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Merge pull request #14984 from kencooke/master

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Case 21384: Avatar mouth blendshapes are not working correctly (master)
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Anthony Thibault 2019-02-22 16:29:07 -08:00 committed by GitHub
commit 1213084fc6
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3 changed files with 21 additions and 10 deletions
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5
interface/src/scripting/Audio.cpp
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@ -27,8 +27,9 @@ QString Audio::HMD { "VR" };
Setting::Handle<bool> enableNoiseReductionSetting { QStringList { Audio::AUDIO, "NoiseReduction" }, true }; Setting::Handle<bool> enableNoiseReductionSetting { QStringList { Audio::AUDIO, "NoiseReduction" }, true };
float Audio::loudnessToLevel(float loudness) { float Audio::loudnessToLevel(float loudness) {
float level = 6.02059991f * fastLog2f(loudness); // level in dBFS float level = loudness * (1/32768.0f); // level in [0, 1]
level = (level + 48.0f) * (1/39.0f); // map [-48, -9] dBFS to [0, 1] level = 6.02059991f * fastLog2f(level); // convert to dBFS
level = (level + 48.0f) * (1/42.0f); // map [-48, -6] dBFS to [0, 1]
return glm::clamp(level, 0.0f, 1.0f); return glm::clamp(level, 0.0f, 1.0f);
} }

20
libraries/audio-client/src/AudioClient.cpp
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@ -175,7 +175,7 @@ static float computeLoudness(int16_t* samples, int numSamples, int numChannels,
const int32_t CLIPPING_THRESHOLD = 32392; // -0.1 dBFS const int32_t CLIPPING_THRESHOLD = 32392; // -0.1 dBFS
const int32_t CLIPPING_DETECTION = 3; // consecutive samples over threshold const int32_t CLIPPING_DETECTION = 3; // consecutive samples over threshold
float scale = numSamples ? 1.0f / (numSamples * 32768.0f) : 0.0f; float scale = numSamples ? 1.0f / numSamples : 0.0f;
int32_t loudness = 0; int32_t loudness = 0;
isClipping = false; isClipping = false;
@ -249,6 +249,8 @@ AudioClient::AudioClient() :
_outputBufferSizeFrames("audioOutputBufferFrames", DEFAULT_BUFFER_FRAMES), _outputBufferSizeFrames("audioOutputBufferFrames", DEFAULT_BUFFER_FRAMES),
_sessionOutputBufferSizeFrames(_outputBufferSizeFrames.get()), _sessionOutputBufferSizeFrames(_outputBufferSizeFrames.get()),
_outputStarveDetectionEnabled("audioOutputStarveDetectionEnabled", DEFAULT_STARVE_DETECTION_ENABLED), _outputStarveDetectionEnabled("audioOutputStarveDetectionEnabled", DEFAULT_STARVE_DETECTION_ENABLED),
_lastRawInputLoudness(0.0f),
_lastSmoothedRawInputLoudness(0.0f),
_lastInputLoudness(0.0f), _lastInputLoudness(0.0f),
_timeSinceLastClip(-1.0f), _timeSinceLastClip(-1.0f),
_muted(false), _muted(false),
@ -1144,6 +1146,9 @@ void AudioClient::handleAudioInput(QByteArray& audioBuffer) {
emit inputReceived(audioBuffer); emit inputReceived(audioBuffer);
} }
// loudness after mute/gate
_lastInputLoudness = (_muted || !audioGateOpen) ? 0.0f : _lastRawInputLoudness;
// detect gate opening and closing // detect gate opening and closing
bool openedInLastBlock = !_audioGateOpen && audioGateOpen; // the gate just opened bool openedInLastBlock = !_audioGateOpen && audioGateOpen; // the gate just opened
bool closedInLastBlock = _audioGateOpen && !audioGateOpen; // the gate just closed bool closedInLastBlock = _audioGateOpen && !audioGateOpen; // the gate just closed
@ -1222,12 +1227,15 @@ void AudioClient::handleMicAudioInput() {
// detect loudness and clipping on the raw input // detect loudness and clipping on the raw input
bool isClipping = false; bool isClipping = false;
float inputLoudness = computeLoudness(inputAudioSamples.get(), inputSamplesRequired, _inputFormat.channelCount(), isClipping); float loudness = computeLoudness(inputAudioSamples.get(), inputSamplesRequired, _inputFormat.channelCount(), isClipping);
_lastRawInputLoudness = loudness;
float tc = (inputLoudness > _lastInputLoudness) ? 0.378f : 0.967f; // 10ms attack, 300ms release @ 100Hz // envelope detection
inputLoudness += tc * (_lastInputLoudness - inputLoudness); float tc = (loudness > _lastSmoothedRawInputLoudness) ? 0.378f : 0.967f; // 10ms attack, 300ms release @ 100Hz
_lastInputLoudness = inputLoudness; loudness += tc * (_lastSmoothedRawInputLoudness - loudness);
_lastSmoothedRawInputLoudness = loudness;
// clipping indicator
if (isClipping) { if (isClipping) {
_timeSinceLastClip = 0.0f; _timeSinceLastClip = 0.0f;
} else if (_timeSinceLastClip >= 0.0f) { } else if (_timeSinceLastClip >= 0.0f) {
@ -1235,7 +1243,7 @@ void AudioClient::handleMicAudioInput() {
} }
isClipping = (_timeSinceLastClip >= 0.0f) && (_timeSinceLastClip < 2.0f); // 2 second hold time isClipping = (_timeSinceLastClip >= 0.0f) && (_timeSinceLastClip < 2.0f); // 2 second hold time
emit inputLoudnessChanged(_lastInputLoudness, isClipping); emit inputLoudnessChanged(_lastSmoothedRawInputLoudness, isClipping);
if (!_muted) { if (!_muted) {
possibleResampling(_inputToNetworkResampler, possibleResampling(_inputToNetworkResampler,

6
libraries/audio-client/src/AudioClient.h
View file

@ -127,7 +127,7 @@ public:
const QAudioFormat& getOutputFormat() const { return _outputFormat; } const QAudioFormat& getOutputFormat() const { return _outputFormat; }
float getLastInputLoudness() const { return _lastInputLoudness; } // TODO: relative to noise floor? float getLastInputLoudness() const { return _lastInputLoudness; }
float getTimeSinceLastClip() const { return _timeSinceLastClip; } float getTimeSinceLastClip() const { return _timeSinceLastClip; }
float getAudioAverageInputLoudness() const { return _lastInputLoudness; } float getAudioAverageInputLoudness() const { return _lastInputLoudness; }
@ -355,7 +355,9 @@ private:
StDev _stdev; StDev _stdev;
QElapsedTimer _timeSinceLastReceived; QElapsedTimer _timeSinceLastReceived;
float _lastInputLoudness; float _lastRawInputLoudness; // before mute/gate
float _lastSmoothedRawInputLoudness;
float _lastInputLoudness; // after mute/gate
float _timeSinceLastClip; float _timeSinceLastClip;
int _totalInputAudioSamples; int _totalInputAudioSamples;