AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M131 release

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Ongoing fixes and improvements, transient suppressor is gone. Also,
dropping isac because it doesn't seem to be useful, and is just build
system deadweight now.

Upstream references:

  Version: 131.0.6778.200
  WebRTC: 79aff54b0fa9238ce3518dd9eaf9610cd6f22e82
  Chromium: 2a19506ad24af755f2a215a4c61f775393e0db42
This commit is contained in:
Arun Raghavan
2024-12-24 19:32:07 -05:00
parent 8bdb53d91c
commit b5c48b97f6
263 changed files with 4628 additions and 20416 deletions
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4
webrtc/modules/audio_processing/agc2/saturation_protector_buffer.h
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@ -12,8 +12,8 @@
#define MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_BUFFER_H_
#include <array>
#include <optional>
#include "absl/types/optional.h"
#include "modules/audio_processing/agc2/agc2_common.h"
namespace webrtc {
@ -43,7 +43,7 @@ class SaturationProtectorBuffer {
// Returns the oldest item in the buffer. Returns an empty value if the
// buffer is empty.
absl::optional<float> Front() const;
std::optional<float> Front() const;
private:
int FrontIndex() const;