/* * 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 #include #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 CreatePeerConnectionFactory( rtc::Thread* network_thread, rtc::Thread* worker_thread, rtc::Thread* signaling_thread, rtc::scoped_refptr default_adm, rtc::scoped_refptr audio_encoder_factory, rtc::scoped_refptr audio_decoder_factory, std::unique_ptr video_encoder_factory, std::unique_ptr video_decoder_factory, rtc::scoped_refptr audio_mixer, rtc::scoped_refptr 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( dependencies.task_queue_factory.get()); dependencies.trials = std::make_unique(); 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