diff options
| author | Jonathan "Geenz" Goodman <geenz@lindenlab.com> | 2025-10-17 16:56:48 -0400 |
|---|---|---|
| committer | GitHub <noreply@github.com> | 2025-10-17 16:56:48 -0400 |
| commit | aec7bf19ebffd9d6b60c68e31de723eabd6aa98a (patch) | |
| tree | 9ee0b36fa08a99444260baaf5ea9b019758bb125 /indra/newview/llvoicewebrtc.cpp | |
| parent | e81b1de01e2b28f83cd2c85691428211bb9105e9 (diff) | |
| parent | 57a9e51360aebf142bbbdc2663f68ebacfb7d8f5 (diff) | |
Merge pull request #4714 from secondlife/release/2025.07
Release/2025.07
Diffstat (limited to 'indra/newview/llvoicewebrtc.cpp')
| -rw-r--r-- | indra/newview/llvoicewebrtc.cpp | 174 |
1 files changed, 127 insertions, 47 deletions
diff --git a/indra/newview/llvoicewebrtc.cpp b/indra/newview/llvoicewebrtc.cpp index eae71ca454..72a60c506b 100644 --- a/indra/newview/llvoicewebrtc.cpp +++ b/indra/newview/llvoicewebrtc.cpp @@ -82,9 +82,15 @@ const std::string WEBRTC_VOICE_SERVER_TYPE = "webrtc"; namespace { - const F32 MAX_AUDIO_DIST = 50.0f; - const F32 VOLUME_SCALE_WEBRTC = 0.01f; - const F32 LEVEL_SCALE_WEBRTC = 0.008f; + const F32 MAX_AUDIO_DIST = 50.0f; + const F32 VOLUME_SCALE_WEBRTC = 0.01f; + const F32 TUNING_LEVEL_SCALE = 0.01f; + const F32 TUNING_LEVEL_START_POINT = 0.8f; + const F32 LEVEL_SCALE = 0.005f; + const F32 LEVEL_START_POINT = 0.18f; + const uint32_t SET_HIDDEN_RESTORE_DELAY_MS = 200; // 200 ms to unmute again after hiding during teleport + const uint32_t MUTE_FADE_DELAY_MS = 500; // 20ms fade followed by 480ms silence gets rid of the click just after unmuting. + // This is because the buffers and processing is cleared by the silence. const F32 SPEAKING_AUDIO_LEVEL = 0.30; @@ -201,7 +207,6 @@ bool LLWebRTCVoiceClient::sShuttingDown = false; LLWebRTCVoiceClient::LLWebRTCVoiceClient() : mHidden(false), - mTuningMode(false), mTuningMicGain(0.0), mTuningSpeakerVolume(50), // Set to 50 so the user can hear themselves when he sets his mic volume mDevicesListUpdated(false), @@ -268,6 +273,11 @@ void LLWebRTCVoiceClient::cleanupSingleton() void LLWebRTCVoiceClient::init(LLPumpIO* pump) { // constructor will set up LLVoiceClient::getInstance() + initWebRTC(); +} + +void LLWebRTCVoiceClient::initWebRTC() +{ llwebrtc::init(this); mWebRTCDeviceInterface = llwebrtc::getDeviceInterface(); @@ -283,8 +293,11 @@ void LLWebRTCVoiceClient::terminate() return; } + LL_INFOS("Voice") << "Terminating WebRTC" << LL_ENDL; + mVoiceEnabled = false; llwebrtc::terminate(); + mWebRTCDeviceInterface = nullptr; sShuttingDown = true; } @@ -348,25 +361,45 @@ void LLWebRTCVoiceClient::updateSettings() static LLCachedControl<std::string> sOutputDevice(gSavedSettings, "VoiceOutputAudioDevice"); setRenderDevice(sOutputDevice); - LL_INFOS("Voice") << "Input device: " << std::quoted(sInputDevice()) << ", output device: " << std::quoted(sOutputDevice()) << LL_ENDL; + LL_INFOS("Voice") << "Input device: " << std::quoted(sInputDevice()) << ", output device: " << std::quoted(sOutputDevice()) + << LL_ENDL; static LLCachedControl<F32> sMicLevel(gSavedSettings, "AudioLevelMic"); setMicGain(sMicLevel); llwebrtc::LLWebRTCDeviceInterface::AudioConfig config; + bool audioConfigChanged = false; + static LLCachedControl<bool> sEchoCancellation(gSavedSettings, "VoiceEchoCancellation", true); - config.mEchoCancellation = sEchoCancellation; + if (sEchoCancellation != config.mEchoCancellation) + { + config.mEchoCancellation = sEchoCancellation; + audioConfigChanged = true; + } static LLCachedControl<bool> sAGC(gSavedSettings, "VoiceAutomaticGainControl", true); - config.mAGC = sAGC; + if (sAGC != config.mAGC) + { + config.mAGC = sAGC; + audioConfigChanged = true; + } - static LLCachedControl<U32> sNoiseSuppressionLevel(gSavedSettings, + static LLCachedControl<U32> sNoiseSuppressionLevel( + gSavedSettings, "VoiceNoiseSuppressionLevel", llwebrtc::LLWebRTCDeviceInterface::AudioConfig::ENoiseSuppressionLevel::NOISE_SUPPRESSION_LEVEL_VERY_HIGH); - config.mNoiseSuppressionLevel = (llwebrtc::LLWebRTCDeviceInterface::AudioConfig::ENoiseSuppressionLevel)(U32)sNoiseSuppressionLevel; - - mWebRTCDeviceInterface->setAudioConfig(config); + auto noiseSuppressionLevel = + (llwebrtc::LLWebRTCDeviceInterface::AudioConfig::ENoiseSuppressionLevel)(U32)sNoiseSuppressionLevel; + if (noiseSuppressionLevel != config.mNoiseSuppressionLevel) + { + config.mNoiseSuppressionLevel = noiseSuppressionLevel; + audioConfigChanged = true; + } + if (audioConfigChanged) + { + mWebRTCDeviceInterface->setAudioConfig(config); + } } } @@ -664,7 +697,10 @@ LLVoiceDeviceList& LLWebRTCVoiceClient::getCaptureDevices() void LLWebRTCVoiceClient::setCaptureDevice(const std::string& name) { - mWebRTCDeviceInterface->setCaptureDevice(name); + if (mWebRTCDeviceInterface) + { + mWebRTCDeviceInterface->setCaptureDevice(name); + } } void LLWebRTCVoiceClient::setDevicesListUpdated(bool state) { @@ -695,21 +731,38 @@ void LLWebRTCVoiceClient::OnDevicesChangedImpl(const llwebrtc::LLWebRTCVoiceDevi std::string outputDevice = gSavedSettings.getString("VoiceOutputAudioDevice"); LL_DEBUGS("Voice") << "Setting devices to-input: '" << inputDevice << "' output: '" << outputDevice << "'" << LL_ENDL; - clearRenderDevices(); - for (auto &device : render_devices) + + // only set the render device if the device list has changed. + if (mRenderDevices.size() != render_devices.size() || !std::equal(mRenderDevices.begin(), + mRenderDevices.end(), + render_devices.begin(), + [](const LLVoiceDevice& a, const llwebrtc::LLWebRTCVoiceDevice& b) { + return a.display_name == b.mDisplayName && a.full_name == b.mID; })) { - addRenderDevice(LLVoiceDevice(device.mDisplayName, device.mID)); + clearRenderDevices(); + for (auto& device : render_devices) + { + addRenderDevice(LLVoiceDevice(device.mDisplayName, device.mID)); + } + setRenderDevice(outputDevice); } - setRenderDevice(outputDevice); - clearCaptureDevices(); - for (auto &device : capture_devices) + // only set the capture device if the device list has changed. + if (mCaptureDevices.size() != capture_devices.size() ||!std::equal(mCaptureDevices.begin(), + mCaptureDevices.end(), + capture_devices.begin(), + [](const LLVoiceDevice& a, const llwebrtc::LLWebRTCVoiceDevice& b) + { return a.display_name == b.mDisplayName && a.full_name == b.mID; })) { - LL_DEBUGS("Voice") << "Checking capture device:'" << device.mID << "'" << LL_ENDL; + clearCaptureDevices(); + for (auto& device : capture_devices) + { + LL_DEBUGS("Voice") << "Checking capture device:'" << device.mID << "'" << LL_ENDL; - addCaptureDevice(LLVoiceDevice(device.mDisplayName, device.mID)); + addCaptureDevice(LLVoiceDevice(device.mDisplayName, device.mID)); + } + setCaptureDevice(inputDevice); } - setCaptureDevice(inputDevice); setDevicesListUpdated(true); } @@ -734,7 +787,10 @@ LLVoiceDeviceList& LLWebRTCVoiceClient::getRenderDevices() void LLWebRTCVoiceClient::setRenderDevice(const std::string& name) { - mWebRTCDeviceInterface->setRenderDevice(name); + if (mWebRTCDeviceInterface) + { + mWebRTCDeviceInterface->setRenderDevice(name); + } } void LLWebRTCVoiceClient::tuningStart() @@ -762,7 +818,14 @@ bool LLWebRTCVoiceClient::inTuningMode() void LLWebRTCVoiceClient::tuningSetMicVolume(float volume) { - mTuningMicGain = volume; + if (volume != mTuningMicGain) + { + mTuningMicGain = volume; + if (mWebRTCDeviceInterface) + { + mWebRTCDeviceInterface->setTuningMicGain(volume); + } + } } void LLWebRTCVoiceClient::tuningSetSpeakerVolume(float volume) @@ -774,21 +837,10 @@ void LLWebRTCVoiceClient::tuningSetSpeakerVolume(float volume) } } -float LLWebRTCVoiceClient::getAudioLevel() -{ - if (mIsInTuningMode) - { - return (1.0f - mWebRTCDeviceInterface->getTuningAudioLevel() * LEVEL_SCALE_WEBRTC) * mTuningMicGain / 2.1f; - } - else - { - return (1.0f - mWebRTCDeviceInterface->getPeerConnectionAudioLevel() * LEVEL_SCALE_WEBRTC) * mMicGain / 2.1f; - } -} - float LLWebRTCVoiceClient::tuningGetEnergy(void) { - return getAudioLevel(); + float rms = mWebRTCDeviceInterface->getTuningAudioLevel(); + return TUNING_LEVEL_START_POINT - TUNING_LEVEL_SCALE * rms; } bool LLWebRTCVoiceClient::deviceSettingsAvailable() @@ -824,6 +876,11 @@ void LLWebRTCVoiceClient::setHidden(bool hidden) if (inSpatialChannel()) { + if (mWebRTCDeviceInterface) + { + mWebRTCDeviceInterface->setMute(mHidden || mMuteMic, + mHidden ? 0 : SET_HIDDEN_RESTORE_DELAY_MS); // delay 200ms so as to not pile up mutes/unmutes. + } if (mHidden) { // get out of the channel entirely @@ -990,7 +1047,6 @@ void LLWebRTCVoiceClient::updatePosition(void) { if (participant->mRegion != region->getRegionID()) { participant->mRegion = region->getRegionID(); - setMuteMic(mMuteMic); } } } @@ -1115,13 +1171,14 @@ void LLWebRTCVoiceClient::sendPositionUpdate(bool force) // Update our own volume on our participant, so it'll show up // in the UI. This is done on all sessions, so switching // sessions retains consistent volume levels. -void LLWebRTCVoiceClient::updateOwnVolume() { - F32 audio_level = 0.0; - if (!mMuteMic && !mTuningMode) +void LLWebRTCVoiceClient::updateOwnVolume() +{ + F32 audio_level = 0.0f; + if (!mMuteMic) { - audio_level = getAudioLevel(); + float rms = mWebRTCDeviceInterface->getPeerConnectionAudioLevel(); + audio_level = LEVEL_START_POINT - LEVEL_SCALE * rms; } - sessionState::for_each(boost::bind(predUpdateOwnVolume, _1, audio_level)); } @@ -1518,6 +1575,17 @@ void LLWebRTCVoiceClient::setMuteMic(bool muted) } mMuteMic = muted; + + if (mIsInTuningMode) + { + return; + } + + if (mWebRTCDeviceInterface) + { + mWebRTCDeviceInterface->setMute(muted, muted ? MUTE_FADE_DELAY_MS : 0); // delay for 40ms on mute to allow buffers to empty + } + // when you're hidden, your mic is always muted. if (!mHidden) { @@ -1556,7 +1624,10 @@ void LLWebRTCVoiceClient::setMicGain(F32 gain) if (gain != mMicGain) { mMicGain = gain; - mWebRTCDeviceInterface->setPeerConnectionGain(gain); + if (mWebRTCDeviceInterface) + { + mWebRTCDeviceInterface->setMicGain(gain); + } } } @@ -1740,6 +1811,15 @@ void LLWebRTCVoiceClient::onChangeDetailed(const LLMute& mute) } } +void LLWebRTCVoiceClient::userAuthorized(const std::string& user_id, const LLUUID& agentID) +{ + if (sShuttingDown) + { + sShuttingDown = false; // was terminated, restart + initWebRTC(); + } +} + void LLWebRTCVoiceClient::predSetUserMute(const LLWebRTCVoiceClient::sessionStatePtr_t &session, const LLUUID &id, bool mute) { session->setUserMute(id, mute); @@ -2308,7 +2388,6 @@ void LLVoiceWebRTCConnection::processIceUpdatesCoro(connectionPtr_t connection) return; } - bool iceCompleted = false; LLSD body; if (!connection->mIceCandidates.empty() || connection->mIceCompleted) { @@ -2347,7 +2426,6 @@ void LLVoiceWebRTCConnection::processIceUpdatesCoro(connectionPtr_t connection) LLSD body_candidate; body_candidate["completed"] = true; body["candidate"] = body_candidate; - iceCompleted = connection->mIceCompleted; connection->mIceCompleted = false; } @@ -2902,9 +2980,13 @@ bool LLVoiceWebRTCConnection::connectionStateMachine() } // else was already posted by llwebrtc::terminate(). break; + } + case VOICE_STATE_WAIT_FOR_CLOSE: break; + case VOICE_STATE_CLOSED: + { if (!mShutDown) { mVoiceConnectionState = VOICE_STATE_START_SESSION; @@ -2980,7 +3062,6 @@ void LLVoiceWebRTCConnection::OnDataReceivedImpl(const std::string &data, bool b return; } boost::json::object voice_data = voice_data_parsed.as_object(); - bool new_participant = false; boost::json::object mute; boost::json::object user_gain; for (auto &participant_elem : voice_data) @@ -3033,7 +3114,6 @@ void LLVoiceWebRTCConnection::OnDataReceivedImpl(const std::string &data, bool b } } - new_participant |= joined; if (!participant && joined && (primary || !isSpatial())) { participant = LLWebRTCVoiceClient::getInstance()->addParticipantByID(mChannelID, agent_id, mRegionID); |
