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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8
webrtc/modules/audio_processing/agc/loudness_histogram.h
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@ -25,7 +25,7 @@ class LoudnessHistogram {
static LoudnessHistogram* Create();
// Create a sliding LoudnessHistogram, i.e. the histogram represents the last
// |window_size| samples.
// `window_size` samples.
static LoudnessHistogram* Create(int window_size);
~LoudnessHistogram();
@ -49,7 +49,7 @@ class LoudnessHistogram {
LoudnessHistogram();
explicit LoudnessHistogram(int window);
// Find the histogram bin associated with the given |rms|.
// Find the histogram bin associated with the given `rms`.
int GetBinIndex(double rms);
void RemoveOldestEntryAndUpdate();
@ -63,10 +63,10 @@ class LoudnessHistogram {
// Number of times the histogram is updated
int num_updates_;
// Audio content, this should be equal to the sum of the components of
// |bin_count_q10_|.
// `bin_count_q10_`.
int64_t audio_content_q10_;
// LoudnessHistogram of input RMS in Q10 with |kHistSize_| bins. In each
// LoudnessHistogram of input RMS in Q10 with `kHistSize_` bins. In each
// 'Update(),' we increment the associated histogram-bin with the given
// probability. The increment is implemented in Q10 to avoid rounding errors.
int64_t bin_count_q10_[kHistSize];