diff options
Diffstat (limited to 'plugins/audio_webrtc/audio_webrtc_producer.cxx')
| -rwxr-xr-x | plugins/audio_webrtc/audio_webrtc_producer.cxx | 315 |
1 files changed, 156 insertions, 159 deletions
diff --git a/plugins/audio_webrtc/audio_webrtc_producer.cxx b/plugins/audio_webrtc/audio_webrtc_producer.cxx index 02c5aeb..3d95d06 100755 --- a/plugins/audio_webrtc/audio_webrtc_producer.cxx +++ b/plugins/audio_webrtc/audio_webrtc_producer.cxx @@ -1,17 +1,17 @@ /* Copyright (C) 2012 Doubango Telecom <http://www.doubango.org> -* +* * This file is part of Open Source Doubango Framework. * * DOUBANGO 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 3 of the License, or * (at your option) any later version. -* +* * DOUBANGO 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 DOUBANGO. */ @@ -24,147 +24,146 @@ #include "tsk_memory.h" #include "tsk_debug.h" -typedef struct audio_producer_webrtc_s -{ - TDAV_DECLARE_PRODUCER_AUDIO; - - bool isMuted; - audio_webrtc_instance_handle_t* audioInstHandle; - struct{ - void* ptr; - int size; - int index; - } buffer; +typedef struct audio_producer_webrtc_s { + TDAV_DECLARE_PRODUCER_AUDIO; + + bool isMuted; + audio_webrtc_instance_handle_t* audioInstHandle; + struct { + void* ptr; + int size; + int index; + } buffer; } audio_producer_webrtc_t; int audio_producer_webrtc_handle_data_10ms(const audio_producer_webrtc_t* _self, const void* audioSamples, int nSamples, int nBytesPerSample, int samplesPerSec, int nChannels) { - if(!_self || !audioSamples || !nSamples){ - DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter"); - return -1; - } - if((nSamples != (samplesPerSec / 100))){ - DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Not producing 10ms samples (nSamples=%d, samplesPerSec=%d)", nSamples, samplesPerSec); - return -2; - } - if((nBytesPerSample != (TMEDIA_PRODUCER(_self)->audio.bits_per_sample >> 3))){ - DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("%d not valid bytes/samples", nBytesPerSample); - return -3; - } - if((nChannels != TMEDIA_PRODUCER(_self)->audio.channels)){ - DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("%d not the expected number of channels", nChannels); - return -4; - } - - int nSamplesInBits = (nSamples * nBytesPerSample); - if(_self->buffer.index + nSamplesInBits > _self->buffer.size){ - DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Buffer overflow"); - return -5; - } - - audio_producer_webrtc_t* self = const_cast<audio_producer_webrtc_t*>(_self); - - memcpy((((uint8_t*)self->buffer.ptr) + self->buffer.index), audioSamples, nSamplesInBits); - self->buffer.index += nSamplesInBits; - - if(self->buffer.index == self->buffer.size){ - self->buffer.index = 0; - if(TMEDIA_PRODUCER(self)->enc_cb.callback){ - if(self->isMuted){ - memset(self->buffer.ptr, 0, self->buffer.size); - } - TMEDIA_PRODUCER(self)->enc_cb.callback(TMEDIA_PRODUCER(self)->enc_cb.callback_data, self->buffer.ptr, self->buffer.size); - } - } - - return 0; + if(!_self || !audioSamples || !nSamples) { + DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter"); + return -1; + } + if((nSamples != (samplesPerSec / 100))) { + DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Not producing 10ms samples (nSamples=%d, samplesPerSec=%d)", nSamples, samplesPerSec); + return -2; + } + if((nBytesPerSample != (TMEDIA_PRODUCER(_self)->audio.bits_per_sample >> 3))) { + DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("%d not valid bytes/samples", nBytesPerSample); + return -3; + } + if((nChannels != TMEDIA_PRODUCER(_self)->audio.channels)) { + DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("%d not the expected number of channels", nChannels); + return -4; + } + + int nSamplesInBits = (nSamples * nBytesPerSample); + if(_self->buffer.index + nSamplesInBits > _self->buffer.size) { + DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Buffer overflow"); + return -5; + } + + audio_producer_webrtc_t* self = const_cast<audio_producer_webrtc_t*>(_self); + + memcpy((((uint8_t*)self->buffer.ptr) + self->buffer.index), audioSamples, nSamplesInBits); + self->buffer.index += nSamplesInBits; + + if(self->buffer.index == self->buffer.size) { + self->buffer.index = 0; + if(TMEDIA_PRODUCER(self)->enc_cb.callback) { + if(self->isMuted) { + memset(self->buffer.ptr, 0, self->buffer.size); + } + TMEDIA_PRODUCER(self)->enc_cb.callback(TMEDIA_PRODUCER(self)->enc_cb.callback_data, self->buffer.ptr, self->buffer.size); + } + } + + return 0; } /* ============ Media Producer Interface ================= */ static int audio_producer_webrtc_set(tmedia_producer_t* _self, const tmedia_param_t* param) -{ - audio_producer_webrtc_t* self = (audio_producer_webrtc_t*)_self; - if(param->plugin_type == tmedia_ppt_producer){ - if(param->value_type == tmedia_pvt_int32){ - if(tsk_striequals(param->key, "mute")){ - self->isMuted = (TSK_TO_INT32((uint8_t*)param->value) != 0); - return 0; - } - } - } - return tdav_producer_audio_set(TDAV_PRODUCER_AUDIO(self), param); +{ + audio_producer_webrtc_t* self = (audio_producer_webrtc_t*)_self; + if(param->plugin_type == tmedia_ppt_producer) { + if(param->value_type == tmedia_pvt_int32) { + if(tsk_striequals(param->key, "mute")) { + self->isMuted = (TSK_TO_INT32((uint8_t*)param->value) != 0); + return 0; + } + } + } + return tdav_producer_audio_set(TDAV_PRODUCER_AUDIO(self), param); } static int audio_producer_webrtc_prepare(tmedia_producer_t* _self, const tmedia_codec_t* codec) { - audio_producer_webrtc_t* self = (audio_producer_webrtc_t*)_self; - if(!self || !codec){ - DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - // create audio instance - if(!(self->audioInstHandle = audio_webrtc_instance_create(TMEDIA_PRODUCER(self)->session_id))){ - DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Failed to create audio instance handle"); - return -2; - } - - // check that ptime is mutiple of 10 - if((codec->plugin->audio.ptime % 10)){ - DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("ptime=%d not multiple of 10", codec->plugin->audio.ptime); - return -3; - } - - // init input parameters from the codec - TMEDIA_PRODUCER(self)->audio.channels = codec->plugin->audio.channels; - TMEDIA_PRODUCER(self)->audio.rate = codec->plugin->rate; - TMEDIA_PRODUCER(self)->audio.ptime = codec->plugin->audio.ptime; - - // prepare playout device and update output parameters - int ret; - ret = audio_webrtc_instance_prepare_producer(self->audioInstHandle, &_self); - - // now that the producer is prepared we can initialize internal buffer using device caps - if(ret == 0){ - // allocate buffer - int xsize = ((TMEDIA_PRODUCER(self)->audio.ptime * TMEDIA_PRODUCER(self)->audio.rate) / 1000) * (TMEDIA_PRODUCER(self)->audio.bits_per_sample >> 3); - if(!(self->buffer.ptr = tsk_realloc(self->buffer.ptr, xsize))){ - DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Failed to allocate buffer with size = %d", xsize); - self->buffer.size = 0; - return -1; - } - self->buffer.size = xsize; - self->buffer.index = 0; - } - return ret; + audio_producer_webrtc_t* self = (audio_producer_webrtc_t*)_self; + if(!self || !codec) { + DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + // create audio instance + if(!(self->audioInstHandle = audio_webrtc_instance_create(TMEDIA_PRODUCER(self)->session_id))) { + DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Failed to create audio instance handle"); + return -2; + } + + // check that ptime is mutiple of 10 + if((codec->plugin->audio.ptime % 10)) { + DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("ptime=%d not multiple of 10", codec->plugin->audio.ptime); + return -3; + } + + // init input parameters from the codec + TMEDIA_PRODUCER(self)->audio.channels = codec->plugin->audio.channels; + TMEDIA_PRODUCER(self)->audio.rate = codec->plugin->rate; + TMEDIA_PRODUCER(self)->audio.ptime = codec->plugin->audio.ptime; + + // prepare playout device and update output parameters + int ret; + ret = audio_webrtc_instance_prepare_producer(self->audioInstHandle, &_self); + + // now that the producer is prepared we can initialize internal buffer using device caps + if(ret == 0) { + // allocate buffer + int xsize = ((TMEDIA_PRODUCER(self)->audio.ptime * TMEDIA_PRODUCER(self)->audio.rate) / 1000) * (TMEDIA_PRODUCER(self)->audio.bits_per_sample >> 3); + if(!(self->buffer.ptr = tsk_realloc(self->buffer.ptr, xsize))) { + DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Failed to allocate buffer with size = %d", xsize); + self->buffer.size = 0; + return -1; + } + self->buffer.size = xsize; + self->buffer.index = 0; + } + return ret; } static int audio_producer_webrtc_start(tmedia_producer_t* _self) { - audio_producer_webrtc_t* self = (audio_producer_webrtc_t*)_self; - if(!self){ - DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter"); - return -1; - } + audio_producer_webrtc_t* self = (audio_producer_webrtc_t*)_self; + if(!self) { + DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter"); + return -1; + } - return audio_webrtc_instance_start_producer(self->audioInstHandle); + return audio_webrtc_instance_start_producer(self->audioInstHandle); } static int audio_producer_webrtc_pause(tmedia_producer_t* self) { - return 0; + return 0; } static int audio_producer_webrtc_stop(tmedia_producer_t* _self) { - audio_producer_webrtc_t* self = (audio_producer_webrtc_t*)_self; - if(!self){ - DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter"); - return -1; - } + audio_producer_webrtc_t* self = (audio_producer_webrtc_t*)_self; + if(!self) { + DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter"); + return -1; + } - return audio_webrtc_instance_stop_producer(self->audioInstHandle); + return audio_webrtc_instance_stop_producer(self->audioInstHandle); } @@ -174,54 +173,52 @@ static int audio_producer_webrtc_stop(tmedia_producer_t* _self) /* constructor */ static tsk_object_t* audio_producer_webrtc_ctor(tsk_object_t *_self, va_list * app) { - audio_producer_webrtc_t *self = (audio_producer_webrtc_t *)_self; - if(self){ - /* init base */ - tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(self)); - /* init self */ - - } - return self; + audio_producer_webrtc_t *self = (audio_producer_webrtc_t *)_self; + if(self) { + /* init base */ + tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(self)); + /* init self */ + + } + return self; } /* destructor */ static tsk_object_t* audio_producer_webrtc_dtor(tsk_object_t *_self) -{ - audio_producer_webrtc_t *self = (audio_producer_webrtc_t *)_self; - if(self){ - /* stop */ - audio_producer_webrtc_stop(TMEDIA_PRODUCER(self)); - /* deinit self */ - if(self->audioInstHandle){ - audio_webrtc_instance_destroy(&self->audioInstHandle); - } - TSK_FREE(self->buffer.ptr); - - /* deinit base */ - tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(self)); - } - - return self; +{ + audio_producer_webrtc_t *self = (audio_producer_webrtc_t *)_self; + if(self) { + /* stop */ + audio_producer_webrtc_stop(TMEDIA_PRODUCER(self)); + /* deinit self */ + if(self->audioInstHandle) { + audio_webrtc_instance_destroy(&self->audioInstHandle); + } + TSK_FREE(self->buffer.ptr); + + /* deinit base */ + tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(self)); + } + + return self; } /* object definition */ -static const tsk_object_def_t audio_producer_webrtc_def_s = -{ - sizeof(audio_producer_webrtc_t), - audio_producer_webrtc_ctor, - audio_producer_webrtc_dtor, - tdav_producer_audio_cmp, +static const tsk_object_def_t audio_producer_webrtc_def_s = { + sizeof(audio_producer_webrtc_t), + audio_producer_webrtc_ctor, + audio_producer_webrtc_dtor, + tdav_producer_audio_cmp, }; /* plugin definition*/ -static const tmedia_producer_plugin_def_t audio_producer_webrtc_plugin_def_s = -{ - &audio_producer_webrtc_def_s, - - tmedia_audio, - "WebRTC audio producer", - - audio_producer_webrtc_set, - audio_producer_webrtc_prepare, - audio_producer_webrtc_start, - audio_producer_webrtc_pause, - audio_producer_webrtc_stop +static const tmedia_producer_plugin_def_t audio_producer_webrtc_plugin_def_s = { + &audio_producer_webrtc_def_s, + + tmedia_audio, + "WebRTC audio producer", + + audio_producer_webrtc_set, + audio_producer_webrtc_prepare, + audio_producer_webrtc_start, + audio_producer_webrtc_pause, + audio_producer_webrtc_stop }; const tmedia_producer_plugin_def_t *audio_producer_webrtc_plugin_def_t = &audio_producer_webrtc_plugin_def_s;
\ No newline at end of file |
