Use h265

submodules/TelegramVoip/Sources/OngoingCallContext.swift

@@ -514,11 +514,13 @@ public final class OngoingCallContext {
             }
         }))
         
-        self.signalingDataDisposable = (callSessionManager.callSignalingData(internalId: internalId)
+        self.signalingDataDisposable = (callSessionManager.callSignalingData(internalId: internalId)).start(next: { [weak self] data in
-        |> deliverOn(self.queue)).start(next: { [weak self] data in
+            print("data received")
-            self?.withContext { context in
+            queue.async {
-                if let context = context as? OngoingCallThreadLocalContextWebrtc {
+                self?.withContext { context in
-                    context.addSignaling(data)
+                    if let context = context as? OngoingCallThreadLocalContextWebrtc {
+                        context.addSignaling(data)
+                    }
                 }
             }
         })

submodules/TgVoipWebrtc/Impl/CodecsApple.mm

@@ -91,7 +91,7 @@
             return nil;
         }
         
-        [_videoCapturer startCaptureWithDevice:frontCamera format:bestFormat fps:27];
+        [_videoCapturer startCaptureWithDevice:frontCamera format:bestFormat fps:30];
     }
     return self;
 }

submodules/TgVoipWebrtc/Impl/Manager.cpp

@@ -51,13 +51,13 @@ Manager::~Manager() {
 
 void Manager::start() {
     auto weakThis = std::weak_ptr<Manager>(shared_from_this());
-    _networkManager.reset(new ThreadLocalObject<NetworkManager>(getNetworkThread(), [encryptionKey = _encryptionKey, enableP2P = _enableP2P, thread = _thread, weakThis]() {
+    _networkManager.reset(new ThreadLocalObject<NetworkManager>(getNetworkThread(), [encryptionKey = _encryptionKey, enableP2P = _enableP2P, thread = _thread, weakThis, signalingDataEmitted = _signalingDataEmitted]() {
         return new NetworkManager(
             getNetworkThread(),
             encryptionKey,
             enableP2P,
             [thread, weakThis](const NetworkManager::State &state) {
-                thread->Invoke<void>(RTC_FROM_HERE, [weakThis, state]() {
+                thread->PostTask(RTC_FROM_HERE, [weakThis, state]() {
                     auto strongThis = weakThis.lock();
                     if (strongThis == nullptr) {
                         return;
@@ -86,14 +86,8 @@ void Manager::start() {
                     });
                 });
             },
-            [thread, weakThis](const std::vector<uint8_t> &data) {
+            [signalingDataEmitted](const std::vector<uint8_t> &data) {
-                thread->PostTask(RTC_FROM_HERE, [weakThis, data]() {
+                signalingDataEmitted(data);
-                    auto strongThis = weakThis.lock();
-                    if (strongThis == nullptr) {
-                        return;
-                    }
-                    strongThis->_signalingDataEmitted(data);
-                });
             }
         );
     }));

submodules/TgVoipWebrtc/Impl/MediaManager.cpp

@@ -73,9 +73,6 @@ static std::vector<cricket::VideoCodec> AssignPayloadTypesAndDefaultCodecs(std::
      input_formats.push_back(flexfec_format);
      }*/
     
-    bool found = false;
-    bool useVP9 = true;
-    
     std::vector<cricket::VideoCodec> output_codecs;
     for (const webrtc::SdpVideoFormat& format : input_formats) {
         cricket::VideoCodec codec(format);
@@ -83,19 +80,6 @@ static std::vector<cricket::VideoCodec> AssignPayloadTypesAndDefaultCodecs(std::
         AddDefaultFeedbackParams(&codec);
         output_codecs.push_back(codec);
         
-        if (useVP9 && codec.name == cricket::kVp9CodecName) {
-            if (!found) {
-                outCodecId = codec.id;
-                found = true;
-            }
-        }
-        if (!useVP9 && codec.name == cricket::kH264CodecName) {
-            if (!found) {
-                outCodecId = codec.id;
-                found = true;
-            }
-        }
-        
         // Increment payload type.
         ++payload_type;
         if (payload_type > kLastDynamicPayloadType) {
@@ -122,18 +106,24 @@ static std::vector<cricket::VideoCodec> AssignPayloadTypesAndDefaultCodecs(std::
 
 static absl::optional<cricket::VideoCodec> selectVideoCodec(std::vector<cricket::VideoCodec> &codecs) {
     bool useVP9 = false;
+    bool useH265 = true;
     
     for (auto &codec : codecs) {
         if (useVP9) {
             if (codec.name == cricket::kVp9CodecName) {
                 return absl::optional<cricket::VideoCodec>(codec);
             }
+        } else if (useH265) {
+            if (codec.name == cricket::kH265CodecName) {
+                return absl::optional<cricket::VideoCodec>(codec);
+            }
         } else {
             if (codec.name == cricket::kH264CodecName) {
                 return absl::optional<cricket::VideoCodec>(codec);
             }
         }
     }
+    
     return absl::optional<cricket::VideoCodec>();
 }
 
@@ -210,7 +200,7 @@ _taskQueueFactory(webrtc::CreateDefaultTaskQueueFactory()) {
     const uint8_t opusMaxBitrateKbps = 32;
     const uint8_t opusStartBitrateKbps = 6;
     const uint8_t opusPTimeMs = 120;
-    const int opusExtensionSequence = 1;
+    const int extensionSequenceOne = 1;
     
     cricket::AudioCodec opusCodec(opusSdpPayload, opusSdpName, opusClockrate, opusSdpBitrate, opusSdpChannels);
     opusCodec.AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc));
@@ -222,7 +212,7 @@ _taskQueueFactory(webrtc::CreateDefaultTaskQueueFactory()) {
 
     cricket::AudioSendParameters audioSendPrameters;
     audioSendPrameters.codecs.push_back(opusCodec);
-    audioSendPrameters.extensions.emplace_back(webrtc::RtpExtension::kTransportSequenceNumberUri, opusExtensionSequence);
+    audioSendPrameters.extensions.emplace_back(webrtc::RtpExtension::kTransportSequenceNumberUri, extensionSequenceOne);
     audioSendPrameters.options.echo_cancellation = false;
     //audioSendPrameters.options.experimental_ns = false;
     audioSendPrameters.options.noise_suppression = false;
@@ -237,7 +227,7 @@ _taskQueueFactory(webrtc::CreateDefaultTaskQueueFactory()) {
     
     cricket::AudioRecvParameters audioRecvParameters;
     audioRecvParameters.codecs.emplace_back(opusSdpPayload, opusSdpName, opusClockrate, opusSdpBitrate, opusSdpChannels);
-    audioRecvParameters.extensions.emplace_back(webrtc::RtpExtension::kTransportSequenceNumberUri, opusExtensionSequence);
+    audioRecvParameters.extensions.emplace_back(webrtc::RtpExtension::kTransportSequenceNumberUri, extensionSequenceOne);
     audioRecvParameters.rtcp.reduced_size = true;
     audioRecvParameters.rtcp.remote_estimate = true;
     _audioChannel->AddRecvStream(cricket::StreamParams::CreateLegacy(_ssrcAudio.incoming));
@@ -253,15 +243,14 @@ _taskQueueFactory(webrtc::CreateDefaultTaskQueueFactory()) {
         auto codec = videoCodec.value();
         
         codec.SetParam(cricket::kCodecParamMinBitrate, 64);
-        codec.SetParam(cricket::kCodecParamStartBitrate, 256);
+        codec.SetParam(cricket::kCodecParamStartBitrate, 512);
         codec.SetParam(cricket::kCodecParamMaxBitrate, 2500);
         
         _videoCapturer = makeVideoCapturer(_nativeVideoSource);
         
         cricket::VideoSendParameters videoSendParameters;
         videoSendParameters.codecs.push_back(codec);
-        const int videoExtensionSequence = 1;
+        videoSendParameters.extensions.emplace_back(webrtc::RtpExtension::kTransportSequenceNumberUri, extensionSequenceOne);
-        videoSendParameters.extensions.emplace_back(webrtc::RtpExtension::kTransportSequenceNumberUri, videoExtensionSequence);
         //send_parameters.max_bandwidth_bps = 800000;
         //send_parameters.rtcp.reduced_size = true;
         videoSendParameters.rtcp.remote_estimate = true;
@@ -273,7 +262,7 @@ _taskQueueFactory(webrtc::CreateDefaultTaskQueueFactory()) {
         
         cricket::VideoRecvParameters videoRecvParameters;
         videoRecvParameters.codecs.emplace_back(codec);
-        videoRecvParameters.extensions.emplace_back(webrtc::RtpExtension::kTransportSequenceNumberUri, videoExtensionSequence);
+        videoRecvParameters.extensions.emplace_back(webrtc::RtpExtension::kTransportSequenceNumberUri, extensionSequenceOne);
         //recv_parameters.rtcp.reduced_size = true;
         videoRecvParameters.rtcp.remote_estimate = true;
         _videoChannel->AddRecvStream(cricket::StreamParams::CreateLegacy(_ssrcVideo.incoming));

submodules/TgVoipWebrtc/Impl/ThreadLocalObject.h

@@ -21,48 +21,31 @@ private:
     
 public:
     ThreadLocalObject(rtc::Thread *thread, std::function<T *()> generator) :
-    _thread(thread) {
+    _thread(thread),
+    _valueHolder(new ThreadLocalObject::ValueHolder<T>()) {
         assert(_thread != nullptr);
-        _valueHolder = new ThreadLocalObject::ValueHolder<T>();
+        _thread->PostTask(RTC_FROM_HERE, [valueHolder = _valueHolder, generator](){
-        //ValueHolder<T> *valueHolder = _valueHolder;
+            valueHolder->_value.reset(generator());
-        _thread->Invoke<void>(RTC_FROM_HERE, [this, generator](){
-            this->_valueHolder->_value.reset(generator());
         });
     }
     
     ~ThreadLocalObject() {
-        ValueHolder<T> *valueHolder = _valueHolder;
+        _thread->PostTask(RTC_FROM_HERE, [valueHolder = _valueHolder](){
-        _thread->Invoke<void>(RTC_FROM_HERE, [this](){
+            valueHolder->_value.reset();
-            this->_valueHolder->_value.reset();
         });
-        delete valueHolder;
     }
     
     template <class FunctorT>
     void perform(FunctorT&& functor) {
-        //ValueHolder<T> *valueHolder = _valueHolder;
+        _thread->PostTask(RTC_FROM_HERE, [valueHolder = _valueHolder, f = std::forward<FunctorT>(functor)](){
-        /*_thread->PostTask(RTC_FROM_HERE, [valueHolder, f = std::forward<std::function<void(T &)>>(f)](){
+            assert(valueHolder->_value != nullptr);
-            T *value = valueHolder->_value;
+            f(valueHolder->_value.get());
-            assert(value != nullptr);
-            f(*value);
-        });*/
-        _thread->Invoke<void>(RTC_FROM_HERE, [this, f = std::forward<FunctorT>(functor)](){
-            assert(_valueHolder->_value != nullptr);
-            f(_valueHolder->_value.get());
-        });
-    }
-    
-    template <class FunctorT>
-    void performSync(FunctorT&& functor) {
-        _thread->Invoke<void>(RTC_FROM_HERE, [this, f = std::forward<FunctorT>(functor)](){
-            assert(_valueHolder->_value != nullptr);
-            f(_valueHolder->_value.get());
         });
     }
     
 private:
     rtc::Thread *_thread;
-    ValueHolder<T> *_valueHolder;
+    std::shared_ptr<ValueHolder<T>> _valueHolder;
 };
 
 #ifdef TGVOIP_NAMESPACE

third-party/webrtc/build-webrtc-bazel.sh

@@ -23,7 +23,7 @@ if [ "$ARCH" == "x64" ]; then
   OUT_DIR="ios_sim"
 fi
 
-gn gen out/$OUT_DIR --args="use_xcode_clang=true "" target_cpu=\"$ARCH\""' target_os="ios" is_debug=false is_component_build=false rtc_include_tests=false use_rtti=true rtc_use_x11=false use_custom_libcxx=false use_custom_libcxx_for_host=false rtc_build_ssl=false rtc_build_examples=false rtc_build_tools=false ios_deployment_target="9.0" ios_enable_code_signing=false is_unsafe_developer_build=false rtc_enable_protobuf=false rtc_include_builtin_video_codecs=true rtc_build_libvpx=true rtc_libvpx_build_vp9=true rtc_use_gtk=false rtc_use_metal_rendering=true rtc_ssl_root="//openssl"'
+gn gen out/$OUT_DIR --args="use_xcode_clang=true "" target_cpu=\"$ARCH\""' target_os="ios" is_debug=false is_component_build=false rtc_include_tests=false use_rtti=true rtc_use_x11=false use_custom_libcxx=false use_custom_libcxx_for_host=false rtc_build_ssl=false rtc_build_examples=false rtc_build_tools=false ios_deployment_target="9.0" ios_enable_code_signing=false is_unsafe_developer_build=false rtc_enable_protobuf=false rtc_include_builtin_video_codecs=true rtc_build_libvpx=true rtc_libvpx_build_vp9=true rtc_use_gtk=false rtc_use_metal_rendering=true'
 ninja -C out/$OUT_DIR framework_objc_static
 
 popd

third-party/webrtc/build-webrtc-buck.sh

@@ -27,7 +27,7 @@ if [ "$ARCH" == "x64" ]; then
   OUT_DIR="ios_sim"
 fi
 
-buildtools/mac/gn gen out/$OUT_DIR --args="use_xcode_clang=true "" target_cpu=\"$ARCH\""' target_os="ios" is_debug=false is_component_build=false rtc_include_tests=false use_rtti=true rtc_use_x11=false use_custom_libcxx=false use_custom_libcxx_for_host=false rtc_build_ssl=false rtc_build_examples=false rtc_build_tools=false ios_deployment_target="9.0" ios_enable_code_signing=false is_unsafe_developer_build=false rtc_enable_protobuf=false rtc_include_builtin_video_codecs=true rtc_build_libvpx=true rtc_libvpx_build_vp9=true rtc_use_gtk=false rtc_use_metal_rendering=true rtc_ssl_root="//openssl"'
+buildtools/mac/gn gen out/$OUT_DIR --args="use_xcode_clang=true "" target_cpu=\"$ARCH\""' target_os="ios" is_debug=false is_component_build=false rtc_include_tests=false use_rtti=true rtc_use_x11=false use_custom_libcxx=false use_custom_libcxx_for_host=false rtc_build_ssl=false rtc_build_examples=false rtc_build_tools=false ios_deployment_target="9.0" ios_enable_code_signing=false is_unsafe_developer_build=false rtc_enable_protobuf=false rtc_include_builtin_video_codecs=true rtc_build_libvpx=true rtc_libvpx_build_vp9=true rtc_use_gtk=false rtc_use_metal_rendering=true'
 ninja -C out/$OUT_DIR framework_objc_static
 
 popd

third-party/webrtc/webrtc-ios

@@ -1 +1 @@
-Subproject commit 625c777c8ad260cbc4bf9a6409d5b4382d92914b
+Subproject commit cf2c8a8364b4cfda7ea9eb448671033351851130