| /* |
| * 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 "RTCRtpCapabilities+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 "api/audio/audio_device.h" |
| #include "api/audio/audio_processing.h" |
| #include "api/audio/builtin_audio_processing_builder.h" |
| #include "api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "api/create_modular_peer_connection_factory.h" |
| #include "api/enable_media.h" |
| #include "api/environment/environment_factory.h" |
| #include "api/rtc_event_log/rtc_event_log_factory.h" |
| #include "api/task_queue/default_task_queue_factory.h" |
| #import "components/video_codec/RTCVideoDecoderFactoryH264.h" |
| #import "components/video_codec/RTCVideoEncoderFactoryH264.h" |
| #include "media/base/media_constants.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<webrtc::Thread> _networkThread; |
| std::unique_ptr<webrtc::Thread> _workerThread; |
| std::unique_ptr<webrtc::Thread> _signalingThread; |
| BOOL _hasStartedAecDump; |
| } |
| |
| @synthesize nativeFactory = _nativeFactory; |
| |
| - (instancetype)init { |
| webrtc::PeerConnectionFactoryDependencies dependencies; |
| dependencies.audio_encoder_factory = |
| webrtc::CreateBuiltinAudioEncoderFactory(); |
| dependencies.audio_decoder_factory = |
| webrtc::CreateBuiltinAudioDecoderFactory(); |
| dependencies.video_encoder_factory = webrtc::ObjCToNativeVideoEncoderFactory( |
| [[RTC_OBJC_TYPE(RTCVideoEncoderFactoryH264) alloc] init]); |
| dependencies.video_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory( |
| [[RTC_OBJC_TYPE(RTCVideoDecoderFactoryH264) alloc] init]); |
| dependencies.env = webrtc::CreateEnvironment(); |
| #ifdef WEBRTC_IOS |
| dependencies.adm = webrtc::CreateAudioDeviceModule(*dependencies.env); |
| #endif |
| return [self initWithMediaAndDependencies:dependencies]; |
| } |
| |
| - (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 |
| webrtc::PeerConnectionFactoryDependencies dependencies; |
| dependencies.env = webrtc::CreateEnvironment(); |
| dependencies.audio_encoder_factory = |
| webrtc::CreateBuiltinAudioEncoderFactory(); |
| dependencies.audio_decoder_factory = |
| webrtc::CreateBuiltinAudioDecoderFactory(); |
| if (encoderFactory) { |
| dependencies.video_encoder_factory = |
| webrtc::ObjCToNativeVideoEncoderFactory(encoderFactory); |
| } |
| if (decoderFactory) { |
| dependencies.video_decoder_factory = |
| webrtc::ObjCToNativeVideoDecoderFactory(decoderFactory); |
| } |
| if (audioDevice) { |
| dependencies.adm = |
| webrtc::CreateAudioDeviceModule(*dependencies.env, audioDevice); |
| #ifdef WEBRTC_IOS |
| } else { |
| dependencies.adm = webrtc::CreateAudioDeviceModule(*dependencies.env); |
| #endif |
| } |
| return [self initWithMediaAndDependencies:dependencies]; |
| #endif |
| } |
| |
| - (instancetype)initWithNativeDependencies: |
| (webrtc::PeerConnectionFactoryDependencies &)dependencies { |
| self = [super init]; |
| if (self) { |
| _networkThread = webrtc::Thread::CreateWithSocketServer(); |
| _networkThread->SetName("network_thread", _networkThread.get()); |
| BOOL result = _networkThread->Start(); |
| RTC_DCHECK(result) << "Failed to start network thread."; |
| |
| _workerThread = webrtc::Thread::Create(); |
| _workerThread->SetName("worker_thread", _workerThread.get()); |
| result = _workerThread->Start(); |
| RTC_DCHECK(result) << "Failed to start worker thread."; |
| |
| _signalingThread = webrtc::Thread::Create(); |
| _signalingThread->SetName("signaling_thread", _signalingThread.get()); |
| result = _signalingThread->Start(); |
| RTC_DCHECK(result) << "Failed to start signaling thread."; |
| |
| // Set fields that are relevant both to 'no media' and 'with media' |
| // scenarios. |
| dependencies.network_thread = _networkThread.get(); |
| dependencies.worker_thread = _workerThread.get(); |
| dependencies.signaling_thread = _signalingThread.get(); |
| if (!dependencies.env.has_value()) { |
| dependencies.env = webrtc::CreateEnvironment(); |
| } |
| if (dependencies.network_monitor_factory == nullptr && |
| dependencies.env->field_trials().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)initWithNoMedia { |
| webrtc::PeerConnectionFactoryDependencies default_deps; |
| return [self initWithNativeDependencies:default_deps]; |
| } |
| |
| - (instancetype) |
| initWithNativeAudioEncoderFactory: |
| (webrtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory |
| nativeAudioDecoderFactory: |
| (webrtc::scoped_refptr<webrtc::AudioDecoderFactory>) |
| audioDecoderFactory |
| nativeVideoEncoderFactory: |
| (std::unique_ptr<webrtc::VideoEncoderFactory>) |
| videoEncoderFactory |
| nativeVideoDecoderFactory: |
| (std::unique_ptr<webrtc::VideoDecoderFactory>) |
| videoDecoderFactory |
| audioDeviceModule: |
| (webrtc::AudioDeviceModule *)audioDeviceModule |
| audioProcessingModule: |
| (webrtc::scoped_refptr<webrtc::AudioProcessing>) |
| audioProcessingModule { |
| webrtc::PeerConnectionFactoryDependencies dependencies; |
| dependencies.audio_encoder_factory = std::move(audioEncoderFactory); |
| dependencies.audio_decoder_factory = std::move(audioDecoderFactory); |
| dependencies.video_encoder_factory = std::move(videoEncoderFactory); |
| dependencies.video_decoder_factory = std::move(videoDecoderFactory); |
| dependencies.adm = std::move(audioDeviceModule); |
| if (audioProcessingModule != nullptr) { |
| dependencies.audio_processing_builder = |
| CustomAudioProcessing(std::move(audioProcessingModule)); |
| } |
| return [self initWithMediaAndDependencies:dependencies]; |
| } |
| |
| - (instancetype) |
| initWithNativeAudioEncoderFactory: |
| (webrtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory |
| nativeAudioDecoderFactory: |
| (webrtc::scoped_refptr<webrtc::AudioDecoderFactory>) |
| audioDecoderFactory |
| nativeVideoEncoderFactory: |
| (std::unique_ptr<webrtc::VideoEncoderFactory>) |
| videoEncoderFactory |
| nativeVideoDecoderFactory: |
| (std::unique_ptr<webrtc::VideoDecoderFactory>) |
| videoDecoderFactory |
| audioDeviceModule: |
| (webrtc::AudioDeviceModule *)audioDeviceModule |
| audioProcessingModule: |
| (webrtc::scoped_refptr<webrtc::AudioProcessing>) |
| audioProcessingModule |
| networkControllerFactory: |
| (std::unique_ptr<webrtc::NetworkControllerFactoryInterface>) |
| networkControllerFactory { |
| webrtc::PeerConnectionFactoryDependencies dependencies; |
| dependencies.adm = std::move(audioDeviceModule); |
| dependencies.audio_encoder_factory = std::move(audioEncoderFactory); |
| dependencies.audio_decoder_factory = std::move(audioDecoderFactory); |
| dependencies.video_encoder_factory = std::move(videoEncoderFactory); |
| dependencies.video_decoder_factory = std::move(videoDecoderFactory); |
| if (audioProcessingModule != nullptr) { |
| dependencies.audio_processing_builder = |
| CustomAudioProcessing(std::move(audioProcessingModule)); |
| } |
| dependencies.network_controller_factory = std::move(networkControllerFactory); |
| return [self initWithMediaAndDependencies:dependencies]; |
| } |
| |
| - (instancetype)initWithMediaAndDependencies: |
| (webrtc::PeerConnectionFactoryDependencies &)dependencies { |
| #ifndef WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE |
| if (dependencies.audio_processing_builder == nullptr) { |
| dependencies.audio_processing_builder = |
| std::make_unique<webrtc::BuiltinAudioProcessingBuilder>(); |
| } |
| #endif |
| if (dependencies.event_log_factory == nullptr) { |
| dependencies.event_log_factory = |
| std::make_unique<webrtc::RtcEventLogFactory>(); |
| } |
| webrtc::EnableMedia(dependencies); |
| return [self initWithNativeDependencies:dependencies]; |
| } |
| |
| - (RTC_OBJC_TYPE(RTCRtpCapabilities) *)rtpSenderCapabilitiesForKind: |
| (NSString *)kind { |
| webrtc::MediaType mediaType = [[self class] mediaTypeForKind:kind]; |
| |
| webrtc::RtpCapabilities rtpCapabilities = |
| _nativeFactory->GetRtpSenderCapabilities(mediaType); |
| return [[RTC_OBJC_TYPE(RTCRtpCapabilities) alloc] |
| initWithNativeRtpCapabilities:rtpCapabilities]; |
| } |
| |
| - (RTC_OBJC_TYPE(RTCRtpCapabilities) *)rtpReceiverCapabilitiesForKind: |
| (NSString *)kind { |
| webrtc::MediaType mediaType = [[self class] mediaTypeForKind:kind]; |
| |
| webrtc::RtpCapabilities rtpCapabilities = |
| _nativeFactory->GetRtpReceiverCapabilities(mediaType); |
| return [[RTC_OBJC_TYPE(RTCRtpCapabilities) alloc] |
| initWithNativeRtpCapabilities:rtpCapabilities]; |
| } |
| |
| - (RTC_OBJC_TYPE(RTCAudioSource) *)audioSourceWithConstraints: |
| (nullable RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints { |
| std::unique_ptr<webrtc::MediaConstraints> nativeConstraints; |
| if (constraints) { |
| nativeConstraints = constraints.nativeConstraints; |
| } |
| webrtc::AudioOptions options; |
| CopyConstraintsIntoAudioOptions(nativeConstraints.get(), &options); |
| |
| webrtc::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; |
| } |
| |
| - (webrtc::Thread *)signalingThread { |
| return _signalingThread.get(); |
| } |
| |
| - (webrtc::Thread *)workerThread { |
| return _workerThread.get(); |
| } |
| |
| - (webrtc::Thread *)networkThread { |
| return _networkThread.get(); |
| } |
| |
| #pragma mark - Private |
| |
| + (webrtc::MediaType)mediaTypeForKind:(NSString *)kind { |
| if (kind == kRTCMediaStreamTrackKindAudio) { |
| return webrtc::MediaType::AUDIO; |
| } else if (kind == kRTCMediaStreamTrackKindVideo) { |
| return webrtc::MediaType::VIDEO; |
| } else { |
| RTC_DCHECK_NOTREACHED(); |
| return webrtc::MediaType::UNSUPPORTED; |
| } |
| } |
| |
| @end |
