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Code Issues Releases Wiki Activity

Implement media devices tracking on Windows.

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This commit is contained in:
John Preston
2024-01-22 20:40:55 +04:00
parent 30e694420a
commit d3778f92d2
18 changed files with 551 additions and 354 deletions
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78
Telegram/SourceFiles/calls/group/calls_group_call.cpp
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@@ -29,7 +29,6 @@ https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
#include "base/global_shortcuts.h"
#include "base/random.h"
#include "webrtc/webrtc_video_track.h"
#include "webrtc/webrtc_media_devices.h"
#include "webrtc/webrtc_create_adm.h"
#include <tgcalls/group/GroupInstanceCustomImpl.h>
@@ -52,14 +51,6 @@ constexpr auto kFixSpeakingLargeVideoDuration = 3 * crl::time(1000);
constexpr auto kFullAsMediumsCount = 4; // 1 Full is like 4 Mediums.
constexpr auto kMaxMediumQualities = 16; // 4 Fulls or 16 Mediums.
[[nodiscard]] std::unique_ptr<Webrtc::MediaDevices> CreateMediaDevices() {
const auto &settings = Core::App().settings();
return Webrtc::CreateMediaDevices(
settings.callInputDeviceId(),
settings.callOutputDeviceId(),
settings.callVideoInputDeviceId());
}
[[nodiscard]] const Data::GroupCallParticipant *LookupParticipant(
not_null<PeerData*> peer,
CallId id,
@@ -590,12 +581,27 @@ GroupCall::GroupCall(
, _scheduleDate(info.scheduleDate)
, _lastSpokeCheckTimer([=] { checkLastSpoke(); })
, _checkJoinedTimer([=] { checkJoined(); })
, _playbackDeviceId(
&Core::App().mediaDevices(),
Webrtc::DeviceType::Playback,
Webrtc::DeviceIdValueWithFallback(
Core::App().settings().callPlaybackDeviceIdValue(),
Core::App().settings().playbackDeviceIdValue()))
, _captureDeviceId(
&Core::App().mediaDevices(),
Webrtc::DeviceType::Capture,
Webrtc::DeviceIdValueWithFallback(
Core::App().settings().callCaptureDeviceIdValue(),
Core::App().settings().captureDeviceIdValue()))
, _cameraDeviceId(
&Core::App().mediaDevices(),
Webrtc::DeviceType::Camera,
Webrtc::DeviceIdOrDefault(Core::App().settings().cameraDeviceIdValue()))
, _pushToTalkCancelTimer([=] { pushToTalkCancel(); })
, _connectingSoundTimer([=] { playConnectingSoundOnce(); })
, _listenersHidden(info.rtmp)
, _rtmp(info.rtmp)
, _rtmpVolume(Group::kDefaultVolume)
, _mediaDevices(CreateMediaDevices()) {
, _rtmpVolume(Group::kDefaultVolume) {
_muted.value(
) | rpl::combine_previous(
) | rpl::start_with_next([=](MuteState previous, MuteState state) {
@@ -2058,28 +2064,22 @@ void GroupCall::applyOtherParticipantUpdate(
}
void GroupCall::setupMediaDevices() {
_mediaDevices->audioInputId(
) | rpl::start_with_next([=](QString id) {
_audioInputId = id;
if (_instance) {
_instance->setAudioInputDevice(id.toStdString());
}
_playbackDeviceId.changes() | rpl::filter([=] {
return _instance != nullptr;
}) | rpl::start_with_next([=](const QString &deviceId) {
_instance->setAudioOutputDevice(deviceId.toStdString());
}, _lifetime);
_mediaDevices->audioOutputId(
) | rpl::start_with_next([=](QString id) {
_audioOutputId = id;
if (_instance) {
_instance->setAudioOutputDevice(id.toStdString());
}
_captureDeviceId.changes() | rpl::filter([=] {
return _instance != nullptr;
}) | rpl::start_with_next([=](const QString &deviceId) {
_instance->setAudioInputDevice(deviceId.toStdString());
}, _lifetime);
_mediaDevices->videoInputId(
) | rpl::start_with_next([=](QString id) {
_cameraInputId = id;
if (_cameraCapture) {
_cameraCapture->switchToDevice(id.toStdString(), false);
}
_cameraDeviceId.changes() | rpl::filter([=] {
return _cameraCapture != nullptr;
}) | rpl::start_with_next([=](const QString &deviceId) {
_cameraCapture->switchToDevice(deviceId.toStdString(), false);
}, _lifetime);
}
@@ -2117,7 +2117,7 @@ bool GroupCall::emitShareCameraError() {
return emitError(Error::DisabledNoCamera);
} else if (mutedByAdmin()) {
return emitError(Error::MutedNoCamera);
} else if (Webrtc::GetVideoInputList().empty()) {
} else if (_cameraDeviceId.current().isEmpty()) {
return emitError(Error::NoCamera);
}
return false;
@@ -2126,7 +2126,7 @@ bool GroupCall::emitShareCameraError() {
void GroupCall::emitShareCameraError(Error error) {
_cameraState = Webrtc::VideoState::Inactive;
if (error == Error::CameraFailed
&& Webrtc::GetVideoInputList().empty()) {
&& _cameraDeviceId.current().isEmpty()) {
error = Error::NoCamera;
}
_errors.fire_copy(error);
@@ -2180,7 +2180,7 @@ void GroupCall::setupOutgoingVideo() {
return;
} else if (!_cameraCapture) {
_cameraCapture = _delegate->groupCallGetVideoCapture(
_cameraInputId);
_cameraDeviceId.current());
if (!_cameraCapture) {
return emitShareCameraError(Error::CameraFailed);
}
@@ -2192,7 +2192,7 @@ void GroupCall::setupOutgoingVideo() {
});
} else {
_cameraCapture->switchToDevice(
_cameraInputId.toStdString(),
_cameraDeviceId.current().toStdString(),
false);
}
if (_instance) {
@@ -2360,8 +2360,8 @@ bool GroupCall::tryCreateController() {
}
crl::on_main(weak, [=] { audioLevelsUpdated(data); });
},
.initialInputDeviceId = _audioInputId.toStdString(),
.initialOutputDeviceId = _audioOutputId.toStdString(),
.initialInputDeviceId = _captureDeviceId.current().toStdString(),
.initialOutputDeviceId = _playbackDeviceId.current().toStdString(),
.createAudioDeviceModule = Webrtc::AudioDeviceModuleCreator(
settings.callAudioBackend()),
.videoCapture = _cameraCapture,
@@ -3290,14 +3290,6 @@ void GroupCall::requestVideoQuality(
updateRequestedVideoChannelsDelayed();
}
void GroupCall::setCurrentAudioDevice(bool input, const QString &deviceId) {
if (input) {
_mediaDevices->switchToAudioInput(deviceId);
} else {
_mediaDevices->switchToAudioOutput(deviceId);
}
}
void GroupCall::toggleMute(const Group::MuteRequest &data) {
if (_rtmp) {
_rtmpVolume = data.mute ? 0 : Group::kDefaultVolume;