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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/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.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/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm | 342 |
1 files changed, 342 insertions, 0 deletions
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm new file mode 100644 index 0000000000..c4d89e911d --- /dev/null +++ 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 |