diff options
Diffstat (limited to 'pjmedia/src/pjmedia-audiodev/coreaudio_dev.m')
| -rw-r--r-- | pjmedia/src/pjmedia-audiodev/coreaudio_dev.m | 2115 |
1 files changed, 2115 insertions, 0 deletions
diff --git a/pjmedia/src/pjmedia-audiodev/coreaudio_dev.m b/pjmedia/src/pjmedia-audiodev/coreaudio_dev.m new file mode 100644 index 00000000..b2206824 --- /dev/null +++ b/pjmedia/src/pjmedia-audiodev/coreaudio_dev.m @@ -0,0 +1,2115 @@ +/* $Id$ */ +/* + * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com) + * + * 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. + * + * This program 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 General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ +#include <pjmedia-audiodev/audiodev_imp.h> +#include <pj/assert.h> +#include <pj/log.h> +#include <pj/os.h> + +#if PJMEDIA_AUDIO_DEV_HAS_COREAUDIO + +#include "TargetConditionals.h" +#if TARGET_OS_IPHONE + #define COREAUDIO_MAC 0 +#else + #define COREAUDIO_MAC 1 +#endif + +#include <AudioUnit/AudioUnit.h> +#include <AudioToolbox/AudioConverter.h> +#if COREAUDIO_MAC + #include <CoreAudio/CoreAudio.h> +#else + #include <AVFoundation/AVAudioSession.h> + + #define AudioDeviceID unsigned + + /** + * As in iOS SDK 4 or later, audio route change property listener is + * no longer necessary. Just make surethat your application can receive + * remote control events by adding the code: + * [[UIApplication sharedApplication] + * beginReceivingRemoteControlEvents]; + * Otherwise audio route change (such as headset plug/unplug) will not be + * processed while your application is in the background mode. + */ + #define USE_AUDIO_ROUTE_CHANGE_PROP_LISTENER 0 + + /* Starting iOS SDK 7, Audio Session API is deprecated. */ + #define USE_AUDIO_SESSION_API 0 +#endif + +/* For Mac OS 10.5.x and earlier */ +#if AUDIO_UNIT_VERSION < 1060 + #define AudioComponent Component + #define AudioComponentDescription ComponentDescription + #define AudioComponentInstance ComponentInstance + #define AudioComponentFindNext FindNextComponent + #define AudioComponentInstanceNew OpenAComponent + #define AudioComponentInstanceDispose CloseComponent +#endif + + +#define THIS_FILE "coreaudio_dev.c" + +/* coreaudio device info */ +struct coreaudio_dev_info +{ + pjmedia_aud_dev_info info; + AudioDeviceID dev_id; +}; + +/* linked list of streams */ +struct stream_list +{ + PJ_DECL_LIST_MEMBER(struct stream_list); + struct coreaudio_stream *stream; +}; + +/* coreaudio factory */ +struct coreaudio_factory +{ + pjmedia_aud_dev_factory base; + pj_pool_t *base_pool; + pj_pool_t *pool; + pj_pool_factory *pf; + pj_mutex_t *mutex; + + unsigned dev_count; + struct coreaudio_dev_info *dev_info; + + AudioComponent io_comp; + struct stream_list streams; +}; + +/* Sound stream. */ +struct coreaudio_stream +{ + pjmedia_aud_stream base; /**< Base stream */ + pjmedia_aud_param param; /**< Settings */ + pj_pool_t *pool; /**< Memory pool. */ + struct coreaudio_factory *cf; + struct stream_list list_entry; + + pjmedia_aud_rec_cb rec_cb; /**< Capture callback. */ + pjmedia_aud_play_cb play_cb; /**< Playback callback. */ + void *user_data; /**< Application data. */ + + pj_timestamp play_timestamp; + pj_timestamp rec_timestamp; + + pj_int16_t *rec_buf; + unsigned rec_buf_count; + pj_int16_t *play_buf; + unsigned play_buf_count; + + pj_bool_t interrupted; + pj_bool_t quit_flag; + pj_bool_t running; + + pj_bool_t rec_thread_initialized; + pj_thread_desc rec_thread_desc; + pj_thread_t *rec_thread; + + pj_bool_t play_thread_initialized; + pj_thread_desc play_thread_desc; + pj_thread_t *play_thread; + + AudioUnit io_units[2]; + AudioStreamBasicDescription streamFormat; + AudioBufferList *audio_buf; + + AudioConverterRef resample; + pj_int16_t *resample_buf; + void *resample_buf_ptr; + unsigned resample_buf_count; + unsigned resample_buf_size; + +#if !COREAUDIO_MAC + AVAudioSession *sess; +#endif +}; + +/* Static variable */ +static struct coreaudio_factory *cf_instance = NULL; + +/* Prototypes */ +static pj_status_t ca_factory_init(pjmedia_aud_dev_factory *f); +static pj_status_t ca_factory_destroy(pjmedia_aud_dev_factory *f); +static pj_status_t ca_factory_refresh(pjmedia_aud_dev_factory *f); +static unsigned ca_factory_get_dev_count(pjmedia_aud_dev_factory *f); +static pj_status_t ca_factory_get_dev_info(pjmedia_aud_dev_factory *f, + unsigned index, + pjmedia_aud_dev_info *info); +static pj_status_t ca_factory_default_param(pjmedia_aud_dev_factory *f, + unsigned index, + pjmedia_aud_param *param); +static pj_status_t ca_factory_create_stream(pjmedia_aud_dev_factory *f, + const pjmedia_aud_param *param, + pjmedia_aud_rec_cb rec_cb, + pjmedia_aud_play_cb play_cb, + void *user_data, + pjmedia_aud_stream **p_aud_strm); + +static pj_status_t ca_stream_get_param(pjmedia_aud_stream *strm, + pjmedia_aud_param *param); +static pj_status_t ca_stream_get_cap(pjmedia_aud_stream *strm, + pjmedia_aud_dev_cap cap, + void *value); +static pj_status_t ca_stream_set_cap(pjmedia_aud_stream *strm, + pjmedia_aud_dev_cap cap, + const void *value); +static pj_status_t ca_stream_start(pjmedia_aud_stream *strm); +static pj_status_t ca_stream_stop(pjmedia_aud_stream *strm); +static pj_status_t ca_stream_destroy(pjmedia_aud_stream *strm); +static pj_status_t create_audio_unit(AudioComponent io_comp, + AudioDeviceID dev_id, + pjmedia_dir dir, + struct coreaudio_stream *strm, + AudioUnit *io_unit); +#if !COREAUDIO_MAC && USE_AUDIO_SESSION_API != 0 +static void interruptionListener(void *inClientData, UInt32 inInterruption); +static void propListener(void * inClientData, + AudioSessionPropertyID inID, + UInt32 inDataSize, + const void * inData); +#endif + +/* Operations */ +static pjmedia_aud_dev_factory_op factory_op = +{ + &ca_factory_init, + &ca_factory_destroy, + &ca_factory_get_dev_count, + &ca_factory_get_dev_info, + &ca_factory_default_param, + &ca_factory_create_stream, + &ca_factory_refresh +}; + +static pjmedia_aud_stream_op stream_op = +{ + &ca_stream_get_param, + &ca_stream_get_cap, + &ca_stream_set_cap, + &ca_stream_start, + &ca_stream_stop, + &ca_stream_destroy +}; + + +/**************************************************************************** + * Factory operations + */ +/* + * Init coreaudio audio driver. + */ +pjmedia_aud_dev_factory* pjmedia_coreaudio_factory(pj_pool_factory *pf) +{ + struct coreaudio_factory *f; + pj_pool_t *pool; + + pool = pj_pool_create(pf, "core audio base", 1000, 1000, NULL); + f = PJ_POOL_ZALLOC_T(pool, struct coreaudio_factory); + f->pf = pf; + f->base_pool = pool; + f->base.op = &factory_op; + + return &f->base; +} + + +/* API: init factory */ +static pj_status_t ca_factory_init(pjmedia_aud_dev_factory *f) +{ + struct coreaudio_factory *cf = (struct coreaudio_factory*)f; + AudioComponentDescription desc; + pj_status_t status; +#if !COREAUDIO_MAC + unsigned i; +#endif + + pj_list_init(&cf->streams); + status = pj_mutex_create_recursive(cf->base_pool, + "coreaudio", + &cf->mutex); + if (status != PJ_SUCCESS) + return status; + + desc.componentType = kAudioUnitType_Output; +#if COREAUDIO_MAC + desc.componentSubType = kAudioUnitSubType_HALOutput; +#else + desc.componentSubType = kAudioUnitSubType_RemoteIO; +#endif + desc.componentManufacturer = kAudioUnitManufacturer_Apple; + desc.componentFlags = 0; + desc.componentFlagsMask = 0; + + cf->io_comp = AudioComponentFindNext(NULL, &desc); + if (cf->io_comp == NULL) + return PJMEDIA_EAUD_INIT; // cannot find IO unit; + + status = ca_factory_refresh(f); + if (status != PJ_SUCCESS) + return status; + +#if !COREAUDIO_MAC + cf->pool = pj_pool_create(cf->pf, "core audio", 1000, 1000, NULL); + cf->dev_count = 1; + cf->dev_info = (struct coreaudio_dev_info*) + pj_pool_calloc(cf->pool, cf->dev_count, + sizeof(struct coreaudio_dev_info)); + for (i = 0; i < cf->dev_count; i++) { + struct coreaudio_dev_info *cdi; + + cdi = &cf->dev_info[i]; + pj_bzero(cdi, sizeof(*cdi)); + cdi->dev_id = 0; + strcpy(cdi->info.name, "iPhone IO device"); + strcpy(cdi->info.driver, "apple"); + cdi->info.input_count = 1; + cdi->info.output_count = 1; + cdi->info.default_samples_per_sec = 8000; + + /* Set the device capabilities here */ + cdi->info.caps = PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY | + PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY | + PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING | +#if USE_AUDIO_SESSION_API != 0 + PJMEDIA_AUD_DEV_CAP_INPUT_ROUTE | + PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE | +#endif + PJMEDIA_AUD_DEV_CAP_EC; + cdi->info.routes = PJMEDIA_AUD_DEV_ROUTE_LOUDSPEAKER | + PJMEDIA_AUD_DEV_ROUTE_EARPIECE | + PJMEDIA_AUD_DEV_ROUTE_BLUETOOTH; + + PJ_LOG(4, (THIS_FILE, " dev_id %d: %s (in=%d, out=%d) %dHz", + i, + cdi->info.name, + cdi->info.input_count, + cdi->info.output_count, + cdi->info.default_samples_per_sec)); + } + +#if USE_AUDIO_SESSION_API != 0 + { + OSStatus ostatus; + + /* Initialize the Audio Session */ + ostatus = AudioSessionInitialize(NULL, NULL, interruptionListener, + NULL); + if (ostatus != kAudioSessionNoError) { + PJ_LOG(4, (THIS_FILE, + "Warning: cannot initialize audio session services (%i)", + ostatus)); + } + + /* Listen for audio routing change notifications. */ +#if USE_AUDIO_ROUTE_CHANGE_PROP_LISTENER != 0 + ostatus = AudioSessionAddPropertyListener( + kAudioSessionProperty_AudioRouteChange, + propListener, cf); + if (ostatus != kAudioSessionNoError) { + PJ_LOG(4, (THIS_FILE, + "Warning: cannot listen for audio route change " + "notifications (%i)", ostatus)); + } +#endif + } +#endif + + cf_instance = cf; +#endif + + PJ_LOG(4, (THIS_FILE, "core audio initialized")); + + return PJ_SUCCESS; +} + +/* API: destroy factory */ +static pj_status_t ca_factory_destroy(pjmedia_aud_dev_factory *f) +{ + struct coreaudio_factory *cf = (struct coreaudio_factory*)f; + pj_pool_t *pool; + + pj_assert(cf); + pj_assert(cf->base_pool); + pj_assert(pj_list_empty(&cf->streams)); + +#if !COREAUDIO_MAC +#if USE_AUDIO_SESSION_API != 0 && USE_AUDIO_ROUTE_CHANGE_PROP_LISTENER != 0 + AudioSessionRemovePropertyListenerWithUserData( + kAudioSessionProperty_AudioRouteChange, propListener, cf); +#endif +#endif + + if (cf->pool) { + pj_pool_release(cf->pool); + cf->pool = NULL; + } + + if (cf->mutex) { + pj_mutex_lock(cf->mutex); + cf_instance = NULL; + pj_mutex_unlock(cf->mutex); + pj_mutex_destroy(cf->mutex); + cf->mutex = NULL; + } + + pool = cf->base_pool; + cf->base_pool = NULL; + pj_pool_release(pool); + + return PJ_SUCCESS; +} + +/* API: refresh the device list */ +static pj_status_t ca_factory_refresh(pjmedia_aud_dev_factory *f) +{ +#if !COREAUDIO_MAC + /* iPhone doesn't support refreshing the device list */ + PJ_UNUSED_ARG(f); + return PJ_SUCCESS; +#else + struct coreaudio_factory *cf = (struct coreaudio_factory*)f; + unsigned i; + unsigned dev_count; + AudioObjectPropertyAddress addr; + AudioDeviceID *dev_ids; + UInt32 buf_size, dev_size, size = sizeof(AudioDeviceID); + AudioBufferList *buf = NULL; + OSStatus ostatus; + + if (cf->pool != NULL) { + pj_pool_release(cf->pool); + cf->pool = NULL; + } + + cf->dev_count = 0; + cf->pool = pj_pool_create(cf->pf, "core audio", 1000, 1000, NULL); + + /* Find out how many audio devices there are */ + addr.mSelector = kAudioHardwarePropertyDevices; + addr.mScope = kAudioObjectPropertyScopeGlobal; + addr.mElement = kAudioObjectPropertyElementMaster; + ostatus = AudioObjectGetPropertyDataSize(kAudioObjectSystemObject, &addr, + 0, NULL, &dev_size); + if (ostatus != noErr) { + dev_size = 0; + } + + /* Calculate the number of audio devices available */ + dev_count = dev_size / size; + if (dev_count==0) { + PJ_LOG(4,(THIS_FILE, "core audio found no sound devices")); + /* Enabling this will cause pjsua-lib initialization to fail when + * there is no sound device installed in the system, even when pjsua + * has been run with --null-audio. Moreover, it might be better to + * think that the core audio backend initialization is successful, + * regardless there is no audio device installed, as later application + * can check it using get_dev_count(). + return PJMEDIA_EAUD_NODEV; + */ + return PJ_SUCCESS; + } + PJ_LOG(4, (THIS_FILE, "core audio detected %d devices", + dev_count)); + + /* Get all the audio device IDs */ + dev_ids = (AudioDeviceID *)pj_pool_calloc(cf->pool, dev_size, size); + if (!dev_ids) + return PJ_ENOMEM; + pj_bzero(dev_ids, dev_count); + ostatus = AudioObjectGetPropertyData(kAudioObjectSystemObject, &addr, + 0, NULL, + &dev_size, (void *)dev_ids); + if (ostatus != noErr ) { + /* This should not happen since we have successfully retrieved + * the property data size before + */ + return PJMEDIA_EAUD_INIT; + } + + if (dev_size > 1) { + AudioDeviceID dev_id = kAudioObjectUnknown; + unsigned idx = 0; + + /* Find default audio input device */ + addr.mSelector = kAudioHardwarePropertyDefaultInputDevice; + addr.mScope = kAudioObjectPropertyScopeGlobal; + addr.mElement = kAudioObjectPropertyElementMaster; + size = sizeof(dev_id); + + ostatus = AudioObjectGetPropertyData(kAudioObjectSystemObject, + &addr, 0, NULL, + &size, (void *)&dev_id); + if (ostatus == noErr && dev_id != dev_ids[idx]) { + AudioDeviceID temp_id = dev_ids[idx]; + + for (i = idx + 1; i < dev_size; i++) { + if (dev_ids[i] == dev_id) { + dev_ids[idx++] = dev_id; + dev_ids[i] = temp_id; + break; + } + } + } + + /* Find default audio output device */ + addr.mSelector = kAudioHardwarePropertyDefaultOutputDevice; + ostatus = AudioObjectGetPropertyData(kAudioObjectSystemObject, + &addr, 0, NULL, + &size, (void *)&dev_id); + if (ostatus == noErr && dev_id != dev_ids[idx]) { + AudioDeviceID temp_id = dev_ids[idx]; + + for (i = idx + 1; i < dev_size; i++) { + if (dev_ids[i] == dev_id) { + dev_ids[idx] = dev_id; + dev_ids[i] = temp_id; + break; + } + } + } + } + + /* Build the devices' info */ + cf->dev_info = (struct coreaudio_dev_info*) + pj_pool_calloc(cf->pool, dev_count, + sizeof(struct coreaudio_dev_info)); + buf_size = 0; + for (i = 0; i < dev_count; i++) { + struct coreaudio_dev_info *cdi; + Float64 sampleRate; + + cdi = &cf->dev_info[i]; + pj_bzero(cdi, sizeof(*cdi)); + cdi->dev_id = dev_ids[i]; + + /* Get device name */ + addr.mSelector = kAudioDevicePropertyDeviceName; + addr.mScope = kAudioObjectPropertyScopeGlobal; + addr.mElement = kAudioObjectPropertyElementMaster; + size = sizeof(cdi->info.name); + AudioObjectGetPropertyData(cdi->dev_id, &addr, + 0, NULL, + &size, (void *)cdi->info.name); + + strcpy(cdi->info.driver, "core audio"); + + /* Get the number of input channels */ + addr.mSelector = kAudioDevicePropertyStreamConfiguration; + addr.mScope = kAudioDevicePropertyScopeInput; + size = 0; + ostatus = AudioObjectGetPropertyDataSize(cdi->dev_id, &addr, + 0, NULL, &size); + if (ostatus == noErr && size > 0) { + + if (size > buf_size) { + buf = pj_pool_alloc(cf->pool, size); + buf_size = size; + } + if (buf) { + UInt32 idx; + + /* Get the input stream configuration */ + ostatus = AudioObjectGetPropertyData(cdi->dev_id, &addr, + 0, NULL, + &size, buf); + if (ostatus == noErr) { + /* Count the total number of input channels in + * the stream + */ + for (idx = 0; idx < buf->mNumberBuffers; idx++) { + cdi->info.input_count += + buf->mBuffers[idx].mNumberChannels; + } + } + } + } + + /* Get the number of output channels */ + addr.mScope = kAudioDevicePropertyScopeOutput; + size = 0; + ostatus = AudioObjectGetPropertyDataSize(cdi->dev_id, &addr, + 0, NULL, &size); + if (ostatus == noErr && size > 0) { + + if (size > buf_size) { + buf = pj_pool_alloc(cf->pool, size); + buf_size = size; + } + if (buf) { + UInt32 idx; + + /* Get the output stream configuration */ + ostatus = AudioObjectGetPropertyData(cdi->dev_id, &addr, + 0, NULL, + &size, buf); + if (ostatus == noErr) { + /* Count the total number of output channels in + * the stream + */ + for (idx = 0; idx < buf->mNumberBuffers; idx++) { + cdi->info.output_count += + buf->mBuffers[idx].mNumberChannels; + } + } + } + } + + /* Get default sample rate */ + addr.mSelector = kAudioDevicePropertyNominalSampleRate; + addr.mScope = kAudioObjectPropertyScopeGlobal; + size = sizeof(Float64); + ostatus = AudioObjectGetPropertyData (cdi->dev_id, &addr, + 0, NULL, + &size, &sampleRate); + cdi->info.default_samples_per_sec = (ostatus == noErr ? + sampleRate: + 16000); + + /* Set device capabilities here */ + if (cdi->info.input_count > 0) { + cdi->info.caps |= PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY; + } + if (cdi->info.output_count > 0) { + cdi->info.caps |= PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY; + addr.mSelector = kAudioDevicePropertyVolumeScalar; + addr.mScope = kAudioDevicePropertyScopeOutput; + if (AudioObjectHasProperty(cdi->dev_id, &addr)) { + cdi->info.caps |= PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING; + } + } + + cf->dev_count++; + + PJ_LOG(4, (THIS_FILE, " dev_id %d: %s (in=%d, out=%d) %dHz", + i, + cdi->info.name, + cdi->info.input_count, + cdi->info.output_count, + cdi->info.default_samples_per_sec)); + } + + return PJ_SUCCESS; +#endif +} + +/* API: get number of devices */ +static unsigned ca_factory_get_dev_count(pjmedia_aud_dev_factory *f) +{ + struct coreaudio_factory *cf = (struct coreaudio_factory*)f; + return cf->dev_count; +} + +/* API: get device info */ +static pj_status_t ca_factory_get_dev_info(pjmedia_aud_dev_factory *f, + unsigned index, + pjmedia_aud_dev_info *info) +{ + struct coreaudio_factory *cf = (struct coreaudio_factory*)f; + + PJ_ASSERT_RETURN(index < cf->dev_count, PJMEDIA_EAUD_INVDEV); + + pj_memcpy(info, &cf->dev_info[index].info, sizeof(*info)); + + return PJ_SUCCESS; +} + +/* API: create default device parameter */ +static pj_status_t ca_factory_default_param(pjmedia_aud_dev_factory *f, + unsigned index, + pjmedia_aud_param *param) +{ + struct coreaudio_factory *cf = (struct coreaudio_factory*)f; + struct coreaudio_dev_info *di = &cf->dev_info[index]; + + PJ_ASSERT_RETURN(index < cf->dev_count, PJMEDIA_EAUD_INVDEV); + + pj_bzero(param, sizeof(*param)); + if (di->info.input_count && di->info.output_count) { + param->dir = PJMEDIA_DIR_CAPTURE_PLAYBACK; + param->rec_id = index; + param->play_id = index; + } else if (di->info.input_count) { + param->dir = PJMEDIA_DIR_CAPTURE; + param->rec_id = index; + param->play_id = PJMEDIA_AUD_INVALID_DEV; + } else if (di->info.output_count) { + param->dir = PJMEDIA_DIR_PLAYBACK; + param->play_id = index; + param->rec_id = PJMEDIA_AUD_INVALID_DEV; + } else { + return PJMEDIA_EAUD_INVDEV; + } + + /* Set the mandatory settings here */ + param->clock_rate = di->info.default_samples_per_sec; + param->channel_count = 1; + param->samples_per_frame = di->info.default_samples_per_sec * 20 / 1000; + param->bits_per_sample = 16; + + /* Set the param for device capabilities here */ + param->flags = PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY | + PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY; + param->input_latency_ms = PJMEDIA_SND_DEFAULT_REC_LATENCY; + param->output_latency_ms = PJMEDIA_SND_DEFAULT_PLAY_LATENCY; + + return PJ_SUCCESS; +} + +OSStatus resampleProc(AudioConverterRef inAudioConverter, + UInt32 *ioNumberDataPackets, + AudioBufferList *ioData, + AudioStreamPacketDescription **outDataPacketDescription, + void *inUserData) +{ + struct coreaudio_stream *strm = (struct coreaudio_stream*)inUserData; + + if (*ioNumberDataPackets > strm->resample_buf_size) + *ioNumberDataPackets = strm->resample_buf_size; + + ioData->mNumberBuffers = 1; + ioData->mBuffers[0].mNumberChannels = strm->streamFormat.mChannelsPerFrame; + ioData->mBuffers[0].mData = strm->resample_buf_ptr; + ioData->mBuffers[0].mDataByteSize = *ioNumberDataPackets * + strm->streamFormat.mChannelsPerFrame * + strm->param.bits_per_sample >> 3; + + return noErr; +} + +static OSStatus resample_callback(void *inRefCon, + AudioUnitRenderActionFlags *ioActionFlags, + const AudioTimeStamp *inTimeStamp, + UInt32 inBusNumber, + UInt32 inNumberFrames, + AudioBufferList *ioData) +{ + struct coreaudio_stream *strm = (struct coreaudio_stream*)inRefCon; + OSStatus ostatus; + pj_status_t status = 0; + unsigned nsamples; + AudioBufferList *buf = strm->audio_buf; + pj_int16_t *input; + UInt32 resampleSize; + + pj_assert(!strm->quit_flag); + + /* Known cases of callback's thread: + * - The thread may be changed in the middle of a session + * it happens when plugging/unplugging headphone. + * - The same thread may be reused in consecutive sessions. The first + * session will leave TLS set, but release the TLS data address, + * so the second session must re-register the callback's thread. + */ + if (strm->rec_thread_initialized == 0 || !pj_thread_is_registered()) + { + pj_bzero(strm->rec_thread_desc, sizeof(pj_thread_desc)); + status = pj_thread_register("ca_rec", strm->rec_thread_desc, + &strm->rec_thread); + strm->rec_thread_initialized = 1; + PJ_LOG(5,(THIS_FILE, "Recorder thread started, (%i frames)", + inNumberFrames)); + } + + buf->mBuffers[0].mData = NULL; + buf->mBuffers[0].mDataByteSize = inNumberFrames * + strm->streamFormat.mChannelsPerFrame; + /* Render the unit to get input data */ + ostatus = AudioUnitRender(strm->io_units[0], + ioActionFlags, + inTimeStamp, + inBusNumber, + inNumberFrames, + buf); + + if (ostatus != noErr) { + PJ_LOG(5, (THIS_FILE, "Core audio unit render error %i", ostatus)); + goto on_break; + } + input = (pj_int16_t *)buf->mBuffers[0].mData; + + resampleSize = strm->resample_buf_size; + nsamples = inNumberFrames * strm->param.channel_count + + strm->resample_buf_count; + + if (nsamples >= resampleSize) { + pjmedia_frame frame; + UInt32 resampleOutput = strm->param.samples_per_frame / + strm->streamFormat.mChannelsPerFrame; + AudioBufferList ab; + + frame.type = PJMEDIA_FRAME_TYPE_AUDIO; + frame.buf = (void*) strm->rec_buf; + frame.size = strm->param.samples_per_frame * + strm->param.bits_per_sample >> 3; + frame.bit_info = 0; + + ab.mNumberBuffers = 1; + ab.mBuffers[0].mNumberChannels = strm->streamFormat.mChannelsPerFrame; + ab.mBuffers[0].mData = strm->rec_buf; + ab.mBuffers[0].mDataByteSize = frame.size; + + /* If buffer is not empty, combine the buffer with the just incoming + * samples, then call put_frame. + */ + if (strm->resample_buf_count) { + unsigned chunk_count = resampleSize - strm->resample_buf_count; + pjmedia_copy_samples(strm->resample_buf + strm->resample_buf_count, + input, chunk_count); + + /* Do the resample */ + + strm->resample_buf_ptr = strm->resample_buf; + ostatus = AudioConverterFillComplexBuffer(strm->resample, + resampleProc, + strm, + &resampleOutput, + &ab, + NULL); + if (ostatus != noErr) { + goto on_break; + } + frame.timestamp.u64 = strm->rec_timestamp.u64; + + status = (*strm->rec_cb)(strm->user_data, &frame); + + input = input + chunk_count; + nsamples -= resampleSize; + strm->resample_buf_count = 0; + strm->rec_timestamp.u64 += strm->param.samples_per_frame / + strm->param.channel_count; + } + + + /* Give all frames we have */ + while (nsamples >= resampleSize && status == 0) { + frame.timestamp.u64 = strm->rec_timestamp.u64; + + /* Do the resample */ + strm->resample_buf_ptr = input; + ab.mBuffers[0].mDataByteSize = frame.size; + resampleOutput = strm->param.samples_per_frame / + strm->streamFormat.mChannelsPerFrame; + ostatus = AudioConverterFillComplexBuffer(strm->resample, + resampleProc, + strm, + &resampleOutput, + &ab, + NULL); + if (ostatus != noErr) { + goto on_break; + } + + status = (*strm->rec_cb)(strm->user_data, &frame); + + input = (pj_int16_t*) input + resampleSize; + nsamples -= resampleSize; + strm->rec_timestamp.u64 += strm->param.samples_per_frame / + strm->param.channel_count; + } + + /* Store the remaining samples into the buffer */ + if (nsamples && status == 0) { + strm->resample_buf_count = nsamples; + pjmedia_copy_samples(strm->resample_buf, input, + nsamples); + } + + } else { + /* Not enough samples, let's just store them in the buffer */ + pjmedia_copy_samples(strm->resample_buf + strm->resample_buf_count, + input, + inNumberFrames * strm->param.channel_count); + strm->resample_buf_count += inNumberFrames * + strm->param.channel_count; + } + + return noErr; + +on_break: + return -1; +} + +static OSStatus input_callback(void *inRefCon, + AudioUnitRenderActionFlags *ioActionFlags, + const AudioTimeStamp *inTimeStamp, + UInt32 inBusNumber, + UInt32 inNumberFrames, + AudioBufferList *ioData) +{ + struct coreaudio_stream *strm = (struct coreaudio_stream*)inRefCon; + OSStatus ostatus; + pj_status_t status = 0; + unsigned nsamples; + AudioBufferList *buf = strm->audio_buf; + pj_int16_t *input; + + pj_assert(!strm->quit_flag); + + /* Known cases of callback's thread: + * - The thread may be changed in the middle of a session + * it happens when plugging/unplugging headphone. + * - The same thread may be reused in consecutive sessions. The first + * session will leave TLS set, but release the TLS data address, + * so the second session must re-register the callback's thread. + */ + if (strm->rec_thread_initialized == 0 || !pj_thread_is_registered()) + { + pj_bzero(strm->rec_thread_desc, sizeof(pj_thread_desc)); + status = pj_thread_register("ca_rec", strm->rec_thread_desc, + &strm->rec_thread); + strm->rec_thread_initialized = 1; + PJ_LOG(5,(THIS_FILE, "Recorder thread started, (%i frames)", + inNumberFrames)); + } + + buf->mBuffers[0].mData = NULL; + buf->mBuffers[0].mDataByteSize = inNumberFrames * + strm->streamFormat.mChannelsPerFrame; + /* Render the unit to get input data */ + ostatus = AudioUnitRender(strm->io_units[0], + ioActionFlags, + inTimeStamp, + inBusNumber, + inNumberFrames, + buf); + + if (ostatus != noErr) { + PJ_LOG(5, (THIS_FILE, "Core audio unit render error %i", ostatus)); + goto on_break; + } + input = (pj_int16_t *)buf->mBuffers[0].mData; + + /* Calculate number of samples we've got */ + nsamples = inNumberFrames * strm->param.channel_count + + strm->rec_buf_count; + if (nsamples >= strm->param.samples_per_frame) { + pjmedia_frame frame; + + frame.type = PJMEDIA_FRAME_TYPE_AUDIO; + frame.size = strm->param.samples_per_frame * + strm->param.bits_per_sample >> 3; + frame.bit_info = 0; + + /* If buffer is not empty, combine the buffer with the just incoming + * samples, then call put_frame. + */ + if (strm->rec_buf_count) { + unsigned chunk_count = 0; + + chunk_count = strm->param.samples_per_frame - strm->rec_buf_count; + pjmedia_copy_samples(strm->rec_buf + strm->rec_buf_count, + input, chunk_count); + + frame.buf = (void*) strm->rec_buf; + frame.timestamp.u64 = strm->rec_timestamp.u64; + + status = (*strm->rec_cb)(strm->user_data, &frame); + + input = input + chunk_count; + nsamples -= strm->param.samples_per_frame; + strm->rec_buf_count = 0; + strm->rec_timestamp.u64 += strm->param.samples_per_frame / + strm->param.channel_count; + } + + /* Give all frames we have */ + while (nsamples >= strm->param.samples_per_frame && status == 0) { + frame.buf = (void*) input; + frame.timestamp.u64 = strm->rec_timestamp.u64; + + status = (*strm->rec_cb)(strm->user_data, &frame); + + input = (pj_int16_t*) input + strm->param.samples_per_frame; + nsamples -= strm->param.samples_per_frame; + strm->rec_timestamp.u64 += strm->param.samples_per_frame / + strm->param.channel_count; + } + + /* Store the remaining samples into the buffer */ + if (nsamples && status == 0) { + strm->rec_buf_count = nsamples; + pjmedia_copy_samples(strm->rec_buf, input, + nsamples); + } + + } else { + /* Not enough samples, let's just store them in the buffer */ + pjmedia_copy_samples(strm->rec_buf + strm->rec_buf_count, + input, + inNumberFrames * strm->param.channel_count); + strm->rec_buf_count += inNumberFrames * strm->param.channel_count; + } + + return noErr; + +on_break: + return -1; +} + +static OSStatus output_renderer(void *inRefCon, + AudioUnitRenderActionFlags *ioActionFlags, + const AudioTimeStamp *inTimeStamp, + UInt32 inBusNumber, + UInt32 inNumberFrames, + AudioBufferList *ioData) +{ + struct coreaudio_stream *stream = (struct coreaudio_stream*)inRefCon; + pj_status_t status = 0; + unsigned nsamples_req = inNumberFrames * stream->param.channel_count; + pj_int16_t *output = ioData->mBuffers[0].mData; + + pj_assert(!stream->quit_flag); + + /* Known cases of callback's thread: + * - The thread may be changed in the middle of a session + * it happens when plugging/unplugging headphone. + * - The same thread may be reused in consecutive sessions. The first + * session will leave TLS set, but release the TLS data address, + * so the second session must re-register the callback's thread. + */ + if (stream->play_thread_initialized == 0 || !pj_thread_is_registered()) + { + pj_bzero(stream->play_thread_desc, sizeof(pj_thread_desc)); + status = pj_thread_register("coreaudio", stream->play_thread_desc, + &stream->play_thread); + stream->play_thread_initialized = 1; + PJ_LOG(5,(THIS_FILE, "Player thread started, (%i frames)", + inNumberFrames)); + } + + + /* Check if any buffered samples */ + if (stream->play_buf_count) { + /* samples buffered >= requested by sound device */ + if (stream->play_buf_count >= nsamples_req) { + pjmedia_copy_samples((pj_int16_t*)output, stream->play_buf, + nsamples_req); + stream->play_buf_count -= nsamples_req; + pjmedia_move_samples(stream->play_buf, + stream->play_buf + nsamples_req, + stream->play_buf_count); + nsamples_req = 0; + + return noErr; + } + + /* samples buffered < requested by sound device */ + pjmedia_copy_samples((pj_int16_t*)output, stream->play_buf, + stream->play_buf_count); + nsamples_req -= stream->play_buf_count; + output = (pj_int16_t*)output + stream->play_buf_count; + stream->play_buf_count = 0; + } + + /* Fill output buffer as requested */ + while (nsamples_req && status == 0) { + pjmedia_frame frame; + + frame.type = PJMEDIA_FRAME_TYPE_AUDIO; + frame.size = stream->param.samples_per_frame * + stream->param.bits_per_sample >> 3; + frame.timestamp.u64 = stream->play_timestamp.u64; + frame.bit_info = 0; + + if (nsamples_req >= stream->param.samples_per_frame) { + frame.buf = output; + status = (*stream->play_cb)(stream->user_data, &frame); + if (status != PJ_SUCCESS) + goto on_break; + + if (frame.type != PJMEDIA_FRAME_TYPE_AUDIO) + pj_bzero(frame.buf, frame.size); + + nsamples_req -= stream->param.samples_per_frame; + output = (pj_int16_t*)output + stream->param.samples_per_frame; + } else { + frame.buf = stream->play_buf; + status = (*stream->play_cb)(stream->user_data, &frame); + if (status != PJ_SUCCESS) + goto on_break; + + if (frame.type != PJMEDIA_FRAME_TYPE_AUDIO) + pj_bzero(frame.buf, frame.size); + + pjmedia_copy_samples((pj_int16_t*)output, stream->play_buf, + nsamples_req); + stream->play_buf_count = stream->param.samples_per_frame - + nsamples_req; + pjmedia_move_samples(stream->play_buf, + stream->play_buf+nsamples_req, + stream->play_buf_count); + nsamples_req = 0; + } + + stream->play_timestamp.u64 += stream->param.samples_per_frame / + stream->param.channel_count; + } + + return noErr; + +on_break: + return -1; +} + +#if !COREAUDIO_MAC && USE_AUDIO_SESSION_API != 0 +static void propListener(void *inClientData, + AudioSessionPropertyID inID, + UInt32 inDataSize, + const void * inData) +{ + struct coreaudio_factory *cf = (struct coreaudio_factory*)inClientData; + struct stream_list *it, *itBegin; + CFDictionaryRef routeDictionary; + CFNumberRef reason; + SInt32 reasonVal; + pj_assert(cf); + + if (inID != kAudioSessionProperty_AudioRouteChange) + return; + + routeDictionary = (CFDictionaryRef)inData; + reason = (CFNumberRef) + CFDictionaryGetValue( + routeDictionary, + CFSTR(kAudioSession_AudioRouteChangeKey_Reason)); + CFNumberGetValue(reason, kCFNumberSInt32Type, &reasonVal); + + if (reasonVal != kAudioSessionRouteChangeReason_OldDeviceUnavailable) { + PJ_LOG(3, (THIS_FILE, "ignoring audio route change...")); + return; + } + + PJ_LOG(3, (THIS_FILE, "audio route changed")); + + pj_mutex_lock(cf->mutex); + itBegin = &cf->streams; + for (it = itBegin->next; it != itBegin; it = it->next) { + if (it->stream->interrupted) + continue; + + /* + status = ca_stream_stop((pjmedia_aud_stream *)it->stream); + status = ca_stream_start((pjmedia_aud_stream *)it->stream); + if (status != PJ_SUCCESS) { + PJ_LOG(3, (THIS_FILE, + "Error: failed to restart the audio unit (%i)", + status)); + continue; + } + PJ_LOG(3, (THIS_FILE, "core audio unit successfully restarted")); + */ + } + pj_mutex_unlock(cf->mutex); +} + +static void interruptionListener(void *inClientData, UInt32 inInterruption) +{ + struct stream_list *it, *itBegin; + pj_status_t status; + pj_thread_desc thread_desc; + pj_thread_t *thread; + + /* Register the thread with PJLIB, this is must for any external threads + * which need to use the PJLIB framework. + */ + if (!pj_thread_is_registered()) { + pj_bzero(thread_desc, sizeof(pj_thread_desc)); + status = pj_thread_register("intListener", thread_desc, &thread); + } + + PJ_LOG(3, (THIS_FILE, "Session interrupted! --- %s ---", + inInterruption == kAudioSessionBeginInterruption ? + "Begin Interruption" : "End Interruption")); + + if (!cf_instance) + return; + + pj_mutex_lock(cf_instance->mutex); + itBegin = &cf_instance->streams; + for (it = itBegin->next; it != itBegin; it = it->next) { + if (inInterruption == kAudioSessionEndInterruption && + it->stream->interrupted == PJ_TRUE) + { + UInt32 audioCategory; + OSStatus ostatus; + + /* Make sure that your application can receive remote control + * events by adding the code: + * [[UIApplication sharedApplication] + * beginReceivingRemoteControlEvents]; + * Otherwise audio unit will fail to restart while your + * application is in the background mode. + */ + /* Make sure we set the correct audio category before restarting */ + audioCategory = kAudioSessionCategory_PlayAndRecord; + ostatus = AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, + sizeof(audioCategory), + &audioCategory); + if (ostatus != kAudioSessionNoError) { + PJ_LOG(4, (THIS_FILE, + "Warning: cannot set the audio session category (%i)", + ostatus)); + } + + /* Restart the stream */ + status = ca_stream_start((pjmedia_aud_stream*)it->stream); + if (status != PJ_SUCCESS) { + PJ_LOG(3, (THIS_FILE, + "Error: failed to restart the audio unit (%i)", + status)); + continue; + } + PJ_LOG(3, (THIS_FILE, "core audio unit successfully resumed" + " after interruption")); + } else if (inInterruption == kAudioSessionBeginInterruption && + it->stream->running == PJ_TRUE) + { + status = ca_stream_stop((pjmedia_aud_stream*)it->stream); + it->stream->interrupted = PJ_TRUE; + } + } + pj_mutex_unlock(cf_instance->mutex); +} + +#endif + +#if COREAUDIO_MAC +/* Internal: create audio converter for resampling the recorder device */ +static pj_status_t create_audio_resample(struct coreaudio_stream *strm, + AudioStreamBasicDescription *desc) +{ + OSStatus ostatus; + + pj_assert(strm->streamFormat.mSampleRate != desc->mSampleRate); + pj_assert(NULL == strm->resample); + pj_assert(NULL == strm->resample_buf); + + /* Create the audio converter */ + ostatus = AudioConverterNew(desc, &strm->streamFormat, &strm->resample); + if (ostatus != noErr) { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + + /* + * Allocate the buffer required to hold enough input data + */ + strm->resample_buf_size = (unsigned)(desc->mSampleRate * + strm->param.samples_per_frame / + strm->param.clock_rate); + strm->resample_buf = (pj_int16_t*) + pj_pool_alloc(strm->pool, + strm->resample_buf_size * + strm->param.bits_per_sample >> 3); + if (!strm->resample_buf) + return PJ_ENOMEM; + strm->resample_buf_count = 0; + + return PJ_SUCCESS; +} +#endif + +/* Internal: create audio unit for recorder/playback device */ +static pj_status_t create_audio_unit(AudioComponent io_comp, + AudioDeviceID dev_id, + pjmedia_dir dir, + struct coreaudio_stream *strm, + AudioUnit *io_unit) +{ + OSStatus ostatus; +#if !COREAUDIO_MAC + /* We want to be able to open playback and recording streams */ + strm->sess = [AVAudioSession sharedInstance]; + if ([strm->sess setCategory:AVAudioSessionCategoryPlayAndRecord + error:nil] != YES) + { + PJ_LOG(4, (THIS_FILE, + "Warning: cannot set the audio session category")); + } +#endif + + /* Create an audio unit to interface with the device */ + ostatus = AudioComponentInstanceNew(io_comp, io_unit); + if (ostatus != noErr) { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + + /* Set audio unit's properties for capture device */ + if (dir & PJMEDIA_DIR_CAPTURE) { + UInt32 enable = 1; + + /* Enable input */ + ostatus = AudioUnitSetProperty(*io_unit, + kAudioOutputUnitProperty_EnableIO, + kAudioUnitScope_Input, + 1, + &enable, + sizeof(enable)); + if (ostatus != noErr) { + PJ_LOG(4, (THIS_FILE, + "Warning: cannot enable IO of capture device %d", + dev_id)); + } + + /* Disable output */ + if (!(dir & PJMEDIA_DIR_PLAYBACK)) { + enable = 0; + ostatus = AudioUnitSetProperty(*io_unit, + kAudioOutputUnitProperty_EnableIO, + kAudioUnitScope_Output, + 0, + &enable, + sizeof(enable)); + if (ostatus != noErr) { + PJ_LOG(4, (THIS_FILE, + "Warning: cannot disable IO of capture device %d", + dev_id)); + } + } + } + + /* Set audio unit's properties for playback device */ + if (dir & PJMEDIA_DIR_PLAYBACK) { + UInt32 enable = 1; + + /* Enable output */ + ostatus = AudioUnitSetProperty(*io_unit, + kAudioOutputUnitProperty_EnableIO, + kAudioUnitScope_Output, + 0, + &enable, + sizeof(enable)); + if (ostatus != noErr) { + PJ_LOG(4, (THIS_FILE, + "Warning: cannot enable IO of playback device %d", + dev_id)); + } + + } + +#if COREAUDIO_MAC + PJ_LOG(5, (THIS_FILE, "Opening device %d", dev_id)); + ostatus = AudioUnitSetProperty(*io_unit, + kAudioOutputUnitProperty_CurrentDevice, + kAudioUnitScope_Global, + 0, + &dev_id, + sizeof(dev_id)); + if (ostatus != noErr) { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } +#endif + + if (dir & PJMEDIA_DIR_CAPTURE) { +#if COREAUDIO_MAC + AudioStreamBasicDescription deviceFormat; + UInt32 size; + + /* + * Keep the sample rate from the device, otherwise we will confuse + * AUHAL + */ + size = sizeof(AudioStreamBasicDescription); + ostatus = AudioUnitGetProperty(*io_unit, + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Input, + 1, + &deviceFormat, + &size); + if (ostatus != noErr) { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + strm->streamFormat.mSampleRate = deviceFormat.mSampleRate; +#endif + + /* When setting the stream format, we have to make sure the sample + * rate is supported. Setting an unsupported sample rate will cause + * AudioUnitRender() to fail later. + */ + ostatus = AudioUnitSetProperty(*io_unit, + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Output, + 1, + &strm->streamFormat, + sizeof(strm->streamFormat)); + if (ostatus != noErr) { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + +#if COREAUDIO_MAC + strm->streamFormat.mSampleRate = strm->param.clock_rate; + size = sizeof(AudioStreamBasicDescription); + ostatus = AudioUnitGetProperty (*io_unit, + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Output, + 1, + &deviceFormat, + &size); + if (ostatus == noErr) { + if (strm->streamFormat.mSampleRate != deviceFormat.mSampleRate) { + pj_status_t rc = create_audio_resample(strm, &deviceFormat); + if (PJ_SUCCESS != rc) + return rc; + } + } else { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } +#endif + } + + if (dir & PJMEDIA_DIR_PLAYBACK) { + AURenderCallbackStruct output_cb; + + /* Set the stream format */ + ostatus = AudioUnitSetProperty(*io_unit, + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Input, + 0, + &strm->streamFormat, + sizeof(strm->streamFormat)); + if (ostatus != noErr) { + PJ_LOG(4, (THIS_FILE, + "Warning: cannot set playback stream format of dev %d", + dev_id)); + } + + /* Set render callback */ + output_cb.inputProc = output_renderer; + output_cb.inputProcRefCon = strm; + ostatus = AudioUnitSetProperty(*io_unit, + kAudioUnitProperty_SetRenderCallback, + kAudioUnitScope_Input, + 0, + &output_cb, + sizeof(output_cb)); + if (ostatus != noErr) { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + + /* Allocate playback buffer */ + strm->play_buf = (pj_int16_t*)pj_pool_alloc(strm->pool, + strm->param.samples_per_frame * + strm->param.bits_per_sample >> 3); + if (!strm->play_buf) + return PJ_ENOMEM; + strm->play_buf_count = 0; + } + + if (dir & PJMEDIA_DIR_CAPTURE) { + AURenderCallbackStruct input_cb; +#if COREAUDIO_MAC + AudioBuffer *ab; + UInt32 size, buf_size; +#endif + + /* Set input callback */ + input_cb.inputProc = strm->resample ? resample_callback : + input_callback; + input_cb.inputProcRefCon = strm; + ostatus = AudioUnitSetProperty( + *io_unit, + kAudioOutputUnitProperty_SetInputCallback, + kAudioUnitScope_Global, + 0, + &input_cb, + sizeof(input_cb)); + if (ostatus != noErr) { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + +#if COREAUDIO_MAC + /* Get device's buffer frame size */ + size = sizeof(UInt32); + ostatus = AudioUnitGetProperty(*io_unit, + kAudioDevicePropertyBufferFrameSize, + kAudioUnitScope_Global, + 0, + &buf_size, + &size); + if (ostatus != noErr) + { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + + /* Allocate audio buffer */ + strm->audio_buf = (AudioBufferList*)pj_pool_alloc(strm->pool, + sizeof(AudioBufferList) + sizeof(AudioBuffer)); + if (!strm->audio_buf) + return PJ_ENOMEM; + + strm->audio_buf->mNumberBuffers = 1; + ab = &strm->audio_buf->mBuffers[0]; + ab->mNumberChannels = strm->streamFormat.mChannelsPerFrame; + ab->mDataByteSize = buf_size * ab->mNumberChannels * + strm->param.bits_per_sample >> 3; + ab->mData = pj_pool_alloc(strm->pool, + ab->mDataByteSize); + if (!ab->mData) + return PJ_ENOMEM; + +#else + /* We will let AudioUnitRender() to allocate the buffer + * for us later + */ + strm->audio_buf = (AudioBufferList*)pj_pool_alloc(strm->pool, + sizeof(AudioBufferList) + sizeof(AudioBuffer)); + if (!strm->audio_buf) + return PJ_ENOMEM; + + strm->audio_buf->mNumberBuffers = 1; + strm->audio_buf->mBuffers[0].mNumberChannels = + strm->streamFormat.mChannelsPerFrame; + +#endif + + /* Allocate recording buffer */ + strm->rec_buf = (pj_int16_t*)pj_pool_alloc(strm->pool, + strm->param.samples_per_frame * + strm->param.bits_per_sample >> 3); + if (!strm->rec_buf) + return PJ_ENOMEM; + strm->rec_buf_count = 0; + } + + /* Initialize the audio unit */ + ostatus = AudioUnitInitialize(*io_unit); + if (ostatus != noErr) { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + + return PJ_SUCCESS; +} + +/* API: create stream */ +static pj_status_t ca_factory_create_stream(pjmedia_aud_dev_factory *f, + const pjmedia_aud_param *param, + pjmedia_aud_rec_cb rec_cb, + pjmedia_aud_play_cb play_cb, + void *user_data, + pjmedia_aud_stream **p_aud_strm) +{ + struct coreaudio_factory *cf = (struct coreaudio_factory*)f; + pj_pool_t *pool; + struct coreaudio_stream *strm; + pj_status_t status; + + /* Create and Initialize stream descriptor */ + pool = pj_pool_create(cf->pf, "coreaudio-dev", 1000, 1000, NULL); + PJ_ASSERT_RETURN(pool != NULL, PJ_ENOMEM); + + strm = PJ_POOL_ZALLOC_T(pool, struct coreaudio_stream); + pj_list_init(&strm->list_entry); + strm->list_entry.stream = strm; + strm->cf = cf; + pj_memcpy(&strm->param, param, sizeof(*param)); + strm->pool = pool; + strm->rec_cb = rec_cb; + strm->play_cb = play_cb; + strm->user_data = user_data; + + /* Set the stream format */ + strm->streamFormat.mSampleRate = param->clock_rate; + strm->streamFormat.mFormatID = kAudioFormatLinearPCM; + strm->streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger + | kLinearPCMFormatFlagIsPacked; + strm->streamFormat.mBitsPerChannel = strm->param.bits_per_sample; + strm->streamFormat.mChannelsPerFrame = param->channel_count; + strm->streamFormat.mBytesPerFrame = strm->streamFormat.mChannelsPerFrame + * strm->param.bits_per_sample >> 3; + strm->streamFormat.mFramesPerPacket = 1; + strm->streamFormat.mBytesPerPacket = strm->streamFormat.mBytesPerFrame * + strm->streamFormat.mFramesPerPacket; + + /* Apply input/output routes settings before we create the audio units */ + if (param->flags & PJMEDIA_AUD_DEV_CAP_INPUT_ROUTE) { + ca_stream_set_cap(&strm->base, + PJMEDIA_AUD_DEV_CAP_INPUT_ROUTE, + ¶m->input_route); + } + if (param->flags & PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE) { + ca_stream_set_cap(&strm->base, + PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE, + ¶m->output_route); + } + if (param->flags & PJMEDIA_AUD_DEV_CAP_EC) { + ca_stream_set_cap(&strm->base, + PJMEDIA_AUD_DEV_CAP_EC, + ¶m->ec_enabled); + } else { + pj_bool_t ec = PJ_FALSE; + ca_stream_set_cap(&strm->base, + PJMEDIA_AUD_DEV_CAP_EC, &ec); + } + + strm->io_units[0] = strm->io_units[1] = NULL; + if (param->dir == PJMEDIA_DIR_CAPTURE_PLAYBACK && + param->rec_id == param->play_id) + { + /* If both input and output are on the same device, only create + * one audio unit to interface with the device. + */ + status = create_audio_unit(cf->io_comp, + cf->dev_info[param->rec_id].dev_id, + param->dir, strm, &strm->io_units[0]); + if (status != PJ_SUCCESS) + goto on_error; + } else { + unsigned nunits = 0; + + if (param->dir & PJMEDIA_DIR_CAPTURE) { + status = create_audio_unit(cf->io_comp, + cf->dev_info[param->rec_id].dev_id, + PJMEDIA_DIR_CAPTURE, + strm, &strm->io_units[nunits++]); + if (status != PJ_SUCCESS) + goto on_error; + } + if (param->dir & PJMEDIA_DIR_PLAYBACK) { + + status = create_audio_unit(cf->io_comp, + cf->dev_info[param->play_id].dev_id, + PJMEDIA_DIR_PLAYBACK, + strm, &strm->io_units[nunits++]); + if (status != PJ_SUCCESS) + goto on_error; + } + } + + /* Apply the remaining settings */ + if (param->flags & PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY) { + ca_stream_get_cap(&strm->base, + PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY, + &strm->param.input_latency_ms); + } + if (param->flags & PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY) { + ca_stream_get_cap(&strm->base, + PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY, + &strm->param.output_latency_ms); + } + if (param->flags & PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING) { + ca_stream_set_cap(&strm->base, + PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING, + ¶m->output_vol); + } + + pj_mutex_lock(strm->cf->mutex); + pj_assert(pj_list_empty(&strm->list_entry)); + pj_list_insert_after(&strm->cf->streams, &strm->list_entry); + pj_mutex_unlock(strm->cf->mutex); + + /* Done */ + strm->base.op = &stream_op; + *p_aud_strm = &strm->base; + + return PJ_SUCCESS; + + on_error: + ca_stream_destroy((pjmedia_aud_stream *)strm); + return status; +} + +/* API: Get stream info. */ +static pj_status_t ca_stream_get_param(pjmedia_aud_stream *s, + pjmedia_aud_param *pi) +{ + struct coreaudio_stream *strm = (struct coreaudio_stream*)s; + + PJ_ASSERT_RETURN(strm && pi, PJ_EINVAL); + + pj_memcpy(pi, &strm->param, sizeof(*pi)); + + /* Update the device capabilities' values */ + if (ca_stream_get_cap(s, PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY, + &pi->input_latency_ms) == PJ_SUCCESS) + { + pi->flags |= PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY; + } + if (ca_stream_get_cap(s, PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY, + &pi->output_latency_ms) == PJ_SUCCESS) + { + pi->flags |= PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY; + } + if (ca_stream_get_cap(s, PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING, + &pi->output_vol) == PJ_SUCCESS) + { + pi->flags |= PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING; + } + if (ca_stream_get_cap(s, PJMEDIA_AUD_DEV_CAP_INPUT_ROUTE, + &pi->input_route) == PJ_SUCCESS) + { + pi->flags |= PJMEDIA_AUD_DEV_CAP_INPUT_ROUTE; + } + if (ca_stream_get_cap(s, PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE, + &pi->output_route) == PJ_SUCCESS) + { + pi->flags |= PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE; + } + if (ca_stream_get_cap(s, PJMEDIA_AUD_DEV_CAP_EC, + &pi->ec_enabled) == PJ_SUCCESS) + { + pi->flags |= PJMEDIA_AUD_DEV_CAP_EC; + } + + return PJ_SUCCESS; +} + +/* API: get capability */ +static pj_status_t ca_stream_get_cap(pjmedia_aud_stream *s, + pjmedia_aud_dev_cap cap, + void *pval) +{ + struct coreaudio_stream *strm = (struct coreaudio_stream*)s; + + PJ_UNUSED_ARG(strm); + + PJ_ASSERT_RETURN(s && pval, PJ_EINVAL); + + if (cap==PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY && + (strm->param.dir & PJMEDIA_DIR_CAPTURE)) + { +#if COREAUDIO_MAC + UInt32 latency, size = sizeof(UInt32); + + /* Recording latency */ + if (AudioUnitGetProperty (strm->io_units[0], + kAudioDevicePropertyLatency, + kAudioUnitScope_Input, + 1, + &latency, + &size) == noErr) + { + UInt32 latency2; + if (AudioUnitGetProperty (strm->io_units[0], + kAudioDevicePropertyBufferFrameSize, + kAudioUnitScope_Input, + 1, + &latency2, + &size) == noErr) + { + strm->param.input_latency_ms = (latency + latency2) * 1000 / + strm->param.clock_rate; + strm->param.input_latency_ms++; + } + } +#else + if ([strm->sess respondsToSelector:@selector(inputLatency)]) { + strm->param.input_latency_ms = + (unsigned)(([strm->sess inputLatency] + + [strm->sess IOBufferDuration]) * 1000); + strm->param.input_latency_ms++; + } else + return PJMEDIA_EAUD_INVCAP; +#endif + + *(unsigned*)pval = strm->param.input_latency_ms; + return PJ_SUCCESS; + } else if (cap==PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY && + (strm->param.dir & PJMEDIA_DIR_PLAYBACK)) + { +#if COREAUDIO_MAC + UInt32 latency, size = sizeof(UInt32); + AudioUnit *io_unit = strm->io_units[1] ? &strm->io_units[1] : + &strm->io_units[0]; + + /* Playback latency */ + if (AudioUnitGetProperty (*io_unit, + kAudioDevicePropertyLatency, + kAudioUnitScope_Output, + 0, + &latency, + &size) == noErr) + { + UInt32 latency2; + if (AudioUnitGetProperty (*io_unit, + kAudioDevicePropertyBufferFrameSize, + kAudioUnitScope_Output, + 0, + &latency2, + &size) == noErr) + { + strm->param.output_latency_ms = (latency + latency2) * 1000 / + strm->param.clock_rate; + strm->param.output_latency_ms++; + } + } +#else + if ([strm->sess respondsToSelector:@selector(outputLatency)]) { + strm->param.output_latency_ms = + (unsigned)(([strm->sess outputLatency] + + [strm->sess IOBufferDuration]) * 1000); + strm->param.output_latency_ms++; + } else + return PJMEDIA_EAUD_INVCAP; +#endif + *(unsigned*)pval = (++strm->param.output_latency_ms * 2); + return PJ_SUCCESS; + } else if (cap==PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING && + (strm->param.dir & PJMEDIA_DIR_PLAYBACK)) + { +#if COREAUDIO_MAC + OSStatus ostatus; + Float32 volume; + UInt32 size = sizeof(Float32); + + /* Output volume setting */ + ostatus = AudioUnitGetProperty (strm->io_units[1] ? strm->io_units[1] : + strm->io_units[0], + kAudioDevicePropertyVolumeScalar, + kAudioUnitScope_Output, + 0, + &volume, + &size); + if (ostatus != noErr) + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + + *(unsigned*)pval = (unsigned)(volume * 100); + return PJ_SUCCESS; +#else + if ([strm->sess respondsToSelector:@selector(outputVolume)]) { + *(unsigned*)pval = (unsigned)([strm->sess outputVolume] * 100); + return PJ_SUCCESS; + } else + return PJMEDIA_EAUD_INVCAP; +#endif + +#if !COREAUDIO_MAC +#if USE_AUDIO_SESSION_API != 0 + } else if (cap==PJMEDIA_AUD_DEV_CAP_INPUT_ROUTE && + (strm->param.dir & PJMEDIA_DIR_CAPTURE)) + { + UInt32 btooth, size = sizeof(UInt32); + OSStatus ostatus; + + ostatus = AudioSessionGetProperty ( + kAudioSessionProperty_OverrideCategoryEnableBluetoothInput, + &size, &btooth); + if (ostatus != kAudioSessionNoError) { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + + *(pjmedia_aud_dev_route*)pval = btooth? + PJMEDIA_AUD_DEV_ROUTE_BLUETOOTH: + PJMEDIA_AUD_DEV_ROUTE_DEFAULT; + return PJ_SUCCESS; + } else if (cap==PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE && + (strm->param.dir & PJMEDIA_DIR_PLAYBACK)) + { + CFStringRef route; + UInt32 size = sizeof(CFStringRef); + OSStatus ostatus; + + ostatus = AudioSessionGetProperty (kAudioSessionProperty_AudioRoute, + &size, &route); + if (ostatus != kAudioSessionNoError) { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + + if (!route) { + *(pjmedia_aud_dev_route*)pval = PJMEDIA_AUD_DEV_ROUTE_DEFAULT; + } else if (CFStringHasPrefix(route, CFSTR("Headset"))) { + *(pjmedia_aud_dev_route*)pval = PJMEDIA_AUD_DEV_ROUTE_EARPIECE; + } else { + *(pjmedia_aud_dev_route*)pval = PJMEDIA_AUD_DEV_ROUTE_DEFAULT; + } + + CFRelease(route); + + return PJ_SUCCESS; +#endif + } else if (cap==PJMEDIA_AUD_DEV_CAP_EC) { + AudioComponentDescription desc; + OSStatus ostatus; + + ostatus = AudioComponentGetDescription(strm->cf->io_comp, &desc); + if (ostatus != noErr) { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + + *(pj_bool_t*)pval = (desc.componentSubType == + kAudioUnitSubType_VoiceProcessingIO); + return PJ_SUCCESS; +#endif + } else { + return PJMEDIA_EAUD_INVCAP; + } +} + +/* API: set capability */ +static pj_status_t ca_stream_set_cap(pjmedia_aud_stream *s, + pjmedia_aud_dev_cap cap, + const void *pval) +{ + struct coreaudio_stream *strm = (struct coreaudio_stream*)s; + + PJ_ASSERT_RETURN(s && pval, PJ_EINVAL); + +#if COREAUDIO_MAC + if (cap==PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING && + (strm->param.dir & PJMEDIA_DIR_PLAYBACK)) + { + OSStatus ostatus; + Float32 volume = *(unsigned*)pval; + + /* Output volume setting */ + volume /= 100.0; + ostatus = AudioUnitSetProperty (strm->io_units[1] ? strm->io_units[1] : + strm->io_units[0], + kAudioDevicePropertyVolumeScalar, + kAudioUnitScope_Output, + 0, + &volume, + sizeof(Float32)); + if (ostatus != noErr) { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + strm->param.output_vol = *(unsigned*)pval; + return PJ_SUCCESS; + } + +#else + if (cap==PJMEDIA_AUD_DEV_CAP_EC) { + AudioComponentDescription desc; + AudioComponent io_comp; + + desc.componentType = kAudioUnitType_Output; + desc.componentSubType = (*(pj_bool_t*)pval)? + kAudioUnitSubType_VoiceProcessingIO : + kAudioUnitSubType_RemoteIO; + desc.componentManufacturer = kAudioUnitManufacturer_Apple; + desc.componentFlags = 0; + desc.componentFlagsMask = 0; + + io_comp = AudioComponentFindNext(NULL, &desc); + if (io_comp == NULL) + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(-1); + strm->cf->io_comp = io_comp; + strm->param.ec_enabled = *(pj_bool_t*)pval; + + PJ_LOG(4, (THIS_FILE, "Using %s audio unit", + (desc.componentSubType == + kAudioUnitSubType_RemoteIO? "RemoteIO": + "VoiceProcessingIO"))); + + return PJ_SUCCESS; + } else if ((cap==PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY && + (strm->param.dir & PJMEDIA_DIR_CAPTURE)) || + (cap==PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY && + (strm->param.dir & PJMEDIA_DIR_PLAYBACK))) + { + NSTimeInterval duration = *(unsigned *)pval; + unsigned latency; + + /* For low-latency audio streaming, you can set this value to + * as low as 5 ms (the default is 23ms). However, lowering the + * latency may cause a decrease in audio quality. + */ + duration /= 1000; + if ([strm->sess setPreferredIOBufferDuration:duration error:nil] + != YES) + { + PJ_LOG(4, (THIS_FILE, + "Error: cannot set the preferred buffer duration")); + return PJMEDIA_EAUD_INVOP; + } + + ca_stream_get_cap(s, PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY, &latency); + ca_stream_get_cap(s, PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY, &latency); + + return PJ_SUCCESS; + } + +#if USE_AUDIO_SESSION_API != 0 + + else if (cap==PJMEDIA_AUD_DEV_CAP_INPUT_ROUTE && + (strm->param.dir & PJMEDIA_DIR_CAPTURE)) + { + UInt32 btooth = *(pjmedia_aud_dev_route*)pval == + PJMEDIA_AUD_DEV_ROUTE_BLUETOOTH ? 1 : 0; + OSStatus ostatus; + + ostatus = AudioSessionSetProperty ( + kAudioSessionProperty_OverrideCategoryEnableBluetoothInput, + sizeof(btooth), &btooth); + if (ostatus != kAudioSessionNoError) { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + strm->param.input_route = *(pjmedia_aud_dev_route*)pval; + return PJ_SUCCESS; + } else if (cap==PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE && + (strm->param.dir & PJMEDIA_DIR_PLAYBACK)) + { + OSStatus ostatus; + UInt32 route = *(pjmedia_aud_dev_route*)pval == + PJMEDIA_AUD_DEV_ROUTE_LOUDSPEAKER ? + kAudioSessionOverrideAudioRoute_Speaker : + kAudioSessionOverrideAudioRoute_None; + + ostatus = AudioSessionSetProperty ( + kAudioSessionProperty_OverrideAudioRoute, + sizeof(route), &route); + if (ostatus != kAudioSessionNoError) { + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + strm->param.output_route = *(pjmedia_aud_dev_route*)pval; + return PJ_SUCCESS; + } +#endif +#endif + + return PJMEDIA_EAUD_INVCAP; +} + +/* API: Start stream. */ +static pj_status_t ca_stream_start(pjmedia_aud_stream *strm) +{ + struct coreaudio_stream *stream = (struct coreaudio_stream*)strm; + OSStatus ostatus; + UInt32 i; + + if (stream->running) + return PJ_SUCCESS; + + stream->quit_flag = 0; + stream->interrupted = PJ_FALSE; + stream->rec_buf_count = 0; + stream->play_buf_count = 0; + stream->resample_buf_count = 0; + + if (stream->resample) { + ostatus = AudioConverterReset(stream->resample); + if (ostatus != noErr) + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + +#if !COREAUDIO_MAC + if ([stream->sess setActive:true error:nil] != YES) { + PJ_LOG(4, (THIS_FILE, "Warning: cannot activate audio session")); + } +#endif + + for (i = 0; i < 2; i++) { + if (stream->io_units[i] == NULL) break; + ostatus = AudioOutputUnitStart(stream->io_units[i]); + if (ostatus != noErr) { + if (i == 1) + AudioOutputUnitStop(stream->io_units[0]); + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + } + + stream->running = PJ_TRUE; + + PJ_LOG(4, (THIS_FILE, "core audio stream started")); + + return PJ_SUCCESS; +} + +/* API: Stop stream. */ +static pj_status_t ca_stream_stop(pjmedia_aud_stream *strm) +{ + struct coreaudio_stream *stream = (struct coreaudio_stream*)strm; + OSStatus ostatus; + unsigned i; + int should_deactivate; + struct stream_list *it, *itBegin; + + if (!stream->running) + return PJ_SUCCESS; + + for (i = 0; i < 2; i++) { + if (stream->io_units[i] == NULL) break; + ostatus = AudioOutputUnitStop(stream->io_units[i]); + if (ostatus != noErr) { + if (i == 0 && stream->io_units[1]) + AudioOutputUnitStop(stream->io_units[1]); + return PJMEDIA_AUDIODEV_ERRNO_FROM_COREAUDIO(ostatus); + } + } + + /* Check whether we need to deactivate the audio session. */ + pj_mutex_lock(stream->cf->mutex); + pj_assert(!pj_list_empty(&stream->cf->streams)); + pj_assert(!pj_list_empty(&stream->list_entry)); + stream->running = PJ_FALSE; + should_deactivate = PJ_TRUE; + itBegin = &stream->cf->streams; + for (it = itBegin->next; it != itBegin; it = it->next) { + if (it->stream->running) { + should_deactivate = PJ_FALSE; + break; + } + } + pj_mutex_unlock(stream->cf->mutex); + +#if !COREAUDIO_MAC + if (should_deactivate) { + if ([stream->sess setActive:false error:nil] != YES) { + PJ_LOG(4, (THIS_FILE, "Warning: cannot deactivate audio session")); + } + } +#endif + + stream->quit_flag = 1; + stream->play_thread_initialized = 0; + stream->rec_thread_initialized = 0; + pj_bzero(stream->rec_thread_desc, sizeof(pj_thread_desc)); + pj_bzero(stream->play_thread_desc, sizeof(pj_thread_desc)); + + PJ_LOG(4, (THIS_FILE, "core audio stream stopped")); + + return PJ_SUCCESS; +} + + +/* API: Destroy stream. */ +static pj_status_t ca_stream_destroy(pjmedia_aud_stream *strm) +{ + struct coreaudio_stream *stream = (struct coreaudio_stream*)strm; + unsigned i; + + PJ_ASSERT_RETURN(stream != NULL, PJ_EINVAL); + + ca_stream_stop(strm); + + for (i = 0; i < 2; i++) { + if (stream->io_units[i]) { + AudioUnitUninitialize(stream->io_units[i]); + AudioComponentInstanceDispose(stream->io_units[i]); + stream->io_units[i] = NULL; + } + } + + if (stream->resample) + AudioConverterDispose(stream->resample); + + pj_mutex_lock(stream->cf->mutex); + if (!pj_list_empty(&stream->list_entry)) + pj_list_erase(&stream->list_entry); + pj_mutex_unlock(stream->cf->mutex); + + pj_pool_release(stream->pool); + + return PJ_SUCCESS; +} + +#endif /* PJMEDIA_AUDIO_DEV_HAS_COREAUDIO */ |