doubango-android/plugins/audio_webrtc/audio_webrtc.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.h"

#include "audio_webrtc_consumer.h"
#include "audio_webrtc_producer.h"
#include "audio_webrtc_transport.h"

#include <webrtc/audio_device_config.h>
#include <webrtc/audio_device_impl.h>

#include "tinymedia/tmedia_consumer.h"
#include "tinymedia/tmedia_producer.h"

#include "tsk_list.h"
#include "tsk_safeobj.h"
#include "tsk_debug.h"

using namespace webrtc;

#define kAudioDeviceModuleId 444

#if DOUBANGO_AUDIO_WEBRTC_UNDER_ANDROID
// https://groups.google.com/group/android-ndk/browse_thread/thread/a1667f28162cf69b/8ef3a171df7f8dfe
extern "C"
{
    void *__dso_handle = NULL;
}
#endif

typedef enum PLUGIN_INDEX_E {
    PLUGIN_INDEX_AUDIO_CONSUMER,
    PLUGIN_INDEX_AUDIO_PRODUCER,
    PLUGIN_INDEX_COUNT
}
PLUGIN_INDEX_T;


int __plugin_get_def_count()
{
    return PLUGIN_INDEX_COUNT;
}

tsk_plugin_def_type_t __plugin_get_def_type_at(int index)
{
    switch(index) {
    case PLUGIN_INDEX_AUDIO_CONSUMER:
        return tsk_plugin_def_type_consumer;
    case PLUGIN_INDEX_AUDIO_PRODUCER:
        return tsk_plugin_def_type_producer;
    default: {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("No plugin at index %d", index);
        return tsk_plugin_def_type_none;
    }
    }
}

tsk_plugin_def_media_type_t	__plugin_get_def_media_type_at(int index)
{
    switch(index) {
    case PLUGIN_INDEX_AUDIO_CONSUMER:
    case PLUGIN_INDEX_AUDIO_PRODUCER: {
        return tsk_plugin_def_media_type_audio;
    }
    default: {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("No plugin at index %d", index);
        return tsk_plugin_def_media_type_none;
    }
    }
}

tsk_plugin_def_ptr_const_t __plugin_get_def_at(int index)
{
    switch(index) {
    case PLUGIN_INDEX_AUDIO_CONSUMER: {
        return audio_consumer_webrtc_plugin_def_t;
    }
    case PLUGIN_INDEX_AUDIO_PRODUCER: {
        return audio_producer_webrtc_plugin_def_t;
    }
    default: {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("No plugin at index %d", index);
        return tsk_null;
    }
    }
}

//
// WebRTC AudioInstance
//

typedef struct audio_webrtc_instance_s {
    TSK_DECLARE_OBJECT;

    uint64_t sessionId;

    bool isStarted;

    bool isConsumerPrepared;
    bool isConsumerStarted;
    bool isProducerPrepared;
    bool isProducerStarted;

    bool isSpeakerAvailable;
    bool isPlayoutAvailable;
    bool isRecordingAvailable;

    AudioDeviceModule* device;
    AudioTransportImpl* transport;

    TSK_DECLARE_SAFEOBJ;
}
audio_webrtc_instance_t;
typedef tsk_list_t audio_webrtc_instances_L_t;

static audio_webrtc_instances_L_t* __audioInstances = tsk_null;

static tsk_object_t* audio_webrtc_instance_ctor(tsk_object_t * self, va_list * app)
{
    audio_webrtc_instance_t* audioInstance = (audio_webrtc_instance_t*)self;
    if(audioInstance) {
        tsk_safeobj_init(audioInstance);
    }
    return self;
}
static tsk_object_t* audio_webrtc_instance_dtor(tsk_object_t * self)
{
    DOUBANGO_AUDIO_WEBRTC_DEBUG_INFO("Audio Instance destroyed");
    audio_webrtc_instance_t* audioInstance = (audio_webrtc_instance_t*)self;
    if(audioInstance) {
        tsk_safeobj_lock(audioInstance);
        if(audioInstance->device) {
            audioInstance->device->RegisterAudioCallback(tsk_null);
            audioInstance->device->Terminate();
            audioInstance->device->Release();//FIXME: must be deleted?
            audioInstance->device = tsk_null;
        }
        if(audioInstance->transport) {
            delete audioInstance->transport;
            audioInstance->transport = tsk_null;
        }
        tsk_safeobj_unlock(audioInstance);

        tsk_safeobj_deinit(audioInstance);
    }
    return self;
}
static int audio_webrtc_instance_cmp(const tsk_object_t *_ai1, const tsk_object_t *_ai2)
{
    return ((int)_ai1 - (int)_ai2);
}
static const tsk_object_def_t audio_webrtc_instance_def_s = {
    sizeof(audio_webrtc_instance_t),
    audio_webrtc_instance_ctor,
    audio_webrtc_instance_dtor,
    audio_webrtc_instance_cmp,
};
const tsk_object_def_t *audio_webrtc_instance_def_t = &audio_webrtc_instance_def_s;


audio_webrtc_instance_handle_t* audio_webrtc_instance_create(uint64_t sessionId)
{
    audio_webrtc_instance_t* audioInstance = tsk_null;

    // create list used to hold instances
    if(!__audioInstances && !(__audioInstances = tsk_list_create())) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Failed to create new list");
        return tsk_null;
    }

    //= lock the list
    tsk_list_lock(__audioInstances);

    // find the instance from the list
    const tsk_list_item_t* item;
    tsk_list_foreach(item, __audioInstances) {
        if(((audio_webrtc_instance_t*)item->data)->sessionId == sessionId) {
            audioInstance = (audio_webrtc_instance_t*)tsk_object_ref(item->data);
            break;
        }
    }

    if(!audioInstance) {
        audio_webrtc_instance_t* _audioInstance;
        if(!(_audioInstance = (audio_webrtc_instance_t*)tsk_object_new(&audio_webrtc_instance_def_s))) {
            DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Failed to create new audio instance");
            goto done;
        }

        if(!(_audioInstance->device = AudioDeviceModuleImpl::Create(kAudioDeviceModuleId))) {
            DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Failed to create audio device");
            TSK_OBJECT_SAFE_FREE(_audioInstance);
            goto done;
        }
        _audioInstance->device->AddRef();

        if(!(_audioInstance->transport = new AudioTransportImpl(_audioInstance->device))) {
            DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Failed to create audio transport");
            TSK_OBJECT_SAFE_FREE(_audioInstance);
            goto done;
        }
        if((_audioInstance->device->RegisterAudioCallback(_audioInstance->transport))) {
            DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("AudioDeviceModule::RegisterAudioCallback() failed");
            TSK_OBJECT_SAFE_FREE(_audioInstance);
            goto done;
        }

        if((_audioInstance->device->Init())) {
            DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("AudioDeviceModule::Init() failed");
            TSK_OBJECT_SAFE_FREE(_audioInstance);
            goto done;
        }

        _audioInstance->sessionId = sessionId;
        audioInstance = _audioInstance;
        tsk_list_push_back_data(__audioInstances, (void**)&_audioInstance);
    }

done:
    //= unlock the list
    tsk_list_unlock(__audioInstances);

    return audioInstance;
}

int audio_webrtc_instance_prepare_consumer(audio_webrtc_instance_handle_t* _self, tmedia_consumer_t** _consumer)
{
    audio_webrtc_instance_t* self = (audio_webrtc_instance_t*)_self;
    if(!self || !self->device || !self->transport || !_consumer || !*_consumer) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("invalid parameter");
        return -1;
    }

    if(self->isConsumerPrepared) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_WARN("Consumer already prepared");
        return 0;
    }

    int ret;
    bool _bool;

    tsk_safeobj_lock(self);

    self->transport->SetConsumer((const struct audio_consumer_webrtc_s*)*_consumer);

    if((ret = self->device->SetPlayoutDevice(DOUBANGO_AUDIO_WEBRTC_DEVICE_DEFAULT))) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("AudioDeviceModule->SetPlayoutDevice(%d) failed", DOUBANGO_AUDIO_WEBRTC_DEVICE_DEFAULT);
    }

    if((ret = self->device->SpeakerIsAvailable(&_bool))) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("SpeakerIsAvailable() failed with error code=%d", ret);
    }
    else {
        if(!_bool) {
            DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("SpeakerIsAvailable() returned false");
        }
        self->isSpeakerAvailable = _bool;
    }

    if((ret = self->device->InitSpeaker())) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("InitSpeaker() failed with error code=%d", ret);
    }

    if((ret = self->device->PlayoutIsAvailable(&_bool))) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("PlayoutIsAvailable() failed with error code =%d", ret);
    }
    else {
        if(!_bool) {
            DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("PlayoutIsAvailable() returned false");
        }
        self->isPlayoutAvailable = _bool;
    }

    if((ret = self->device->SetStereoPlayout(((*_consumer)->audio.in.channels == 2)))) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("SetStereoPlayout(%d==2) failed with error code=%d", (*_consumer)->audio.in.channels, ret);
    }

    //if((ret = self->device->SetPlayoutBuffer(AudioDeviceModule::kFixedBufferSize, (*_consumer)->audio.ptime))){
    //	DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("SetPlayoutBuffer(%d ms) failed with error code=%d", (*_consumer)->audio.ptime, ret);
    //}
    // always request 10ms buffers. In all cases WebRTC don't support anything else
    if((ret = self->device->SetPlayoutBuffer(AudioDeviceModule::kFixedBufferSize, 10))) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("SetPlayoutBuffer(%d ms) failed with error code=%d", 10, ret);
    }

    uint32_t playoutSampleRate = (*_consumer)->audio.out.rate ? (*_consumer)->audio.out.rate : (*_consumer)->audio.in.rate;
    if((ret = self->device->SetPlayoutSampleRate(playoutSampleRate))) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("SetPlayoutSampleRate(%d) failed with error code=%d", playoutSampleRate, ret);
    }

    if((ret = self->device->InitPlayout())) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("AudioDeviceModule::InitPlayout() failed with error code = %d", ret);
        goto done;
    }

    // init output parameters
    if((ret = self->device->StereoPlayout(&_bool))) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("StereoPlayout() failed with error code=%d", ret);
    }
    else {
        (*_consumer)->audio.out.channels = (_bool ? 2 : 1);
    }
    if((ret = self->device->PlayoutSampleRate(&playoutSampleRate))) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("PlayoutSampleRate() failed with error code=%d", ret);
    }
    else {
        (*_consumer)->audio.out.rate = playoutSampleRate;
    }

done:
    tsk_safeobj_unlock(self);

    self->isConsumerPrepared = (ret == 0);

    return ret;
}

int audio_webrtc_instance_prepare_producer(audio_webrtc_instance_handle_t* _self, tmedia_producer_t** _producer)
{
    audio_webrtc_instance_t* self = (audio_webrtc_instance_t*)_self;
    if(!self || !self->device || !self->transport || !_producer || !*_producer) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("invalid parameter");
        return -1;
    }

    if(self->isProducerPrepared) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_WARN("Producer already prepared");
        return 0;
    }

    int ret;
    bool _bool;

    tsk_safeobj_lock(self);

    self->transport->SetProducer((const struct audio_producer_webrtc_s*)*_producer);

    if((ret = self->device->SetRecordingDevice(DOUBANGO_AUDIO_WEBRTC_DEVICE_DEFAULT))) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("AudioDeviceModule->SetRecordingDevice(%d) failed", DOUBANGO_AUDIO_WEBRTC_DEVICE_DEFAULT);
    }

    if((ret = self->device->RecordingIsAvailable(&_bool))) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("RecordingIsAvailable() failed with error code =%d", ret);
    }
    else {
        if(!_bool) {
            DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("RecordingIsAvailable() returned false");
        }
        self->isRecordingAvailable = _bool;
    }

    if((ret = self->device->MicrophoneIsAvailable(&_bool))) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("MicrophoneIsAvailable() failed with error code =%d", ret);
    }
    else {
        if(!_bool) {
            DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("MicrophoneIsAvailable() returned false");
        }
        else {
            if((ret = self->device->InitMicrophone())) {
                DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("InitMicrophone() failed with error code =%d", ret);
            }
        }
    }

    if((ret = self->device->SetStereoRecording(((*_producer)->audio.channels == 2)))) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("SetStereoRecording(%d==2) failed with error code=%d", (*_producer)->audio.channels, ret);
    }

    uint32_t recordingSampleRate = (*_producer)->audio.rate;
    if((ret = self->device->SetRecordingSampleRate(recordingSampleRate))) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("SetRecordingSampleRate(%d) failed with error code=%d", recordingSampleRate, ret);
    }

    if((ret = self->device->InitRecording())) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("AudioDeviceModule::InitRecording() failed with error code = %d", ret);
        goto done;
    }

    // init output parameters
    if((ret = self->device->StereoRecording(&_bool))) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("StereoRecording() failed with error code=%d", ret);
    }
    else {
        (*_producer)->audio.channels = (_bool ? 2 : 1);
    }
    if((ret = self->device->RecordingSampleRate(&recordingSampleRate))) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("RecordingSampleRate() failed with error code=%d", ret);
    }
    else {
        (*_producer)->audio.rate = recordingSampleRate;
    }

done:
    tsk_safeobj_unlock(self);

    self->isProducerPrepared = (ret == 0);

    return ret;
}

int audio_webrtc_instance_start_consumer(audio_webrtc_instance_handle_t* _self)
{
    audio_webrtc_instance_t* self = (audio_webrtc_instance_t*)_self;
    if(!self || !self->device || !self->transport) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("invalid parameter");
        return -1;
    }

    tsk_safeobj_lock(self);
    if(!self->isConsumerPrepared) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Consumer not prepared");
        goto done;
    }

    if(self->isConsumerStarted) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_WARN("Consumer already started");
        goto done;
    }

    if(self->isPlayoutAvailable) {
        int ret;
        if((ret = self->device->StartPlayout())) {
            DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("StartPlayout() failed with error code = %d", ret);
        }

        self->isConsumerStarted = self->device->Playing();
        DOUBANGO_AUDIO_WEBRTC_DEBUG_INFO("isPlaying=%s", (self->isConsumerPrepared ? "true" : "false"));
    }

done:
    tsk_safeobj_unlock(self);
    return (self->isConsumerStarted ? 0 : -1);
}

int audio_webrtc_instance_start_producer(audio_webrtc_instance_handle_t* _self)
{
    audio_webrtc_instance_t* self = (audio_webrtc_instance_t*)_self;
    if(!self || !self->device || !self->transport) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("invalid parameter");
        return -1;
    }

    tsk_safeobj_lock(self);
    if(!self->isProducerPrepared) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Producer not prepared");
        goto done;
    }

    if(self->isProducerStarted) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_WARN("Consumer already started");
        goto done;
    }

    if(self->isRecordingAvailable) {
        int ret;
        if((ret = self->device->StartRecording())) {
            DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("StartRecording() failed with error code = %d", ret);
        }

        self->isProducerStarted = self->device->Recording();
        DOUBANGO_AUDIO_WEBRTC_DEBUG_INFO("isRecording=%s", (self->isProducerStarted ? "true" : "false"));
    }

done:
    tsk_safeobj_unlock(self);
    return (self->isProducerStarted ? 0 : -1);
    return 0;
}

int audio_webrtc_instance_stop_consumer(audio_webrtc_instance_handle_t* _self)
{
    audio_webrtc_instance_t* self = (audio_webrtc_instance_t*)_self;
    if(!self || !self->device || !self->transport) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("invalid parameter");
        return -1;
    }

    tsk_safeobj_lock(self);

    if(!self->isConsumerStarted) {
        goto done;
    }

    int ret;
    if((ret = self->device->StopPlayout())) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("StopPlayout() failed with error code = %d", ret);
    }
    else {
        self->isConsumerStarted = self->device->Playing();
    }

done:
    tsk_safeobj_unlock(self);
    return (self->isConsumerStarted ? -1 : 0);
}

int audio_webrtc_instance_stop_producer(audio_webrtc_instance_handle_t* _self)
{
    audio_webrtc_instance_t* self = (audio_webrtc_instance_t*)_self;
    if(!self || !self->device || !self->transport) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("invalid parameter");
        return -1;
    }

    tsk_safeobj_lock(self);

    if(!self->isProducerStarted) {
        goto done;
    }

    int ret;
    if((ret = self->device->StopRecording())) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("StopRecording() failed with error code = %d", ret);
    }
    else {
        self->isProducerStarted = self->device->Recording();
    }

done:
    tsk_safeobj_unlock(self);
    return (self->isProducerStarted ? -1 : 0);
}

int audio_webrtc_instance_destroy(audio_webrtc_instance_handle_t** _self)
{
    if(!_self || !*_self) {
        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
        return -1;
    }
    tsk_list_lock(__audioInstances);
    if(tsk_object_get_refcount(*_self)==1) {
        tsk_list_remove_item_by_data(__audioInstances, *_self);
    }
    else {
        tsk_object_unref(*_self);
    }
    tsk_list_unlock(__audioInstances);
    *_self = tsk_null;
    return 0;
}