Improve things for OpenAL devices management.

Telegram/SourceFiles/calls/calls_call.cpp

@@ -430,20 +430,20 @@ void Call::setMuted(bool mute) {
 void Call::setupMediaDevices() {
 	_playbackDeviceId.changes() | rpl::filter([=] {
 		return _instance && _setDeviceIdCallback;
-	}) | rpl::start_with_next([=](const QString &deviceId) {
-		_setDeviceIdCallback(
-			Webrtc::DeviceType::Playback,
-			deviceId);
-		_instance->setAudioOutputDevice(deviceId.toStdString());
+	}) | rpl::start_with_next([=](const Webrtc::DeviceResolvedId &deviceId) {
+		_setDeviceIdCallback(deviceId);
+
+		// Value doesn't matter here, just trigger reading of the new value.
+		_instance->setAudioOutputDevice(deviceId.value.toStdString());
 	}, _lifetime);
 
 	_captureDeviceId.changes() | rpl::filter([=] {
 		return _instance && _setDeviceIdCallback;
-	}) | rpl::start_with_next([=](const QString &deviceId) {
-		_setDeviceIdCallback(
-			Webrtc::DeviceType::Capture,
-			deviceId);
-		_instance->setAudioInputDevice(deviceId.toStdString());
+	}) | rpl::start_with_next([=](const Webrtc::DeviceResolvedId &deviceId) {
+		_setDeviceIdCallback(deviceId);
+
+		// Value doesn't matter here, just trigger reading of the new value.
+		_instance->setAudioInputDevice(deviceId.value.toStdString());
 	}, _lifetime);
 }
 
@@ -497,10 +497,11 @@ void Call::setupOutgoingVideo() {
 	_cameraDeviceId.changes(
 	) | rpl::filter([=] {
 		return !_videoCaptureIsScreencast;
-	}) | rpl::start_with_next([=](QString deviceId) {
-		_videoCaptureDeviceId = deviceId;
+	}) | rpl::start_with_next([=](Webrtc::DeviceResolvedId deviceId) {
+		const auto &id = deviceId.value;
+		_videoCaptureDeviceId = id;
 		if (_videoCapture) {
-			_videoCapture->switchToDevice(deviceId.toStdString(), false);
+			_videoCapture->switchToDevice(id.toStdString(), false);
 			if (_instance) {
 				_instance->sendVideoDeviceUpdated();
 			}
@@ -904,24 +905,25 @@ void Call::createAndStartController(const MTPDphoneCall &call) {
 	const auto playbackDeviceIdInitial = _playbackDeviceId.current();
 	const auto captureDeviceIdInitial = _captureDeviceId.current();
 	const auto saveSetDeviceIdCallback = [=](
-			Fn<void(Webrtc::DeviceType, QString)> setDeviceIdCallback) {
-		setDeviceIdCallback(
-			Webrtc::DeviceType::Playback,
-			playbackDeviceIdInitial);
-		setDeviceIdCallback(
-			Webrtc::DeviceType::Capture,
-			captureDeviceIdInitial);
+			Fn<void(Webrtc::DeviceResolvedId)> setDeviceIdCallback) {
+		setDeviceIdCallback(playbackDeviceIdInitial);
+		setDeviceIdCallback(captureDeviceIdInitial);
 		crl::on_main(weak, [=] {
 			_setDeviceIdCallback = std::move(setDeviceIdCallback);
 			const auto playback = _playbackDeviceId.current();
 			if (_instance && playback != playbackDeviceIdInitial) {
-				_setDeviceIdCallback(Webrtc::DeviceType::Playback, playback);
-				_instance->setAudioOutputDevice(playback.toStdString());
+				_setDeviceIdCallback(playback);
+
+				// Value doesn't matter here, just trigger reading of the...
+				_instance->setAudioOutputDevice(
+					playback.value.toStdString());
 			}
 			const auto capture = _captureDeviceId.current();
 			if (_instance && capture != captureDeviceIdInitial) {
-				_setDeviceIdCallback(Webrtc::DeviceType::Capture, capture);
-				_instance->setAudioInputDevice(capture.toStdString());
+				_setDeviceIdCallback(capture);
+
+				// Value doesn't matter here, just trigger reading of the...
+				_instance->setAudioInputDevice(capture.value.toStdString());
 			}
 		});
 	};
@@ -944,8 +946,8 @@ void Call::createAndStartController(const MTPDphoneCall &call) {
 			std::move(encryptionKeyValue),
 			(_type == Type::Outgoing)),
 		.mediaDevicesConfig = tgcalls::MediaDevicesConfig{
-			.audioInputId = captureDeviceIdInitial.toStdString(),
-			.audioOutputId = playbackDeviceIdInitial.toStdString(),
+			.audioInputId = captureDeviceIdInitial.value.toStdString(),
+			.audioOutputId = playbackDeviceIdInitial.value.toStdString(),
 			.inputVolume = 1.f,//settings.callInputVolume() / 100.f,
 			.outputVolume = 1.f,//settings.callOutputVolume() / 100.f,
 		},
@@ -1223,7 +1225,7 @@ void Call::toggleCameraSharing(bool enabled) {
 	}
 	_delegate->callRequestPermissionsOrFail(crl::guard(this, [=] {
 		toggleScreenSharing(std::nullopt);
-		_videoCaptureDeviceId = _cameraDeviceId.current();
+		_videoCaptureDeviceId = _cameraDeviceId.current().value;
 		if (_videoCapture) {
 			_videoCapture->switchToDevice(
 				_videoCaptureDeviceId.toStdString(),

Telegram/SourceFiles/calls/calls_call.h

@@ -12,7 +12,7 @@ https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
 #include "base/bytes.h"
 #include "mtproto/sender.h"
 #include "mtproto/mtproto_auth_key.h"
-#include "webrtc/webrtc_device_id.h"
+#include "webrtc/webrtc_device_resolver.h"
 
 namespace Media {
 namespace Audio {
@@ -271,10 +271,10 @@ private:
 	base::DelayedCallTimer _finishByTimeoutTimer;
 	base::Timer _discardByTimeoutTimer;
 
-	Fn<void(Webrtc::DeviceType, QString)> _setDeviceIdCallback;
-	Webrtc::DeviceId _playbackDeviceId;
-	Webrtc::DeviceId _captureDeviceId;
-	Webrtc::DeviceId _cameraDeviceId;
+	Fn<void(Webrtc::DeviceResolvedId)> _setDeviceIdCallback;
+	Webrtc::DeviceResolver _playbackDeviceId;
+	Webrtc::DeviceResolver _captureDeviceId;
+	Webrtc::DeviceResolver _cameraDeviceId;
 
 	rpl::variable<bool> _muted = false;
 

Telegram/SourceFiles/calls/group/calls_group_call.cpp

@@ -2066,22 +2066,26 @@ void GroupCall::applyOtherParticipantUpdate(
 void GroupCall::setupMediaDevices() {
 	_playbackDeviceId.changes() | rpl::filter([=] {
 		return _instance && _setDeviceIdCallback;
-	}) | rpl::start_with_next([=](const QString &deviceId) {
-		_setDeviceIdCallback(Webrtc::DeviceType::Playback, deviceId);
-		_instance->setAudioOutputDevice(deviceId.toStdString());
+	}) | rpl::start_with_next([=](const Webrtc::DeviceResolvedId &deviceId) {
+		_setDeviceIdCallback(deviceId);
+
+		// Value doesn't matter here, just trigger reading of the new value.
+		_instance->setAudioOutputDevice(deviceId.value.toStdString());
 	}, _lifetime);
 
 	_captureDeviceId.changes() | rpl::filter([=] {
 		return _instance && _setDeviceIdCallback;
-	}) | rpl::start_with_next([=](const QString &deviceId) {
-		_setDeviceIdCallback(Webrtc::DeviceType::Capture, deviceId);
-		_instance->setAudioInputDevice(deviceId.toStdString());
+	}) | rpl::start_with_next([=](const Webrtc::DeviceResolvedId &deviceId) {
+		_setDeviceIdCallback(deviceId);
+
+		// Value doesn't matter here, just trigger reading of the new value.
+		_instance->setAudioInputDevice(deviceId.value.toStdString());
 	}, _lifetime);
 
 	_cameraDeviceId.changes() | rpl::filter([=] {
 		return _cameraCapture != nullptr;
-	}) | rpl::start_with_next([=](const QString &deviceId) {
-		_cameraCapture->switchToDevice(deviceId.toStdString(), false);
+	}) | rpl::start_with_next([=](const Webrtc::DeviceResolvedId &deviceId) {
+		_cameraCapture->switchToDevice(deviceId.value.toStdString(), false);
 	}, _lifetime);
 }
 
@@ -2119,7 +2123,7 @@ bool GroupCall::emitShareCameraError() {
 		return emitError(Error::DisabledNoCamera);
 	} else if (mutedByAdmin()) {
 		return emitError(Error::MutedNoCamera);
-	} else if (_cameraDeviceId.current().isEmpty()) {
+	} else if (_cameraDeviceId.current().value.isEmpty()) {
 		return emitError(Error::NoCamera);
 	}
 	return false;
@@ -2128,7 +2132,7 @@ bool GroupCall::emitShareCameraError() {
 void GroupCall::emitShareCameraError(Error error) {
 	_cameraState = Webrtc::VideoState::Inactive;
 	if (error == Error::CameraFailed
-		&& _cameraDeviceId.current().isEmpty()) {
+		&& _cameraDeviceId.current().value.isEmpty()) {
 		error = Error::NoCamera;
 	}
 	_errors.fire_copy(error);
@@ -2182,7 +2186,7 @@ void GroupCall::setupOutgoingVideo() {
 				return;
 			} else if (!_cameraCapture) {
 				_cameraCapture = _delegate->groupCallGetVideoCapture(
-					_cameraDeviceId.current());
+					_cameraDeviceId.current().value);
 				if (!_cameraCapture) {
 					return emitShareCameraError(Error::CameraFailed);
 				}
@@ -2194,7 +2198,7 @@ void GroupCall::setupOutgoingVideo() {
 				});
 			} else {
 				_cameraCapture->switchToDevice(
-					_cameraDeviceId.current().toStdString(),
+					_cameraDeviceId.current().value.toStdString(),
 					false);
 			}
 			if (_instance) {
@@ -2343,24 +2347,25 @@ bool GroupCall::tryCreateController() {
 	const auto playbackDeviceIdInitial = _playbackDeviceId.current();
 	const auto captureDeviceIdInitial = _captureDeviceId.current();
 	const auto saveSetDeviceIdCallback = [=](
-			Fn<void(Webrtc::DeviceType, QString)> setDeviceIdCallback) {
-		setDeviceIdCallback(
-			Webrtc::DeviceType::Playback,
-			playbackDeviceIdInitial);
-		setDeviceIdCallback(
-			Webrtc::DeviceType::Capture,
-			captureDeviceIdInitial);
+			Fn<void(Webrtc::DeviceResolvedId)> setDeviceIdCallback) {
+		setDeviceIdCallback(playbackDeviceIdInitial);
+		setDeviceIdCallback(captureDeviceIdInitial);
 		crl::on_main(weak, [=] {
 			_setDeviceIdCallback = std::move(setDeviceIdCallback);
 			const auto playback = _playbackDeviceId.current();
 			if (_instance && playback != playbackDeviceIdInitial) {
-				_setDeviceIdCallback(Webrtc::DeviceType::Playback, playback);
-				_instance->setAudioOutputDevice(playback.toStdString());
+				_setDeviceIdCallback(playback);
+
+				// Value doesn't matter here, just trigger reading of the...
+				_instance->setAudioOutputDevice(
+					playback.value.toStdString());
 			}
 			const auto capture = _captureDeviceId.current();
 			if (_instance && capture != captureDeviceIdInitial) {
-				_setDeviceIdCallback(Webrtc::DeviceType::Capture, capture);
-				_instance->setAudioInputDevice(capture.toStdString());
+				_setDeviceIdCallback(capture);
+
+				// Value doesn't matter here, just trigger reading of the...
+				_instance->setAudioInputDevice(capture.value.toStdString());
 			}
 		});
 	};
@@ -2387,8 +2392,8 @@ bool GroupCall::tryCreateController() {
 			}
 			crl::on_main(weak, [=] { audioLevelsUpdated(data); });
 		},
-		.initialInputDeviceId = captureDeviceIdInitial.toStdString(),
-		.initialOutputDeviceId = playbackDeviceIdInitial.toStdString(),
+		.initialInputDeviceId = captureDeviceIdInitial.value.toStdString(),
+		.initialOutputDeviceId = playbackDeviceIdInitial.value.toStdString(),
 		.createAudioDeviceModule = Webrtc::AudioDeviceModuleCreator(
 			saveSetDeviceIdCallback),
 		.videoCapture = _cameraCapture,

Telegram/SourceFiles/calls/group/calls_group_call.h

@@ -12,7 +12,7 @@ https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
 #include "base/bytes.h"
 #include "mtproto/sender.h"
 #include "mtproto/mtproto_auth_key.h"
-#include "webrtc/webrtc_device_id.h"
+#include "webrtc/webrtc_device_resolver.h"
 
 class History;
 
@@ -667,10 +667,10 @@ private:
 
 	crl::time _lastSendProgressUpdate = 0;
 
-	Fn<void(Webrtc::DeviceType, QString)> _setDeviceIdCallback;
-	Webrtc::DeviceId _playbackDeviceId;
-	Webrtc::DeviceId _captureDeviceId;
-	Webrtc::DeviceId _cameraDeviceId;
+	Fn<void(Webrtc::DeviceResolvedId)> _setDeviceIdCallback;
+	Webrtc::DeviceResolver _playbackDeviceId;
+	Webrtc::DeviceResolver _captureDeviceId;
+	Webrtc::DeviceResolver _cameraDeviceId;
 
 	std::shared_ptr<GlobalShortcutManager> _shortcutManager;
 	std::shared_ptr<GlobalShortcutValue> _pushToTalk;

Telegram/SourceFiles/calls/group/calls_group_settings.cpp

@@ -42,6 +42,7 @@ https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
 #include "core/application.h"
 #include "core/core_settings.h"
 #include "webrtc/webrtc_audio_input_tester.h"
+#include "webrtc/webrtc_device_resolver.h"
 #include "settings/settings_calls.h"
 #include "main/main_session.h"
 #include "apiwrap.h"
@@ -249,7 +250,7 @@ void SettingsBox(
 	const auto weakBox = Ui::MakeWeak(box);
 
 	struct State {
-		std::unique_ptr<Webrtc::DeviceId> computedDeviceId;
+		std::unique_ptr<Webrtc::DeviceResolver> deviceId;
 		std::unique_ptr<Webrtc::AudioInputTester> micTester;
 		Ui::LevelMeter *micTestLevel = nullptr;
 		float micLevel = 0.;
@@ -770,14 +771,14 @@ void SettingsBox(
 		box->setShowFinishedCallback([=] {
 			// Means we finished showing the box.
 			crl::on_main(box, [=] {
-				state->computedDeviceId = std::make_unique<Webrtc::DeviceId>(
+				state->deviceId = std::make_unique<Webrtc::DeviceResolver>(
 					&Core::App().mediaDevices(),
 					Webrtc::DeviceType::Capture,
 					Webrtc::DeviceIdValueWithFallback(
 						Core::App().settings().callCaptureDeviceIdValue(),
 						Core::App().settings().captureDeviceIdValue()));
 				state->micTester = std::make_unique<Webrtc::AudioInputTester>(
-					state->computedDeviceId->value());
+					state->deviceId->value());
 				state->levelUpdateTimer.callEach(kMicTestUpdateInterval);
 			});
 		});
@@ -884,11 +885,13 @@ std::pair<Fn<void()>, rpl::lifetime> ShareInviteLinkAction(
 MicLevelTester::MicLevelTester(Fn<void()> show)
 : _show(std::move(show))
 , _timer([=] { check(); })
-, _tester(
-	std::make_unique<Webrtc::AudioInputTester>(
-		Webrtc::DeviceIdValueWithFallback(
-			Core::App().settings().callCaptureDeviceIdValue(),
-			Core::App().settings().captureDeviceIdValue()))) {
+, _deviceId(std::make_unique<Webrtc::DeviceResolver>(
+	&Core::App().mediaDevices(),
+	Webrtc::DeviceType::Capture,
+	Webrtc::DeviceIdValueWithFallback(
+		Core::App().settings().callCaptureDeviceIdValue(),
+		Core::App().settings().captureDeviceIdValue())))
+, _tester(std::make_unique<Webrtc::AudioInputTester>(_deviceId->value())) {
 	_timer.callEach(kMicrophoneTooltipCheckInterval);
 }
 

Telegram/SourceFiles/calls/group/calls_group_settings.h

@@ -11,6 +11,7 @@ https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
 
 namespace Webrtc {
 class AudioInputTester;
+class DeviceResolver;
 } // namespace Webrtc
 
 namespace Calls {
@@ -38,6 +39,7 @@ private:
 
 	Fn<void()> _show;
 	base::Timer _timer;
+	std::unique_ptr<Webrtc::DeviceResolver> _deviceId;
 	std::unique_ptr<Webrtc::AudioInputTester> _tester;
 	int _loudCount = 0;
 	int _quietCount = 0;

Telegram/SourceFiles/media/audio/media_audio.cpp

@@ -39,6 +39,9 @@ constexpr auto kWaveformCounterBufferSize = 256 * 1024;
 QMutex AudioMutex;
 ALCdevice *AudioDevice = nullptr;
 ALCcontext *AudioContext = nullptr;
+Webrtc::DeviceResolvedId AudioDeviceLastUsedId{
+	.type = Webrtc::DeviceType::Playback
+};
 
 auto VolumeMultiplierAll = 1.;
 auto VolumeMultiplierSong = 1.;
@@ -89,8 +92,12 @@ void DestroyPlaybackDevice() {
 bool CreatePlaybackDevice() {
 	if (AudioDevice) return true;
 
-	const auto id = Current().playbackDeviceId().toStdString();
-	AudioDevice = alcOpenDevice(id.c_str());
+	AudioDeviceLastUsedId = Current().playbackDeviceId();
+
+	const auto id = AudioDeviceLastUsedId.isDefault()
+		? std::string()
+		: AudioDeviceLastUsedId.value.toStdString();
+	AudioDevice = alcOpenDevice(id.empty() ? nullptr : id.c_str());
 	if (!AudioDevice) {
 		LOG(("Audio Error: Could not create default playback device, refreshing.."));
 		crl::on_main([] {
@@ -1380,6 +1387,20 @@ void DetachFromDevice(not_null<Audio::Instance*> instance) {
 	}
 }
 
+bool DetachIfDeviceChanged(
+		not_null<Audio::Instance*> instance,
+		const Webrtc::DeviceResolvedId &nowDeviceId) {
+	QMutexLocker lock(&AudioMutex);
+	if (AudioDeviceLastUsedId == nowDeviceId) {
+		return false;
+	}
+	Audio::ClosePlaybackDevice(instance);
+	if (mixer()) {
+		mixer()->reattachIfNeeded();
+	}
+	return true;
+}
+
 } // namespace internal
 
 } // namespace Player

Telegram/SourceFiles/media/audio/media_audio.h

@@ -30,6 +30,10 @@ struct TimePoint;
 } // namespace Streaming
 } // namespace Media
 
+namespace Webrtc {
+struct DeviceResolvedId;
+} // namespace Webrtc
+
 namespace Media {
 namespace Audio {
 
@@ -378,6 +382,9 @@ bool CheckAudioDeviceConnected();
 
 // Thread: Main. Locks: AudioMutex.
 void DetachFromDevice(not_null<Audio::Instance*> instance);
+bool DetachIfDeviceChanged(
+	not_null<Audio::Instance*> instance,
+	const Webrtc::DeviceResolvedId &nowDeviceId);
 
 // Thread: Any.
 QMutex *audioPlayerMutex();

Telegram/SourceFiles/media/audio/media_audio_capture.cpp

@@ -85,7 +85,10 @@ public:
 	Inner(QThread *thread);
 	~Inner();
 
-	void start(QString id, Fn<void(Update)> updated, Fn<void()> error);
+	void start(
+		Webrtc::DeviceResolvedId id,
+		Fn<void(Update)> updated,
+		Fn<void()> error);
 	void stop(Fn<void(Result&&)> callback = nullptr);
 	void pause(bool value, Fn<void(Result&&)> callback);
 
@@ -295,7 +298,7 @@ void Instance::Inner::fail() {
 }
 
 void Instance::Inner::start(
-		QString id,
+		Webrtc::DeviceResolvedId id,
 		Fn<void(Update)> updated,
 		Fn<void()> error) {
 	_updated = std::move(updated);
@@ -305,9 +308,9 @@ void Instance::Inner::start(
 	}
 
 	// Start OpenAL Capture
-	const auto utf = id.toStdString();
+	const auto utf = id.isDefault() ? std::string() : id.value.toStdString();
 	d->device = alcCaptureOpenDevice(
-		utf.c_str(),
+		utf.empty() ? nullptr : utf.c_str(),
 		kCaptureFrequency,
 		AL_FORMAT_MONO16,
 		kCaptureFrequency / 5);

Telegram/SourceFiles/media/audio/media_audio_track.cpp

@@ -272,17 +272,19 @@ Instance::Instance()
 		Player::internal::DetachFromDevice(this);
 	});
 
-	_playbackDeviceId.changes() | rpl::start_with_next([=] {
-		_detachFromDeviceForce = false;
-		Player::internal::DetachFromDevice(this);
+	_playbackDeviceId.changes(
+	) | rpl::start_with_next([=](Webrtc::DeviceResolvedId id) {
+		if (Player::internal::DetachIfDeviceChanged(this, id)) {
+			_detachFromDeviceForce = false;
+		}
 	}, _lifetime);
 }
 
-QString Instance::playbackDeviceId() const {
-	return _playbackDeviceId.current();
+Webrtc::DeviceResolvedId Instance::playbackDeviceId() const {
+	return _playbackDeviceId.threadSafeCurrent();
 }
 
-QString Instance::captureDeviceId() const {
+Webrtc::DeviceResolvedId Instance::captureDeviceId() const {
 	return _captureDeviceId.current();
 }
 

Telegram/SourceFiles/media/audio/media_audio_track.h

@@ -9,7 +9,7 @@ https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
 
 #include "base/timer.h"
 #include "base/bytes.h"
-#include "webrtc/webrtc_device_id.h"
+#include "webrtc/webrtc_device_resolver.h"
 
 namespace Core {
 class FileLocation;
@@ -95,8 +95,11 @@ public:
 	// Thread: Main.
 	Instance();
 
-	[[nodiscard]] QString playbackDeviceId() const;
-	[[nodiscard]] QString captureDeviceId() const;
+	// Thread: Any. Must be locked: AudioMutex.
+	[[nodiscard]] Webrtc::DeviceResolvedId playbackDeviceId() const;
+
+	// Thread: Main.
+	[[nodiscard]] Webrtc::DeviceResolvedId captureDeviceId() const;
 
 	[[nodiscard]] std::unique_ptr<Track> createTrack();
 
@@ -119,8 +122,8 @@ private:
 
 private:
 	std::set<Track*> _tracks;
-	Webrtc::DeviceId _playbackDeviceId;
-	Webrtc::DeviceId _captureDeviceId;
+	Webrtc::DeviceResolver _playbackDeviceId;
+	Webrtc::DeviceResolver _captureDeviceId;
 
 	base::Timer _updateTimer;
 

Telegram/SourceFiles/settings/settings_calls.cpp

@@ -365,7 +365,7 @@ void Calls::initCaptureButton(
 	});
 
 	struct LevelState {
-		std::unique_ptr<Webrtc::DeviceId> computedDeviceId;
+		std::unique_ptr<Webrtc::DeviceResolver> deviceId;
 		std::unique_ptr<Webrtc::AudioInputTester> tester;
 		base::Timer timer;
 		Ui::Animations::Simple animation;
@@ -388,18 +388,18 @@ void Calls::initCaptureButton(
 	});
 	_testingMicrophone.value() | rpl::start_with_next([=](bool testing) {
 		if (testing) {
-			state->computedDeviceId = std::make_unique<Webrtc::DeviceId>(
+			state->deviceId = std::make_unique<Webrtc::DeviceResolver>(
 				&Core::App().mediaDevices(),
 				Webrtc::DeviceType::Capture,
 				rpl::duplicate(resolvedId));
 			state->tester = std::make_unique<AudioInputTester>(
-				state->computedDeviceId->value());
+				state->deviceId->value());
 			state->timer.callEach(kMicTestUpdateInterval);
 		} else {
 			state->timer.cancel();
 			state->animation.stop();
 			state->tester = nullptr;
-			state->computedDeviceId = nullptr;
+			state->deviceId = nullptr;
 		}
 	}, level->lifetime());
 }

Telegram/lib_webrtc

@@ -1 +1 @@
-Subproject commit 16b8f6ee0a1b4a1852266f1b3fc727f6a82c3716
+Subproject commit 72b1aa0405e14beef0b596c9bc748eb8905a7ef8