Use audio device enumeration from lib_webrtc.

Telegram/SourceFiles/calls/calls_call.cpp

@@ -24,7 +24,7 @@ https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
 #include "base/platform/base_platform_info.h"
 #include "calls/calls_panel.h"
 #include "webrtc/webrtc_video_track.h"
-#include "webrtc/webrtc_camera_utilities.h"
+#include "webrtc/webrtc_media_devices.h"
 #include "data/data_user.h"
 #include "data/data_session.h"
 #include "facades.h"
@@ -350,7 +350,7 @@ void Call::setupOutgoingVideo() {
 	_videoOutgoing->stateValue(
 	) | rpl::start_with_next([=](Webrtc::VideoState state) {
 		if (state != Webrtc::VideoState::Inactive
-			&& Webrtc::GetCamerasList().empty()) {
+			&& Webrtc::GetVideoInputList().empty()) {
 			_videoOutgoing->setState(Webrtc::VideoState::Inactive);
 		} else if (_state.current() != State::Established
 			&& state != started
@@ -710,6 +710,8 @@ void Call::createAndStartController(const MTPDphoneCall &call) {
 	auto encryptionKeyValue = std::make_shared<std::array<uint8_t, 256>>();
 	memcpy(encryptionKeyValue->data(), _authKey.data(), 256);
 
+	const auto &settings = Core::App().settings();
+
 	const auto weak = base::make_weak(this);
 	tgcalls::Descriptor descriptor = {
 		.config = tgcalls::Config{
@@ -726,6 +728,12 @@ void Call::createAndStartController(const MTPDphoneCall &call) {
 		.encryptionKey = tgcalls::EncryptionKey(
 			std::move(encryptionKeyValue),
 			(_type == Type::Outgoing)),
+		.mediaDevicesConfig = tgcalls::MediaDevicesConfig{
+			.audioInputId = settings.callInputDeviceID().toStdString(),
+			.audioOutputId = settings.callOutputDeviceID().toStdString(),
+			.inputVolume = settings.callInputVolume() / 100.f,
+			.outputVolume = settings.callOutputVolume() / 100.f,
+		},
 		.videoCapture = _videoCapture,
 		.stateUpdated = [=](tgcalls::State state) {
 			crl::on_main(weak, [=] {
@@ -810,14 +818,6 @@ void Call::createAndStartController(const MTPDphoneCall &call) {
 	}
 
 	raw->setIncomingVideoOutput(_videoIncoming->sink());
-
-	const auto &settings = Core::App().settings();
-	raw->setAudioOutputDevice(
-		settings.callOutputDeviceID().toStdString());
-	raw->setAudioInputDevice(
-		settings.callInputDeviceID().toStdString());
-	raw->setOutputVolume(settings.callOutputVolume() / 100.0f);
-	raw->setInputVolume(settings.callInputVolume() / 100.0f);
 	raw->setAudioOutputDuckingEnabled(settings.callAudioDuckingEnabled());
 }
 

Telegram/SourceFiles/core/core_settings.h

@@ -176,13 +176,17 @@ public:
 		_autoLock = value;
 	}
 	[[nodiscard]] QString callOutputDeviceID() const {
-		return _callOutputDeviceID;
+		return _callOutputDeviceID.isEmpty()
+			? u"default"_q
+			: _callOutputDeviceID;
 	}
 	void setCallOutputDeviceID(const QString &value) {
 		_callOutputDeviceID = value;
 	}
 	[[nodiscard]] QString callInputDeviceID() const {
-		return _callInputDeviceID;
+		return _callInputDeviceID.isEmpty()
+			? u"default"_q
+			: _callInputDeviceID;
 	}
 	void setCallInputDeviceID(const QString &value) {
 		_callInputDeviceID = value;

Telegram/SourceFiles/platform/win/specific_win.cpp

@@ -593,6 +593,7 @@ bool OpenSystemSettings(SystemSettingsType type) {
 	if (type == SystemSettingsType::Audio) {
 		crl::on_main([] {
 			WinExec("control.exe mmsys.cpl", SW_SHOW);
+			//QDesktopServices::openUrl(QUrl("ms-settings:sound"));
 		});
 	}
 	return true;

Telegram/SourceFiles/settings/settings_calls.cpp

@@ -26,6 +26,7 @@ https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
 #include "core/application.h"
 #include "core/core_settings.h"
 #include "calls/calls_instance.h"
+#include "webrtc/webrtc_media_devices.h"
 #include "facades.h"
 
 #ifdef slots
@@ -68,10 +69,10 @@ void Calls::setupContent() {
 
 	const auto content = Ui::CreateChild<Ui::VerticalLayout>(this);
 	const auto getId = [](const auto &device) {
-		return QString::fromStdString(device.id);
+		return device.id;
 	};
 	const auto getName = [](const auto &device) {
-		return QString::fromStdString(device.displayName);
+		return device.name;
 	};
 
 	const auto &settings = Core::App().settings();
@@ -79,7 +80,7 @@ void Calls::setupContent() {
 		if (settings.callOutputDeviceID() == qsl("default")) {
 			return tr::lng_settings_call_device_default(tr::now);
 		}
-		const auto &list = VoIP::EnumerateAudioOutputs();
+		const auto list = Webrtc::GetAudioOutputList();
 		const auto i = ranges::find(
 			list,
 			settings.callOutputDeviceID(),
@@ -93,7 +94,7 @@ void Calls::setupContent() {
 		if (settings.callInputDeviceID() == qsl("default")) {
 			return tr::lng_settings_call_device_default(tr::now);
 		}
-		const auto &list = VoIP::EnumerateAudioInputs();
+		const auto list = Webrtc::GetAudioInputList();
 		const auto i = ranges::find(
 			list,
 			settings.callInputDeviceID(),
@@ -115,7 +116,7 @@ void Calls::setupContent() {
 		),
 		st::settingsButton
 	)->addClickHandler([=] {
-		const auto &devices = VoIP::EnumerateAudioOutputs();
+		const auto &devices = Webrtc::GetAudioOutputList();
 		const auto options = ranges::view::concat(
 			ranges::view::single(tr::lng_settings_call_device_default(tr::now)),
 			devices | ranges::view::transform(getName)
@@ -132,11 +133,10 @@ void Calls::setupContent() {
 			const auto deviceId = option
 				? devices[option - 1].id
 				: "default";
-			Core::App().settings().setCallOutputDeviceID(
-				QString::fromStdString(deviceId));
+			Core::App().settings().setCallOutputDeviceID(deviceId);
 			Core::App().saveSettingsDelayed();
 			if (const auto call = Core::App().calls().currentCall()) {
-				call->setCurrentAudioDevice(false, deviceId);
+				call->setCurrentAudioDevice(false, deviceId.toStdString());
 			}
 		});
 		Ui::show(Box<SingleChoiceBox>(
@@ -171,7 +171,7 @@ void Calls::setupContent() {
 	};
 	outputSlider->resize(st::settingsAudioVolumeSlider.seekSize);
 	outputSlider->setPseudoDiscrete(
-		201,
+		101,
 		[](int val) { return val; },
 		settings.callOutputVolume(),
 		updateOutputVolume);
@@ -191,7 +191,7 @@ void Calls::setupContent() {
 		),
 		st::settingsButton
 	)->addClickHandler([=] {
-		const auto &devices = VoIP::EnumerateAudioInputs();
+		const auto devices = Webrtc::GetAudioInputList();
 		const auto options = ranges::view::concat(
 			ranges::view::single(tr::lng_settings_call_device_default(tr::now)),
 			devices | ranges::view::transform(getName)
@@ -208,14 +208,13 @@ void Calls::setupContent() {
 			const auto deviceId = option
 				? devices[option - 1].id
 				: "default";
-			Core::App().settings().setCallInputDeviceID(
-				QString::fromStdString(deviceId));
+			Core::App().settings().setCallInputDeviceID(deviceId);
 			Core::App().saveSettingsDelayed();
 			if (_micTester) {
 				stopTestingMicrophone();
 			}
 			if (const auto call = Core::App().calls().currentCall()) {
-				call->setCurrentAudioDevice(true, deviceId);
+				call->setCurrentAudioDevice(true, deviceId.toStdString());
 			}
 		});
 		Ui::show(Box<SingleChoiceBox>(

Telegram/ThirdParty/libtgvoip

@@ -1 +1 @@
-Subproject commit 6e82b6e45664c1f80b9039256c99bebc76d34672
+Subproject commit 7563a96b8f8e86b7a5fd1ce783388adf29bf4cf9

Telegram/ThirdParty/tgcalls

@@ -1 +1 @@
-Subproject commit 3e4c5e2c23b644fe9b2444e291c32d7efa31db41
+Subproject commit c6c78d68729ce16cff2bee868939780788443963

Telegram/cmake/lib_tgcalls.cmake

@@ -170,6 +170,8 @@ remove_target_sources(lib_tgcalls ${tgcalls_loc}
     platform/android/VideoCameraCapturer.h
     platform/android/VideoCapturerInterfaceImpl.cpp
     platform/android/VideoCapturerInterfaceImpl.h
+    reference/InstanceImplReference.cpp
+    reference/InstanceImplReference.h
 )
 
 target_include_directories(lib_tgcalls

Telegram/lib_webrtc

@@ -1 +1 @@
-Subproject commit 4d0f0b8ef41b7a7f8014cebb1dbc5549b06fde6f
+Subproject commit 32957e855993b20c95fa714518ba4bc55ebcdd32