Add test implementation of webrtc calls.

Telegram/CMakeLists.txt

@@ -318,6 +318,8 @@ PRIVATE
     calls/calls_controller.cpp
     calls/calls_controller.h
     calls/calls_controller_tgvoip.h
+    calls/calls_controller_webrtc.cpp
+    calls/calls_controller_webrtc.h
     calls/calls_emoji_fingerprint.cpp
     calls/calls_emoji_fingerprint.h
     calls/calls_instance.cpp

Telegram/Patches/build_ffmpeg_win.sh

@@ -13,6 +13,8 @@ pacman --noconfirm -S pkg-config
 PKG_CONFIG_PATH="/mingw64/lib/pkgconfig:$PKG_CONFIG_PATH"
 
 ./configure --toolchain=msvc \
+--extra-cflags="-DCONFIG_SAFE_BITSTREAM_READER=1" \
+--extra-cxxflags="-DCONFIG_SAFE_BITSTREAM_READER=1" \
 --extra-ldflags="-libpath:$FullExecPath/../opus/win32/VS2015/Win32/Release" \
 --disable-programs \
 --disable-doc \

Telegram/SourceFiles/api/api_updates.cpp

@@ -1726,8 +1726,9 @@ void Updates::feedUpdate(const MTPUpdate &update) {
 		auto &d = update.c_updateEncryptedMessagesRead();
 	} break;
 
-	case mtpc_updatePhoneCall: {
+	case mtpc_updatePhoneCall:
-		Core::App().calls().handleUpdate(&session(), update.c_updatePhoneCall());
+	case mtpc_updatePhoneCallSignalingData: {
+		Core::App().calls().handleUpdate(&session(), update);
 	} break;
 
 	case mtpc_updateUserBlocked: {

Telegram/SourceFiles/calls/calls_call.cpp

@@ -32,6 +32,7 @@ namespace {
 constexpr auto kMinLayer = 65;
 constexpr auto kHangupTimeoutMs = 5000;
 constexpr auto kSha256Size = 32;
+const auto kDefaultVersion = "2.4.4"_q;
 
 void AppendEndpoint(
 		std::vector<TgVoipEndpoint> &list,
@@ -71,10 +72,6 @@ uint64 ComputeFingerprint(bytes::const_span authKey) {
 		| (gsl::to_integer<uint64>(hash[12]));
 }
 
-[[nodiscard]] std::vector<std::string> CollectVersions() {
-	return { TgVoip::getVersion() };
-}
-
 [[nodiscard]] QVector<MTPstring> WrapVersions(
 		const std::vector<std::string> &data) {
 	auto result = QVector<MTPstring>();
@@ -86,13 +83,11 @@ uint64 ComputeFingerprint(bytes::const_span authKey) {
 }
 
 [[nodiscard]] QVector<MTPstring> CollectVersionsForApi() {
-	return WrapVersions(CollectVersions());
+	return WrapVersions(CollectControllerVersions());
 }
 
 } // namespace
 
-Call::Delegate::~Delegate() = default;
-
 Call::Call(
 	not_null<Delegate*> delegate,
 	not_null<UserData*> user,
@@ -170,7 +165,7 @@ void Call::startOutgoing() {
 			MTP_flags(MTPDphoneCallProtocol::Flag::f_udp_p2p
 				| MTPDphoneCallProtocol::Flag::f_udp_reflector),
 			MTP_int(kMinLayer),
-			MTP_int(TgVoip::getConnectionMaxLayer()),
+			MTP_int(ControllerMaxLayer()),
 			MTP_vector(CollectVersionsForApi()))
 	)).done([=](const MTPphone_PhoneCall &result) {
 		Expects(result.type() == mtpc_phone_phoneCall);
@@ -251,7 +246,7 @@ void Call::actuallyAnswer() {
 			MTP_flags(MTPDphoneCallProtocol::Flag::f_udp_p2p
 				| MTPDphoneCallProtocol::Flag::f_udp_reflector),
 			MTP_int(kMinLayer),
-			MTP_int(TgVoip::getConnectionMaxLayer()),
+			MTP_int(ControllerMaxLayer()),
 			MTP_vector(CollectVersionsForApi()))
 	)).done([=](const MTPphone_PhoneCall &result) {
 		Expects(result.type() == mtpc_phone_phoneCall);
@@ -322,6 +317,25 @@ void Call::startWaitingTrack() {
 	_waitingTrack->playInLoop();
 }
 
+void Call::sendSignalingData(const QByteArray &data) {
+	_api.request(MTPphone_SendSignalingData(
+		MTP_inputPhoneCall(
+			MTP_long(_id),
+			MTP_long(_accessHash)),
+		MTP_bytes(data)
+	)).done([=](const MTPBool &result) {
+		if (!mtpIsTrue(result)) {
+			finish(FinishType::Failed);
+		}
+	}).fail([=](const RPCError &error) {
+		handleRequestError(error);
+	}).send();
+}
+
+void Call::displayNextFrame(QImage frame) {
+	_frames.fire(std::move(frame));
+}
+
 float64 Call::getWaitingSoundPeakValue() const {
 	if (_waitingTrack) {
 		auto when = crl::now() + kSoundSampleMs / 4;
@@ -462,6 +476,14 @@ bool Call::handleUpdate(const MTPPhoneCall &call) {
 	Unexpected("phoneCall type inside an existing call handleUpdate()");
 }
 
+bool Call::handleSignalingData(
+		const MTPDupdatePhoneCallSignalingData &data) {
+	if (data.vphone_call_id().v != _id || !_controller) {
+		return false;
+	}
+	return _controller->receiveSignalingData(data.vdata().v);
+}
+
 void Call::confirmAcceptedCall(const MTPDphoneCallAccepted &call) {
 	Expects(_type == Type::Outgoing);
 
@@ -494,9 +516,9 @@ void Call::confirmAcceptedCall(const MTPDphoneCallAccepted &call) {
 			MTP_flags(MTPDphoneCallProtocol::Flag::f_udp_p2p
 				| MTPDphoneCallProtocol::Flag::f_udp_reflector),
 			MTP_int(kMinLayer),
-			MTP_int(TgVoip::getConnectionMaxLayer()),
+			MTP_int(ControllerMaxLayer()),
 			MTP_vector(CollectVersionsForApi()))
-	)).done([this](const MTPphone_PhoneCall &result) {
+	)).done([=](const MTPphone_PhoneCall &result) {
 		Expects(result.type() == mtpc_phone_phoneCall);
 
 		auto &call = result.c_phone_phoneCall();
@@ -508,7 +530,7 @@ void Call::confirmAcceptedCall(const MTPDphoneCallAccepted &call) {
 		}
 
 		createAndStartController(call.vphone_call().c_phoneCall());
-	}).fail([this](const RPCError &error) {
+	}).fail([=](const RPCError &error) {
 		handleRequestError(error);
 	}).send();
 }
@@ -597,14 +619,20 @@ void Call::createAndStartController(const MTPDphoneCall &call) {
 		return static_cast<uint8_t>(byte);
 	}) | ranges::to_vector;
 
+	const auto version = call.vprotocol().match([&](
+			const MTPDphoneCallProtocol &data) {
+		return data.vlibrary_versions().v;
+	}).value(0, MTP_bytes(kDefaultVersion)).v;
 	_controller = MakeController(
-		"2.4.4",
+		version.toStdString(),
 		config,
 		TgVoipPersistentState(),
 		endpoints,
 		proxy.host.empty() ? nullptr : &proxy,
 		TgVoipNetworkType::Unknown,
-		encryptionKey);
+		encryptionKey,
+		[=](QByteArray data) { sendSignalingData(data); },
+		[=](QImage frame) { displayNextFrame(frame); });
 
 	const auto raw = _controller.get();
 	raw->setOnStateUpdated([=](TgVoipState state) {

Telegram/SourceFiles/calls/calls_call.h

@@ -48,7 +48,7 @@ public:
 		virtual void playSound(Sound sound) = 0;
 		virtual void requestMicrophonePermissionOrFail(Fn<void()> result) = 0;
 
-		virtual ~Delegate();
+		virtual ~Delegate() = default;
 
 	};
 
@@ -70,6 +70,7 @@ public:
 
 	void start(bytes::const_span random);
 	bool handleUpdate(const MTPPhoneCall &call);
+	bool handleSignalingData(const MTPDupdatePhoneCallSignalingData &data);
 
 	enum State {
 		Starting,
@@ -110,6 +111,10 @@ public:
 		return _muteChanged;
 	}
 
+	rpl::producer<QImage> frames() const {
+		return _frames.events();
+	}
+
 	crl::time getDurationMs() const;
 	float64 getWaitingSoundPeakValue() const;
 
@@ -144,6 +149,8 @@ private:
 	void startOutgoing();
 	void startIncoming();
 	void startWaitingTrack();
+	void sendSignalingData(const QByteArray &data);
+	void displayNextFrame(QImage frame);
 
 	void generateModExpFirst(bytes::const_span randomSeed);
 	void handleControllerStateChange(
@@ -181,6 +188,7 @@ private:
 
 	bool _mute = false;
 	base::Observable<bool> _muteChanged;
+	rpl::event_stream<QImage> _frames;
 
 	DhConfig _dhConfig;
 	bytes::vector _ga;

Telegram/SourceFiles/calls/calls_controller.cpp

@@ -8,6 +8,7 @@ https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
 #include "calls/calls_controller.h"
 
 #include "calls/calls_controller_tgvoip.h"
+#include "calls/calls_controller_webrtc.h"
 
 namespace Calls {
 
@@ -18,7 +19,20 @@ namespace Calls {
 		const std::vector<TgVoipEndpoint> &endpoints,
 		const TgVoipProxy *proxy,
 		TgVoipNetworkType initialNetworkType,
-		const TgVoipEncryptionKey &encryptionKey) {
+		const TgVoipEncryptionKey &encryptionKey,
+		Fn<void(QByteArray)> sendSignalingData,
+		Fn<void(QImage)> displayNextFrame) {
+	if (version == WebrtcController::Version()) {
+		return std::make_unique<WebrtcController>(
+			config,
+			persistentState,
+			endpoints,
+			proxy,
+			initialNetworkType,
+			encryptionKey,
+			std::move(sendSignalingData),
+			std::move(displayNextFrame));
+	}
 	return std::make_unique<TgVoipController>(
 		config,
 		persistentState,
@@ -28,4 +42,12 @@ namespace Calls {
 		encryptionKey);
 }
 
+std::vector<std::string> CollectControllerVersions() {
+	return { WebrtcController::Version(), TgVoipController::Version() };
+}
+
+int ControllerMaxLayer() {
+	return TgVoip::getConnectionMaxLayer();
+}
+
 } // namespace Calls

Telegram/SourceFiles/calls/calls_controller.h

@@ -26,6 +26,7 @@ public:
 	virtual void setInputVolume(float level) = 0;
 	virtual void setOutputVolume(float level) = 0;
 	virtual void setAudioOutputDuckingEnabled(bool enabled) = 0;
+	virtual bool receiveSignalingData(const QByteArray &data) = 0;
 
 	virtual std::string getLastError() = 0;
 	virtual std::string getDebugInfo() = 0;
@@ -48,6 +49,11 @@ public:
 	const std::vector<TgVoipEndpoint> &endpoints,
 	const TgVoipProxy *proxy,
 	TgVoipNetworkType initialNetworkType,
-	const TgVoipEncryptionKey &encryptionKey);
+	const TgVoipEncryptionKey &encryptionKey,
+	Fn<void(QByteArray)> sendSignalingData,
+	Fn<void(QImage)> displayNextFrame);
+
+[[nodiscard]] std::vector<std::string> CollectControllerVersions();
+[[nodiscard]] int ControllerMaxLayer();
 
 } // namespace Calls

Telegram/SourceFiles/calls/calls_controller_tgvoip.h

@@ -33,7 +33,7 @@ public:
 		return TgVoip::getVersion();
 	}
 
-	[[nodiscard]] std::string version() override {
+	std::string version() override {
 		return Version();
 	}
 	void setNetworkType(TgVoipNetworkType networkType) override {
@@ -63,6 +63,9 @@ public:
 	void setAudioOutputDuckingEnabled(bool enabled) override {
 		_impl->setAudioOutputDuckingEnabled(enabled);
 	}
+	bool receiveSignalingData(const QByteArray &data) override {
+		return false;
+	}
 	std::string getLastError() override {
 		return _impl->getLastError();
 	}
@@ -81,8 +84,7 @@ public:
 	void setOnStateUpdated(Fn<void(TgVoipState)> onStateUpdated) override {
 		_impl->setOnStateUpdated(std::move(onStateUpdated));
 	}
-	void setOnSignalBarsUpdated(
+	void setOnSignalBarsUpdated(Fn<void(int)> onSignalBarsUpdated) override {
-			Fn<void(int)> onSignalBarsUpdated) override {
 		_impl->setOnSignalBarsUpdated(std::move(onSignalBarsUpdated));
 	}
 	TgVoipFinalState stop() override {

Telegram/SourceFiles/calls/calls_controller_webrtc.cpp

@@ -0,0 +1,175 @@
+/*
+This file is part of Telegram Desktop,
+the official desktop application for the Telegram messaging service.
+
+For license and copyright information please follow this link:
+https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
+*/
+#include "calls/calls_controller_webrtc.h"
+
+#include "webrtc/webrtc_call_context.h"
+
+namespace Calls {
+namespace {
+
+using namespace Webrtc;
+
+[[nodiscard]] CallConnectionDescription ConvertEndpoint(const TgVoipEndpoint &data) {
+	return CallConnectionDescription{
+		.ip = QString::fromStdString(data.host.ipv4),
+		.ipv6 = QString::fromStdString(data.host.ipv6),
+		.peerTag = QByteArray(
+			reinterpret_cast<const char*>(data.peerTag),
+			base::array_size(data.peerTag)),
+		.connectionId = data.endpointId,
+		.port = data.port,
+	};
+}
+
+[[nodiscard]] CallContext::Config MakeContextConfig(
+		const TgVoipConfig &config,
+		const TgVoipPersistentState &persistentState,
+		const std::vector<TgVoipEndpoint> &endpoints,
+		const TgVoipProxy *proxy,
+		TgVoipNetworkType initialNetworkType,
+		const TgVoipEncryptionKey &encryptionKey,
+		Fn<void(QByteArray)> sendSignalingData,
+		Fn<void(QImage)> displayNextFrame) {
+	Expects(!endpoints.empty());
+
+	auto result = CallContext::Config{
+		.proxy = (proxy
+			? ProxyServer{
+				.host = QString::fromStdString(proxy->host),
+				.username = QString::fromStdString(proxy->login),
+				.password = QString::fromStdString(proxy->password),
+				.port = proxy->port }
+			: ProxyServer()),
+		.dataSaving = (config.dataSaving != TgVoipDataSaving::Never),
+		.key = QByteArray(
+			reinterpret_cast<const char*>(encryptionKey.value.data()),
+			encryptionKey.value.size()),
+		.outgoing = encryptionKey.isOutgoing,
+		.primary = ConvertEndpoint(endpoints.front()),
+		.alternatives = endpoints | ranges::view::drop(
+			1
+		) | ranges::view::transform(ConvertEndpoint) | ranges::to_vector,
+		.maxLayer = config.maxApiLayer,
+		.allowP2P = config.enableP2P,
+		.sendSignalingData = std::move(sendSignalingData),
+		.displayNextFrame = std::move(displayNextFrame),
+	};
+	return result;
+}
+
+} // namespace
+
+WebrtcController::WebrtcController(
+	const TgVoipConfig &config,
+	const TgVoipPersistentState &persistentState,
+	const std::vector<TgVoipEndpoint> &endpoints,
+	const TgVoipProxy *proxy,
+	TgVoipNetworkType initialNetworkType,
+	const TgVoipEncryptionKey &encryptionKey,
+	Fn<void(QByteArray)> sendSignalingData,
+	Fn<void(QImage)> displayNextFrame)
+: _impl(std::make_unique<CallContext>(MakeContextConfig(
+		config,
+		persistentState,
+		endpoints,
+		proxy,
+		initialNetworkType,
+		encryptionKey,
+		std::move(sendSignalingData),
+		std::move(displayNextFrame)))) {
+}
+
+WebrtcController::~WebrtcController() = default;
+
+std::string WebrtcController::Version() {
+	return CallContext::Version().toStdString();
+}
+
+std::string WebrtcController::version() {
+	return Version();
+}
+
+void WebrtcController::setNetworkType(TgVoipNetworkType networkType) {
+}
+
+void WebrtcController::setMuteMicrophone(bool muteMicrophone) {
+	_impl->setIsMuted(muteMicrophone);
+}
+
+void WebrtcController::setAudioOutputGainControlEnabled(bool enabled) {
+}
+
+void WebrtcController::setEchoCancellationStrength(int strength) {
+}
+
+void WebrtcController::setAudioInputDevice(std::string id) {
+}
+
+void WebrtcController::setAudioOutputDevice(std::string id) {
+}
+
+void WebrtcController::setInputVolume(float level) {
+}
+
+void WebrtcController::setOutputVolume(float level) {
+}
+
+void WebrtcController::setAudioOutputDuckingEnabled(bool enabled) {
+}
+
+bool WebrtcController::receiveSignalingData(const QByteArray &data) {
+	return _impl->receiveSignalingData(data);
+}
+
+std::string WebrtcController::getLastError() {
+	return {};
+}
+
+std::string WebrtcController::getDebugInfo() {
+	return _impl->getDebugInfo().toStdString();
+}
+
+int64_t WebrtcController::getPreferredRelayId() {
+	return 0;
+}
+
+TgVoipTrafficStats WebrtcController::getTrafficStats() {
+	return {};
+}
+
+TgVoipPersistentState WebrtcController::getPersistentState() {
+	return TgVoipPersistentState{};
+}
+
+void WebrtcController::setOnStateUpdated(
+		Fn<void(TgVoipState)> onStateUpdated) {
+	_stateUpdatedLifetime.destroy();
+	_impl->state().changes(
+	) | rpl::start_with_next([=](CallState state) {
+		onStateUpdated([&] {
+			switch (state) {
+			case CallState::Initializing: return TgVoipState::WaitInit;
+			case CallState::Reconnecting: return TgVoipState::Reconnecting;
+			case CallState::Connected: return TgVoipState::Established;
+			case CallState::Failed: return TgVoipState::Failed;
+			}
+			Unexpected("State value in Webrtc::CallContext::state.");
+		}());
+	}, _stateUpdatedLifetime);
+}
+
+void WebrtcController::setOnSignalBarsUpdated(
+	Fn<void(int)> onSignalBarsUpdated) {
+}
+
+TgVoipFinalState WebrtcController::stop() {
+	_impl->stop();
+	return TgVoipFinalState();
+}
+
+} // namespace Calls

Telegram/SourceFiles/calls/calls_controller_webrtc.h

@@ -0,0 +1,60 @@
+/*
+This file is part of Telegram Desktop,
+the official desktop application for the Telegram messaging service.
+
+For license and copyright information please follow this link:
+https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
+*/
+#pragma once
+
+#include "calls/calls_controller.h"
+
+namespace Webrtc {
+class CallContext;
+} // namespace Webrtc
+
+namespace Calls {
+
+class WebrtcController final : public Controller {
+public:
+	WebrtcController(
+		const TgVoipConfig &config,
+		const TgVoipPersistentState &persistentState,
+		const std::vector<TgVoipEndpoint> &endpoints,
+		const TgVoipProxy *proxy,
+		TgVoipNetworkType initialNetworkType,
+		const TgVoipEncryptionKey &encryptionKey,
+		Fn<void(QByteArray)> sendSignalingData,
+		Fn<void(QImage)> displayNextFrame);
+	~WebrtcController();
+
+	[[nodiscard]] static std::string Version();
+
+	std::string version() override;
+	void setNetworkType(TgVoipNetworkType networkType) override;
+	void setMuteMicrophone(bool muteMicrophone) override;
+	void setAudioOutputGainControlEnabled(bool enabled) override;
+	void setEchoCancellationStrength(int strength) override;
+	void setAudioInputDevice(std::string id) override;
+	void setAudioOutputDevice(std::string id) override;
+	void setInputVolume(float level) override;
+	void setOutputVolume(float level) override;
+	void setAudioOutputDuckingEnabled(bool enabled) override;
+	bool receiveSignalingData(const QByteArray &data) override;
+	std::string getLastError() override;
+	std::string getDebugInfo() override;
+	int64_t getPreferredRelayId() override;
+	TgVoipTrafficStats getTrafficStats() override;
+	TgVoipPersistentState getPersistentState() override;
+	void setOnStateUpdated(Fn<void(TgVoipState)> onStateUpdated) override;
+	void setOnSignalBarsUpdated(Fn<void(int)> onSignalBarsUpdated) override;
+	TgVoipFinalState stop() override;
+
+private:
+	const std::unique_ptr<Webrtc::CallContext> _impl;
+
+	rpl::lifetime _stateUpdatedLifetime;
+
+};
+
+} // namespace Calls

Telegram/SourceFiles/calls/calls_instance.cpp

@@ -232,13 +232,19 @@ void Instance::refreshServerConfig(not_null<Main::Session*> session) {
 		UpdateConfig(std::string(json.data(), json.size()));
 	}).fail([=](const RPCError &error) {
 		_serverConfigRequestSession = nullptr;
-		}).send();
+	}).send();
 }
 
 void Instance::handleUpdate(
 		not_null<Main::Session*> session,
-		const MTPDupdatePhoneCall& update) {
+		const MTPUpdate &update) {
-	handleCallUpdate(session, update.vphone_call());
+	update.match([&](const MTPDupdatePhoneCall &data) {
+		handleCallUpdate(session, data.vphone_call());
+	}, [&](const MTPDupdatePhoneCallSignalingData &data) {
+		handleSignalingData(data);
+	}, [](const auto &) {
+		Unexpected("Update type in Calls::Instance::handleUpdate.");
+	});
 }
 
 void Instance::showInfoPanel(not_null<Call*> call) {
@@ -291,6 +297,14 @@ void Instance::handleCallUpdate(
 	}
 }
 
+void Instance::handleSignalingData(
+		const MTPDupdatePhoneCallSignalingData &data) {
+	if (!_currentCall || !_currentCall->handleSignalingData(data)) {
+		DEBUG_LOG(("API Warning: unexpected call signaling data %1"
+			).arg(data.vphone_call_id().v));
+	}
+}
+
 bool Instance::alreadyInCall() {
 	return (_currentCall && _currentCall->state() != Call::State::Busy);
 }

Telegram/SourceFiles/calls/calls_instance.h

@@ -35,7 +35,7 @@ public:
 	void startOutgoingCall(not_null<UserData*> user);
 	void handleUpdate(
 		not_null<Main::Session*> session,
-		const MTPDupdatePhoneCall &update);
+		const MTPUpdate &update);
 	void showInfoPanel(not_null<Call*> call);
 	[[nodiscard]] Call *currentCall() const;
 	[[nodiscard]] rpl::producer<Call*> currentCallValue() const;
@@ -59,6 +59,8 @@ private:
 	void destroyCurrentPanel();
 	void requestMicrophonePermissionOrFail(Fn<void()> onSuccess) override;
 
+	void handleSignalingData(const MTPDupdatePhoneCallSignalingData &data);
+
 	void refreshDhConfig();
 	void refreshServerConfig(not_null<Main::Session*> session);
 	bytes::const_span updateDhConfig(const MTPmessages_DhConfig &data);

Telegram/SourceFiles/calls/calls_panel.cpp

@@ -400,6 +400,11 @@ void Panel::initControls() {
 
 	_decline->finishAnimating();
 	_cancel->finishAnimating();
+
+	_call->frames() | rpl::start_with_next([=](QImage frame) {
+		_videoFrame = std::move(frame);
+		update();
+	}, lifetime());
 }
 
 void Panel::reinitControls() {
@@ -742,6 +747,17 @@ void Panel::paintEvent(QPaintEvent *e) {
 		p.fillRect(0, _contentTop, width(), height() - _contentTop, brush);
 	}
 
+	if (!_videoFrame.isNull()) {
+		const auto to = rect().marginsRemoved(_padding);
+		p.save();
+		p.setClipRect(to);
+		const auto big = _videoFrame.size().scaled(to.size(), Qt::KeepAspectRatioByExpanding);
+		const auto pos = QPoint((to.width() - big.width()) / 2, (to.height() - big.height()) / 2);
+		auto hq = PainterHighQualityEnabler(p);
+		p.drawImage(QRect(pos, big), _videoFrame);
+		p.restore();
+	}
+
 	if (_signalBars->isDisplayed()) {
 		paintSignalBarsBg(p);
 	}

Telegram/SourceFiles/calls/calls_panel.h

@@ -158,6 +158,8 @@ private:
 	QPixmap _bottomCache;
 	QPixmap _cache;
 
+	QImage _videoFrame;
+
 };
 
 } // namespace Calls

Telegram/lib_webrtc

@@ -1 +1 @@
-Subproject commit 2b1cd3b5b0c909834aea00e0590435a63d318468
+Subproject commit eb5be18405d47a226f6a197280176f82b3e903bd