doubango/plugins/audio_webrtc/audio_webrtc_producer.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_producer.h"
#include "audio_webrtc.h"

#include "tinydav/audio/tdav_producer_audio.h"

#include "tsk_string.h"
#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;
}
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;
}


/* ============ 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);
}
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;
}

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;
    }

    return audio_webrtc_instance_start_producer(self->audioInstHandle);
}

static int audio_producer_webrtc_pause(tmedia_producer_t* self)
{
    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;
    }

    return audio_webrtc_instance_stop_producer(self->audioInstHandle);
}


//
//	WebRTC audio producer object definition
//
/* 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;
}
/* 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;
}
/* 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,
};
/* 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
};
const tmedia_producer_plugin_def_t *audio_producer_webrtc_plugin_def_t = &audio_producer_webrtc_plugin_def_s;