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Merge pull request #16040 from kencooke/audio-webrtc-improvements

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Better detection of audio clipping during echo cancellation
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Ken Cooke 2019-08-08 09:17:58 -07:00 committed by GitHub
commit f445737550
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1 changed files with 27 additions and 25 deletions
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52
libraries/audio-client/src/AudioClient.cpp
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@ -170,27 +170,21 @@ static void channelDownmix(int16_t* source, int16_t* dest, int numSamples) {
}
}
static float computeLoudness(int16_t* samples, int numSamples, int numChannels, bool& isClipping) {
static bool detectClipping(int16_t* samples, int numSamples, int numChannels) {
const int32_t CLIPPING_THRESHOLD = 32392; // -0.1 dBFS
const int32_t CLIPPING_DETECTION = 3; // consecutive samples over threshold
const int CLIPPING_DETECTION = 3; // consecutive samples over threshold
float scale = numSamples ? 1.0f / numSamples : 0.0f;
int32_t loudness = 0;
isClipping = false;
bool isClipping = false;
if (numChannels == 2) {
int32_t oversLeft = 0;
int32_t oversRight = 0;
int oversLeft = 0;
int oversRight = 0;
for (int i = 0; i < numSamples/2; i++) {
int32_t left = std::abs((int32_t)samples[2*i+0]);
int32_t right = std::abs((int32_t)samples[2*i+1]);
loudness += left;
loudness += right;
if (left > CLIPPING_THRESHOLD) {
isClipping |= (++oversLeft >= CLIPPING_DETECTION);
} else {
@ -203,13 +197,11 @@ static float computeLoudness(int16_t* samples, int numSamples, int numChannels,
}
}
} else {
int32_t overs = 0;
int overs = 0;
for (int i = 0; i < numSamples; i++) {
int32_t sample = std::abs((int32_t)samples[i]);
loudness += sample;
if (sample > CLIPPING_THRESHOLD) {
isClipping |= (++overs >= CLIPPING_DETECTION);
} else {
@ -218,6 +210,17 @@ static float computeLoudness(int16_t* samples, int numSamples, int numChannels,
}
}
return isClipping;
}
static float computeLoudness(int16_t* samples, int numSamples) {
float scale = numSamples ? 1.0f / numSamples : 0.0f;
int32_t loudness = 0;
for (int i = 0; i < numSamples; i++) {
loudness += std::abs((int32_t)samples[i]);
}
return (float)loudness * scale;
}
@ -1404,6 +1407,15 @@ void AudioClient::handleMicAudioInput() {
_inputRingBuffer.readSamples(inputAudioSamples.get(), inputSamplesRequired);
// detect clipping on the raw input
bool isClipping = detectClipping(inputAudioSamples.get(), inputSamplesRequired, _inputFormat.channelCount());
if (isClipping) {
_timeSinceLastClip = 0.0f;
} else if (_timeSinceLastClip >= 0.0f) {
_timeSinceLastClip += AudioConstants::NETWORK_FRAME_SECS;
}
isClipping = (_timeSinceLastClip >= 0.0f) && (_timeSinceLastClip < 2.0f); // 2 second hold time
#if defined(WEBRTC_ENABLED)
if (_isAECEnabled) {
processWebrtcNearEnd(inputAudioSamples.get(), inputSamplesRequired / _inputFormat.channelCount(),
@ -1411,9 +1423,7 @@ void AudioClient::handleMicAudioInput() {
}
#endif
// detect loudness and clipping on the raw input
bool isClipping = false;
float loudness = computeLoudness(inputAudioSamples.get(), inputSamplesRequired, _inputFormat.channelCount(), isClipping);
float loudness = computeLoudness(inputAudioSamples.get(), inputSamplesRequired);
_lastRawInputLoudness = loudness;
// envelope detection
@ -1421,14 +1431,6 @@ void AudioClient::handleMicAudioInput() {
loudness += tc * (_lastSmoothedRawInputLoudness - loudness);
_lastSmoothedRawInputLoudness = loudness;
// clipping indicator
if (isClipping) {
_timeSinceLastClip = 0.0f;
} else if (_timeSinceLastClip >= 0.0f) {
_timeSinceLastClip += AudioConstants::NETWORK_FRAME_SECS;
}
isClipping = (_timeSinceLastClip >= 0.0f) && (_timeSinceLastClip < 2.0f); // 2 second hold time
emit inputLoudnessChanged(_lastSmoothedRawInputLoudness, isClipping);
if (!_muted) {