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Improve things for OpenAL devices management.

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This commit is contained in:
John Preston 2024-02-01 10:19:46 +04:00
parent 0945e04f6b
commit 2f40a44b5c
13 changed files with 140 additions and 92 deletions
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56
Telegram/SourceFiles/calls/calls_call.cpp
View file

@ -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(),

10
Telegram/SourceFiles/calls/calls_call.h
View file

@ -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;

55
Telegram/SourceFiles/calls/group/calls_group_call.cpp
View file

@ -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,

10
Telegram/SourceFiles/calls/group/calls_group_call.h
View file

@ -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;

19
Telegram/SourceFiles/calls/group/calls_group_settings.cpp
View file

@ -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);
}

2
Telegram/SourceFiles/calls/group/calls_group_settings.h
View file

@ -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;

25
Telegram/SourceFiles/media/audio/media_audio.cpp
View file

@ -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

7
Telegram/SourceFiles/media/audio/media_audio.h
View file

@ -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();

11
Telegram/SourceFiles/media/audio/media_audio_capture.cpp
View file

@ -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);

14
Telegram/SourceFiles/media/audio/media_audio_track.cpp
View file

@ -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();
}

13
Telegram/SourceFiles/media/audio/media_audio_track.h
View file

@ -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;

8
Telegram/SourceFiles/settings/settings_calls.cpp
View file

@ -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());
}

2
Telegram/lib_webrtc

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