Temp

submodules/TelegramVoip/BUILD

@@ -14,6 +14,7 @@ swift_library(
         "//submodules/TelegramUIPreferences:TelegramUIPreferences",
         "//submodules/TgVoip:TgVoip",
         "//submodules/TgVoipWebrtcCustom:TgVoipWebrtcCustom",
+        "//submodules/TgVoipWebrtc:TgVoipWebrtc",
     ],
     visibility = [
         "//visibility:public",

submodules/TelegramVoip/Sources/OngoingCallContext.swift

@@ -7,16 +7,16 @@ import Postbox
 import TelegramUIPreferences
 
 import TgVoip
-//import TgVoipWebrtc
+import TgVoipWebrtc
 import TgVoipWebrtcCustom
 
 private func callConnectionDescription(_ connection: CallSessionConnection) -> OngoingCallConnectionDescription {
     return OngoingCallConnectionDescription(connectionId: connection.id, ip: connection.ip, ipv6: connection.ipv6, port: connection.port, peerTag: connection.peerTag)
 }
 
-/*private func callConnectionDescriptionWebrtc(_ connection: CallSessionConnection) -> OngoingCallConnectionDescriptionWebrtc {
+private func callConnectionDescriptionWebrtc(_ connection: CallSessionConnection) -> OngoingCallConnectionDescriptionWebrtc {
     return OngoingCallConnectionDescriptionWebrtc(connectionId: connection.id, ip: connection.ip, ipv6: connection.ipv6, port: connection.port, peerTag: connection.peerTag)
-}*/
+}
 
 private func callConnectionDescriptionWebrtcCustom(_ connection: CallSessionConnection) -> OngoingCallConnectionDescriptionWebrtcCustom {
     return OngoingCallConnectionDescriptionWebrtcCustom(connectionId: connection.id, ip: connection.ip, ipv6: connection.ipv6, port: connection.port, peerTag: connection.peerTag)
@@ -80,11 +80,11 @@ private let setupLogs: Bool = {
             Logger.shared.log("TGVOIP", value)
         }
     })
-    /*OngoingCallThreadLocalContextWebrtc.setupLoggingFunction({ value in
+    OngoingCallThreadLocalContextWebrtc.setupLoggingFunction({ value in
         if let value = value {
             Logger.shared.log("TGVOIP", value)
         }
-    })*/
+    })
     OngoingCallThreadLocalContextWebrtcCustom.setupLoggingFunction({ value in
         if let value = value {
             Logger.shared.log("TGVOIP", value)
@@ -100,7 +100,7 @@ public enum OngoingCallContextState {
     case failed
 }
 
-private final class OngoingCallThreadLocalContextQueueImpl: NSObject, OngoingCallThreadLocalContextQueue/*, OngoingCallThreadLocalContextQueueWebrtc*/ , OngoingCallThreadLocalContextQueueWebrtcCustom {
+private final class OngoingCallThreadLocalContextQueueImpl: NSObject, OngoingCallThreadLocalContextQueue, OngoingCallThreadLocalContextQueueWebrtc , OngoingCallThreadLocalContextQueueWebrtcCustom {
     private let queue: Queue
     
     init(queue: Queue) {
@@ -144,7 +144,7 @@ private func ongoingNetworkTypeForType(_ type: NetworkType) -> OngoingCallNetwor
     }
 }
 
-/*private func ongoingNetworkTypeForTypeWebrtc(_ type: NetworkType) -> OngoingCallNetworkTypeWebrtc {
+private func ongoingNetworkTypeForTypeWebrtc(_ type: NetworkType) -> OngoingCallNetworkTypeWebrtc {
     switch type {
         case .none:
             return .wifi
@@ -162,7 +162,7 @@ private func ongoingNetworkTypeForType(_ type: NetworkType) -> OngoingCallNetwor
                     return .cellularLte
             }
     }
-}*/
+}
 
 private func ongoingNetworkTypeForTypeWebrtcCustom(_ type: NetworkType) -> OngoingCallNetworkTypeWebrtcCustom {
     switch type {
@@ -197,7 +197,7 @@ private func ongoingDataSavingForType(_ type: VoiceCallDataSaving) -> OngoingCal
     }
 }
 
-/*private func ongoingDataSavingForTypeWebrtc(_ type: VoiceCallDataSaving) -> OngoingCallDataSavingWebrtc {
+private func ongoingDataSavingForTypeWebrtc(_ type: VoiceCallDataSaving) -> OngoingCallDataSavingWebrtc {
     switch type {
         case .never:
             return .never
@@ -208,7 +208,7 @@ private func ongoingDataSavingForType(_ type: VoiceCallDataSaving) -> OngoingCal
         default:
             return .never
     }
-}*/
+}
 
 private func ongoingDataSavingForTypeWebrtcCustom(_ type: VoiceCallDataSaving) -> OngoingCallDataSavingWebrtcCustom {
     switch type {
@@ -266,7 +266,7 @@ extension OngoingCallThreadLocalContext: OngoingCallThreadLocalContextProtocol {
     }
 }
 
-/*extension OngoingCallThreadLocalContextWebrtc: OngoingCallThreadLocalContextProtocol {
+extension OngoingCallThreadLocalContextWebrtc: OngoingCallThreadLocalContextProtocol {
     func nativeSetNetworkType(_ type: NetworkType) {
         self.setNetworkType(ongoingNetworkTypeForTypeWebrtc(type))
     }
@@ -290,7 +290,7 @@ extension OngoingCallThreadLocalContext: OngoingCallThreadLocalContextProtocol {
     func nativeGetDerivedState() -> Data {
         return self.getDerivedState()
     }
-}*/
+}
 
 extension OngoingCallThreadLocalContextWebrtcCustom: OngoingCallThreadLocalContextProtocol {
     func nativeSetNetworkType(_ type: NetworkType) {
@@ -335,7 +335,7 @@ private extension OngoingCallContextState {
     }
 }
 
-/*private extension OngoingCallContextState {
+private extension OngoingCallContextState {
     init(_ state: OngoingCallStateWebrtc) {
         switch state {
         case .initializing:
@@ -350,7 +350,7 @@ private extension OngoingCallContextState {
             self = .failed
         }
     }
-}*/
+}
 
 private extension OngoingCallContextState {
     init(_ state: OngoingCallStateWebrtcCustom) {
@@ -401,7 +401,8 @@ public final class OngoingCallContext {
     public static func versions(includeExperimental: Bool) -> [String] {
         var result: [String] = [OngoingCallThreadLocalContext.version()]
         if includeExperimental {
-            result.append(OngoingCallThreadLocalContextWebrtcCustom.version())
+            result.append(OngoingCallThreadLocalContextWebrtc.version())
+            //result.append(OngoingCallThreadLocalContextWebrtcCustom.version())
         }
         return result
     }
@@ -453,7 +454,7 @@ public final class OngoingCallContext {
                             context.nativeSetNetworkType(networkType)
                         }
                     })
-                }/* else if version == OngoingCallThreadLocalContextWebrtc.version() {
+                } else if version == OngoingCallThreadLocalContextWebrtc.version() {
                     var voipProxyServer: VoipProxyServerWebrtc?
                     if let proxyServer = proxyServer {
                         switch proxyServer.connection {
@@ -479,7 +480,7 @@ public final class OngoingCallContext {
                             context.nativeSetNetworkType(networkType)
                         }
                     })
-                }*/ else {
+                } else {
                     var voipProxyServer: VoipProxyServer?
                     if let proxyServer = proxyServer {
                         switch proxyServer.connection {

submodules/TgVoip/BUILD

@@ -41,6 +41,23 @@ replace_symbols = [
     "WebRtcAgc_CalculateGainTable",
     "WebRtcAgc_InitVad",
     "WebRtcAgc_ProcessVad",
+    "TimeDiff",
+    "TimeAfter",
+    "TimeMicros",
+    "TimeUTCMicros",
+    "SystemTimeNanos",
+    "TimeNanos",
+    "SystemTimeMillis",
+    "TimeMillis",
+    "TimeUTCMillis",
+    "GetClockForTesting",
+    "TimestampWrapAroundHandler",
+    "Time32",
+    "TmToSeconds",
+    "TimeDiff32",
+    "TimestampWrapAroundHandler",
+    "g_clock",
+    "SetClockForTesting",
 ]
 
 objc_library(

submodules/TgVoipWebrtc/BUILD

@@ -25,6 +25,7 @@ objc_library(
         "-DWEBRTC_IOS",
         "-DWEBRTC_MAC",
         "-DWEBRTC_POSIX",
+        "-std=c++14",
     ],
     includes = [
         "PublicHeaders",

submodules/TgVoipWebrtc/Impl/Controller.cpp

@@ -3,7 +3,7 @@
 #include "Layer92.h"
 
 #include "modules/rtp_rtcp/source/rtp_utility.h"
-#include "rtc_base/time_utils.cc"
+#include "rtc_base/time_utils.h"
 #include "rtc_base/message_handler.h"
 
 #include <memory>
@@ -89,13 +89,7 @@ void Controller::NewMessage(const message::Base& msg) {
         state = State::Established;
         SignalNewState(state);
         thread->PostTask(RTC_FROM_HERE, [this]() {
-#ifdef TGVOIP_PREPROCESSED_OUTPUT
-            preproc = std::make_unique<MediaEngineWebrtc>(not is_outgoing, false, true);
-            preproc->Play.connect(this, &Controller::Preprocessed);
-#endif
             media = std::make_unique<MediaEngineWebrtc>(is_outgoing);
-            media->Record.connect(this, &Controller::Record);
-            media->Play.connect(this, &Controller::Play);
             media->Send.connect(this, &Controller::SendRtp);
         });
         StartRepeating([this]() {
@@ -144,9 +138,6 @@ void Controller::StopRepeating() {
 void Controller::SetFail() {
     thread->PostTask(RTC_FROM_HERE, [this]() {
         media = nullptr;
-#ifdef TGVOIP_PREPROCESSED_OUTPUT
-        preproc = nullptr;
-#endif
     });
     if (state != State::Failed) {
         state = State::Failed;
@@ -155,29 +146,7 @@ void Controller::SetFail() {
     StopRepeating();
 }
 
-void Controller::Play(const int16_t *data, size_t size) {
-    SignalPlay(data, size);
-}
-
-void Controller::Record(int16_t *data, size_t size) {
-    SignalRecord(data, size);
-    last_send_time = rtc::TimeMillis();
-}
-
-#ifdef TGVOIP_PREPROCESSED_OUTPUT
-void Controller::Preprocessed(const int16_t *data, size_t size) {
-    if (rtc::TimeMillis() - last_send_time < 100)
-        SignalPreprocessed(data, size);
-}
-#endif
-
 void Controller::SendRtp(rtc::CopyOnWriteBuffer packet) {
-#ifdef TGVOIP_PREPROCESSED_OUTPUT
-    thread->PostTask(RTC_FROM_HERE, [this, packet]() {
-        if (preproc)
-            preproc->Receive(packet);
-    });
-#endif
     message::RtpStream msg;
     msg.data = packet;
     msg.network_type = local_network_type;

submodules/TgVoipWebrtc/Impl/Controller.h

@@ -41,20 +41,12 @@ public:
     static std::map<message::NetworkType, MediaEngineWebrtc::NetworkParams> network_params;
     static MediaEngineWebrtc::NetworkParams default_network_params;
     static MediaEngineWebrtc::NetworkParams datasaving_network_params;
-    sigslot::signal2<int16_t *, size_t> SignalRecord;
-#ifdef TGVOIP_PREPROCESSED_OUTPUT
-    sigslot::signal2<const int16_t *, size_t> SignalPreprocessed;
-#endif
-    sigslot::signal2<const int16_t *, size_t> SignalPlay;
     sigslot::signal1<State> SignalNewState;
 
 private:
     std::unique_ptr<rtc::Thread> thread;
     std::unique_ptr<Connector> connector;
     std::unique_ptr<MediaEngineWebrtc> media;
-#ifdef TGVOIP_PREPROCESSED_OUTPUT
-    std::unique_ptr<MediaEngineWebrtc> preproc;
-#endif
     State state;
     webrtc::RepeatingTaskHandle repeatable;
     int64_t last_recv_time;
@@ -73,9 +65,6 @@ private:
     void SetFail();
     void Play(const int16_t *data, size_t size);
     void Record(int16_t *data, size_t size);
-#ifdef TGVOIP_PREPROCESSED_OUTPUT
-    void Preprocessed(const int16_t *data, size_t size);
-#endif
     void SendRtp(rtc::CopyOnWriteBuffer packet);
     void UpdateNetworkParams(const message::RtpStream& rtp);
 };

submodules/TgVoipWebrtc/Impl/MediaEngineBase.h

@@ -14,8 +14,6 @@ public:
 
     sigslot::signal1<rtc::CopyOnWriteBuffer> Send;
     virtual void Receive(rtc::CopyOnWriteBuffer) = 0;
-    sigslot::signal2<const int16_t *, size_t> Play;
-    sigslot::signal2<int16_t *, size_t> Record;
 };
 
 #endif //DEMO_MEDIAENGINEBASE_H

submodules/TgVoipWebrtc/Impl/MediaEngineWebrtc.cpp

@@ -12,6 +12,7 @@
 #include "modules/audio_device/include/audio_device_default.h"
 #include "rtc_base/task_utils/repeating_task.h"
 #include "system_wrappers/include/field_trial.h"
+#include "api/video/builtin_video_bitrate_allocator_factory.h"
 
 #if WEBRTC_ENABLE_PROTOBUF
 #include "modules/audio_coding/audio_network_adaptor/config.pb.h"
@@ -28,12 +29,16 @@ const uint8_t sdp_channels = 2;
 const uint32_t sdp_bitrate = 0;
 const uint32_t caller_ssrc = 1;
 const uint32_t called_ssrc = 2;
+const uint32_t caller_ssrc_video = 1;
+const uint32_t called_ssrc_video = 2;
 const int extension_sequence = 1;
 }
 
 MediaEngineWebrtc::MediaEngineWebrtc(bool outgoing, bool send, bool recv)
 : ssrc_send(outgoing ? caller_ssrc : called_ssrc)
 , ssrc_recv(outgoing ? called_ssrc : caller_ssrc)
+, ssrc_send_video(outgoing ? caller_ssrc_video : called_ssrc_video)
+, ssrc_recv_video(outgoing ? called_ssrc_video : caller_ssrc_video)
 , event_log(std::make_unique<webrtc::RtcEventLogNull>())
 , task_queue_factory(webrtc::CreateDefaultTaskQueueFactory())
 , data_sender(*this) {
@@ -41,17 +46,10 @@ MediaEngineWebrtc::MediaEngineWebrtc(bool outgoing, bool send, bool recv)
             "WebRTC-Audio-SendSideBwe/Enabled/"
             "WebRTC-Audio-Allocation/min:6kbps,max:32kbps/"
             "WebRTC-Audio-OpusMinPacketLossRate/Enabled-1/"
-//            "WebRTC-Audio-OpusPlcUsePrevDecodedSamples/Enabled/"
-//            "WebRTC-Audio-NewOpusPacketLossRateOptimization/Enabled-1-20-1.0/"
-//            "WebRTC-SendSideBwe-WithOverhead/Enabled/"
-//            "WebRTC-Bwe-SeparateAudioPackets/enabled:true,packet_threshold:15,time_threshold:1000ms/"
-//            "WebRTC-Audio-AlrProbing/Disabled/"
     );
+    video_bitrate_allocator_factory = webrtc::CreateBuiltinVideoBitrateAllocatorFactory();
     cricket::MediaEngineDependencies media_deps;
     media_deps.task_queue_factory = task_queue_factory.get();
-#ifdef TGVOIP_USE_CALLBACK_AUDIO_IO
-    media_deps.adm = new rtc::RefCountedObject<webrtc::webrtc_impl::AudioDeviceModuleDefault<webrtc::AudioDeviceModule>>();
-#endif
     media_deps.audio_encoder_factory = webrtc::CreateAudioEncoderFactory<webrtc::AudioEncoderOpus>();
     media_deps.audio_decoder_factory = webrtc::CreateAudioDecoderFactory<webrtc::AudioDecoderOpus>();
     media_deps.audio_processing = webrtc::AudioProcessingBuilder().Create();
@@ -62,12 +60,9 @@ MediaEngineWebrtc::MediaEngineWebrtc(bool outgoing, bool send, bool recv)
     call_config.trials = &field_trials;
     call_config.audio_state = media_engine->voice().GetAudioState();
     call.reset(webrtc::Call::Create(call_config));
-#ifdef TGVOIP_USE_CALLBACK_AUDIO_IO
-    audio_processor = std::make_unique<AudioProcessor>(call_config.audio_state->audio_transport(),
-            task_queue_factory.get(), *this, send, recv);
-#endif
     voice_channel.reset(media_engine->voice().CreateMediaChannel(
             call.get(), cricket::MediaConfig(), cricket::AudioOptions(), webrtc::CryptoOptions::NoGcm()));
+    video_channel.reset(media_engine->video().CreateMediaChannel(call.get(), cricket::MediaConfig(), cricket::VideoOptions(), webrtc::CryptoOptions::NoGcm(), video_bitrate_allocator_factory.get()));
     if (send) {
         voice_channel->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc_send));
         SetNetworkParams({6, 32, 6, 120, false, false, false});
@@ -76,6 +71,12 @@ MediaEngineWebrtc::MediaEngineWebrtc(bool outgoing, bool send, bool recv)
         voice_channel->OnReadyToSend(true);
         voice_channel->SetSend(true);
     }
+    if (false && send) {
+        video_channel->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc_send_video));
+        video_channel->SetInterface(&data_sender, webrtc::MediaTransportConfig());
+        video_channel->OnReadyToSend(true);
+        video_channel->SetSend(true);
+    }
     if (recv) {
         cricket::AudioRecvParameters recv_parameters;
         recv_parameters.codecs.emplace_back(sdp_payload, sdp_name, clockrate, sdp_bitrate, sdp_channels);
@@ -86,6 +87,16 @@ MediaEngineWebrtc::MediaEngineWebrtc(bool outgoing, bool send, bool recv)
         voice_channel->SetRecvParameters(recv_parameters);
         voice_channel->SetPlayout(true);
     }
+    if (false && recv) {
+        cricket::VideoRecvParameters recv_parameters;
+        //recv_parameters.codecs.emplace_back(sdp_payload, sdp_name, clockrate, sdp_bitrate, sdp_channels);
+        recv_parameters.extensions.emplace_back(webrtc::RtpExtension::kTransportSequenceNumberUri, extension_sequence);
+        recv_parameters.rtcp.reduced_size = true;
+        recv_parameters.rtcp.remote_estimate = true;
+        video_channel->AddRecvStream(cricket::StreamParams::CreateLegacy(ssrc_recv_video));
+        video_channel->SetRecvParameters(recv_parameters);
+        //video_channel->SetPlayout(true);
+    }
 }
 
 MediaEngineWebrtc::~MediaEngineWebrtc() = default;
@@ -107,19 +118,8 @@ void MediaEngineWebrtc::SetNetworkParams(const MediaEngineWebrtc::NetworkParams&
     opus_codec.SetParam(cricket::kCodecParamMaxBitrate, params.max_bitrate_kbps);
     opus_codec.SetParam(cricket::kCodecParamUseInbandFec, 1);
     opus_codec.SetParam(cricket::kCodecParamPTime, params.ptime_ms);
-//    opus_codec.SetParam(cricket::kCodecParamUseDtx, "1");
+
-//    opus_codec.SetParam(cricket::kCodecParamMaxAverageBitrate, 6);
-    std::string config_string;
-#if WEBRTC_ENABLE_PROTOBUF
-    webrtc::audio_network_adaptor::config::ControllerManager cont_conf;
-//    cont_conf.add_controllers()->mutable_bitrate_controller();
-    config_string = cont_conf.SerializeAsString();
-#endif
     cricket::AudioSendParameters send_parameters;
-    if (!config_string.empty()) {
-        send_parameters.options.audio_network_adaptor_config = config_string;
-        send_parameters.options.audio_network_adaptor = true;
-    }
     send_parameters.codecs.push_back(opus_codec);
     send_parameters.extensions.emplace_back(webrtc::RtpExtension::kTransportSequenceNumberUri, extension_sequence);
     send_parameters.options.echo_cancellation = params.echo_cancellation;
@@ -157,48 +157,3 @@ int MediaEngineWebrtc::Sender::SetOption(cricket::MediaChannel::NetworkInterface
 }
 
 MediaEngineWebrtc::Sender::Sender(MediaEngineWebrtc& engine) : engine(engine) {}
-
-MediaEngineWebrtc::AudioProcessor::AudioProcessor(webrtc::AudioTransport *transport_,
-        webrtc::TaskQueueFactory *task_queue_factory, MediaEngineBase& engine_, bool send_, bool recv_)
-        : send(send_)
-        , recv(recv_)
-        , transport(transport_)
-        , delay_us(frame_samples * 1000000 / clockrate)
-        , buf_send(nullptr)
-        , buf_recv(nullptr)
-        , engine(engine_)
-        , task_queue_send(std::make_unique<rtc::TaskQueue>(task_queue_factory->CreateTaskQueue(
-                "AudioProcessorSend", webrtc::TaskQueueFactory::Priority::NORMAL)))
-        , task_queue_recv(std::make_unique<rtc::TaskQueue>(task_queue_factory->CreateTaskQueue(
-                "AudioProcessorRecv", webrtc::TaskQueueFactory::Priority::NORMAL))) {
-    if (send) {
-        buf_send = new int16_t[frame_samples * channels];
-        webrtc::RepeatingTaskHandle::Start(task_queue_send->Get(), [this]() {
-            static uint32_t new_mic_level = 0;
-            memset(buf_send, 0, frame_samples * channels * sample_bytes);
-            engine.Record(buf_send, frame_samples * channels);
-            transport->RecordedDataIsAvailable(buf_send, frame_samples, sample_bytes, channels, clockrate,
-                                               0, 0, 0, false, new_mic_level);
-            return webrtc::TimeDelta::us(delay_us);
-        });
-    }
-    if (recv) {
-        buf_recv = new int16_t[frame_samples * channels];
-        webrtc::RepeatingTaskHandle::Start(task_queue_recv->Get(), [this]() {
-            static int64_t elapsed_time_ms = -1;
-            static int64_t ntp_time_ms = -1;
-            size_t samples_out = 0;
-            transport->NeedMorePlayData(frame_samples, sample_bytes, channels, clockrate, buf_recv,
-                                        samples_out, &elapsed_time_ms, &ntp_time_ms);
-            engine.Play(buf_recv, samples_out * channels);
-            return webrtc::TimeDelta::us(delay_us);
-        });
-    }
-}
-
-MediaEngineWebrtc::AudioProcessor::~AudioProcessor() {
-    task_queue_send = nullptr;
-    task_queue_recv = nullptr;
-    delete[] buf_send;
-    delete[] buf_recv;
-}

submodules/TgVoipWebrtc/Impl/MediaEngineWebrtc.h

@@ -61,6 +61,8 @@ private:
 
     const uint32_t ssrc_send;
     const uint32_t ssrc_recv;
+    const uint32_t ssrc_send_video;
+    const uint32_t ssrc_recv_video;
     std::unique_ptr<webrtc::Call> call;
     std::unique_ptr<cricket::MediaEngineInterface> media_engine;
     std::unique_ptr<webrtc::RtcEventLogNull> event_log;
@@ -69,9 +71,8 @@ private:
     webrtc::LocalAudioSinkAdapter audio_source;
     Sender data_sender;
     std::unique_ptr<cricket::VoiceMediaChannel> voice_channel;
-#ifdef TGVOIP_USE_CALLBACK_AUDIO_IO
+    std::unique_ptr<cricket::VideoMediaChannel> video_channel;
-    std::unique_ptr<AudioProcessor> audio_processor;
+    std::unique_ptr<webrtc::VideoBitrateAllocatorFactory> video_bitrate_allocator_factory;
-#endif
 };