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/* 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.
*/
#include "audio_webrtc_consumer.h"
#include "audio_webrtc.h"
#include "tinydav/audio/tdav_consumer_audio.h"
#include "tsk_string.h"
#include "tsk_memory.h"
#include "tsk_debug.h"
typedef struct audio_consumer_webrtc_s
{
TDAV_DECLARE_CONSUMER_AUDIO;
audio_webrtc_instance_handle_t* audioInstHandle;
struct{
void* ptr;
bool isFull;
int size;
int index;
} buffer;
}
audio_consumer_webrtc_t;
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)
{
nSamplesOut = 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_CONSUMER(_self)->audio.bits_per_sample >> 3))){
DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("%d not valid bytes/samples", nBytesPerSample);
return -3;
}
if((nChannels != TMEDIA_CONSUMER(_self)->audio.out.channels)){
DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("%d not the expected number of channels", nChannels);
return -4;
}
audio_consumer_webrtc_t* self = const_cast<audio_consumer_webrtc_t*>(_self);
if(self->buffer.index == self->buffer.size){
tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(self));
self->buffer.index = 0;
if((tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(self), self->buffer.ptr, self->buffer.size)) != self->buffer.size){
nSamplesOut = 0;
return 0;
}
}
int nSamplesInBits = (nSamples * nBytesPerSample);
if(_self->buffer.index + nSamplesInBits <= _self->buffer.size){
memcpy(audioSamples, (((uint8_t*)self->buffer.ptr) + self->buffer.index), nSamplesInBits);
}
self->buffer.index += nSamplesInBits;
TSK_CLAMP(0, self->buffer.index, self->buffer.size);
nSamplesOut = nSamples;
return 0;
}
/* ============ Media Consumer Interface ================= */
static int audio_consumer_webrtc_set(tmedia_consumer_t* self, const tmedia_param_t* param)
{
audio_consumer_webrtc_t* webrtc = (audio_consumer_webrtc_t*)self;
int ret = tdav_consumer_audio_set(TDAV_CONSUMER_AUDIO(self), param);
if(ret == 0){
if(tsk_striequals(param->key, "volume")){
}
}
return ret;
}
static int audio_consumer_webrtc_prepare(tmedia_consumer_t* _self, const tmedia_codec_t* codec)
{
audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
if(!self){
DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
return -1;
}
// create audio instance
if(!(self->audioInstHandle = audio_webrtc_instance_create(TMEDIA_CONSUMER(self)->session_id))){
DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Failed to create audio instance handle");
return -1;
}
// initialize input parameters from the codec information
TMEDIA_CONSUMER(self)->audio.ptime = codec->plugin->audio.ptime;
TMEDIA_CONSUMER(self)->audio.in.channels = codec->plugin->audio.channels;
TMEDIA_CONSUMER(self)->audio.in.rate = codec->plugin->rate;
// prepare playout device and update output parameters
int ret = audio_webrtc_instance_prepare_consumer(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_CONSUMER(self)->audio.ptime * TMEDIA_CONSUMER(self)->audio.out.rate) / 1000) * (TMEDIA_CONSUMER(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;
}
memset(self->buffer.ptr, 0, xsize);
self->buffer.size = xsize;
self->buffer.index = 0;
self->buffer.isFull = false;
}
return ret;
}
static int audio_consumer_webrtc_start(tmedia_consumer_t* _self)
{
audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
if(!self){
DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
return -1;
}
return audio_webrtc_instance_start_consumer(self->audioInstHandle);
}
static int audio_consumer_webrtc_consume(tmedia_consumer_t* _self, const void* buffer, tsk_size_t size, const tsk_object_t* proto_hdr)
{
audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
if(!self || !buffer || !size){
DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("1Invalid parameter");
return -1;
}
/* buffer is already decoded */
return tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(self), buffer, size, proto_hdr);
}
static int audio_consumer_webrtc_pause(tmedia_consumer_t* self)
{
return 0;
}
static int audio_consumer_webrtc_stop(tmedia_consumer_t* _self)
{
audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
if(!self){
DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
return -1;
}
return audio_webrtc_instance_stop_consumer(self->audioInstHandle);
}
//
// WebRTC audio consumer object definition
//
/* constructor */
static tsk_object_t* audio_consumer_webrtc_ctor(tsk_object_t *_self, va_list * app)
{
audio_consumer_webrtc_t *self = (audio_consumer_webrtc_t *)_self;
if(self){
/* init base */
tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(self));
/* init self */
}
return self;
}
/* destructor */
static tsk_object_t* audio_consumer_webrtc_dtor(tsk_object_t *_self)
{
audio_consumer_webrtc_t *self = (audio_consumer_webrtc_t *)_self;
if(self){
/* stop */
audio_consumer_webrtc_stop(TMEDIA_CONSUMER(self));
/* deinit self */
if(self->audioInstHandle){
audio_webrtc_instance_destroy(&self->audioInstHandle);
}
TSK_FREE(self->buffer.ptr);
/* deinit base */
tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(self));
}
return self;
}
/* object definition */
static const tsk_object_def_t audio_consumer_webrtc_def_s =
{
sizeof(audio_consumer_webrtc_t),
audio_consumer_webrtc_ctor,
audio_consumer_webrtc_dtor,
tdav_consumer_audio_cmp,
};
/* plugin definition*/
static const tmedia_consumer_plugin_def_t audio_consumer_webrtc_plugin_def_s =
{
&audio_consumer_webrtc_def_s,
tmedia_audio,
"WebRTC audio consumer",
audio_consumer_webrtc_set,
audio_consumer_webrtc_prepare,
audio_consumer_webrtc_start,
audio_consumer_webrtc_consume,
audio_consumer_webrtc_pause,
audio_consumer_webrtc_stop
};
const tmedia_consumer_plugin_def_t *audio_consumer_webrtc_plugin_def_t = &audio_consumer_webrtc_plugin_def_s;
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