSize;
+
+};
+
+}
+
+
+#endif // _SOUNDFILE_H_
+
+
diff --git a/src/sound/WAVAudioFile.cpp b/src/sound/WAVAudioFile.cpp
new file mode 100644
index 0000000..4e3b3bd
--- /dev/null
+++ b/src/sound/WAVAudioFile.cpp
@@ -0,0 +1,255 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+ 
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+ 
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+ 
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "WAVAudioFile.h"
+#include "RealTime.h"
+
+#if (__GNUC__ < 3)
+#include <strstream>
+#define stringstream strstream
+#else
+#include <sstream>
+#endif
+
+using std::cout;
+using std::cerr;
+using std::endl;
+
+//#define DEBUG_DECODE 1
+
+namespace Rosegarden
+{
+
+WAVAudioFile::WAVAudioFile(const unsigned int &id,
+                           const std::string &name,
+                           const std::string &fileName):
+        RIFFAudioFile(id, name, fileName)
+{
+    m_type = WAV;
+}
+
+WAVAudioFile::WAVAudioFile(const std::string &fileName,
+                           unsigned int channels = 1,
+                           unsigned int sampleRate = 48000,
+                           unsigned int bytesPerSecond = 6000,
+                           unsigned int bytesPerFrame = 2,
+                           unsigned int bitsPerSample = 16):
+        RIFFAudioFile(fileName, channels, sampleRate, bytesPerSecond, bytesPerFrame, bitsPerSample)
+{
+    m_type = WAV;
+}
+
+WAVAudioFile::~WAVAudioFile()
+{}
+
+bool
+WAVAudioFile::open()
+{
+    // if already open
+    if (m_inFile && (*m_inFile))
+        return true;
+
+    m_inFile = new std::ifstream(m_fileName.c_str(),
+                                 std::ios::in | std::ios::binary);
+
+    if (!(*m_inFile)) {
+        m_type = UNKNOWN;
+        return false;
+    }
+
+    // Get the file size and store it for comparison later
+    m_fileSize = m_fileInfo->size();
+
+    try {
+        parseHeader();
+    } catch (BadSoundFileException e) {
+        std::cerr << "ERROR: WAVAudioFile::open(): parseHeader: " << e.getMessage() << endl;
+        return false;
+    }
+
+    return true;
+}
+
+// Open the file for writing, write out the header and move
+// to the data chunk to accept samples.  We fill in all the
+// totals when we close().
+//
+bool
+WAVAudioFile::write()
+{
+    // close if we're open
+    if (m_outFile) {
+        m_outFile->close();
+        delete m_outFile;
+    }
+
+    // open for writing
+    m_outFile = new std::ofstream(m_fileName.c_str(),
+                                  std::ios::out | std::ios::binary);
+
+    if (!(*m_outFile))
+        return false;
+
+    // write out format header chunk and prepare for sample writing
+    //
+    writeFormatChunk();
+
+    return true;
+}
+
+void
+WAVAudioFile::close()
+{
+    if (m_outFile == 0)
+        return ;
+
+    m_outFile->seekp(0, std::ios::end);
+    unsigned int totalSize = m_outFile->tellp();
+
+    // seek to first length position
+    m_outFile->seekp(4, std::ios::beg);
+
+    // write complete file size minus 8 bytes to here
+    putBytes(m_outFile, getLittleEndianFromInteger(totalSize - 8, 4));
+
+    // reseek from start forward 40
+    m_outFile->seekp(40, std::ios::beg);
+
+    // write the data chunk size to end
+    putBytes(m_outFile, getLittleEndianFromInteger(totalSize - 44, 4));
+
+    m_outFile->close();
+
+    delete m_outFile;
+    m_outFile = 0;
+}
+
+// Set the AudioFile meta data according to WAV file format specification.
+//
+void
+WAVAudioFile::parseHeader()
+{
+    // Read the format chunk and populate the file data.  A plain WAV
+    // file only has this chunk.  Exceptions tumble through.
+    //
+    readFormatChunk();
+
+}
+
+std::streampos
+WAVAudioFile::getDataOffset()
+{
+    return 0;
+}
+
+bool
+WAVAudioFile::decode(const unsigned char *ubuf,
+                     size_t sourceBytes,
+                     size_t targetSampleRate,
+                     size_t targetChannels,
+                     size_t nframes,
+                     std::vector<float *> &target,
+                     bool adding)
+{
+    size_t sourceChannels = getChannels();
+    size_t sourceSampleRate = getSampleRate();
+    size_t fileFrames = sourceBytes / getBytesPerFrame();
+
+    int bitsPerSample = getBitsPerSample();
+    if (bitsPerSample != 8 &&
+            bitsPerSample != 16 &&
+            bitsPerSample != 24 &&
+            bitsPerSample != 32) { // 32-bit is IEEE-float (enforced in RIFFAudioFile)
+        std::cerr << "WAVAudioFile::decode: unsupported " <<
+        bitsPerSample << "-bit sample size" << std::endl;
+        return false;
+    }
+
+#ifdef DEBUG_DECODE
+    std::cerr << "WAVAudioFile::decode: " << sourceBytes << " bytes -> " << nframes << " frames, SSR " << getSampleRate() << ", TSR " << targetSampleRate << ", sch " << getChannels() << ", tch " << targetChannels << std::endl;
+#endif
+
+    // If we're reading a stereo file onto a mono target, we mix the
+    // two channels.  If we're reading mono to stereo, we duplicate
+    // the mono channel.  Otherwise if the numbers of channels differ,
+    // we just copy across the ones that do match and zero the rest.
+
+    bool reduceToMono = (targetChannels == 1 && sourceChannels == 2);
+
+    for (size_t ch = 0; ch < sourceChannels; ++ch) {
+
+        if (!reduceToMono || ch == 0) {
+            if (ch >= targetChannels)
+                break;
+            if (!adding)
+                memset(target[ch], 0, nframes * sizeof(float));
+        }
+
+        int tch = ch; // target channel for this data
+        if (reduceToMono && ch == 1) {
+            tch = 0;
+        }
+
+        float ratio = 1.0;
+        if (sourceSampleRate != targetSampleRate) {
+            ratio = float(sourceSampleRate) / float(targetSampleRate);
+        }
+
+        for (size_t i = 0; i < nframes; ++i) {
+
+            size_t j = i;
+            if (sourceSampleRate != targetSampleRate) {
+                j = size_t(i * ratio);
+            }
+            if (j >= fileFrames)
+                j = fileFrames - 1;
+
+	    float sample = convertBytesToSample
+		(&ubuf[(bitsPerSample / 8) * (ch + j * sourceChannels)]);
+
+            target[tch][i] += sample;
+        }
+    }
+
+    // Now deal with any excess target channels
+
+    for (int ch = sourceChannels; ch < targetChannels; ++ch) {
+        if (ch == 1 && targetChannels == 2) {
+            // copy mono to stereo
+            if (!adding) {
+                memcpy(target[ch], target[ch - 1], nframes * sizeof(float));
+            } else {
+                for (size_t i = 0; i < nframes; ++i) {
+                    target[ch][i] += target[ch - 1][i];
+                }
+            }
+        } else {
+            if (!adding) {
+                memset(target[ch], 0, nframes * sizeof(float));
+            }
+        }
+    }
+
+    return true;
+}
+
+
+}
diff --git a/src/sound/WAVAudioFile.h b/src/sound/WAVAudioFile.h
new file mode 100644
index 0000000..ec57ec6
--- /dev/null
+++ b/src/sound/WAVAudioFile.h
@@ -0,0 +1,93 @@
+// -*- c-basic-offset: 4 -*-
+
+/*
+    Rosegarden
+    A sequencer and musical notation editor.
+
+    This program is Copyright 2000-2008
+        Guillaume Laurent   <glaurent@telegraph-road.org>,
+        Chris Cannam        <cannam@all-day-breakfast.com>,
+        Richard Bown        <bownie@bownie.com>
+
+    The moral right of the authors to claim authorship of this work
+    has been asserted.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+
+// Specialisation of a RIFF file - the WAV defines a format chunk
+// holding audio file meta data and a data chunk with interleaved
+// sample bytes.
+//
+
+#include "RIFFAudioFile.h"
+
+
+#ifndef _WAVAUDIOFILE_H_
+#define _WAVAUDIOFILE_H_
+
+namespace Rosegarden
+{
+
+class WAVAudioFile : public RIFFAudioFile
+{
+public:
+    WAVAudioFile(const unsigned int &id,
+                 const std::string &name,
+                 const std::string &fileName);
+
+    WAVAudioFile(const std::string &fileName,
+                  unsigned int channels,
+                  unsigned int sampleRate,
+                  unsigned int bytesPerSecond,
+                  unsigned int bytesPerSample,
+                  unsigned int bitsPerSample);
+
+    ~WAVAudioFile();
+
+    // Override these methods for the WAV
+    //
+    virtual bool open();
+    virtual bool write();
+    virtual void close();
+
+    // Decode and de-interleave the given samples that were retrieved
+    // from this file or another with the same format as it.  Place
+    // the results in the given float buffer.  Return true for
+    // success.  This function does crappy resampling if necessary.
+    // 
+    virtual bool decode(const unsigned char *sourceData,
+                        size_t sourceBytes,
+                        size_t targetSampleRate,
+                        size_t targetChannels,
+                        size_t targetFrames,
+                        std::vector<float *> &targetData,
+                        bool addToResultBuffers = false);
+
+    // Get all header information
+    //
+    void parseHeader();
+
+    // Offset to start of sample data
+    //
+    virtual std::streampos getDataOffset();
+
+    // Peak file name
+    //
+    virtual std::string getPeakFilename()
+        { return (m_fileName + std::string(".pk")); }
+
+
+protected:
+
+};
+
+}
+
+
+#endif // _WAVAUDIOFILE_H_