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// -*- 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 ¤tTime);
/**
* 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, TQString 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);
TQStringList getPluginPrograms(InstrumentId, int);
TQString getPluginProgram(InstrumentId, int);
TQString getPluginProgram(InstrumentId, int, int, int);
unsigned long getPluginProgram(InstrumentId, int, TQString);
void setPluginProgram(InstrumentId, int, TQString);
TQString configurePlugin(InstrumentId, int, TQString, TQString);
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 ¤tTime);
/**
* 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 ¤tTime);
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
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