| /* $Id: opensl_dev.c 4435 2013-03-11 06:32:58Z nanang $ */ |
| /* |
| * Copyright (C) 2012-2012 Teluu Inc. (http://www.teluu.com) |
| * Copyright (C) 2010-2012 Regis Montoya (aka r3gis - www.r3gis.fr) |
| * |
| * 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 |
| */ |
| /* This file is the implementation of Android OpenSL ES audio device. |
| * The original code was originally part of CSipSimple |
| * (http://code.google.com/p/csipsimple/) and was kindly donated |
| * by Regis Montoya. |
| */ |
| |
| #include <pjmedia-audiodev/audiodev_imp.h> |
| #include <pj/assert.h> |
| #include <pj/log.h> |
| #include <pj/os.h> |
| #include <pj/string.h> |
| #include <pjmedia/errno.h> |
| |
| #if defined(PJMEDIA_AUDIO_DEV_HAS_OPENSL) && PJMEDIA_AUDIO_DEV_HAS_OPENSL != 0 |
| |
| #include <SLES/OpenSLES.h> |
| |
| #ifdef __ANDROID__ |
| #include <SLES/OpenSLES_Android.h> |
| #include <SLES/OpenSLES_AndroidConfiguration.h> |
| #include <sys/system_properties.h> |
| #include <android/api-level.h> |
| |
| #define W_SLBufferQueueItf SLAndroidSimpleBufferQueueItf |
| #define W_SLBufferQueueState SLAndroidSimpleBufferQueueState |
| #define W_SL_IID_BUFFERQUEUE SL_IID_ANDROIDSIMPLEBUFFERQUEUE |
| #else |
| #define W_SLBufferQueueItf SLBufferQueueItf |
| #define W_SLBufferQueueState SLBufferQueueState |
| #define W_SL_IID_BUFFERQUEUE SL_IID_BUFFERQUEUE |
| #endif |
| |
| #define THIS_FILE "opensl_dev.c" |
| #define DRIVER_NAME "OpenSL" |
| |
| #define NUM_BUFFERS 2 |
| |
| struct opensl_aud_factory |
| { |
| pjmedia_aud_dev_factory base; |
| pj_pool_factory *pf; |
| pj_pool_t *pool; |
| |
| SLObjectItf engineObject; |
| SLEngineItf engineEngine; |
| SLObjectItf outputMixObject; |
| }; |
| |
| /* |
| * Sound stream descriptor. |
| * This struct may be used for both unidirectional or bidirectional sound |
| * streams. |
| */ |
| struct opensl_aud_stream |
| { |
| pjmedia_aud_stream base; |
| pj_pool_t *pool; |
| pj_str_t name; |
| pjmedia_dir dir; |
| pjmedia_aud_param param; |
| |
| void *user_data; |
| pj_bool_t quit_flag; |
| pjmedia_aud_rec_cb rec_cb; |
| pjmedia_aud_play_cb play_cb; |
| |
| pj_timestamp play_timestamp; |
| pj_timestamp rec_timestamp; |
| |
| 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; |
| |
| /* Player */ |
| SLObjectItf playerObj; |
| SLPlayItf playerPlay; |
| SLVolumeItf playerVol; |
| unsigned playerBufferSize; |
| char *playerBuffer[NUM_BUFFERS]; |
| int playerBufIdx; |
| |
| /* Recorder */ |
| SLObjectItf recordObj; |
| SLRecordItf recordRecord; |
| unsigned recordBufferSize; |
| char *recordBuffer[NUM_BUFFERS]; |
| int recordBufIdx; |
| |
| W_SLBufferQueueItf playerBufQ; |
| W_SLBufferQueueItf recordBufQ; |
| }; |
| |
| /* Factory prototypes */ |
| static pj_status_t opensl_init(pjmedia_aud_dev_factory *f); |
| static pj_status_t opensl_destroy(pjmedia_aud_dev_factory *f); |
| static pj_status_t opensl_refresh(pjmedia_aud_dev_factory *f); |
| static unsigned opensl_get_dev_count(pjmedia_aud_dev_factory *f); |
| static pj_status_t opensl_get_dev_info(pjmedia_aud_dev_factory *f, |
| unsigned index, |
| pjmedia_aud_dev_info *info); |
| static pj_status_t opensl_default_param(pjmedia_aud_dev_factory *f, |
| unsigned index, |
| pjmedia_aud_param *param); |
| static pj_status_t opensl_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); |
| |
| /* Stream prototypes */ |
| static pj_status_t strm_get_param(pjmedia_aud_stream *strm, |
| pjmedia_aud_param *param); |
| static pj_status_t strm_get_cap(pjmedia_aud_stream *strm, |
| pjmedia_aud_dev_cap cap, |
| void *value); |
| static pj_status_t strm_set_cap(pjmedia_aud_stream *strm, |
| pjmedia_aud_dev_cap cap, |
| const void *value); |
| static pj_status_t strm_start(pjmedia_aud_stream *strm); |
| static pj_status_t strm_stop(pjmedia_aud_stream *strm); |
| static pj_status_t strm_destroy(pjmedia_aud_stream *strm); |
| |
| static pjmedia_aud_dev_factory_op opensl_op = |
| { |
| &opensl_init, |
| &opensl_destroy, |
| &opensl_get_dev_count, |
| &opensl_get_dev_info, |
| &opensl_default_param, |
| &opensl_create_stream, |
| &opensl_refresh |
| }; |
| |
| static pjmedia_aud_stream_op opensl_strm_op = |
| { |
| &strm_get_param, |
| &strm_get_cap, |
| &strm_set_cap, |
| &strm_start, |
| &strm_stop, |
| &strm_destroy |
| }; |
| |
| /* This callback is called every time a buffer finishes playing. */ |
| void bqPlayerCallback(W_SLBufferQueueItf bq, void *context) |
| { |
| struct opensl_aud_stream *stream = (struct opensl_aud_stream*) context; |
| SLresult result; |
| int status; |
| |
| pj_assert(context != NULL); |
| pj_assert(bq == stream->playerBufQ); |
| |
| if (stream->play_thread_initialized == 0 || !pj_thread_is_registered()) |
| { |
| pj_bzero(stream->play_thread_desc, sizeof(pj_thread_desc)); |
| status = pj_thread_register("opensl_play", stream->play_thread_desc, |
| &stream->play_thread); |
| stream->play_thread_initialized = 1; |
| PJ_LOG(5, (THIS_FILE, "Player thread started")); |
| } |
| |
| if (!stream->quit_flag) { |
| pjmedia_frame frame; |
| char * buf; |
| |
| frame.type = PJMEDIA_FRAME_TYPE_AUDIO; |
| frame.buf = buf = stream->playerBuffer[stream->playerBufIdx++]; |
| frame.size = stream->playerBufferSize; |
| frame.timestamp.u64 = stream->play_timestamp.u64; |
| frame.bit_info = 0; |
| |
| status = (*stream->play_cb)(stream->user_data, &frame); |
| if (status != PJ_SUCCESS || frame.type != PJMEDIA_FRAME_TYPE_AUDIO) |
| pj_bzero(buf, stream->playerBufferSize); |
| |
| stream->play_timestamp.u64 += stream->param.samples_per_frame / |
| stream->param.channel_count; |
| |
| result = (*bq)->Enqueue(bq, buf, stream->playerBufferSize); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Unable to enqueue next player buffer !!! %d", |
| result)); |
| } |
| |
| stream->playerBufIdx %= NUM_BUFFERS; |
| } |
| } |
| |
| /* This callback handler is called every time a buffer finishes recording */ |
| void bqRecorderCallback(W_SLBufferQueueItf bq, void *context) |
| { |
| struct opensl_aud_stream *stream = (struct opensl_aud_stream*) context; |
| SLresult result; |
| int status; |
| |
| pj_assert(context != NULL); |
| pj_assert(bq == stream->recordBufQ); |
| |
| if (stream->rec_thread_initialized == 0 || !pj_thread_is_registered()) |
| { |
| pj_bzero(stream->rec_thread_desc, sizeof(pj_thread_desc)); |
| status = pj_thread_register("opensl_rec", stream->rec_thread_desc, |
| &stream->rec_thread); |
| stream->rec_thread_initialized = 1; |
| PJ_LOG(5, (THIS_FILE, "Recorder thread started")); |
| } |
| |
| if (!stream->quit_flag) { |
| pjmedia_frame frame; |
| char *buf; |
| |
| frame.type = PJMEDIA_FRAME_TYPE_AUDIO; |
| frame.buf = buf = stream->recordBuffer[stream->recordBufIdx++]; |
| frame.size = stream->recordBufferSize; |
| frame.timestamp.u64 = stream->rec_timestamp.u64; |
| frame.bit_info = 0; |
| |
| status = (*stream->rec_cb)(stream->user_data, &frame); |
| |
| stream->rec_timestamp.u64 += stream->param.samples_per_frame / |
| stream->param.channel_count; |
| |
| /* And now enqueue next buffer */ |
| result = (*bq)->Enqueue(bq, buf, stream->recordBufferSize); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Unable to enqueue next record buffer !!! %d", |
| result)); |
| } |
| |
| stream->recordBufIdx %= NUM_BUFFERS; |
| } |
| } |
| |
| pj_status_t opensl_to_pj_error(SLresult code) |
| { |
| switch(code) { |
| case SL_RESULT_SUCCESS: |
| return PJ_SUCCESS; |
| case SL_RESULT_PRECONDITIONS_VIOLATED: |
| case SL_RESULT_PARAMETER_INVALID: |
| case SL_RESULT_CONTENT_CORRUPTED: |
| case SL_RESULT_FEATURE_UNSUPPORTED: |
| return PJMEDIA_EAUD_INVOP; |
| case SL_RESULT_MEMORY_FAILURE: |
| case SL_RESULT_BUFFER_INSUFFICIENT: |
| return PJ_ENOMEM; |
| case SL_RESULT_RESOURCE_ERROR: |
| case SL_RESULT_RESOURCE_LOST: |
| case SL_RESULT_CONTROL_LOST: |
| return PJMEDIA_EAUD_NOTREADY; |
| case SL_RESULT_CONTENT_UNSUPPORTED: |
| return PJ_ENOTSUP; |
| default: |
| return PJMEDIA_EAUD_ERR; |
| } |
| } |
| |
| /* Init Android audio driver. */ |
| pjmedia_aud_dev_factory* pjmedia_opensl_factory(pj_pool_factory *pf) |
| { |
| struct opensl_aud_factory *f; |
| pj_pool_t *pool; |
| |
| pool = pj_pool_create(pf, "opensles", 256, 256, NULL); |
| f = PJ_POOL_ZALLOC_T(pool, struct opensl_aud_factory); |
| f->pf = pf; |
| f->pool = pool; |
| f->base.op = &opensl_op; |
| |
| return &f->base; |
| } |
| |
| /* API: Init factory */ |
| static pj_status_t opensl_init(pjmedia_aud_dev_factory *f) |
| { |
| struct opensl_aud_factory *pa = (struct opensl_aud_factory*)f; |
| SLresult result; |
| |
| /* Create engine */ |
| result = slCreateEngine(&pa->engineObject, 0, NULL, 0, NULL, NULL); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot create engine %d ", result)); |
| return opensl_to_pj_error(result); |
| } |
| |
| /* Realize the engine */ |
| result = (*pa->engineObject)->Realize(pa->engineObject, SL_BOOLEAN_FALSE); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot realize engine")); |
| opensl_destroy(f); |
| return opensl_to_pj_error(result); |
| } |
| |
| /* Get the engine interface, which is needed in order to create |
| * other objects. |
| */ |
| result = (*pa->engineObject)->GetInterface(pa->engineObject, |
| SL_IID_ENGINE, |
| &pa->engineEngine); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot get engine interface")); |
| opensl_destroy(f); |
| return opensl_to_pj_error(result); |
| } |
| |
| /* Create output mix */ |
| result = (*pa->engineEngine)->CreateOutputMix(pa->engineEngine, |
| &pa->outputMixObject, |
| 0, NULL, NULL); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot create output mix")); |
| opensl_destroy(f); |
| return opensl_to_pj_error(result); |
| } |
| |
| /* Realize the output mix */ |
| result = (*pa->outputMixObject)->Realize(pa->outputMixObject, |
| SL_BOOLEAN_FALSE); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot realize output mix")); |
| opensl_destroy(f); |
| return opensl_to_pj_error(result); |
| } |
| |
| PJ_LOG(4,(THIS_FILE, "OpenSL sound library initialized")); |
| return PJ_SUCCESS; |
| } |
| |
| /* API: Destroy factory */ |
| static pj_status_t opensl_destroy(pjmedia_aud_dev_factory *f) |
| { |
| struct opensl_aud_factory *pa = (struct opensl_aud_factory*)f; |
| pj_pool_t *pool; |
| |
| PJ_LOG(4,(THIS_FILE, "OpenSL sound library shutting down..")); |
| |
| /* Destroy Output Mix object */ |
| if (pa->outputMixObject) { |
| (*pa->outputMixObject)->Destroy(pa->outputMixObject); |
| pa->outputMixObject = NULL; |
| } |
| |
| /* Destroy engine object, and invalidate all associated interfaces */ |
| if (pa->engineObject) { |
| (*pa->engineObject)->Destroy(pa->engineObject); |
| pa->engineObject = NULL; |
| pa->engineEngine = NULL; |
| } |
| |
| pool = pa->pool; |
| pa->pool = NULL; |
| pj_pool_release(pool); |
| |
| return PJ_SUCCESS; |
| } |
| |
| /* API: refresh the list of devices */ |
| static pj_status_t opensl_refresh(pjmedia_aud_dev_factory *f) |
| { |
| PJ_UNUSED_ARG(f); |
| return PJ_SUCCESS; |
| } |
| |
| /* API: Get device count. */ |
| static unsigned opensl_get_dev_count(pjmedia_aud_dev_factory *f) |
| { |
| PJ_UNUSED_ARG(f); |
| return 1; |
| } |
| |
| /* API: Get device info. */ |
| static pj_status_t opensl_get_dev_info(pjmedia_aud_dev_factory *f, |
| unsigned index, |
| pjmedia_aud_dev_info *info) |
| { |
| PJ_UNUSED_ARG(f); |
| |
| pj_bzero(info, sizeof(*info)); |
| |
| pj_ansi_strcpy(info->name, "OpenSL ES Audio"); |
| info->default_samples_per_sec = 8000; |
| info->caps = PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING; |
| info->input_count = 1; |
| info->output_count = 1; |
| |
| return PJ_SUCCESS; |
| } |
| |
| /* API: fill in with default parameter. */ |
| static pj_status_t opensl_default_param(pjmedia_aud_dev_factory *f, |
| unsigned index, |
| pjmedia_aud_param *param) |
| { |
| |
| pjmedia_aud_dev_info adi; |
| pj_status_t status; |
| |
| status = opensl_get_dev_info(f, index, &adi); |
| if (status != PJ_SUCCESS) |
| return status; |
| |
| pj_bzero(param, sizeof(*param)); |
| if (adi.input_count && adi.output_count) { |
| param->dir = PJMEDIA_DIR_CAPTURE_PLAYBACK; |
| param->rec_id = index; |
| param->play_id = index; |
| } else if (adi.input_count) { |
| param->dir = PJMEDIA_DIR_CAPTURE; |
| param->rec_id = index; |
| param->play_id = PJMEDIA_AUD_INVALID_DEV; |
| } else if (adi.output_count) { |
| param->dir = PJMEDIA_DIR_PLAYBACK; |
| param->play_id = index; |
| param->rec_id = PJMEDIA_AUD_INVALID_DEV; |
| } else { |
| return PJMEDIA_EAUD_INVDEV; |
| } |
| |
| param->clock_rate = adi.default_samples_per_sec; |
| param->channel_count = 1; |
| param->samples_per_frame = adi.default_samples_per_sec * 20 / 1000; |
| param->bits_per_sample = 16; |
| param->input_latency_ms = PJMEDIA_SND_DEFAULT_REC_LATENCY; |
| param->output_latency_ms = PJMEDIA_SND_DEFAULT_PLAY_LATENCY; |
| |
| return PJ_SUCCESS; |
| } |
| |
| /* API: create stream */ |
| static pj_status_t opensl_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) |
| { |
| /* Audio sink for recorder and audio source for player */ |
| #ifdef __ANDROID__ |
| SLDataLocator_AndroidSimpleBufferQueue loc_bq = |
| { SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, NUM_BUFFERS }; |
| #else |
| SLDataLocator_BufferQueue loc_bq = |
| { SL_DATALOCATOR_BUFFERQUEUE, NUM_BUFFERS }; |
| #endif |
| struct opensl_aud_factory *pa = (struct opensl_aud_factory*)f; |
| pj_pool_t *pool; |
| struct opensl_aud_stream *stream; |
| pj_status_t status = PJ_SUCCESS; |
| int i, bufferSize; |
| SLresult result; |
| SLDataFormat_PCM format_pcm; |
| |
| /* Only supports for mono channel for now */ |
| PJ_ASSERT_RETURN(param->channel_count == 1, PJ_EINVAL); |
| PJ_ASSERT_RETURN(play_cb && rec_cb && p_aud_strm, PJ_EINVAL); |
| |
| PJ_LOG(4,(THIS_FILE, "Creating OpenSL stream")); |
| |
| pool = pj_pool_create(pa->pf, "openslstrm", 1024, 1024, NULL); |
| if (!pool) |
| return PJ_ENOMEM; |
| |
| stream = PJ_POOL_ZALLOC_T(pool, struct opensl_aud_stream); |
| stream->pool = pool; |
| pj_strdup2_with_null(pool, &stream->name, "OpenSL"); |
| stream->dir = PJMEDIA_DIR_CAPTURE_PLAYBACK; |
| pj_memcpy(&stream->param, param, sizeof(*param)); |
| stream->user_data = user_data; |
| stream->rec_cb = rec_cb; |
| stream->play_cb = play_cb; |
| bufferSize = param->samples_per_frame * param->bits_per_sample / 8; |
| |
| /* Configure audio PCM format */ |
| format_pcm.formatType = SL_DATAFORMAT_PCM; |
| format_pcm.numChannels = param->channel_count; |
| /* Here samples per sec should be supported else we will get an error */ |
| format_pcm.samplesPerSec = (SLuint32) param->clock_rate * 1000; |
| format_pcm.bitsPerSample = (SLuint16) param->bits_per_sample; |
| format_pcm.containerSize = (SLuint16) param->bits_per_sample; |
| format_pcm.channelMask = SL_SPEAKER_FRONT_CENTER; |
| format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN; |
| |
| if (stream->dir & PJMEDIA_DIR_PLAYBACK) { |
| /* Audio source */ |
| SLDataSource audioSrc = {&loc_bq, &format_pcm}; |
| /* Audio sink */ |
| SLDataLocator_OutputMix loc_outmix = {SL_DATALOCATOR_OUTPUTMIX, |
| pa->outputMixObject}; |
| SLDataSink audioSnk = {&loc_outmix, NULL}; |
| /* Audio interface */ |
| #ifdef __ANDROID__ |
| int numIface = 3; |
| const SLInterfaceID ids[3] = {SL_IID_BUFFERQUEUE, |
| SL_IID_VOLUME, |
| SL_IID_ANDROIDCONFIGURATION}; |
| const SLboolean req[3] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, |
| SL_BOOLEAN_TRUE}; |
| SLAndroidConfigurationItf playerConfig; |
| SLint32 streamType = SL_ANDROID_STREAM_VOICE; |
| #else |
| int numIface = 2; |
| const SLInterfaceID ids[2] = {SL_IID_BUFFERQUEUE, |
| SL_IID_VOLUME}; |
| const SLboolean req[2] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE}; |
| #endif |
| |
| /* Create audio player */ |
| result = (*pa->engineEngine)->CreateAudioPlayer(pa->engineEngine, |
| &stream->playerObj, |
| &audioSrc, &audioSnk, |
| numIface, ids, req); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot create audio player: %d", result)); |
| goto on_error; |
| } |
| |
| #ifdef __ANDROID__ |
| /* Set Android configuration */ |
| result = (*stream->playerObj)->GetInterface(stream->playerObj, |
| SL_IID_ANDROIDCONFIGURATION, |
| &playerConfig); |
| if (result == SL_RESULT_SUCCESS && playerConfig) { |
| result = (*playerConfig)->SetConfiguration( |
| playerConfig, SL_ANDROID_KEY_STREAM_TYPE, |
| &streamType, sizeof(SLint32)); |
| } |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(4, (THIS_FILE, "Warning: Unable to set android " |
| "player configuration")); |
| } |
| #endif |
| |
| /* Realize the player */ |
| result = (*stream->playerObj)->Realize(stream->playerObj, |
| SL_BOOLEAN_FALSE); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot realize player : %d", result)); |
| goto on_error; |
| } |
| |
| /* Get the play interface */ |
| result = (*stream->playerObj)->GetInterface(stream->playerObj, |
| SL_IID_PLAY, |
| &stream->playerPlay); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot get play interface")); |
| goto on_error; |
| } |
| |
| /* Get the buffer queue interface */ |
| result = (*stream->playerObj)->GetInterface(stream->playerObj, |
| SL_IID_BUFFERQUEUE, |
| &stream->playerBufQ); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot get buffer queue interface")); |
| goto on_error; |
| } |
| |
| /* Get the volume interface */ |
| result = (*stream->playerObj)->GetInterface(stream->playerObj, |
| SL_IID_VOLUME, |
| &stream->playerVol); |
| |
| /* Register callback on the buffer queue */ |
| result = (*stream->playerBufQ)->RegisterCallback(stream->playerBufQ, |
| bqPlayerCallback, |
| (void *)stream); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot register player callback")); |
| goto on_error; |
| } |
| |
| stream->playerBufferSize = bufferSize; |
| for (i = 0; i < NUM_BUFFERS; i++) { |
| stream->playerBuffer[i] = (char *) |
| pj_pool_alloc(stream->pool, |
| stream->playerBufferSize); |
| } |
| } |
| |
| if (stream->dir & PJMEDIA_DIR_CAPTURE) { |
| /* Audio source */ |
| SLDataLocator_IODevice loc_dev = {SL_DATALOCATOR_IODEVICE, |
| SL_IODEVICE_AUDIOINPUT, |
| SL_DEFAULTDEVICEID_AUDIOINPUT, |
| NULL}; |
| SLDataSource audioSrc = {&loc_dev, NULL}; |
| /* Audio sink */ |
| SLDataSink audioSnk = {&loc_bq, &format_pcm}; |
| /* Audio interface */ |
| #ifdef __ANDROID__ |
| int numIface = 2; |
| const SLInterfaceID ids[2] = {W_SL_IID_BUFFERQUEUE, |
| SL_IID_ANDROIDCONFIGURATION}; |
| const SLboolean req[2] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE}; |
| SLAndroidConfigurationItf recorderConfig; |
| #else |
| int numIface = 1; |
| const SLInterfaceID ids[1] = {W_SL_IID_BUFFERQUEUE}; |
| const SLboolean req[1] = {SL_BOOLEAN_TRUE}; |
| #endif |
| |
| /* Create audio recorder |
| * (requires the RECORD_AUDIO permission) |
| */ |
| result = (*pa->engineEngine)->CreateAudioRecorder(pa->engineEngine, |
| &stream->recordObj, |
| &audioSrc, &audioSnk, |
| numIface, ids, req); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot create recorder: %d", result)); |
| goto on_error; |
| } |
| |
| #ifdef __ANDROID__ |
| /* Set Android configuration */ |
| result = (*stream->recordObj)->GetInterface(stream->recordObj, |
| SL_IID_ANDROIDCONFIGURATION, |
| &recorderConfig); |
| if (result == SL_RESULT_SUCCESS) { |
| SLint32 streamType = SL_ANDROID_RECORDING_PRESET_GENERIC; |
| #if __ANDROID_API__ >= 14 |
| char sdk_version[PROP_VALUE_MAX]; |
| pj_str_t pj_sdk_version; |
| int sdk_v; |
| |
| __system_property_get("ro.build.version.sdk", sdk_version); |
| pj_sdk_version = pj_str(sdk_version); |
| sdk_v = pj_strtoul(&pj_sdk_version); |
| if (sdk_v >= 14) |
| streamType = SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION; |
| PJ_LOG(4, (THIS_FILE, "Recording stream type %d, SDK : %d", |
| streamType, sdk_v)); |
| #endif |
| result = (*recorderConfig)->SetConfiguration( |
| recorderConfig, SL_ANDROID_KEY_RECORDING_PRESET, |
| &streamType, sizeof(SLint32)); |
| } |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(4, (THIS_FILE, "Warning: Unable to set android " |
| "recorder configuration")); |
| } |
| #endif |
| |
| /* Realize the recorder */ |
| result = (*stream->recordObj)->Realize(stream->recordObj, |
| SL_BOOLEAN_FALSE); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot realize recorder : %d", result)); |
| goto on_error; |
| } |
| |
| /* Get the record interface */ |
| result = (*stream->recordObj)->GetInterface(stream->recordObj, |
| SL_IID_RECORD, |
| &stream->recordRecord); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot get record interface")); |
| goto on_error; |
| } |
| |
| /* Get the buffer queue interface */ |
| result = (*stream->recordObj)->GetInterface( |
| stream->recordObj, W_SL_IID_BUFFERQUEUE, |
| &stream->recordBufQ); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot get recorder buffer queue iface")); |
| goto on_error; |
| } |
| |
| /* Register callback on the buffer queue */ |
| result = (*stream->recordBufQ)->RegisterCallback(stream->recordBufQ, |
| bqRecorderCallback, |
| (void *) stream); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot register recorder callback")); |
| goto on_error; |
| } |
| |
| stream->recordBufferSize = bufferSize; |
| for (i = 0; i < NUM_BUFFERS; i++) { |
| stream->recordBuffer[i] = (char *) |
| pj_pool_alloc(stream->pool, |
| stream->recordBufferSize); |
| } |
| |
| } |
| |
| if (param->flags & PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING) { |
| strm_set_cap(&stream->base, PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING, |
| ¶m->output_vol); |
| } |
| |
| /* Done */ |
| stream->base.op = &opensl_strm_op; |
| *p_aud_strm = &stream->base; |
| return PJ_SUCCESS; |
| |
| on_error: |
| strm_destroy(&stream->base); |
| return status; |
| } |
| |
| /* API: Get stream parameters */ |
| static pj_status_t strm_get_param(pjmedia_aud_stream *s, |
| pjmedia_aud_param *pi) |
| { |
| struct opensl_aud_stream *strm = (struct opensl_aud_stream*)s; |
| PJ_ASSERT_RETURN(strm && pi, PJ_EINVAL); |
| pj_memcpy(pi, &strm->param, sizeof(*pi)); |
| |
| if (strm_get_cap(s, PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING, |
| &pi->output_vol) == PJ_SUCCESS) |
| { |
| pi->flags |= PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING; |
| } |
| |
| return PJ_SUCCESS; |
| } |
| |
| /* API: get capability */ |
| static pj_status_t strm_get_cap(pjmedia_aud_stream *s, |
| pjmedia_aud_dev_cap cap, |
| void *pval) |
| { |
| struct opensl_aud_stream *strm = (struct opensl_aud_stream*)s; |
| pj_status_t status = PJMEDIA_EAUD_INVCAP; |
| |
| PJ_ASSERT_RETURN(s && pval, PJ_EINVAL); |
| |
| if (cap==PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING && |
| (strm->param.dir & PJMEDIA_DIR_PLAYBACK)) |
| { |
| if (strm->playerVol) { |
| SLresult res; |
| SLmillibel vol, mvol; |
| |
| res = (*strm->playerVol)->GetMaxVolumeLevel(strm->playerVol, |
| &mvol); |
| if (res == SL_RESULT_SUCCESS) { |
| res = (*strm->playerVol)->GetVolumeLevel(strm->playerVol, |
| &vol); |
| if (res == SL_RESULT_SUCCESS) { |
| *(int *)pval = ((int)vol - SL_MILLIBEL_MIN) * 100 / |
| ((int)mvol - SL_MILLIBEL_MIN); |
| return PJ_SUCCESS; |
| } |
| } |
| } |
| } |
| |
| return status; |
| } |
| |
| /* API: set capability */ |
| static pj_status_t strm_set_cap(pjmedia_aud_stream *s, |
| pjmedia_aud_dev_cap cap, |
| const void *value) |
| { |
| struct opensl_aud_stream *strm = (struct opensl_aud_stream*)s; |
| |
| PJ_ASSERT_RETURN(s && value, PJ_EINVAL); |
| |
| if (cap==PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING && |
| (strm->param.dir & PJMEDIA_DIR_PLAYBACK)) |
| { |
| if (strm->playerVol) { |
| SLresult res; |
| SLmillibel vol, mvol; |
| |
| res = (*strm->playerVol)->GetMaxVolumeLevel(strm->playerVol, |
| &mvol); |
| if (res == SL_RESULT_SUCCESS) { |
| vol = (SLmillibel)(*(int *)value * |
| ((int)mvol - SL_MILLIBEL_MIN) / 100 + SL_MILLIBEL_MIN); |
| res = (*strm->playerVol)->SetVolumeLevel(strm->playerVol, |
| vol); |
| if (res == SL_RESULT_SUCCESS) |
| return PJ_SUCCESS; |
| } |
| } |
| } |
| |
| return PJMEDIA_EAUD_INVCAP; |
| } |
| |
| /* API: start stream. */ |
| static pj_status_t strm_start(pjmedia_aud_stream *s) |
| { |
| struct opensl_aud_stream *stream = (struct opensl_aud_stream*)s; |
| int i; |
| SLresult result = SL_RESULT_SUCCESS; |
| |
| PJ_LOG(4, (THIS_FILE, "Starting %s stream..", stream->name.ptr)); |
| stream->quit_flag = 0; |
| |
| if (stream->recordBufQ && stream->recordRecord) { |
| /* Enqueue an empty buffer to be filled by the recorder |
| * (for streaming recording, we need to enqueue at least 2 empty |
| * buffers to start things off) |
| */ |
| for (i = 0; i < NUM_BUFFERS; i++) { |
| result = (*stream->recordBufQ)->Enqueue(stream->recordBufQ, |
| stream->recordBuffer[i], |
| stream->recordBufferSize); |
| /* The most likely other result is SL_RESULT_BUFFER_INSUFFICIENT, |
| * which for this code would indicate a programming error |
| */ |
| pj_assert(result == SL_RESULT_SUCCESS); |
| } |
| |
| result = (*stream->recordRecord)->SetRecordState( |
| stream->recordRecord, SL_RECORDSTATE_RECORDING); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot start recorder")); |
| goto on_error; |
| } |
| } |
| |
| if (stream->playerPlay && stream->playerBufQ) { |
| /* Set the player's state to playing */ |
| result = (*stream->playerPlay)->SetPlayState(stream->playerPlay, |
| SL_PLAYSTATE_PLAYING); |
| if (result != SL_RESULT_SUCCESS) { |
| PJ_LOG(3, (THIS_FILE, "Cannot start player")); |
| goto on_error; |
| } |
| |
| for (i = 0; i < NUM_BUFFERS; i++) { |
| pj_bzero(stream->playerBuffer[i], stream->playerBufferSize/100); |
| result = (*stream->playerBufQ)->Enqueue(stream->playerBufQ, |
| stream->playerBuffer[i], |
| stream->playerBufferSize/100); |
| pj_assert(result == SL_RESULT_SUCCESS); |
| } |
| } |
| |
| PJ_LOG(4, (THIS_FILE, "%s stream started", stream->name.ptr)); |
| return PJ_SUCCESS; |
| |
| on_error: |
| if (result != SL_RESULT_SUCCESS) |
| strm_stop(&stream->base); |
| return opensl_to_pj_error(result); |
| } |
| |
| /* API: stop stream. */ |
| static pj_status_t strm_stop(pjmedia_aud_stream *s) |
| { |
| struct opensl_aud_stream *stream = (struct opensl_aud_stream*)s; |
| |
| if (stream->quit_flag) |
| return PJ_SUCCESS; |
| |
| PJ_LOG(4, (THIS_FILE, "Stopping stream")); |
| |
| stream->quit_flag = 1; |
| |
| if (stream->recordBufQ && stream->recordRecord) { |
| /* Stop recording and clear buffer queue */ |
| (*stream->recordRecord)->SetRecordState(stream->recordRecord, |
| SL_RECORDSTATE_STOPPED); |
| (*stream->recordBufQ)->Clear(stream->recordBufQ); |
| } |
| |
| if (stream->playerBufQ && stream->playerPlay) { |
| /* Wait until the PCM data is done playing, the buffer queue callback |
| * will continue to queue buffers until the entire PCM data has been |
| * played. This is indicated by waiting for the count member of the |
| * SLBufferQueueState to go to zero. |
| */ |
| /* |
| SLresult result; |
| W_SLBufferQueueState state; |
| |
| result = (*stream->playerBufQ)->GetState(stream->playerBufQ, &state); |
| while (state.count) { |
| (*stream->playerBufQ)->GetState(stream->playerBufQ, &state); |
| } */ |
| /* Stop player */ |
| (*stream->playerPlay)->SetPlayState(stream->playerPlay, |
| SL_PLAYSTATE_STOPPED); |
| } |
| |
| PJ_LOG(4,(THIS_FILE, "OpenSL stream stopped")); |
| |
| return PJ_SUCCESS; |
| |
| } |
| |
| /* API: destroy stream. */ |
| static pj_status_t strm_destroy(pjmedia_aud_stream *s) |
| { |
| struct opensl_aud_stream *stream = (struct opensl_aud_stream*)s; |
| |
| /* Stop the stream */ |
| strm_stop(s); |
| |
| if (stream->playerObj) { |
| /* Destroy the player */ |
| (*stream->playerObj)->Destroy(stream->playerObj); |
| /* Invalidate all associated interfaces */ |
| stream->playerObj = NULL; |
| stream->playerPlay = NULL; |
| stream->playerBufQ = NULL; |
| stream->playerVol = NULL; |
| } |
| |
| if (stream->recordObj) { |
| /* Destroy the recorder */ |
| (*stream->recordObj)->Destroy(stream->recordObj); |
| /* Invalidate all associated interfaces */ |
| stream->recordObj = NULL; |
| stream->recordRecord = NULL; |
| stream->recordBufQ = NULL; |
| } |
| |
| pj_pool_release(stream->pool); |
| PJ_LOG(4, (THIS_FILE, "OpenSL stream destroyed")); |
| |
| return PJ_SUCCESS; |
| } |
| |
| #endif /* PJMEDIA_AUDIO_DEV_HAS_OPENSL */ |