Add ability to choose calls audio backend.

Telegram/SourceFiles/calls/calls_call.cpp

@@ -780,7 +780,8 @@ void Call::createAndStartController(const MTPDphoneCall &call) {
 				sendSignalingData(bytes);
 			});
 		},
-		.createAudioDeviceModule = Webrtc::AudioDeviceModuleCreator(),
+		.createAudioDeviceModule = Webrtc::AudioDeviceModuleCreator(
+			settings.callAudioBackend()),
 	};
 	if (Logs::DebugEnabled()) {
 		auto callLogFolder = cWorkingDir() + qsl("DebugLogs");

Telegram/SourceFiles/calls/calls_group_call.cpp

@@ -582,7 +582,8 @@ void GroupCall::createAndStartController() {
 		},
 		.initialInputDeviceId = _audioInputId.toStdString(),
 		.initialOutputDeviceId = _audioOutputId.toStdString(),
-		.createAudioDeviceModule = Webrtc::AudioDeviceModuleCreator(),
+		.createAudioDeviceModule = Webrtc::AudioDeviceModuleCreator(
+			settings.callAudioBackend()),
 	};
 	if (Logs::DebugEnabled()) {
 		auto callLogFolder = cWorkingDir() + qsl("DebugLogs");

Telegram/SourceFiles/calls/calls_group_settings.cpp

@@ -480,6 +480,7 @@ void GroupCallSettingsBox(
 		// Means we finished showing the box.
 		crl::on_main(box, [=] {
 			state->micTester = std::make_unique<Webrtc::AudioInputTester>(
+				Core::App().settings().callAudioBackend(),
 				Core::App().settings().callInputDeviceId());
 			state->levelUpdateTimer.callEach(kMicTestUpdateInterval);
 		});

Telegram/SourceFiles/core/core_settings.cpp

@@ -13,12 +13,14 @@ https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
 #include "window/themes/window_theme.h"
 #include "window/section_widget.h"
 #include "base/platform/base_platform_info.h"
+#include "webrtc/webrtc_create_adm.h"
 #include "facades.h"
 
 namespace Core {
 
 Settings::Settings()
-: _sendSubmitWay(Ui::InputSubmitSettings::Enter)
+: _callAudioBackend(Webrtc::Backend::OpenAL)
+, _sendSubmitWay(Ui::InputSubmitSettings::Enter)
 , _floatPlayerColumn(Window::Column::Second)
 , _floatPlayerCorner(RectPart::TopRight)
 , _dialogsWidthRatio(DefaultDialogsWidthRatio()) {
@@ -112,7 +114,8 @@ QByteArray Settings::serialize() const {
 			<< qint32(_ipRevealWarning ? 1 : 0)
 			<< qint32(_groupCallPushToTalk ? 1 : 0)
 			<< _groupCallPushToTalkShortcut
-			<< qint64(_groupCallPushToTalkDelay);
+			<< qint64(_groupCallPushToTalkDelay)
+			<< qint32(_callAudioBackend);
 	}
 	return result;
 }
@@ -183,6 +186,7 @@ void Settings::addFromSerialized(const QByteArray &serialized) {
 	qint32 groupCallPushToTalk = _groupCallPushToTalk ? 1 : 0;
 	QByteArray groupCallPushToTalkShortcut = _groupCallPushToTalkShortcut;
 	qint64 groupCallPushToTalkDelay = _groupCallPushToTalkDelay;
+	qint32 callAudioBackend = static_cast<qint32>(_callAudioBackend);
 
 	stream >> themesAccentColors;
 	if (!stream.atEnd()) {
@@ -275,6 +279,9 @@ void Settings::addFromSerialized(const QByteArray &serialized) {
 			>> groupCallPushToTalkShortcut
 			>> groupCallPushToTalkDelay;
 	}
+	if (!stream.atEnd()) {
+		stream >> callAudioBackend;
+	}
 	if (stream.status() != QDataStream::Ok) {
 		LOG(("App Error: "
 			"Bad data for Core::Settings::constructFromSerialized()"));
@@ -369,6 +376,12 @@ void Settings::addFromSerialized(const QByteArray &serialized) {
 	_groupCallPushToTalk = (groupCallPushToTalk == 1);
 	_groupCallPushToTalkShortcut = groupCallPushToTalkShortcut;
 	_groupCallPushToTalkDelay = groupCallPushToTalkDelay;
+	auto uncheckedBackend = static_cast<Webrtc::Backend>(callAudioBackend);
+	switch (uncheckedBackend) {
+	case Webrtc::Backend::OpenAL:
+	case Webrtc::Backend::ADM:
+	case Webrtc::Backend::ADM2: _callAudioBackend = uncheckedBackend; break;
+	}
 }
 
 bool Settings::chatWide() const {

Telegram/SourceFiles/core/core_settings.h

@@ -21,6 +21,10 @@ namespace Window {
 enum class Column;
 } // namespace Window
 
+namespace Webrtc {
+enum class Backend;
+} // namespace Webrtc
+
 namespace Core {
 
 class Settings final {
@@ -217,6 +221,12 @@ public:
 	void setCallAudioDuckingEnabled(bool value) {
 		_callAudioDuckingEnabled = value;
 	}
+	[[nodiscard]] Webrtc::Backend callAudioBackend() const {
+		return _callAudioBackend;
+	}
+	void setCallAudioBackend(Webrtc::Backend backend) {
+		_callAudioBackend = backend;
+	}
 	[[nodiscard]] bool groupCallPushToTalk() const {
 		return _groupCallPushToTalk;
 	}
@@ -531,13 +541,14 @@ private:
 	int _callOutputVolume = 100;
 	int _callInputVolume = 100;
 	bool _callAudioDuckingEnabled = true;
+	Webrtc::Backend _callAudioBackend = Webrtc::Backend();
 	bool _groupCallPushToTalk = false;
 	QByteArray _groupCallPushToTalkShortcut;
 	crl::time _groupCallPushToTalkDelay = 20;
 	Window::Theme::AccentColors _themesAccentColors;
 	bool _lastSeenWarningSeen = false;
-	Ui::SendFilesWay _sendFilesWay;
-	Ui::InputSubmitSettings _sendSubmitWay;
+	Ui::SendFilesWay _sendFilesWay = Ui::SendFilesWay();
+	Ui::InputSubmitSettings _sendSubmitWay = Ui::InputSubmitSettings();
 	base::flat_map<QString, QString> _soundOverrides;
 	bool _exeLaunchWarning = true;
 	bool _ipRevealWarning = true;
@@ -553,8 +564,8 @@ private:
 	rpl::variable<bool> _autoDownloadDictionaries = true;
 	rpl::variable<bool> _mainMenuAccountsShown = true;
 	bool _tabbedSelectorSectionEnabled = false; // per-window
-	Window::Column _floatPlayerColumn; // per-window
-	RectPart _floatPlayerCorner; // per-window
+	Window::Column _floatPlayerColumn = Window::Column(); // per-window
+	RectPart _floatPlayerCorner = RectPart(); // per-window
 	bool _thirdSectionInfoEnabled = true; // per-window
 	rpl::event_stream<bool> _thirdSectionInfoEnabledValue; // per-window
 	int _thirdSectionExtendedBy = -1; // per-window

Telegram/SourceFiles/settings/settings_calls.cpp

@@ -30,11 +30,18 @@ https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
 #include "webrtc/webrtc_media_devices.h"
 #include "webrtc/webrtc_video_track.h"
 #include "webrtc/webrtc_audio_input_tester.h"
+#include "webrtc/webrtc_create_adm.h" // Webrtc::Backend.
 #include "tgcalls/VideoCaptureInterface.h"
 #include "facades.h"
+#include "app.h" // App::restart().
 #include "styles/style_layers.h"
 
 namespace Settings {
+namespace {
+
+using namespace Webrtc;
+
+} // namespace
 
 Calls::Calls(
 	QWidget *parent,
@@ -58,7 +65,7 @@ void Calls::setupContent() {
 	const auto content = Ui::CreateChild<Ui::VerticalLayout>(this);
 
 	const auto &settings = Core::App().settings();
-	const auto cameras = Webrtc::GetVideoInputList();
+	const auto cameras = GetVideoInputList();
 	if (!cameras.empty()) {
 		const auto hasCall = (Core::App().calls().currentCall() != nullptr);
 
@@ -66,16 +73,16 @@ void Calls::setupContent() {
 		const auto capturer = capturerOwner.get();
 		content->lifetime().add([owner = std::move(capturerOwner)]{});
 
-		const auto track = content->lifetime().make_state<Webrtc::VideoTrack>(
+		const auto track = content->lifetime().make_state<VideoTrack>(
 			(hasCall
-				? Webrtc::VideoState::Inactive
-				: Webrtc::VideoState::Active));
+				? VideoState::Inactive
+				: VideoState::Active));
 
 		const auto currentCameraName = [&] {
 			const auto i = ranges::find(
 				cameras,
 				settings.callVideoInputDeviceId(),
-				&Webrtc::VideoInput::id);
+				&VideoInput::id);
 			return (i != end(cameras))
 				? i->name
 				: tr::lng_settings_call_device_default(tr::now);
@@ -93,15 +100,15 @@ void Calls::setupContent() {
 			),
 			st::settingsButton
 		)->addClickHandler([=] {
-			const auto &devices = Webrtc::GetVideoInputList();
+			const auto &devices = GetVideoInputList();
 			const auto options = ranges::view::concat(
 				ranges::view::single(tr::lng_settings_call_device_default(tr::now)),
-				devices | ranges::view::transform(&Webrtc::VideoInput::name)
+				devices | ranges::view::transform(&VideoInput::name)
 			) | ranges::to_vector;
 			const auto i = ranges::find(
 				devices,
 				Core::App().settings().callVideoInputDeviceId(),
-				&Webrtc::VideoInput::id);
+				&VideoInput::id);
 			const auto currentOption = (i != end(devices))
 				? int(i - begin(devices) + 1)
 				: 0;
@@ -159,11 +166,11 @@ void Calls::setupContent() {
 		Core::App().calls().currentCallValue(
 		) | rpl::start_with_next([=](::Calls::Call *value) {
 			if (value) {
-				track->setState(Webrtc::VideoState::Inactive);
+				track->setState(VideoState::Inactive);
 				bubbleWrap->resize(bubbleWrap->width(), 0);
 			} else {
 				capturer->setPreferredAspectRatio(0.);
-				track->setState(Webrtc::VideoState::Active);
+				track->setState(VideoState::Active);
 				capturer->setOutput(track->sink());
 			}
 		}, content->lifetime());
@@ -252,6 +259,22 @@ void Calls::setupContent() {
 //	}, content->lifetime());
 //#endif // Q_OS_MAC && !OS_MAC_STORE
 
+	const auto backend = [&]() -> QString {
+		using namespace Webrtc;
+		switch (settings.callAudioBackend()) {
+		case Backend::OpenAL: return "OpenAL";
+		case Backend::ADM: return "WebRTC ADM";
+		case Backend::ADM2: return "WebRTC ADM2";
+		}
+		Unexpected("Value in backend.");
+	}();
+	AddButton(
+		content,
+		rpl::single("Call audio backend: " + backend),
+		st::settingsButton
+	)->addClickHandler([] {
+		Ui::show(ChooseAudioBackendBox());
+	});
 	AddButton(
 		content,
 		tr::lng_settings_call_open_system_prefs(),
@@ -300,27 +323,30 @@ void Calls::requestPermissionAndStartTestingMicrophone() {
 
 void Calls::startTestingMicrophone() {
 	_levelUpdateTimer.callEach(kMicTestUpdateInterval);
-	_micTester = std::make_unique<Webrtc::AudioInputTester>(
+	_micTester = std::make_unique<AudioInputTester>(
+		Core::App().settings().callAudioBackend(),
 		Core::App().settings().callInputDeviceId());
 }
 
 QString CurrentAudioOutputName() {
-	const auto list = Webrtc::GetAudioOutputList();
+	const auto &settings = Core::App().settings();
+	const auto list = GetAudioOutputList(settings.callAudioBackend());
 	const auto i = ranges::find(
 		list,
-		Core::App().settings().callOutputDeviceId(),
-		&Webrtc::AudioOutput::id);
+		settings.callOutputDeviceId(),
+		&AudioOutput::id);
 	return (i != end(list))
 		? i->name
 		: tr::lng_settings_call_device_default(tr::now);
 }
 
 QString CurrentAudioInputName() {
-	const auto list = Webrtc::GetAudioInputList();
+	const auto &settings = Core::App().settings();
+	const auto list = GetAudioInputList(settings.callAudioBackend());
 	const auto i = ranges::find(
 		list,
-		Core::App().settings().callInputDeviceId(),
-		&Webrtc::AudioInput::id);
+		settings.callInputDeviceId(),
+		&AudioInput::id);
 	return (i != end(list))
 		? i->name
 		: tr::lng_settings_call_device_default(tr::now);
@@ -330,21 +356,22 @@ object_ptr<SingleChoiceBox> ChooseAudioOutputBox(
 		Fn<void(QString id, QString name)> chosen,
 		const style::Checkbox *st,
 		const style::Radio *radioSt) {
-	const auto &devices = Webrtc::GetAudioOutputList();
+	const auto &settings = Core::App().settings();
+	const auto list = GetAudioOutputList(settings.callAudioBackend());
 	const auto options = ranges::view::concat(
 		ranges::view::single(tr::lng_settings_call_device_default(tr::now)),
-		devices | ranges::view::transform(&Webrtc::AudioOutput::name)
+		list | ranges::view::transform(&AudioOutput::name)
 	) | ranges::to_vector;
 	const auto i = ranges::find(
-		devices,
-		Core::App().settings().callOutputDeviceId(),
-		&Webrtc::AudioOutput::id);
-	const auto currentOption = (i != end(devices))
-		? int(i - begin(devices) + 1)
+		list,
+		settings.callOutputDeviceId(),
+		&AudioOutput::id);
+	const auto currentOption = (i != end(list))
+		? int(i - begin(list) + 1)
 		: 0;
 	const auto save = [=](int option) {
 		const auto deviceId = option
-			? devices[option - 1].id
+			? list[option - 1].id
 			: "default";
 		Core::App().calls().setCurrentAudioDevice(false, deviceId);
 		chosen(deviceId, options[option]);
@@ -362,21 +389,22 @@ object_ptr<SingleChoiceBox> ChooseAudioInputBox(
 		Fn<void(QString id, QString name)> chosen,
 		const style::Checkbox *st,
 		const style::Radio *radioSt) {
-	const auto devices = Webrtc::GetAudioInputList();
+	const auto &settings = Core::App().settings();
+	const auto list = GetAudioInputList(settings.callAudioBackend());
 	const auto options = ranges::view::concat(
 		ranges::view::single(tr::lng_settings_call_device_default(tr::now)),
-		devices | ranges::view::transform(&Webrtc::AudioInput::name)
+		list | ranges::view::transform(&AudioInput::name)
 	) | ranges::to_vector;
 	const auto i = ranges::find(
-		devices,
+		list,
 		Core::App().settings().callInputDeviceId(),
-		&Webrtc::AudioInput::id);
-	const auto currentOption = (i != end(devices))
-		? int(i - begin(devices) + 1)
+		&AudioInput::id);
+	const auto currentOption = (i != end(list))
+		? int(i - begin(list) + 1)
 		: 0;
 	const auto save = [=](int option) {
 		const auto deviceId = option
-			? devices[option - 1].id
+			? list[option - 1].id
 			: "default";
 		Core::App().calls().setCurrentAudioDevice(true, deviceId);
 		chosen(deviceId, options[option]);
@@ -390,5 +418,33 @@ object_ptr<SingleChoiceBox> ChooseAudioInputBox(
 		radioSt);
 }
 
+object_ptr<SingleChoiceBox> ChooseAudioBackendBox(
+		const style::Checkbox *st,
+		const style::Radio *radioSt) {
+	const auto &settings = Core::App().settings();
+	const auto list = GetAudioInputList(settings.callAudioBackend());
+	const auto options = std::vector<QString>{
+		"OpenAL",
+		"Webrtc ADM",
+#ifdef Q_OS_WIN
+		"Webrtc ADM2",
+#endif // Q_OS_WIN
+	};
+	const auto currentOption = static_cast<int>(settings.callAudioBackend());
+	const auto save = [=](int option) {
+		Core::App().settings().setCallAudioBackend(
+			static_cast<Webrtc::Backend>(option));
+		Core::App().saveSettings();
+		App::restart();
+	};
+	return Box<SingleChoiceBox>(
+		rpl::single<QString>("Calls audio backend"),
+		options,
+		currentOption,
+		save,
+		st,
+		radioSt);
+}
+
 } // namespace Settings
 

Telegram/SourceFiles/settings/settings_calls.h

@@ -69,6 +69,9 @@ inline constexpr auto kMicTestAnimationDuration = crl::time(200);
 	Fn<void(QString id, QString name)> chosen,
 	const style::Checkbox *st = nullptr,
 	const style::Radio *radioSt = nullptr);
+[[nodiscard]] object_ptr<SingleChoiceBox> ChooseAudioBackendBox(
+	const style::Checkbox *st = nullptr,
+	const style::Radio *radioSt = nullptr);
 
 } // namespace Settings
 

Telegram/lib_webrtc

@@ -1 +1 @@
-Subproject commit bada95202ae45a650d76d9572b20fc9cb03365ad
+Subproject commit 4dd8c26bd1748a677df34785e62e031785e93e23