webrtc_m130/api/create_peerconnection_factory.cc

79 lines
3.1 KiB
C++
Raw Permalink Normal View History

/*
* 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"
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
#include <memory>
#include <utility>
#include "api/audio/audio_device.h"
#include "api/audio/audio_mixer.h"
#include "api/audio/audio_processing.h"
#include "api/audio/builtin_audio_processing_builder.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/enable_media.h"
#include "api/field_trials_view.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/scoped_refptr.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.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) {
PeerConnectionFactoryDependencies dependencies;
dependencies.network_thread = network_thread;
dependencies.worker_thread = worker_thread;
dependencies.signaling_thread = signaling_thread;
dependencies.event_log_factory = std::make_unique<RtcEventLogFactory>();
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 != nullptr) {
dependencies.audio_processing_builder =
CustomAudioProcessing(std::move(audio_processing));
} else {
#ifndef WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE
dependencies.audio_processing_builder =
std::make_unique<BuiltinAudioProcessingBuilder>();
#endif
}
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