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SDL_paudio.c
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/*
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SDL - Simple DirectMedia Layer
Copyright (C) 1997-2006 Sam Lantinga
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This library is free software; you can redistribute it and/or
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modify it under the terms of the GNU Lesser General Public
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License as published by the Free Software Foundation; either
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version 2.1 of the License, or (at your option) any later version.
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This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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Lesser General Public License for more details.
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You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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Carsten Griwodz
griff@kom.tu-darmstadt.de
based on linux/SDL_dspaudio.c by Sam Lantinga
*/
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#include "SDL_config.h"
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/* Allow access to a raw mixing buffer */
#include <errno.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/time.h>
#include <sys/ioctl.h>
#include <sys/stat.h>
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#include "SDL_timer.h"
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#include "SDL_audio.h"
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#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "../SDL_audiodev_c.h"
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#include "SDL_paudio.h"
#define DEBUG_AUDIO 1
/* 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>
/* The tag name used by paud audio */
#define Paud_DRIVER_NAME "paud"
/* Open the audio device for playback, and don't block if busy */
/* #define OPEN_FLAGS (O_WRONLY|O_NONBLOCK) */
#define OPEN_FLAGS O_WRONLY
/* Audio driver functions */
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static int Paud_OpenAudio(_THIS, SDL_AudioSpec * spec);
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static void Paud_WaitAudio(_THIS);
static void Paud_PlayAudio(_THIS);
static Uint8 *Paud_GetAudioBuf(_THIS);
static void Paud_CloseAudio(_THIS);
/* Audio driver bootstrap functions */
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static int
Audio_Available(void)
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{
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int fd;
int available;
available = 0;
fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0);
if (fd >= 0) {
available = 1;
close(fd);
}
return (available);
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}
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static void
Audio_DeleteDevice(SDL_AudioDevice * device)
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{
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SDL_free(device->hidden);
SDL_free(device);
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}
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static SDL_AudioDevice *
Audio_CreateDevice(int devindex)
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{
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SDL_AudioDevice *this;
/* Initialize all variables that we clean on shutdown */
this = (SDL_AudioDevice *) SDL_malloc(sizeof(SDL_AudioDevice));
if (this) {
SDL_memset(this, 0, (sizeof *this));
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
}
if ((this == NULL) || (this->hidden == NULL)) {
SDL_OutOfMemory();
if (this) {
SDL_free(this);
}
return (0);
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
audio_fd = -1;
/* Set the function pointers */
this->OpenAudio = Paud_OpenAudio;
this->WaitAudio = Paud_WaitAudio;
this->PlayAudio = Paud_PlayAudio;
this->GetAudioBuf = Paud_GetAudioBuf;
this->CloseAudio = Paud_CloseAudio;
this->free = Audio_DeleteDevice;
return this;
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}
AudioBootStrap Paud_bootstrap = {
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Paud_DRIVER_NAME, "AIX Paudio",
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Audio_Available, Audio_CreateDevice, 0
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};
/* This function waits until it is possible to write a full sound buffer */
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static void
Paud_WaitAudio(_THIS)
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{
fd_set fdset;
/* See if we need to use timed audio synchronization */
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if (frame_ticks) {
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/* Use timer for general audio synchronization */
Sint32 ticks;
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ticks = ((Sint32) (next_frame - SDL_GetTicks())) - FUDGE_TICKS;
if (ticks > 0) {
SDL_Delay(ticks);
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}
} else {
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audio_buffer paud_bufinfo;
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/* Use select() for audio synchronization */
struct timeval timeout;
FD_ZERO(&fdset);
FD_SET(audio_fd, &fdset);
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if (ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0) {
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#ifdef DEBUG_AUDIO
fprintf(stderr, "Couldn't get audio buffer information\n");
#endif
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timeout.tv_sec = 10;
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timeout.tv_usec = 0;
} else {
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long ms_in_buf = paud_bufinfo.write_buf_time;
timeout.tv_sec = ms_in_buf / 1000;
ms_in_buf = ms_in_buf - timeout.tv_sec * 1000;
timeout.tv_usec = ms_in_buf * 1000;
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#ifdef DEBUG_AUDIO
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fprintf(stderr,
"Waiting for write_buf_time=%ld,%ld\n",
timeout.tv_sec, timeout.tv_usec);
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#endif
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}
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#ifdef DEBUG_AUDIO
fprintf(stderr, "Waiting for audio to get ready\n");
#endif
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if (select(audio_fd + 1, NULL, &fdset, NULL, &timeout) <= 0) {
const char *message =
"Audio timeout - buggy audio driver? (disabled)";
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/*
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* In general we should never print to the screen,
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* but in this case we have no other way of letting
* the user know what happened.
*/
fprintf(stderr, "SDL: %s - %s\n", strerror(errno), message);
this->enabled = 0;
/* Don't try to close - may hang */
audio_fd = -1;
#ifdef DEBUG_AUDIO
fprintf(stderr, "Done disabling audio\n");
#endif
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Ready!\n");
#endif
}
}
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static void
Paud_PlayAudio(_THIS)
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{
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int written;
/* Write the audio data, checking for EAGAIN on broken audio drivers */
do {
written = write(audio_fd, mixbuf, mixlen);
if ((written < 0) && ((errno == 0) || (errno == EAGAIN))) {
SDL_Delay(1); /* Let a little CPU time go by */
}
}
while ((written < 0) &&
((errno == 0) || (errno == EAGAIN) || (errno == EINTR)));
/* If timer synchronization is enabled, set the next write frame */
if (frame_ticks) {
next_frame += frame_ticks;
}
/* If we couldn't write, assume fatal error for now */
if (written < 0) {
this->enabled = 0;
}
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#ifdef DEBUG_AUDIO
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fprintf(stderr, "Wrote %d bytes of audio data\n", written);
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#endif
}
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static Uint8 *
Paud_GetAudioBuf(_THIS)
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{
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return mixbuf;
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}
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static void
Paud_CloseAudio(_THIS)
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{
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if (mixbuf != NULL) {
SDL_FreeAudioMem(mixbuf);
mixbuf = NULL;
}
if (audio_fd >= 0) {
close(audio_fd);
audio_fd = -1;
}
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}
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static int
Paud_OpenAudio(_THIS, SDL_AudioSpec * spec)
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{
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char audiodev[1024];
int format;
int bytes_per_sample;
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SDL_AudioFormat test_format;
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audio_init paud_init;
audio_buffer paud_bufinfo;
audio_status paud_status;
audio_control paud_control;
audio_change paud_change;
/* Reset the timer synchronization flag */
frame_ticks = 0.0;
/* Open the audio device */
audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
if (audio_fd < 0) {
SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
return -1;
}
/*
* 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) {
SDL_SetError("Couldn't get audio buffer information");
return -1;
}
mixbuf = NULL;
if (spec->channels > 1)
spec->channels = 2;
else
spec->channels = 1;
/*
* 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 = spec->freq;
paud_init.mode = PCM;
paud_init.operation = PLAY;
paud_init.channels = spec->channels;
/* Try for a closest match on audio format */
format = 0;
for (test_format = SDL_FirstAudioFormat(spec->format);
!format && test_format;) {
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#ifdef DEBUG_AUDIO
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fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
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#endif
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switch (test_format) {
case AUDIO_U8:
bytes_per_sample = 1;
paud_init.bits_per_sample = 8;
paud_init.flags = TWOS_COMPLEMENT | FIXED;
format = 1;
break;
case AUDIO_S8:
bytes_per_sample = 1;
paud_init.bits_per_sample = 8;
paud_init.flags = SIGNED | TWOS_COMPLEMENT | FIXED;
format = 1;
break;
case AUDIO_S16LSB:
bytes_per_sample = 2;
paud_init.bits_per_sample = 16;
paud_init.flags = SIGNED | TWOS_COMPLEMENT | FIXED;
format = 1;
break;
case AUDIO_S16MSB:
bytes_per_sample = 2;
paud_init.bits_per_sample = 16;
paud_init.flags = BIG_ENDIAN | SIGNED | TWOS_COMPLEMENT | FIXED;
format = 1;
break;
case AUDIO_U16LSB:
bytes_per_sample = 2;
paud_init.bits_per_sample = 16;
paud_init.flags = TWOS_COMPLEMENT | FIXED;
format = 1;
break;
case AUDIO_U16MSB:
bytes_per_sample = 2;
paud_init.bits_per_sample = 16;
paud_init.flags = BIG_ENDIAN | TWOS_COMPLEMENT | FIXED;
format = 1;
break;
default:
break;
}
if (!format) {
test_format = SDL_NextAudioFormat();
}
}
if (format == 0) {
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#ifdef DEBUG_AUDIO
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fprintf(stderr, "Couldn't find any hardware audio formats\n");
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#endif
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SDL_SetError("Couldn't find any hardware audio formats");
return -1;
}
spec->format = test_format;
/*
* 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 calculate spec->samples like this because SDL_CalculateAudioSpec()
* will give put paud_bufinfo.write_buf_cap (or paud_bufinfo.write_buf_cap/2)
* into spec->size in return.
*/
if (paud_bufinfo.request_buf_cap == 1) {
spec->samples = paud_bufinfo.write_buf_cap
/ bytes_per_sample / spec->channels;
} else {
spec->samples = paud_bufinfo.write_buf_cap
/ bytes_per_sample / spec->channels / 2;
}
paud_init.bsize = bytes_per_sample * spec->channels;
SDL_CalculateAudioSpec(spec);
/*
* 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 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:
SDL_SetError
("Couldn't set audio format: DSP can't do play requests");
return -1;
break;
case 2:
SDL_SetError
("Couldn't set audio format: DSP can't do record requests");
return -1;
break;
case 4:
SDL_SetError("Couldn't set audio format: request was invalid");
return -1;
break;
case 5:
SDL_SetError
("Couldn't set audio format: conflict with open's flags");
return -1;
break;
case 6:
SDL_SetError
("Couldn't set audio format: out of DSP MIPS or memory");
return -1;
break;
default:
SDL_SetError
("Couldn't set audio format: not documented in sys/audio.h");
return -1;
break;
}
}
/* Allocate mixing buffer */
mixlen = spec->size;
mixbuf = (Uint8 *) SDL_AllocAudioMem(mixlen);
if (mixbuf == NULL) {
return -1;
}
SDL_memset(mixbuf, spec->silence, spec->size);
/*
* Set some paramters: 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) {
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#ifdef DEBUG_AUDIO
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fprintf(stderr, "Can't change audio display settings\n");
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#endif
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}
/*
* 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) {
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#ifdef DEBUG_AUDIO
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fprintf(stderr, "Can't start audio play\n");
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#endif
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SDL_SetError("Can't start audio play");
return -1;
}
/* Check to see if we need to use select() workaround */
{
char *workaround;
workaround = SDL_getenv("SDL_DSP_NOSELECT");
if (workaround) {
frame_ticks = (float) (spec->samples * 1000) / spec->freq;
next_frame = SDL_GetTicks() + frame_ticks;
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}
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}
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/* Get the parent process id (we're the parent of the audio thread) */
parent = getpid();
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/* We're ready to rock and roll. :-) */
return 0;
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}
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/* vi: set ts=4 sw=4 expandtab: */