231 lines
7.6 KiB
C++
Executable File
231 lines
7.6 KiB
C++
Executable File
/* Copyright (C) 2012 Doubango Telecom <http://www.doubango.org>
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*
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* This file is part of Open Source Doubango Framework.
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*
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* DOUBANGO is free software: you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation, either version 3 of the License, or
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* (at your option) any later version.
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*
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* DOUBANGO is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with DOUBANGO.
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*/
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#include "audio_webrtc_consumer.h"
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#include "audio_webrtc.h"
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#include "tinydav/audio/tdav_consumer_audio.h"
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#include "tsk_string.h"
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#include "tsk_memory.h"
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#include "tsk_debug.h"
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typedef struct audio_consumer_webrtc_s {
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TDAV_DECLARE_CONSUMER_AUDIO;
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audio_webrtc_instance_handle_t* audioInstHandle;
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struct {
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void* ptr;
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bool isFull;
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int size;
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int index;
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} buffer;
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}
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audio_consumer_webrtc_t;
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int audio_consumer_webrtc_get_data_10ms(const audio_consumer_webrtc_t* _self, void* audioSamples, int nSamples, int nBytesPerSample, int nChannels, int samplesPerSec, uint32_t &nSamplesOut)
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{
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nSamplesOut = 0;
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if(!_self || !audioSamples || !nSamples) {
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DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
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return -1;
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}
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if((nSamples != (samplesPerSec / 100))) {
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DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Not producing 10ms samples (nSamples=%d, samplesPerSec=%d)", nSamples, samplesPerSec);
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return -2;
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}
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if((nBytesPerSample != (TMEDIA_CONSUMER(_self)->audio.bits_per_sample >> 3))) {
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DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("%d not valid bytes/samples", nBytesPerSample);
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return -3;
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}
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if((nChannels != TMEDIA_CONSUMER(_self)->audio.out.channels)) {
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DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("%d not the expected number of channels", nChannels);
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return -4;
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}
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audio_consumer_webrtc_t* self = const_cast<audio_consumer_webrtc_t*>(_self);
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if(self->buffer.index == self->buffer.size) {
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tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(self));
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self->buffer.index = 0;
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if((tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(self), self->buffer.ptr, self->buffer.size)) != self->buffer.size) {
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nSamplesOut = 0;
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return 0;
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}
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}
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int nSamplesInBits = (nSamples * nBytesPerSample);
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if(_self->buffer.index + nSamplesInBits <= _self->buffer.size) {
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memcpy(audioSamples, (((uint8_t*)self->buffer.ptr) + self->buffer.index), nSamplesInBits);
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}
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self->buffer.index += nSamplesInBits;
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TSK_CLAMP(0, self->buffer.index, self->buffer.size);
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nSamplesOut = nSamples;
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return 0;
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}
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/* ============ Media Consumer Interface ================= */
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static int audio_consumer_webrtc_set(tmedia_consumer_t* self, const tmedia_param_t* param)
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{
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audio_consumer_webrtc_t* webrtc = (audio_consumer_webrtc_t*)self;
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int ret = tdav_consumer_audio_set(TDAV_CONSUMER_AUDIO(self), param);
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if(ret == 0) {
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if(tsk_striequals(param->key, "volume")) {
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}
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}
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return ret;
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}
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static int audio_consumer_webrtc_prepare(tmedia_consumer_t* _self, const tmedia_codec_t* codec)
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{
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audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
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if(!self) {
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DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
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return -1;
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}
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// create audio instance
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if(!(self->audioInstHandle = audio_webrtc_instance_create(TMEDIA_CONSUMER(self)->session_id))) {
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DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Failed to create audio instance handle");
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return -1;
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}
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// initialize input parameters from the codec information
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TMEDIA_CONSUMER(self)->audio.ptime = codec->plugin->audio.ptime;
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TMEDIA_CONSUMER(self)->audio.in.channels = codec->plugin->audio.channels;
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TMEDIA_CONSUMER(self)->audio.in.rate = codec->plugin->rate;
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// prepare playout device and update output parameters
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int ret = audio_webrtc_instance_prepare_consumer(self->audioInstHandle, &_self);
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// now that the producer is prepared we can initialize internal buffer using device caps
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if(ret == 0) {
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// allocate buffer
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int xsize = ((TMEDIA_CONSUMER(self)->audio.ptime * TMEDIA_CONSUMER(self)->audio.out.rate) / 1000) * (TMEDIA_CONSUMER(self)->audio.bits_per_sample >> 3);
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if(!(self->buffer.ptr = tsk_realloc(self->buffer.ptr, xsize))) {
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DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Failed to allocate buffer with size = %d", xsize);
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self->buffer.size = 0;
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return -1;
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}
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memset(self->buffer.ptr, 0, xsize);
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self->buffer.size = xsize;
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self->buffer.index = 0;
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self->buffer.isFull = false;
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}
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return ret;
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}
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static int audio_consumer_webrtc_start(tmedia_consumer_t* _self)
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{
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audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
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if(!self) {
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DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
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return -1;
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}
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return audio_webrtc_instance_start_consumer(self->audioInstHandle);
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}
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static int audio_consumer_webrtc_consume(tmedia_consumer_t* _self, const void* buffer, tsk_size_t size, const tsk_object_t* proto_hdr)
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{
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audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
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if(!self || !buffer || !size) {
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DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("1Invalid parameter");
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return -1;
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}
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/* buffer is already decoded */
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return tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(self), buffer, size, proto_hdr);
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}
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static int audio_consumer_webrtc_pause(tmedia_consumer_t* self)
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{
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return 0;
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}
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static int audio_consumer_webrtc_stop(tmedia_consumer_t* _self)
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{
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audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
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if(!self) {
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DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
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return -1;
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}
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return audio_webrtc_instance_stop_consumer(self->audioInstHandle);
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}
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//
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// WebRTC audio consumer object definition
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//
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/* constructor */
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static tsk_object_t* audio_consumer_webrtc_ctor(tsk_object_t *_self, va_list * app)
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{
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audio_consumer_webrtc_t *self = (audio_consumer_webrtc_t *)_self;
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if(self) {
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/* init base */
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tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(self));
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/* init self */
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}
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return self;
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}
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/* destructor */
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static tsk_object_t* audio_consumer_webrtc_dtor(tsk_object_t *_self)
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{
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audio_consumer_webrtc_t *self = (audio_consumer_webrtc_t *)_self;
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if(self) {
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/* stop */
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audio_consumer_webrtc_stop(TMEDIA_CONSUMER(self));
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/* deinit self */
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if(self->audioInstHandle) {
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audio_webrtc_instance_destroy(&self->audioInstHandle);
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}
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TSK_FREE(self->buffer.ptr);
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/* deinit base */
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tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(self));
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}
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return self;
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}
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/* object definition */
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static const tsk_object_def_t audio_consumer_webrtc_def_s = {
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sizeof(audio_consumer_webrtc_t),
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audio_consumer_webrtc_ctor,
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audio_consumer_webrtc_dtor,
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tdav_consumer_audio_cmp,
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};
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/* plugin definition*/
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static const tmedia_consumer_plugin_def_t audio_consumer_webrtc_plugin_def_s = {
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&audio_consumer_webrtc_def_s,
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tmedia_audio,
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"WebRTC audio consumer",
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audio_consumer_webrtc_set,
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audio_consumer_webrtc_prepare,
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audio_consumer_webrtc_start,
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audio_consumer_webrtc_consume,
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audio_consumer_webrtc_pause,
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audio_consumer_webrtc_stop
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};
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const tmedia_consumer_plugin_def_t *audio_consumer_webrtc_plugin_def_t = &audio_consumer_webrtc_plugin_def_s;
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