Support three-value VideoState.

Telegram/SourceFiles/calls/calls_call.cpp

@@ -143,6 +143,11 @@ uint64 ComputeFingerprint(bytes::const_span authKey) {
 	return WrapVersions(tgcalls::Meta::Versions() | ranges::action::reverse);
 }
 
+[[nodiscard]] webrtc::VideoState StartVideoState(bool enabled) {
+	using State = webrtc::VideoState;
+	return enabled ? State::Active : State::Inactive;
+}
+
 } // namespace
 
 Call::Call(
@@ -154,8 +159,8 @@ Call::Call(
 , _user(user)
 , _api(&_user->session().mtp())
 , _type(type)
-, _videoIncoming(std::make_unique<webrtc::VideoTrack>(video))
+, _videoIncoming(std::make_unique<webrtc::VideoTrack>(StartVideoState(video)))
-, _videoOutgoing(std::make_unique<webrtc::VideoTrack>(video)) {
+, _videoOutgoing(std::make_unique<webrtc::VideoTrack>(StartVideoState(video))) {
 	_discardByTimeoutTimer.setCallback([=] { hangup(); });
 
 	if (_type == Type::Outgoing) {
@@ -338,24 +343,26 @@ void Call::setMuted(bool mute) {
 }
 
 void Call::setupOutgoingVideo() {
-	const auto started = _videoOutgoing->enabled();
+	const auto started = _videoOutgoing->state();
-	_videoOutgoing->enabledValue(
+	_videoOutgoing->stateValue(
-	) | rpl::start_with_next([=](bool enabled) {
+	) | rpl::start_with_next([=](webrtc::VideoState state) {
-		if (_state.current() != State::Established && enabled != started) {
+		if (_state.current() != State::Established
-			_videoOutgoing->setEnabled(started);
+			&& state != started
-		} else if (enabled) {
+			&& !_videoCapture) {
+			_videoOutgoing->setState(started);
+		} else if (state != webrtc::VideoState::Inactive) {
+			// Paused not supported right now.
+			Assert(state == webrtc::VideoState::Active);
 			if (!_videoCapture) {
 				_videoCapture = tgcalls::VideoCaptureInterface::Create();
-				_videoCapture->setVideoOutput(_videoOutgoing->sink());
+				_videoCapture->setOutput(_videoOutgoing->sink());
 			}
 			if (_instance) {
-				_instance->requestVideo(_videoCapture);
+				_instance->setVideoCapture(_videoCapture);
-			} else {
-				_videoState = VideoState::OutgoingRequested;
 			}
-			_videoCapture->setIsVideoEnabled(true);
+			_videoCapture->setState(tgcalls::VideoState::Active);
 		} else if (_videoCapture) {
-			_videoCapture->setIsVideoEnabled(false);
+			_videoCapture->setState(tgcalls::VideoState::Inactive);
 		}
 	}, _lifetime);
 }
@@ -568,6 +575,30 @@ bool Call::handleUpdate(const MTPPhoneCall &call) {
 	Unexpected("phoneCall type inside an existing call handleUpdate()");
 }
 
+void Call::updateRemoteMediaState(
+		tgcalls::AudioState audio,
+		tgcalls::VideoState video) {
+	_remoteAudioState = [&] {
+		using From = tgcalls::AudioState;
+		using To = RemoteAudioState;
+		switch (audio) {
+		case From::Active: return To::Active;
+		case From::Muted: return To::Muted;
+		}
+		Unexpected("Audio state in remoteMediaStateUpdated.");
+	}();
+	_videoIncoming->setState([&] {
+		using From = tgcalls::VideoState;
+		using To = webrtc::VideoState;
+		switch (video) {
+		case From::Inactive: return To::Inactive;
+		case From::Paused: return To::Paused;
+		case From::Active: return To::Active;
+		}
+		Unexpected("Video state in remoteMediaStateUpdated.");
+	}());
+}
+
 bool Call::handleSignalingData(
 		const MTPDupdatePhoneCallSignalingData &data) {
 	if (data.vphone_call_id().v != _id || !_instance) {
@@ -685,10 +716,10 @@ void Call::createAndStartController(const MTPDphoneCall &call) {
 		.encryptionKey = tgcalls::EncryptionKey(
 			std::move(encryptionKeyValue),
 			(_type == Type::Outgoing)),
-		.videoCapture = _videoOutgoing->enabled() ? _videoCapture : nullptr,
+		.videoCapture = _videoCapture,
-		.stateUpdated = [=](tgcalls::State state, tgcalls::VideoState videoState) {
+		.stateUpdated = [=](tgcalls::State state) {
 			crl::on_main(weak, [=] {
-				handleControllerStateChange(state, videoState);
+				handleControllerStateChange(state);
 			});
 		},
 		.signalBarsUpdated = [=](int count) {
@@ -696,9 +727,9 @@ void Call::createAndStartController(const MTPDphoneCall &call) {
 				handleControllerBarCountChange(count);
 			});
 		},
-		.remoteVideoIsActiveUpdated = [=](bool active) {
+		.remoteMediaStateUpdated = [=](tgcalls::AudioState audio, tgcalls::VideoState video) {
 			crl::on_main(weak, [=] {
-				_videoIncoming->setEnabled(active);
+				updateRemoteMediaState(audio, video);
 			});
 		},
 		.signalingDataEmitted = [=](const std::vector<uint8_t> &data) {
@@ -780,22 +811,7 @@ void Call::createAndStartController(const MTPDphoneCall &call) {
 	raw->setAudioOutputDuckingEnabled(settings.callAudioDuckingEnabled());
 }
 
-void Call::handleControllerStateChange(
+void Call::handleControllerStateChange(tgcalls::State state) {
-		tgcalls::State state,
-		tgcalls::VideoState videoState) {
-	_videoState = [&] {
-		switch (videoState) {
-		case tgcalls::VideoState::Possible: return VideoState::Disabled;
-		case tgcalls::VideoState::OutgoingRequested:
-			return VideoState::OutgoingRequested;
-		case tgcalls::VideoState::IncomingRequested:
-		case tgcalls::VideoState::IncomingRequestedAndActive:
-			return VideoState::IncomingRequested;
-		case tgcalls::VideoState::Active: return VideoState::Enabled;
-		}
-		Unexpected("VideoState value in Call::handleControllerStateChange.");
-	}();
-
 	switch (state) {
 	case tgcalls::State::WaitInit: {
 		DEBUG_LOG(("Call Info: State changed to WaitingInit."));
@@ -976,7 +992,8 @@ void Call::finish(FinishType type, const MTPPhoneCallDiscardReason &reason) {
 	auto duration = getDurationMs() / 1000;
 	auto connectionId = _instance ? _instance->getPreferredRelayId() : 0;
 	_finishByTimeoutTimer.call(kHangupTimeoutMs, [this, finalState] { setState(finalState); });
-	const auto flags = (_videoState.current() == VideoState::Enabled)
+	const auto flags = ((_videoIncoming->state() != webrtc::VideoState::Inactive)
+		|| (_videoOutgoing->state() != webrtc::VideoState::Inactive))
 		? MTPphone_DiscardCall::Flag::f_video
 		: MTPphone_DiscardCall::Flag(0);
 	_api.request(MTPphone_DiscardCall(

Telegram/SourceFiles/calls/calls_call.h

@@ -24,9 +24,11 @@ class Instance;
 class VideoCaptureInterface;
 enum class State;
 enum class VideoState;
+enum class AudioState;
 } // namespace tgcalls
 
 namespace webrtc {
+enum class VideoState;
 class VideoTrack;
 } // namespace webrtc
 
@@ -103,17 +105,24 @@ public:
 		return _state.value();
 	}
 
-	enum class VideoState {
+	enum class RemoteAudioState {
-		Disabled,
+		Muted,
-		OutgoingRequested,
+		Active,
-		IncomingRequested,
-		Enabled
 	};
-	[[nodiscard]] VideoState videoState() const {
+	[[nodiscard]] RemoteAudioState remoteAudioState() const {
-		return _videoState.current();
+		return _remoteAudioState.current();
 	}
-	[[nodiscard]] rpl::producer<VideoState> videoStateValue() const {
+	[[nodiscard]] auto remoteAudioStateValue() const
-		return _videoState.value();
+	-> rpl::producer<RemoteAudioState> {
+		return _remoteAudioState.value();
+	}
+
+	[[nodiscard]] webrtc::VideoState remoteVideoState() const {
+		return _remoteVideoState.current();
+	}
+	[[nodiscard]] auto remoteVideoStateValue() const
+	-> rpl::producer<webrtc::VideoState> {
+		return _remoteVideoState.value();
 	}
 
 	static constexpr auto kSignalBarStarting = -1;
@@ -164,14 +173,17 @@ private:
 	};
 	void handleRequestError(const RPCError &error);
 	void handleControllerError(const QString &error);
-	void finish(FinishType type, const MTPPhoneCallDiscardReason &reason = MTP_phoneCallDiscardReasonDisconnect());
+	void finish(
+		FinishType type,
+		const MTPPhoneCallDiscardReason &reason
+			= MTP_phoneCallDiscardReasonDisconnect());
 	void startOutgoing();
 	void startIncoming();
 	void startWaitingTrack();
 	void sendSignalingData(const QByteArray &data);
 
 	void generateModExpFirst(bytes::const_span randomSeed);
-	void handleControllerStateChange(tgcalls::State state, tgcalls::VideoState videoState);
+	void handleControllerStateChange(tgcalls::State state);
 	void handleControllerBarCountChange(int count);
 	void createAndStartController(const MTPDphoneCall &call);
 
@@ -190,13 +202,17 @@ private:
 	void destroyController();
 
 	void setupOutgoingVideo();
+	void updateRemoteMediaState(
+		tgcalls::AudioState audio,
+		tgcalls::VideoState video);
 
 	const not_null<Delegate*> _delegate;
 	const not_null<UserData*> _user;
 	MTP::Sender _api;
 	Type _type = Type::Outgoing;
 	rpl::variable<State> _state = State::Starting;
-	rpl::variable<VideoState> _videoState = VideoState::Disabled;
+	rpl::variable<RemoteAudioState> _remoteAudioState = RemoteAudioState::Active;
+	rpl::variable<webrtc::VideoState> _remoteVideoState;
 	FinishType _finishAfterRequestingCall = FinishType::None;
 	bool _answerAfterDhConfigReceived = false;
 	rpl::variable<int> _signalBarCount = kSignalBarStarting;

Telegram/SourceFiles/calls/calls_panel.cpp

@@ -361,8 +361,10 @@ void Panel::initControls() {
 	});
 	_camera->setClickedCallback([=] {
 		if (_call) {
-			_call->videoOutgoing()->setEnabled(
+			_call->videoOutgoing()->setState(
-				!_call->videoOutgoing()->enabled());
+				(_call->videoOutgoing()->state() == webrtc::VideoState::Active)
+				? webrtc::VideoState::Inactive
+				: webrtc::VideoState::Active);
 		}
 	});
 
@@ -430,9 +432,11 @@ void Panel::reinitWithCall(Call *call) {
 		_mute->setIconOverride(mute ? &st::callUnmuteIcon : nullptr);
 	}, _callLifetime);
 
-	_call->videoOutgoing()->enabledValue(
+	_call->videoOutgoing()->stateValue(
-	) | rpl::start_with_next([=](bool enabled) {
+	) | rpl::start_with_next([=](webrtc::VideoState state) {
-		_camera->setIconOverride(enabled ? nullptr : &st::callNoCameraIcon);
+		_camera->setIconOverride((state == webrtc::VideoState::Active)
+			? nullptr
+			: &st::callNoCameraIcon);
 	}, _callLifetime);
 
 	_call->stateValue(

Telegram/ThirdParty/tgcalls

@@ -1 +1 @@
-Subproject commit c3ba7d2068c0658468c8e924b6f421beface7ccb
+Subproject commit 13a261d8ed3717e1a2a324b9e13a4e351af8fe83

Telegram/lib_webrtc

@@ -1 +1 @@
-Subproject commit 503e551331f45cbccc29cad0bc158f11c85169d3
+Subproject commit 4d13b96b4c4e4be2ada7e460203eea9fecde458d