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Rework VU meter level processing to be closer to Vivox

master
Roxie Linden 2023-12-07 19:47:37 -08:00
parent 60da67c56c
commit c51309ac72
3 changed files with 67 additions and 82 deletions
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6
indra/llwebrtc/llwebrtc.cpp
View File

@ -54,7 +54,7 @@ void LLAudioDeviceObserver::OnCaptureData(const void *audio_samples,
const short *samples = (const short *) audio_samples;
for (size_t index = 0; index < num_samples * num_channels; index++)
{
float sample = (static_cast<float>(samples[index]) / (float) 32768);
float sample = (static_cast<float>(samples[index]) / (float) 32767);
energy += sample * sample;
}
@ -384,9 +384,9 @@ void LLWebRTCImpl::setTuningMode(bool enable)
}
}
float LLWebRTCImpl::getTuningAudioLevel() { return 20 * mTuningAudioDeviceObserver->getMicrophoneEnergy(); }
float LLWebRTCImpl::getTuningAudioLevel() { return -20 * log10f(mTuningAudioDeviceObserver->getMicrophoneEnergy()); }
float LLWebRTCImpl::getPeerAudioLevel() { return 20 * mPeerAudioDeviceObserver->getMicrophoneEnergy(); }
float LLWebRTCImpl::getPeerAudioLevel() { return -20 * log10f(mPeerAudioDeviceObserver->getMicrophoneEnergy()); }
//
// Helpers

134
indra/newview/llvoicewebrtc.cpp
View File

@ -86,9 +86,10 @@ extern void handle_voice_morphing_subscribe();
namespace {
const F32 VOLUME_SCALE_WEBRTC = 0.01f;
const F32 LEVEL_SCALE_WEBRTC = 0.01f;
const F32 SPEAKING_TIMEOUT = 1.f;
const F32 SPEAKING_AUDIO_LEVEL = 0.05;
const F32 SPEAKING_AUDIO_LEVEL = 0.10;
static const std::string VOICE_SERVER_TYPE = "WebRTC";
@ -234,9 +235,7 @@ LLWebRTCVoiceClient::LLWebRTCVoiceClient() :
mSpatialJoiningNum(0),
mTuningMode(false),
mTuningEnergy(0.0f),
mTuningMicVolume(0),
mTuningMicVolumeDirty(true),
mTuningMicGain(0.0),
mTuningSpeakerVolume(50), // Set to 50 so the user can hear himself when he sets his mic volume
mTuningSpeakerVolumeDirty(true),
mDevicesListUpdated(false),
@ -261,8 +260,7 @@ LLWebRTCVoiceClient::LLWebRTCVoiceClient() :
mEarLocation(0),
mSpeakerVolumeDirty(true),
mMicVolume(0),
mMicVolumeDirty(true),
mMicGain(0.0),
mVoiceEnabled(false),
@ -314,25 +312,25 @@ LLWebRTCVoiceClient::LLWebRTCVoiceClient() :
LLWebRTCVoiceClient::~LLWebRTCVoiceClient()
{
if (mAvatarNameCacheConnection.connected())
{
mAvatarNameCacheConnection.disconnect();
}
if (mAvatarNameCacheConnection.connected())
{
mAvatarNameCacheConnection.disconnect();
}
sShuttingDown = true;
}
//---------------------------------------------------
void LLWebRTCVoiceClient::init(LLPumpIO *pump)
void LLWebRTCVoiceClient::init(LLPumpIO* pump)
{
// constructor will set up LLVoiceClient::getInstance()
sPump = pump;
// constructor will set up LLVoiceClient::getInstance()
sPump = pump;
// LLCoros::instance().launch("LLWebRTCVoiceClient::voiceConnectionCoro",
// boost::bind(&LLWebRTCVoiceClient::voiceConnectionCoro, LLWebRTCVoiceClient::getInstance()));
// LLCoros::instance().launch("LLWebRTCVoiceClient::voiceConnectionCoro",
// boost::bind(&LLWebRTCVoiceClient::voiceConnectionCoro, LLWebRTCVoiceClient::getInstance()));
llwebrtc::init();
mWebRTCDeviceInterface = llwebrtc::getDeviceInterface();
mWebRTCDeviceInterface = llwebrtc::getDeviceInterface();
mWebRTCDeviceInterface->setDevicesObserver(this);
}
@ -342,9 +340,9 @@ void LLWebRTCVoiceClient::terminate()
{
return;
}
mRelogRequested = false;
mVoiceEnabled = false;
mRelogRequested = false;
mVoiceEnabled = false;
llwebrtc::terminate();
sShuttingDown = true;
@ -356,10 +354,10 @@ void LLWebRTCVoiceClient::terminate()
void LLWebRTCVoiceClient::cleanUp()
{
LL_DEBUGS("Voice") << LL_ENDL;
mNextAudioSession.reset();
mAudioSession.reset();
sessionState::deleteAllSessions();
sessionState::deleteAllSessions();
LL_DEBUGS("Voice") << "exiting" << LL_ENDL;
}
@ -367,7 +365,7 @@ void LLWebRTCVoiceClient::cleanUp()
const LLVoiceVersionInfo& LLWebRTCVoiceClient::getVersion()
{
return mVoiceVersion;
return mVoiceVersion;
}
//---------------------------------------------------
@ -375,15 +373,15 @@ const LLVoiceVersionInfo& LLWebRTCVoiceClient::getVersion()
void LLWebRTCVoiceClient::updateSettings()
{
setVoiceEnabled(voiceEnabled());
setEarLocation(gSavedSettings.getS32("VoiceEarLocation"));
setEarLocation(gSavedSettings.getS32("VoiceEarLocation"));
std::string inputDevice = gSavedSettings.getString("VoiceInputAudioDevice");
setCaptureDevice(inputDevice);
std::string outputDevice = gSavedSettings.getString("VoiceOutputAudioDevice");
setRenderDevice(outputDevice);
F32 mic_level = gSavedSettings.getF32("AudioLevelMic");
setMicGain(mic_level);
setLipSyncEnabled(gSavedSettings.getBOOL("LipSyncEnabled"));
std::string inputDevice = gSavedSettings.getString("VoiceInputAudioDevice");
setCaptureDevice(inputDevice);
std::string outputDevice = gSavedSettings.getString("VoiceOutputAudioDevice");
setRenderDevice(outputDevice);
F32 mic_level = gSavedSettings.getF32("AudioLevelMic");
setMicGain(mic_level);
setLipSyncEnabled(gSavedSettings.getBOOL("LipSyncEnabled"));
}
@ -395,7 +393,7 @@ void LLWebRTCVoiceClient::predOnConnectionEstablished(const LLWebRTCVoiceClient:
session->OnConnectionEstablished(channelID);
}
void LLWebRTCVoiceClient::predOnConnectionFailure(const LLWebRTCVoiceClient::sessionStatePtr_t &session, std::string channelID)
void LLWebRTCVoiceClient::predOnConnectionFailure(const LLWebRTCVoiceClient::sessionStatePtr_t& session, std::string channelID)
{
session->OnConnectionFailure(channelID);
}
@ -405,7 +403,7 @@ void LLWebRTCVoiceClient::OnConnectionFailure(const std::string& channelID)
sessionState::for_each(boost::bind(predOnConnectionFailure, _1, channelID));
}
void LLWebRTCVoiceClient::OnConnectionEstablished(const std::string &channelID)
void LLWebRTCVoiceClient::OnConnectionEstablished(const std::string& channelID)
{
if (mNextAudioSession && mNextAudioSession->mChannelID == channelID)
{
@ -413,6 +411,10 @@ void LLWebRTCVoiceClient::OnConnectionEstablished(const std::string &channelID)
mNextAudioSession.reset();
LLWebRTCVoiceClient::getInstance()->notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_LOGGED_IN);
}
else if (mAudioSession && mAudioSession->mChannelID == channelID)
{
LLWebRTCVoiceClient::getInstance()->notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_LOGGED_IN);
}
sessionState::for_each(boost::bind(predOnConnectionEstablished, _1, channelID));
}
@ -658,13 +660,7 @@ bool LLWebRTCVoiceClient::inTuningMode()
void LLWebRTCVoiceClient::tuningSetMicVolume(float volume)
{
int scaled_volume = scale_mic_volume(volume);
if(scaled_volume != mTuningMicVolume)
{
mTuningMicVolume = scaled_volume;
mTuningMicVolumeDirty = true;
}
mTuningMicGain = volume;
}
void LLWebRTCVoiceClient::tuningSetSpeakerVolume(float volume)
@ -679,7 +675,7 @@ void LLWebRTCVoiceClient::tuningSetSpeakerVolume(float volume)
float LLWebRTCVoiceClient::tuningGetEnergy(void)
{
return mWebRTCDeviceInterface->getTuningAudioLevel();
return (1.0 - mWebRTCDeviceInterface->getTuningAudioLevel() * LEVEL_SCALE_WEBRTC) * mTuningMicGain/2.1;
}
bool LLWebRTCVoiceClient::deviceSettingsAvailable()
@ -739,10 +735,10 @@ void LLWebRTCVoiceClient::sendPositionAndVolumeUpdate(bool force)
F32 audio_level = 0.0;
uint32_t uint_audio_level = 0.0;
if (!mMuteMic)
if (!mMuteMic && !mTuningMode)
{
audio_level = (F32) mWebRTCDeviceInterface->getPeerAudioLevel() * mMicVolume;
uint_audio_level = (uint32_t) (audio_level * 128);
audio_level = (1.0 - mWebRTCDeviceInterface->getTuningAudioLevel() * LEVEL_SCALE_WEBRTC) * mMicGain / 2.1;
uint_audio_level = (uint32_t) (audio_level*128);
}
@ -810,9 +806,10 @@ void LLWebRTCVoiceClient::sendPositionAndVolumeUpdate(bool force)
void LLWebRTCVoiceClient::updateOwnVolume() {
F32 audio_level = 0.0;
if (!mMuteMic)
if (!mMuteMic && !mTuningMode)
{
audio_level = (F32) mWebRTCDeviceInterface->getPeerAudioLevel();
audio_level = (F32) (1.0 - mWebRTCDeviceInterface->getTuningAudioLevel() * LEVEL_SCALE_WEBRTC) * mMicGain / 2.1;
LL_DEBUGS("Voice") << "Level " << audio_level << LL_ENDL;
}
sessionState::for_each(boost::bind(predUpdateOwnVolume, _1, audio_level));
@ -823,7 +820,7 @@ void LLWebRTCVoiceClient::predUpdateOwnVolume(const LLWebRTCVoiceClient::session
participantStatePtr_t participant = session->findParticipant(gAgentID.asString());
if (participant)
{
participant->mPower = audio_level;
participant->mLevel = audio_level;
participant->mIsSpeaking = audio_level > SPEAKING_AUDIO_LEVEL;
}
}
@ -991,7 +988,7 @@ LLWebRTCVoiceClient::participantState::participantState(const LLUUID& agent_id)
mIsSpeaking(false),
mIsModeratorMuted(false),
mLastSpokeTimestamp(0.f),
mPower(0.f),
mLevel(0.f),
mVolume(LLVoiceClient::VOLUME_DEFAULT),
mUserVolume(0),
mOnMuteList(false),
@ -1763,7 +1760,7 @@ void LLWebRTCVoiceClient::predSetMuteMic(const LLWebRTCVoiceClient::sessionState
participantStatePtr_t participant = session->findParticipant(gAgentID.asString());
if (participant)
{
participant->mPower = 0.0;
participant->mLevel = 0.0;
}
session->setMuteMic(muted);
}
@ -1773,9 +1770,9 @@ void LLWebRTCVoiceClient::predSetSpeakerVolume(const LLWebRTCVoiceClient::sessio
session->setSpeakerVolume(volume);
}
void LLWebRTCVoiceClient::predSetMicGain(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 volume)
void LLWebRTCVoiceClient::predSetMicGain(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 gain)
{
session->setMicGain(volume);
session->setMicGain(gain);
}
void LLWebRTCVoiceClient::setVoiceEnabled(bool enabled)
@ -1875,16 +1872,13 @@ void LLWebRTCVoiceClient::setVoiceVolume(F32 volume)
}
}
void LLWebRTCVoiceClient::setMicGain(F32 volume)
{
int scaled_volume = scale_mic_volume(volume);
if(scaled_volume != mMicVolume)
{
mMicVolume = scaled_volume;
mMicVolumeDirty = true;
}
sessionState::for_each(boost::bind(predSetMicGain, _1, scaled_volume));
void LLWebRTCVoiceClient::setMicGain(F32 gain)
{
if (gain != mMicGain)
{
mMicGain = gain;
sessionState::for_each(boost::bind(predSetMicGain, _1, gain));
}
}
/////////////////////////////
@ -1967,7 +1961,7 @@ F32 LLWebRTCVoiceClient::getCurrentPower(const LLUUID &id)
participantStatePtr_t participant(mAudioSession->findParticipant(id.asString()));
if (participant)
{
result = participant->mPower * 2.0;
result = participant->mLevel;
}
return result;
}
@ -2224,7 +2218,7 @@ LLWebRTCVoiceClient::sessionStatePtr_t LLWebRTCVoiceClient::addSession(const std
LL_DEBUGS("Voice") << "adding new session: CHANNEL " << channel_id << LL_ENDL;
result = sessionState::createSession(channel_id, parcel_local_id);
result->setMuteMic(mMuteMic);
result->setMicGain(mMicVolume);
result->setMicGain(mMicGain);
result->setSpeakerVolume(mSpeakerVolume);
if (LLVoiceClient::instance().getVoiceEffectEnabled())
@ -2639,12 +2633,12 @@ void LLVoiceWebRTCConnection::OnDataReceived(const std::string &data, bool binar
}
else
{
F32 energyRMS = (F32) (voice_data[participant_id].get("p", Json::Value(participant->mPower)).asInt()) / 128;
F32 level = (F32) (voice_data[participant_id].get("p", Json::Value(participant->mLevel)).asInt()) / 128;
// convert to decibles
participant->mPower = energyRMS;
participant->mLevel = level;
/* WebRTC appears to have deprecated VAD, but it's still in the Audio Processing Module so maybe we
can use it at some point when we actually process frames. */
participant->mIsSpeaking = participant->mPower > SPEAKING_AUDIO_LEVEL;
participant->mIsSpeaking = participant->mLevel > SPEAKING_AUDIO_LEVEL;
}
}
}
@ -2898,14 +2892,8 @@ bool LLVoiceWebRTCConnection::connectionStateMachine()
break;
case VOICE_STATE_DISCONNECT:
if (breakVoiceConnection(true))
{
setVoiceConnectionState(VOICE_STATE_WAIT_FOR_EXIT);
}
else
{
setVoiceConnectionState(VOICE_STATE_SESSION_EXIT);
}
breakVoiceConnection(true);
retry = 0; // Connected without issues
break;

9
indra/newview/llvoicewebrtc.h
View File

@ -280,7 +280,7 @@ public:
std::string mDisplayName;
LLFrameTimer mSpeakingTimeout;
F32 mLastSpokeTimestamp;
F32 mPower;
F32 mLevel;
F32 mVolume;
std::string mGroupID;
int mUserVolume;
@ -559,9 +559,7 @@ private:
static void idle(void *user_data);
bool mTuningMode;
float mTuningEnergy;
int mTuningMicVolume;
bool mTuningMicVolumeDirty;
F32 mTuningMicGain;
int mTuningSpeakerVolume;
bool mTuningSpeakerVolumeDirty;
bool mDevicesListUpdated; // set to true when the device list has been updated
@ -647,8 +645,7 @@ private:
bool mSpeakerVolumeDirty;
float mSpeakerVolume;
int mMicVolume;
bool mMicVolumeDirty;
F32 mMicGain;
bool mVoiceEnabled;