Bump to WebRTC M120 release
Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone. We're continuing to carry iSAC even though it's gone upstream, but maybe we'll want to drop that soon.
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@ -21,9 +21,11 @@
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namespace webrtc {
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namespace {
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const int kDefaultLevelDbfs = -18;
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const int kNumAnalysisFrames = 100;
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const double kActivityThreshold = 0.3;
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constexpr int kDefaultLevelDbfs = -18;
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constexpr int kNumAnalysisFrames = 100;
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constexpr double kActivityThreshold = 0.3;
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constexpr int kNum10msFramesInOneSecond = 100;
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constexpr int kMaxSampleRateHz = 384000;
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} // namespace
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@ -35,8 +37,10 @@ Agc::Agc()
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Agc::~Agc() = default;
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void Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
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vad_.ProcessChunk(audio, length, sample_rate_hz);
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void Agc::Process(rtc::ArrayView<const int16_t> audio) {
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const int sample_rate_hz = audio.size() * kNum10msFramesInOneSecond;
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RTC_DCHECK_LE(sample_rate_hz, kMaxSampleRateHz);
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vad_.ProcessChunk(audio.data(), audio.size(), sample_rate_hz);
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const std::vector<double>& rms = vad_.chunkwise_rms();
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const std::vector<double>& probabilities =
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vad_.chunkwise_voice_probabilities();
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@ -48,7 +52,7 @@ void Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
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bool Agc::GetRmsErrorDb(int* error) {
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if (!error) {
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RTC_NOTREACHED();
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RTC_DCHECK_NOTREACHED();
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return false;
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}
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