ref: ed6e41bccb569d38d0cba4dbf84304f5f8c6ed1f
dir: /src/coreaudio.c/
/* AudioCore sound handler
*
* Copyright 2008 Chris Bagwell And Sundry Contributors
*/
#include "sox_i.h"
#include <CoreAudio/CoreAudio.h>
#include <pthread.h>
typedef struct {
AudioDeviceID adid;
pthread_mutex_t mutex;
pthread_cond_t cond;
int device_started;
size_t buf_size;
size_t buf_offset;
float *buffer;
} priv_t;
static OSStatus PlaybackIOProc(AudioDeviceID inDevice UNUSED,
const AudioTimeStamp *inNow UNUSED,
const AudioBufferList *inInputData UNUSED,
const AudioTimeStamp *inInputTime UNUSED,
AudioBufferList *outOutputData,
const AudioTimeStamp *inOutputTime UNUSED,
void *inClientData)
{
sox_format_t *ft = (sox_format_t *)inClientData;
priv_t *ac = (priv_t *)ft->priv;
float *buf = outOutputData->mBuffers[0].mData;
pthread_mutex_lock(&ac->mutex);
memcpy(buf, ac->buffer, ac->buf_offset);
ac->buf_offset = 0;
pthread_mutex_unlock(&ac->mutex);
pthread_cond_signal(&ac->cond);
return kAudioHardwareNoError;
}
static OSStatus RecIOProc(AudioDeviceID inDevice UNUSED,
const AudioTimeStamp *inNow UNUSED,
const AudioBufferList *inInputData,
const AudioTimeStamp *inInputTime UNUSED,
AudioBufferList *outOutputData UNUSED,
const AudioTimeStamp *inOutputTime UNUSED,
void *inClientData)
{
sox_format_t *ft = (sox_format_t *)inClientData;
priv_t *ac = (priv_t *)ft->priv;
float *buf = inInputData->mBuffers[0].mData;
size_t buflen = inInputData->mBuffers[0].mDataByteSize;
float *destbuf = (float *)((unsigned char *)ac->buffer + ac->buf_offset);
int i;
/* mDataByteSize may be non-zero even when mData is NULL, but that is not an error */
if (buf == NULL)
return kAudioHardwareNoError;
if (buflen > (ac->buf_size + ac->buf_offset))
buflen = ac->buf_size - ac->buf_offset;
pthread_mutex_lock(&ac->mutex);
for (i = 0; i < (int)(buflen / sizeof(float)); i += 2) {
destbuf[i] = buf[i];
destbuf[i + 1] = buf[i + 1];
ac->buf_offset += sizeof(float) * 2;
}
pthread_mutex_unlock(&ac->mutex);
pthread_cond_signal(&ac->cond);
return kAudioHardwareNoError;
}
static int setup(sox_format_t *ft, int is_input)
{
priv_t *ac = (priv_t *)ft->priv;
OSStatus status;
UInt32 property_size;
struct AudioStreamBasicDescription stream_desc;
int32_t buf_size;
int rc;
property_size = sizeof(ac->adid);
if (is_input)
status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice,
&property_size, &ac->adid);
else
status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
&property_size, &ac->adid);
if (status || ac->adid == kAudioDeviceUnknown)
{
lsx_fail_errno(ft, SOX_EPERM, "can not open audio device");
return SOX_EOF;
}
/* Query device to get initial values */
property_size = sizeof(struct AudioStreamBasicDescription);
status = AudioDeviceGetProperty(ac->adid, 0, is_input,
kAudioDevicePropertyStreamFormat,
&property_size, &stream_desc);
if (status)
{
lsx_fail_errno(ft, SOX_EPERM, "can not get audio device properties");
return SOX_EOF;
}
if (!(stream_desc.mFormatFlags & kLinearPCMFormatFlagIsFloat))
{
lsx_fail_errno(ft, SOX_EPERM, "audio device does not accept floats");
return SOX_EOF;
}
/* OS X effectively only supports these values. */
ft->signal.channels = 2;
ft->signal.rate = 44100;
ft->encoding.bits_per_sample = 32;
/* TODO: My limited experience with hardware can only get floats working which a fixed sample
* rate and stereo. I know that is a limitiation of audio device I have so this may not be
* standard operating orders. If some hardware supports setting sample rates and channel counts
* then should do that over resampling and mixing.
*/
#if 0
stream_desc.mSampleRate = ft->signal.rate;
stream_desc.mChannelsPerFrame = ft->signal.channels;
/* Write them back */
property_size = sizeof(struct AudioStreamBasicDescription);
status = AudioDeviceSetProperty(ac->adid, NULL, 0, is_input,
kAudioDevicePropertyStreamFormat,
property_size, &stream_desc);
if (status)
{
lsx_fail_errno(ft, SOX_EPERM, "can not set audio device properties");
return SOX_EOF;
}
/* Query device to see if it worked */
property_size = sizeof(struct AudioStreamBasicDescription);
status = AudioDeviceGetProperty(ac->adid, 0, is_input,
kAudioDevicePropertyStreamFormat,
&property_size, &stream_desc);
if (status)
{
lsx_fail_errno(ft, SOX_EPERM, "can not get audio device properties");
return SOX_EOF;
}
#endif
if (stream_desc.mChannelsPerFrame != ft->signal.channels)
{
lsx_debug("audio device did not accept %d channels. Use %d channels instead.", (int)ft->signal.channels,
(int)stream_desc.mChannelsPerFrame);
ft->signal.channels = stream_desc.mChannelsPerFrame;
}
if (stream_desc.mSampleRate != ft->signal.rate)
{
lsx_debug("audio device did not accept %d sample rate. Use %d instead.", (int)ft->signal.rate,
(int)stream_desc.mSampleRate);
ft->signal.rate = stream_desc.mSampleRate;
}
ac->buf_size = sox_globals.bufsiz * sizeof(float);
ac->buf_offset = 0;
ac->buffer = lsx_malloc(ac->buf_size);
buf_size = ac->buf_size;
property_size = sizeof(buf_size);
status = AudioDeviceSetProperty(ac->adid, NULL, 0, is_input,
kAudioDevicePropertyBufferSize,
property_size, &buf_size);
rc = pthread_mutex_init(&ac->mutex, NULL);
if (rc)
{
lsx_fail_errno(ft, SOX_EPERM, "failed initializing mutex");
return SOX_EOF;
}
rc = pthread_cond_init(&ac->cond, NULL);
if (rc)
{
lsx_fail_errno(ft, SOX_EPERM, "failed initializing condition");
return SOX_EOF;
}
ac->device_started = 0;
/* Registers callback with the device without activating it. */
if (is_input)
status = AudioDeviceAddIOProc(ac->adid, RecIOProc, (void *)ft);
else
status = AudioDeviceAddIOProc(ac->adid, PlaybackIOProc, (void *)ft);
return SOX_SUCCESS;
}
static int startread(sox_format_t *ft)
{
return setup(ft, 1);
}
static size_t read_samples(sox_format_t *ft, sox_sample_t *buf, size_t nsamp)
{
priv_t *ac = (priv_t *)ft->priv;
size_t len = nsamp;
size_t samp_left;
OSStatus status;
float *p;
SOX_SAMPLE_LOCALS;
if (!ac->device_started)
{
status = AudioDeviceStart(ac->adid, RecIOProc);
ac->device_started = 1;
}
pthread_mutex_lock(&ac->mutex);
/* Wait until input buffer has been filled by device driver */
while (ac->buf_offset < ac->buf_size)
pthread_cond_wait(&ac->cond, &ac->mutex);
len = ac->buf_offset / sizeof(float);
for (p = ac->buffer, samp_left = len; samp_left > 0; samp_left--, buf++, p++)
*buf = SOX_FLOAT_32BIT_TO_SAMPLE(*p, ft->clips);
ac->buf_offset = 0;
pthread_mutex_unlock(&ac->mutex);
return len;
}
static int stopread(sox_format_t * ft)
{
priv_t *ac = (priv_t *)ft->priv;
AudioDeviceStop(ac->adid, RecIOProc);
return SOX_SUCCESS;
}
static int startwrite(sox_format_t * ft)
{
return setup(ft, 0);
}
static size_t write_samples(sox_format_t *ft, const sox_sample_t *buf, size_t nsamp)
{
priv_t *ac = (priv_t *)ft->priv;
size_t len, written = 0;
size_t samp_left;
OSStatus status;
float *p;
SOX_SAMPLE_LOCALS;
if (!ac->device_started)
{
status = AudioDeviceStart(ac->adid, PlaybackIOProc);
ac->device_started = 1;
}
pthread_mutex_lock(&ac->mutex);
/* globals.bufsize is in samples
* buf_offset is in bytes
* buf_size is in bytes
*/
do {
/* Wait until callback has cleared the buffer. We move in
* lock-step with the callback; we never deal with a partially
* written buffer. */
while (ac->buf_offset != 0)
pthread_cond_wait(&ac->cond, &ac->mutex);
len = nsamp - written;
if (len > (ac->buf_size - ac->buf_offset) / sizeof(float))
len = (ac->buf_size - ac->buf_offset) / sizeof(float);
samp_left = len;
p = ((unsigned char *)ac->buffer) + ac->buf_offset;
while (samp_left--)
*p++ = SOX_SAMPLE_TO_FLOAT_32BIT(*buf++, ft->clips);
ac->buf_offset += len * sizeof(float);
written += len;
} while (written < nsamp);
pthread_mutex_unlock(&ac->mutex);
return written;
}
static int stopwrite(sox_format_t * ft)
{
priv_t *ac = (priv_t *)ft->priv;
AudioDeviceStop(ac->adid, PlaybackIOProc);
return SOX_SUCCESS;
}
LSX_FORMAT_HANDLER(coreaudio)
{
static char const *const names[] = { "coreaudio", NULL };
static unsigned const write_encodings[] = {
SOX_ENCODING_FLOAT, 32, 0,
0};
static sox_format_handler_t const handler = {SOX_LIB_VERSION_CODE,
"Mac AudioCore device driver",
names, SOX_FILE_DEVICE | SOX_FILE_NOSTDIO,
startread, read_samples, stopread,
startwrite, write_samples, stopwrite,
NULL, write_encodings, NULL, sizeof(priv_t)
};
return &handler;
}