Telegram-Android/TMessagesProj/jni/voip/webrtc/api/create_peerconnection_factory.cc
2022-03-13 04:58:00 +03:00

73 lines
3.2 KiB
C++

/*
* Copyright 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "api/create_peerconnection_factory.h"
#include <memory>
#include <utility>
#include "api/call/call_factory_interface.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/transport/field_trial_based_config.h"
#include "media/base/media_engine.h"
#include "media/engine/webrtc_media_engine.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/thread.h"
namespace webrtc {
rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
rtc::scoped_refptr<AudioDeviceModule> default_adm,
rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
std::unique_ptr<VideoEncoderFactory> video_encoder_factory,
std::unique_ptr<VideoDecoderFactory> video_decoder_factory,
rtc::scoped_refptr<AudioMixer> audio_mixer,
rtc::scoped_refptr<AudioProcessing> audio_processing,
AudioFrameProcessor* audio_frame_processor) {
PeerConnectionFactoryDependencies dependencies;
dependencies.network_thread = network_thread;
dependencies.worker_thread = worker_thread;
dependencies.signaling_thread = signaling_thread;
dependencies.task_queue_factory = CreateDefaultTaskQueueFactory();
dependencies.call_factory = CreateCallFactory();
dependencies.event_log_factory = std::make_unique<RtcEventLogFactory>(
dependencies.task_queue_factory.get());
dependencies.trials = std::make_unique<webrtc::FieldTrialBasedConfig>();
cricket::MediaEngineDependencies media_dependencies;
media_dependencies.task_queue_factory = dependencies.task_queue_factory.get();
media_dependencies.adm = std::move(default_adm);
media_dependencies.audio_encoder_factory = std::move(audio_encoder_factory);
media_dependencies.audio_decoder_factory = std::move(audio_decoder_factory);
media_dependencies.audio_frame_processor = audio_frame_processor;
if (audio_processing) {
media_dependencies.audio_processing = std::move(audio_processing);
} else {
media_dependencies.audio_processing = AudioProcessingBuilder().Create();
}
media_dependencies.audio_mixer = std::move(audio_mixer);
media_dependencies.video_encoder_factory = std::move(video_encoder_factory);
media_dependencies.video_decoder_factory = std::move(video_decoder_factory);
media_dependencies.trials = dependencies.trials.get();
dependencies.media_engine =
cricket::CreateMediaEngine(std::move(media_dependencies));
return CreateModularPeerConnectionFactory(std::move(dependencies));
}
} // namespace webrtc