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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6
webrtc/modules/audio_processing/rms_level.h
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@ -34,7 +34,7 @@ class RmsLevel {
int peak;
};
enum : int { kMinLevelDb = 127 };
enum : int { kMinLevelDb = 127, kInaudibleButNotMuted = 126 };
RmsLevel();
~RmsLevel();
@ -47,7 +47,7 @@ class RmsLevel {
void Analyze(rtc::ArrayView<const int16_t> data);
void Analyze(rtc::ArrayView<const float> data);
// If all samples with the given |length| have a magnitude of zero, this is
// If all samples with the given `length` have a magnitude of zero, this is
// a shortcut to avoid some computation.
void AnalyzeMuted(size_t length);
@ -62,7 +62,7 @@ class RmsLevel {
Levels AverageAndPeak();
private:
// Compares |block_size| with |block_size_|. If they are different, calls
// Compares `block_size` with `block_size_`. If they are different, calls
// Reset() and stores the new size.
void CheckBlockSize(size_t block_size);