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authortoma <toma@283d02a7-25f6-0310-bc7c-ecb5cbfe19da>2009-11-25 17:56:58 +0000
committertoma <toma@283d02a7-25f6-0310-bc7c-ecb5cbfe19da>2009-11-25 17:56:58 +0000
commite2de64d6f1beb9e492daf5b886e19933c1fa41dd (patch)
tree9047cf9e6b5c43878d5bf82660adae77ceee097a /mpeglib/lib/util/audio/audioIO_AIX.cpp
downloadtdemultimedia-e2de64d6f1beb9e492daf5b886e19933c1fa41dd.tar.gz
tdemultimedia-e2de64d6f1beb9e492daf5b886e19933c1fa41dd.zip
Copy the KDE 3.5 branch to branches/trinity for new KDE 3.5 features.
BUG:215923 git-svn-id: svn://anonsvn.kde.org/home/kde/branches/trinity/kdemultimedia@1054174 283d02a7-25f6-0310-bc7c-ecb5cbfe19da
Diffstat (limited to 'mpeglib/lib/util/audio/audioIO_AIX.cpp')
-rw-r--r--mpeglib/lib/util/audio/audioIO_AIX.cpp533
1 files changed, 533 insertions, 0 deletions
diff --git a/mpeglib/lib/util/audio/audioIO_AIX.cpp b/mpeglib/lib/util/audio/audioIO_AIX.cpp
new file mode 100644
index 00000000..15316852
--- /dev/null
+++ b/mpeglib/lib/util/audio/audioIO_AIX.cpp
@@ -0,0 +1,533 @@
+/*
+ * AIX audio - griff@acm.org 02aug2000
+ * tested on 43P 260 with builtin audio
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <errno.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/time.h>
+#include <sys/ioctl.h>
+#include <sys/stat.h>
+/* A conflict within AIX 4.3.3 <sys/> headers and probably others as well.
+ * I guess nobody ever uses audio... Shame over AIX header files. */
+#include <sys/machine.h>
+#undef BIG_ENDIAN
+#include <sys/audio.h>
+
+static int audio_fd;
+
+static void debugUpdate( unsigned long& flags, long& bsize );
+
+#ifndef AUDIO_BIG_ENDIAN
+#define AUDIO_BIG_ENDIAN BIG_ENDIAN
+#endif
+
+
+
+int audioConstruct() {
+ printf("audioConstruct AIX ********\n");
+ audio_fd=-1;
+ return true;
+}
+
+
+void audioDestruct() {
+
+}
+
+int audioOpen()
+{
+ char devname[14];
+ for ( int dev=0; dev<4; dev++ )
+ {
+ for ( int chan=1; chan<8; chan++ )
+ {
+ sprintf(devname,"/dev/paud%d/%d",dev,chan);
+ audio_fd = open (devname, O_WRONLY, 0);
+ if ( audio_fd >= 0 )
+ {
+ return 1;
+ }
+ sprintf(devname,"/dev/baud%d/%d",dev,chan);
+ audio_fd = open (devname, O_WRONLY, 0);
+ if ( audio_fd >= 0 )
+ {
+ return 1;
+ }
+ }
+ }
+
+ fprintf(stderr, "Could not open AIX audio device, faking\n" );
+ return 1;
+}
+
+int getAudioBufferSize()
+{
+ audio_buffer paud_bufinfo;
+
+ if( audio_fd < 0 ) return 1024*65;
+
+ if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 )
+ {
+ perror("ioctl getAudioBufferSize using default");
+ return 1024*65;
+ }
+
+ /*
+ * Do you need the total capacity or the current capacity?
+ * This is the total capacity:
+ */
+ return paud_bufinfo.write_buf_cap;
+ /*
+ * This is the current capacity:
+ * return (paud_bufinfo.write_buf_cap - paud_bufinfo.write_buf_size);
+ */
+}
+
+void audioInit(int sampleSize,int frequency, int stereo, int sign, int bigendian )
+{
+ // int format;
+ int bytes_per_sample;
+ audio_init paud_init;
+ audio_buffer paud_bufinfo;
+ // audio_status paud_status;
+ audio_control paud_control;
+ audio_change paud_change;
+
+ if( audio_fd < 0 ) return;
+
+ /*
+ * We can't set the buffer size - just ask the device for the maximum
+ * that we can have.
+ */
+ if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 )
+ {
+ perror("Couldn't get audio buffer information");
+ return;
+ }
+
+ /*
+ * Fields in the audio_init structure:
+ *
+ * Ignored by us:
+ *
+ * paud.loadpath[LOAD_PATH]; * DSP code to load, MWave chip only?
+ * paud.slot_number; * slot number of the adapter
+ * paud.device_id; * adapter identification number
+ *
+ * Input:
+ *
+ * paud.srate; * the sampling rate in Hz
+ * paud.bits_per_sample; * 8, 16, 32, ...
+ * paud.bsize; * block size for this rate
+ * paud.mode; * ADPCM, PCM, MU_LAW, A_LAW, SOURCE_MIX
+ * paud.channels; * 1=mono, 2=stereo
+ * paud.flags; * FIXED - fixed length data
+ * * LEFT_ALIGNED, RIGHT_ALIGNED (var len only)
+ * * TWOS_COMPLEMENT - 2's complement data
+ * * SIGNED - signed? comment seems wrong in sys/audio.h
+ * * BIG_ENDIAN
+ * paud.operation; * PLAY, RECORD
+ *
+ * Output:
+ *
+ * paud.flags; * PITCH - pitch is supported
+ * * INPUT - input is supported
+ * * OUTPUT - output is supported
+ * * MONITOR - monitor is supported
+ * * VOLUME - volume is supported
+ * * VOLUME_DELAY - volume delay is supported
+ * * BALANCE - balance is supported
+ * * BALANCE_DELAY - balance delay is supported
+ * * TREBLE - treble control is supported
+ * * BASS - bass control is supported
+ * * BESTFIT_PROVIDED - best fit returned
+ * * LOAD_CODE - DSP load needed
+ * paud.rc; * NO_PLAY - DSP code can't do play requests
+ * * NO_RECORD - DSP code can't do record requests
+ * * INVALID_REQUEST - request was invalid
+ * * CONFLICT - conflict with open's flags
+ * * OVERLOADED - out of DSP MIPS or memory
+ * paud.position_resolution; * smallest increment for position
+ */
+
+ paud_init.srate = frequency;
+ paud_init.mode = PCM;
+ paud_init.operation = PLAY;
+ paud_init.channels = (stereo?2:1);
+
+ /*
+ * options in AIX:
+ * paud_init.bits_per_sample: 8 | 16
+ * paud_init.flags: AUDIO_BIG_ENDIAN (not used here)
+ * SIGNED (always used here)
+ * TWOS_COMPLEMENT (always on for Linux dsp porting?)
+ * FIXED <- that's right for SDL
+ * or LEFT_ALIGNED <- that's right for mpeglib
+ * or RIGHT_ALIGNED
+ * paud_init.bsize: sample byte size,
+ * bits_per_sample * (stereo?2:1) - for SDL
+ * bits_per_sample * (stereo?2:1) * 2 - for mpeglib
+ */
+ if ( sampleSize == 8 )
+ {
+ /* AFMT_S8 in linux dsp */
+ bytes_per_sample = 2; // why not 1 ?
+ paud_init.bits_per_sample = 8;
+ paud_init.flags = TWOS_COMPLEMENT | LEFT_ALIGNED;
+ }
+ else
+ {
+ /* AFMT_S16_LE in linux dsp */
+ bytes_per_sample = 4; // why not 2 ?
+ paud_init.bits_per_sample = 16;
+ paud_init.flags = TWOS_COMPLEMENT | LEFT_ALIGNED;
+ }
+ if( sign ) paud_init.flags |= SIGNED;
+ if( bigendian ) paud_init.flags |= AUDIO_BIG_ENDIAN;
+
+ paud_init.bsize = bytes_per_sample * (stereo?2:1);
+
+#if 0
+ debugUpdate(paud_init.flags, paud_init.bsize);
+
+ printf("CG: sampleSize = %d\n", sampleSize);
+ printf("CG: frequency = %d\n", frequency);
+ printf("CG: stereo = %s\n", (stereo)?"y":"n");
+ printf("CG: mode = %s\n", "PCM");
+ printf("CG: channels = %d\n", paud_init.channels);
+ printf("CG: bsize = %d\n", paud_init.bsize);
+ printf("CG: bits_per_sample = %d\n", paud_init.bits_per_sample);
+ printf("CG: flags & BIG_ENDIAN = %s\n", ((paud_init.flags&AUDIO_BIG_ENDIAN)?"y":"n"));
+ printf("CG: flags & SIGNED = %s\n", ((paud_init.flags&SIGNED)?"y":"n"));
+ printf("CG: flags & TWOS_COMPLEMENT = %s\n", ((paud_init.flags&TWOS_COMPLEMENT)?"y":"n"));
+ printf("CG: flags & FIXED = %s\n", ((paud_init.flags&FIXED)?"y":"n"));
+ printf("CG: flags & LEFT_ALIGNED = %s\n", ((paud_init.flags&LEFT_ALIGNED)?"y":"n"));
+ printf("CG: flags & RIGHT_ALIGNED = %s\n", ((paud_init.flags&RIGHT_ALIGNED)?"y":"n"));
+#endif
+
+ /*
+ * We know the buffer size and the max number of subsequent writes
+ * that can be pending. If more than one can pend, allow the application
+ * to do something like double buffering between our write buffer and
+ * the device's own buffer that we are filling with write() anyway.
+ *
+ * We can calculate the number of samples that fit into the audio
+ * device buffer if that is necessary:
+ *
+ * samples_capacity = paud_bufinfo.write_buf_cap
+ * / bytes_per_sample
+ * / (stereo?2:1);
+ * if ( paud_bufinfo.request_buf_cap != 1 ) samples_capacity /= 2;
+ */
+
+ /*
+ * The AIX paud device init can't modify the values of the audio_init
+ * structure that we pass to it. So we don't need any recalculation
+ * of this stuff and no reinit call as in linux SDL dsp and dma code.
+ *
+ * /dev/paud supports all of the encoding formats, so we don't need
+ * to do anything like reopening the device, either.
+ */
+ if ( ioctl(audio_fd, AUDIO_INIT, &paud_init) < 0 )
+ {
+ switch ( paud_init.rc )
+ {
+ case 1 :
+ perror("Couldn't set audio format: DSP can't do play requests");
+ return;
+ break;
+ case 2 :
+ perror("Couldn't set audio format: DSP can't do record requests");
+ return;
+ break;
+ case 4 :
+ perror("Couldn't set audio format: request was invalid");
+ return;
+ break;
+ case 5 :
+ perror("Couldn't set audio format: conflict with open's flags");
+ return;
+ break;
+ case 6 :
+ perror("Couldn't set audio format: out of DSP MIPS or memory");
+ return;
+ break;
+ default :
+ perror("Couldn't set audio format: not documented in sys/audio.h");
+ return;
+ break;
+ }
+ }
+
+ /*
+ * Set some parameters: full volume, first speaker that we can find.
+ * Ignore the other settings for now.
+ */
+ paud_change.input = AUDIO_IGNORE; /* the new input source */
+ paud_change.output = OUTPUT_1;
+ /* EXTERNAL_SPEAKER,
+ * INTERNAL_SPEAKER,
+ * OUTPUT_1 */
+ paud_change.monitor = AUDIO_IGNORE; /* the new monitor state */
+ paud_change.volume = 0x7fffffff; /* volume level [0-0x7fffffff] */
+ paud_change.volume_delay = AUDIO_IGNORE; /* the new volume delay */
+ paud_change.balance = 0x3fffffff; /* the new balance */
+ paud_change.balance_delay = AUDIO_IGNORE; /* the new balance delay */
+ paud_change.treble = AUDIO_IGNORE; /* the new treble state */
+ paud_change.bass = AUDIO_IGNORE; /* the new bass state */
+ paud_change.pitch = AUDIO_IGNORE; /* the new pitch state */
+
+ paud_control.ioctl_request = AUDIO_CHANGE;
+ paud_control.request_info = (char*)&paud_change;
+ if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 )
+ {
+ perror("Can't change audio display settings (ignoring)" );
+ }
+
+ /*
+ * Tell the device to expect data. Actual start will wait for
+ * the first write() call.
+ */
+ paud_control.ioctl_request = AUDIO_START;
+ paud_control.position = 0;
+ if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 )
+ {
+ perror("Can't start audio play");
+ return;
+ }
+}
+
+
+void audioSetVolume(int volume)
+{
+ long vol = (long)(volume/100.0) * 0x7fffffff;
+ if( audio_fd < 0 ) return;
+
+ audio_control paud_control;
+ audio_change paud_change;
+
+ paud_change.input = AUDIO_IGNORE; /* the new input source */
+ paud_change.output = OUTPUT_1; /* EXTERNAL_SPEAKER,INTERNAL_SPEAKER,
+ OUTPUT_1 */
+ paud_change.monitor = AUDIO_IGNORE; /* the new monitor state */
+ paud_change.volume = vol; /* volume level [0-0x7fffffff] */
+ paud_change.volume_delay = AUDIO_IGNORE; /* the new volume delay */
+ paud_change.balance = AUDIO_IGNORE; /* the new balance */
+ paud_change.balance_delay = AUDIO_IGNORE; /* the new balance delay */
+ paud_change.treble = AUDIO_IGNORE; /* the new treble state */
+ paud_change.bass = AUDIO_IGNORE; /* the new bass state */
+ paud_change.pitch = AUDIO_IGNORE; /* the new pitch state */
+
+ paud_control.ioctl_request = AUDIO_CHANGE;
+ paud_control.request_info = (char*)&paud_change;
+
+ if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 )
+ {
+ perror("Change audio volume failed");
+ }
+}
+
+void audioFlush()
+{
+ if( audio_fd < 0 ) return;
+
+ if ( ioctl(audio_fd, AUDIO_WAIT, NULL) < 0 )
+ {
+ perror("Flush audio buffers failed");
+ }
+}
+
+void audioClose()
+{
+ if( audio_fd < 0 ) return;
+
+ if ( ioctl(audio_fd, AUDIO_WAIT, NULL) < 0 )
+ {
+ perror("Flush audio buffers failed");
+ }
+ close(audio_fd);
+}
+
+int audioWrite(char *buffer, int count)
+{
+ int written = write(audio_fd, buffer, count);
+ if( written < count )
+ {
+ return count;
+ }
+
+ return written;
+}
+
+int
+getAudioFd()
+{
+ return audio_fd;
+}
+
+int mixerOpen()
+{
+ return true;
+}
+
+void mixerClose()
+{
+}
+
+void mixerSetVolume(int leftVolume,int rightVolume)
+{
+ long balance;
+
+ if( audio_fd < 0 ) return;
+
+ balance = 2 * (leftVolume-rightVolume) / (leftVolume+rightVolume);
+ balance = 0x3fffffff + balance*0x3fffffff;
+
+ audio_control paud_control;
+ audio_change paud_change;
+
+ paud_change.input = AUDIO_IGNORE; /* the new input source */
+ paud_change.output = OUTPUT_1; /* EXTERNAL_SPEAKER,INTERNAL_SPEAKER,
+ OUTPUT_1 */
+ paud_change.monitor = AUDIO_IGNORE; /* the new monitor state */
+ paud_change.volume = AUDIO_IGNORE; /* volume level [0-0x7fffffff] */
+ paud_change.volume_delay = AUDIO_IGNORE; /* the new volume delay */
+ paud_change.balance = balance; /* the new balance */
+ paud_change.balance_delay = AUDIO_IGNORE; /* the new balance delay */
+ paud_change.treble = AUDIO_IGNORE; /* the new treble state */
+ paud_change.bass = AUDIO_IGNORE; /* the new bass state */
+ paud_change.pitch = AUDIO_IGNORE; /* the new pitch state */
+
+ paud_control.ioctl_request = AUDIO_CHANGE;
+ paud_control.request_info = (char*)&paud_change;
+
+ if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 )
+ {
+ perror("Change audio volume failed");
+ }
+}
+
+static void debugUpdate( unsigned long& flags, long& bsize )
+{
+ const char* g;
+
+ g = getenv("AUDIO_BIG_ENDIAN");
+ if ( g )
+ {
+ int i = atoi(g);
+ if ( i == 1 )
+ {
+ flags |= AUDIO_BIG_ENDIAN;
+ }
+ else if ( i == 0 )
+ {
+ flags &= ~AUDIO_BIG_ENDIAN;
+ }
+ else
+ {
+ printf("CG: bad AUDIO_BIG_ENDIAN env variable %s\n", g);
+ }
+ }
+
+ g = getenv("SIGNED");
+ if ( g )
+ {
+ int i = atoi(g);
+ if ( i == 1 )
+ {
+ flags |= SIGNED;
+ }
+ else if ( i == 0 )
+ {
+ flags &= ~SIGNED;
+ }
+ else
+ {
+ printf("CG: bad SIGNED env variable %s\n", g);
+ }
+ }
+
+ g = getenv("TWOS_COMPLEMENT");
+ if ( g )
+ {
+ int i = atoi(g);
+ if ( i == 1 )
+ {
+ flags |= TWOS_COMPLEMENT;
+ }
+ else if ( i == 0 )
+ {
+ flags &= ~TWOS_COMPLEMENT;
+ }
+ else
+ {
+ printf("CG: bad TWOS_COMPLEMENT env variable %s\n", g);
+ }
+ }
+
+ g = getenv("FIXED");
+ if ( g )
+ {
+ int i = atoi(g);
+ if ( i == 1 )
+ {
+ flags |= FIXED;
+ }
+ else if ( i == 0 )
+ {
+ flags &= ~FIXED;
+ }
+ else
+ {
+ printf("CG: bad FIXED env variable %s\n", g);
+ }
+ }
+
+ g = getenv("LEFT_ALIGNED");
+ if ( g )
+ {
+ int i = atoi(g);
+ if ( i == 1 )
+ {
+ flags |= LEFT_ALIGNED;
+ }
+ else if ( i == 0 )
+ {
+ flags &= ~LEFT_ALIGNED;
+ }
+ else
+ {
+ printf("CG: bad LEFT_ALIGNED env variable %s\n", g);
+ }
+ }
+
+ g = getenv("RIGHT_ALIGNED");
+ if ( g )
+ {
+ int i = atoi(g);
+ if ( i == 1 )
+ {
+ flags |= RIGHT_ALIGNED;
+ }
+ else if ( i == 0 )
+ {
+ flags &= ~RIGHT_ALIGNED;
+ }
+ else
+ {
+ printf("CG: bad RIGHT_ALIGNED env variable %s\n", g);
+ }
+ }
+
+ g = getenv("BSIZE");
+ if ( g )
+ {
+ bsize = atoi(g);
+ }
+}
+