diff options
Diffstat (limited to 'pjmedia/src/pjmedia-audiodev/opensl_dev.c')
-rw-r--r-- | pjmedia/src/pjmedia-audiodev/opensl_dev.c | 949 |
1 files changed, 949 insertions, 0 deletions
diff --git a/pjmedia/src/pjmedia-audiodev/opensl_dev.c b/pjmedia/src/pjmedia-audiodev/opensl_dev.c new file mode 100644 index 00000000..a77340b4 --- /dev/null +++ b/pjmedia/src/pjmedia-audiodev/opensl_dev.c @@ -0,0 +1,949 @@ +/* $Id$ */ +/* + * 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 */ |