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Merge pull request #16434 from kencooke/audio-webrtc-warning-spam

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DEV-2619: WebRTC spamming logs
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Shannon Romano 2019-11-01 10:22:35 -07:00 committed by GitHub
commit 295c37aea9
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GPG key ID: 4AEE18F83AFDEB23
1 changed files with 13 additions and 17 deletions
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30
libraries/audio-client/src/AudioClient.cpp
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@ -1187,12 +1187,12 @@ void AudioClient::processWebrtcFarEnd(const int16_t* samples, int numFrames, int
const webrtc::StreamConfig streamConfig = webrtc::StreamConfig(sampleRate, numChannels);
const int numChunk = (int)streamConfig.num_frames();
if (sampleRate > WEBRTC_SAMPLE_RATE_MAX) {
qCWarning(audioclient) << "WebRTC does not support" << sampleRate << "output sample rate.";
return;
}
if (numChannels > WEBRTC_CHANNELS_MAX) {
qCWarning(audioclient) << "WebRTC does not support" << numChannels << "output channels.";
static int32_t lastWarningHash = 0;
if (sampleRate > WEBRTC_SAMPLE_RATE_MAX || numChannels > WEBRTC_CHANNELS_MAX) {
if (lastWarningHash != ((sampleRate << 8) | numChannels)) {
lastWarningHash = ((sampleRate << 8) | numChannels);
qCWarning(audioclient) << "AEC not unsupported for output format: sampleRate =" << sampleRate << "numChannels =" << numChannels;
}
return;
}
@ -1227,18 +1227,14 @@ void AudioClient::processWebrtcFarEnd(const int16_t* samples, int numFrames, int
void AudioClient::processWebrtcNearEnd(int16_t* samples, int numFrames, int numChannels, int sampleRate) {
const webrtc::StreamConfig streamConfig = webrtc::StreamConfig(sampleRate, numChannels);
const int numChunk = (int)streamConfig.num_frames();
assert(numFrames == (int)streamConfig.num_frames()); // WebRTC requires exactly 10ms of input
if (sampleRate > WEBRTC_SAMPLE_RATE_MAX) {
qCWarning(audioclient) << "WebRTC does not support" << sampleRate << "input sample rate.";
return;
}
if (numChannels > WEBRTC_CHANNELS_MAX) {
qCWarning(audioclient) << "WebRTC does not support" << numChannels << "input channels.";
return;
}
if (numFrames != numChunk) {
qCWarning(audioclient) << "WebRTC requires exactly 10ms of input.";
static int32_t lastWarningHash = 0;
if (sampleRate > WEBRTC_SAMPLE_RATE_MAX || numChannels > WEBRTC_CHANNELS_MAX) {
if (lastWarningHash != ((sampleRate << 8) | numChannels)) {
lastWarningHash = ((sampleRate << 8) | numChannels);
qCWarning(audioclient) << "AEC not unsupported for input format: sampleRate =" << sampleRate << "numChannels =" << numChannels;
}
return;
}