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diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
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+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
@@ -0,0 +1,342 @@
+/*
+ * Copyright 2015 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 <memory>
+
+#import "RTCPeerConnectionFactory+Native.h"
+#import "RTCPeerConnectionFactory+Private.h"
+#import "RTCPeerConnectionFactoryOptions+Private.h"
+
+#import "RTCAudioSource+Private.h"
+#import "RTCAudioTrack+Private.h"
+#import "RTCMediaConstraints+Private.h"
+#import "RTCMediaStream+Private.h"
+#import "RTCPeerConnection+Private.h"
+#import "RTCVideoSource+Private.h"
+#import "RTCVideoTrack+Private.h"
+#import "base/RTCLogging.h"
+#import "base/RTCVideoDecoderFactory.h"
+#import "base/RTCVideoEncoderFactory.h"
+#import "helpers/NSString+StdString.h"
+#include "rtc_base/checks.h"
+#include "sdk/objc/native/api/network_monitor_factory.h"
+#include "sdk/objc/native/api/ssl_certificate_verifier.h"
+#include "system_wrappers/include/field_trial.h"
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/rtc_event_log/rtc_event_log_factory.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/transport/field_trial_based_config.h"
+#import "components/video_codec/RTCVideoDecoderFactoryH264.h"
+#import "components/video_codec/RTCVideoEncoderFactoryH264.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 "sdk/objc/native/api/objc_audio_device_module.h"
+#include "sdk/objc/native/api/video_decoder_factory.h"
+#include "sdk/objc/native/api/video_encoder_factory.h"
+#include "sdk/objc/native/src/objc_video_decoder_factory.h"
+#include "sdk/objc/native/src/objc_video_encoder_factory.h"
+
+#if defined(WEBRTC_IOS)
+#import "sdk/objc/native/api/audio_device_module.h"
+#endif
+
+@implementation RTC_OBJC_TYPE (RTCPeerConnectionFactory) {
+ std::unique_ptr<rtc::Thread> _networkThread;
+ std::unique_ptr<rtc::Thread> _workerThread;
+ std::unique_ptr<rtc::Thread> _signalingThread;
+ BOOL _hasStartedAecDump;
+}
+
+@synthesize nativeFactory = _nativeFactory;
+
+- (rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule {
+#if defined(WEBRTC_IOS)
+ return webrtc::CreateAudioDeviceModule();
+#else
+ return nullptr;
+#endif
+}
+
+- (instancetype)init {
+ return [self
+ initWithNativeAudioEncoderFactory:webrtc::CreateBuiltinAudioEncoderFactory()
+ nativeAudioDecoderFactory:webrtc::CreateBuiltinAudioDecoderFactory()
+ nativeVideoEncoderFactory:webrtc::ObjCToNativeVideoEncoderFactory([[RTC_OBJC_TYPE(
+ RTCVideoEncoderFactoryH264) alloc] init])
+ nativeVideoDecoderFactory:webrtc::ObjCToNativeVideoDecoderFactory([[RTC_OBJC_TYPE(
+ RTCVideoDecoderFactoryH264) alloc] init])
+ audioDeviceModule:[self audioDeviceModule].get()
+ audioProcessingModule:nullptr];
+}
+
+- (instancetype)
+ initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+ decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory {
+ return [self initWithEncoderFactory:encoderFactory decoderFactory:decoderFactory audioDevice:nil];
+}
+
+- (instancetype)
+ initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+ decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory
+ audioDevice:(nullable id<RTC_OBJC_TYPE(RTCAudioDevice)>)audioDevice {
+#ifdef HAVE_NO_MEDIA
+ return [self initWithNoMedia];
+#else
+ std::unique_ptr<webrtc::VideoEncoderFactory> native_encoder_factory;
+ std::unique_ptr<webrtc::VideoDecoderFactory> native_decoder_factory;
+ if (encoderFactory) {
+ native_encoder_factory = webrtc::ObjCToNativeVideoEncoderFactory(encoderFactory);
+ }
+ if (decoderFactory) {
+ native_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory(decoderFactory);
+ }
+ rtc::scoped_refptr<webrtc::AudioDeviceModule> audio_device_module;
+ if (audioDevice) {
+ audio_device_module = webrtc::CreateAudioDeviceModule(audioDevice);
+ } else {
+ audio_device_module = [self audioDeviceModule];
+ }
+ return [self initWithNativeAudioEncoderFactory:webrtc::CreateBuiltinAudioEncoderFactory()
+ nativeAudioDecoderFactory:webrtc::CreateBuiltinAudioDecoderFactory()
+ nativeVideoEncoderFactory:std::move(native_encoder_factory)
+ nativeVideoDecoderFactory:std::move(native_decoder_factory)
+ audioDeviceModule:audio_device_module.get()
+ audioProcessingModule:nullptr];
+#endif
+}
+
+- (instancetype)initNative {
+ if (self = [super init]) {
+ _networkThread = rtc::Thread::CreateWithSocketServer();
+ _networkThread->SetName("network_thread", _networkThread.get());
+ BOOL result = _networkThread->Start();
+ RTC_DCHECK(result) << "Failed to start network thread.";
+
+ _workerThread = rtc::Thread::Create();
+ _workerThread->SetName("worker_thread", _workerThread.get());
+ result = _workerThread->Start();
+ RTC_DCHECK(result) << "Failed to start worker thread.";
+
+ _signalingThread = rtc::Thread::Create();
+ _signalingThread->SetName("signaling_thread", _signalingThread.get());
+ result = _signalingThread->Start();
+ RTC_DCHECK(result) << "Failed to start signaling thread.";
+ }
+ return self;
+}
+
+- (instancetype)initWithNoMedia {
+ if (self = [self initNative]) {
+ webrtc::PeerConnectionFactoryDependencies dependencies;
+ dependencies.network_thread = _networkThread.get();
+ dependencies.worker_thread = _workerThread.get();
+ dependencies.signaling_thread = _signalingThread.get();
+ if (webrtc::field_trial::IsEnabled("WebRTC-Network-UseNWPathMonitor")) {
+ dependencies.network_monitor_factory = webrtc::CreateNetworkMonitorFactory();
+ }
+ _nativeFactory = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
+ NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
+ }
+ return self;
+}
+
+- (instancetype)initWithNativeAudioEncoderFactory:
+ (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
+ nativeAudioDecoderFactory:
+ (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
+ nativeVideoEncoderFactory:
+ (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
+ nativeVideoDecoderFactory:
+ (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
+ audioDeviceModule:(webrtc::AudioDeviceModule *)audioDeviceModule
+ audioProcessingModule:
+ (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule {
+ return [self initWithNativeAudioEncoderFactory:audioEncoderFactory
+ nativeAudioDecoderFactory:audioDecoderFactory
+ nativeVideoEncoderFactory:std::move(videoEncoderFactory)
+ nativeVideoDecoderFactory:std::move(videoDecoderFactory)
+ audioDeviceModule:audioDeviceModule
+ audioProcessingModule:audioProcessingModule
+ networkControllerFactory:nullptr];
+}
+- (instancetype)initWithNativeAudioEncoderFactory:
+ (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
+ nativeAudioDecoderFactory:
+ (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
+ nativeVideoEncoderFactory:
+ (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
+ nativeVideoDecoderFactory:
+ (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
+ audioDeviceModule:(webrtc::AudioDeviceModule *)audioDeviceModule
+ audioProcessingModule:
+ (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
+ networkControllerFactory:
+ (std::unique_ptr<webrtc::NetworkControllerFactoryInterface>)
+ networkControllerFactory {
+ if (self = [self initNative]) {
+ webrtc::PeerConnectionFactoryDependencies dependencies;
+ dependencies.network_thread = _networkThread.get();
+ dependencies.worker_thread = _workerThread.get();
+ dependencies.signaling_thread = _signalingThread.get();
+ if (webrtc::field_trial::IsEnabled("WebRTC-Network-UseNWPathMonitor")) {
+ dependencies.network_monitor_factory = webrtc::CreateNetworkMonitorFactory();
+ }
+ dependencies.trials = std::make_unique<webrtc::FieldTrialBasedConfig>();
+ dependencies.task_queue_factory =
+ webrtc::CreateDefaultTaskQueueFactory(dependencies.trials.get());
+ cricket::MediaEngineDependencies media_deps;
+ media_deps.adm = std::move(audioDeviceModule);
+ media_deps.task_queue_factory = dependencies.task_queue_factory.get();
+ media_deps.audio_encoder_factory = std::move(audioEncoderFactory);
+ media_deps.audio_decoder_factory = std::move(audioDecoderFactory);
+ media_deps.video_encoder_factory = std::move(videoEncoderFactory);
+ media_deps.video_decoder_factory = std::move(videoDecoderFactory);
+ if (audioProcessingModule) {
+ media_deps.audio_processing = std::move(audioProcessingModule);
+ } else {
+ media_deps.audio_processing = webrtc::AudioProcessingBuilder().Create();
+ }
+ media_deps.trials = dependencies.trials.get();
+ dependencies.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
+ dependencies.call_factory = webrtc::CreateCallFactory();
+ dependencies.event_log_factory =
+ std::make_unique<webrtc::RtcEventLogFactory>(dependencies.task_queue_factory.get());
+ dependencies.network_controller_factory = std::move(networkControllerFactory);
+ _nativeFactory = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
+ NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
+ }
+ return self;
+}
+
+- (RTC_OBJC_TYPE(RTCAudioSource) *)audioSourceWithConstraints:
+ (nullable RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints {
+ std::unique_ptr<webrtc::MediaConstraints> nativeConstraints;
+ if (constraints) {
+ nativeConstraints = constraints.nativeConstraints;
+ }
+ cricket::AudioOptions options;
+ CopyConstraintsIntoAudioOptions(nativeConstraints.get(), &options);
+
+ rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
+ _nativeFactory->CreateAudioSource(options);
+ return [[RTC_OBJC_TYPE(RTCAudioSource) alloc] initWithFactory:self nativeAudioSource:source];
+}
+
+- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithTrackId:(NSString *)trackId {
+ RTC_OBJC_TYPE(RTCAudioSource) *audioSource = [self audioSourceWithConstraints:nil];
+ return [self audioTrackWithSource:audioSource trackId:trackId];
+}
+
+- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithSource:(RTC_OBJC_TYPE(RTCAudioSource) *)source
+ trackId:(NSString *)trackId {
+ return [[RTC_OBJC_TYPE(RTCAudioTrack) alloc] initWithFactory:self source:source trackId:trackId];
+}
+
+- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSource {
+ return [[RTC_OBJC_TYPE(RTCVideoSource) alloc] initWithFactory:self
+ signalingThread:_signalingThread.get()
+ workerThread:_workerThread.get()];
+}
+
+- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSourceForScreenCast:(BOOL)forScreenCast {
+ return [[RTC_OBJC_TYPE(RTCVideoSource) alloc] initWithFactory:self
+ signalingThread:_signalingThread.get()
+ workerThread:_workerThread.get()
+ isScreenCast:forScreenCast];
+}
+
+- (RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrackWithSource:(RTC_OBJC_TYPE(RTCVideoSource) *)source
+ trackId:(NSString *)trackId {
+ return [[RTC_OBJC_TYPE(RTCVideoTrack) alloc] initWithFactory:self source:source trackId:trackId];
+}
+
+- (RTC_OBJC_TYPE(RTCMediaStream) *)mediaStreamWithStreamId:(NSString *)streamId {
+ return [[RTC_OBJC_TYPE(RTCMediaStream) alloc] initWithFactory:self streamId:streamId];
+}
+
+- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
+ peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+ constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+ delegate:
+ (nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
+ return [[RTC_OBJC_TYPE(RTCPeerConnection) alloc] initWithFactory:self
+ configuration:configuration
+ constraints:constraints
+ certificateVerifier:nil
+ delegate:delegate];
+}
+
+- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
+ peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+ constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+ certificateVerifier:
+ (id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)certificateVerifier
+ delegate:
+ (nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
+ return [[RTC_OBJC_TYPE(RTCPeerConnection) alloc] initWithFactory:self
+ configuration:configuration
+ constraints:constraints
+ certificateVerifier:certificateVerifier
+ delegate:delegate];
+}
+
+- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
+ peerConnectionWithDependencies:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+ constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+ dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies
+ delegate:(id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
+ return [[RTC_OBJC_TYPE(RTCPeerConnection) alloc] initWithDependencies:self
+ configuration:configuration
+ constraints:constraints
+ dependencies:std::move(dependencies)
+ delegate:delegate];
+}
+
+- (void)setOptions:(nonnull RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions) *)options {
+ RTC_DCHECK(options != nil);
+ _nativeFactory->SetOptions(options.nativeOptions);
+}
+
+- (BOOL)startAecDumpWithFilePath:(NSString *)filePath
+ maxSizeInBytes:(int64_t)maxSizeInBytes {
+ RTC_DCHECK(filePath.length);
+ RTC_DCHECK_GT(maxSizeInBytes, 0);
+
+ if (_hasStartedAecDump) {
+ RTCLogError(@"Aec dump already started.");
+ return NO;
+ }
+ FILE *f = fopen(filePath.UTF8String, "wb");
+ if (!f) {
+ RTCLogError(@"Error opening file: %@. Error: %s", filePath, strerror(errno));
+ return NO;
+ }
+ _hasStartedAecDump = _nativeFactory->StartAecDump(f, maxSizeInBytes);
+ return _hasStartedAecDump;
+}
+
+- (void)stopAecDump {
+ _nativeFactory->StopAecDump();
+ _hasStartedAecDump = NO;
+}
+
+- (rtc::Thread *)signalingThread {
+ return _signalingThread.get();
+}
+
+- (rtc::Thread *)workerThread {
+ return _workerThread.get();
+}
+
+@end