path: root/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;
}