Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 129 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..b7f9eb7f30
--- /dev/null
+++ b/third_party/libwebrtc/api/create_peerconnection_factory.cc
@@ -0,0 +1,129 @@
+/*
+ *  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<AudioFrameProcessor> owned_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);
+  if (audio_frame_processor) {
+    media_dependencies.audio_frame_processor = audio_frame_processor;
+  } else if (owned_audio_frame_processor) {
+    media_dependencies.owned_audio_frame_processor =
+        std::move(owned_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));
+}
+
+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) {
+  return CreatePeerConnectionFactory(
+      network_thread, worker_thread, signaling_thread, default_adm,
+      audio_encoder_factory, audio_decoder_factory,
+      std::move(video_encoder_factory), std::move(video_decoder_factory),
+      audio_mixer, audio_processing, audio_frame_processor, nullptr, nullptr);
+}
+
+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> owned_audio_frame_processor,
+    std::unique_ptr<FieldTrialsView> field_trials) {
+  return CreatePeerConnectionFactory(
+      network_thread, worker_thread, signaling_thread, default_adm,
+      audio_encoder_factory, audio_decoder_factory,
+      std::move(video_encoder_factory), std::move(video_decoder_factory),
+      audio_mixer, audio_processing, nullptr,
+      std::move(owned_audio_frame_processor), std::move(field_trials));
+}
+
+}  // namespace webrtc