doubango-android/plugins/audio_webrtc/audio_webrtc_consumer.cxx

/* 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;