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/*
* 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/enable_media.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 "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,
std::unique_ptr<AudioFrameProcessor> audio_frame_processor,
std::unique_ptr<FieldTrialsView> field_trials) {
if (!field_trials) {
field_trials = std::make_unique<webrtc::FieldTrialBasedConfig>();
}
PeerConnectionFactoryDependencies dependencies;
dependencies.network_thread = network_thread;
dependencies.worker_thread = worker_thread;
dependencies.signaling_thread = signaling_thread;
dependencies.task_queue_factory =
CreateDefaultTaskQueueFactory(field_trials.get());
dependencies.event_log_factory = std::make_unique<RtcEventLogFactory>(
dependencies.task_queue_factory.get());
dependencies.trials = std::move(field_trials);
if (network_thread) {
// TODO(bugs.webrtc.org/13145): Add an rtc::SocketFactory* argument.
dependencies.socket_factory = network_thread->socketserver();
}
dependencies.adm = std::move(default_adm);
dependencies.audio_encoder_factory = std::move(audio_encoder_factory);
dependencies.audio_decoder_factory = std::move(audio_decoder_factory);
dependencies.audio_frame_processor = std::move(audio_frame_processor);
if (audio_processing) {
dependencies.audio_processing = std::move(audio_processing);
} else {
dependencies.audio_processing = AudioProcessingBuilder().Create();
}
dependencies.audio_mixer = std::move(audio_mixer);
dependencies.video_encoder_factory = std::move(video_encoder_factory);
dependencies.video_decoder_factory = std::move(video_decoder_factory);
EnableMedia(dependencies);
return CreateModularPeerConnectionFactory(std::move(dependencies));
}
} // namespace webrtc
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