AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M120 release

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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.
This commit is contained in:
Arun Raghavan
2023-12-12 10:42:58 -05:00
parent 9a202fb8c2
commit c6abf6cd3f
479 changed files with 20900 additions and 11996 deletions
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9
webrtc/modules/audio_processing/ns/noise_suppressor.h
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@ -41,12 +41,21 @@ class NoiseSuppressor {
// Applies noise suppression.
void Process(AudioBuffer* audio);
// Specifies whether the capture output will be used. The purpose of this is
// to allow the noise suppressor to deactivate some of the processing when the
// resulting output is anyway not used, for instance when the endpoint is
// muted.
void SetCaptureOutputUsage(bool capture_output_used) {
capture_output_used_ = capture_output_used;
}
private:
const size_t num_bands_;
const size_t num_channels_;
const SuppressionParams suppression_params_;
int32_t num_analyzed_frames_ = -1;
NrFft fft_;
bool capture_output_used_ = true;
struct ChannelState {
ChannelState(const SuppressionParams& suppression_params, size_t num_bands);