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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/create_peerconnection_factory.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/api/create_peerconnection_factory.cc | 83 |
1 files changed, 83 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/create_peerconnection_factory.cc b/third_party/libwebrtc/api/create_peerconnection_factory.cc new file mode 100644 index 0000000000..f9cc7ad3e2 --- /dev/null +++ b/third_party/libwebrtc/api/create_peerconnection_factory.cc @@ -0,0 +1,83 @@ +/* + * 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, + 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.call_factory = CreateCallFactory(); + 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(); + } + 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 |