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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5
webrtc/modules/audio_processing/aec3/echo_audibility.h
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@ -13,7 +13,8 @@
#include <stddef.h>
#include "absl/types/optional.h"
#include <optional>
#include "api/array_view.h"
#include "modules/audio_processing/aec3/block_buffer.h"
#include "modules/audio_processing/aec3/render_buffer.h"
@ -73,7 +74,7 @@ class EchoAudibility {
// values.
bool IsRenderTooLow(const BlockBuffer& block_buffer);
absl::optional<int> render_spectrum_write_prev_;
std::optional<int> render_spectrum_write_prev_;
int render_block_write_prev_;
bool non_zero_render_seen_;
const bool use_render_stationarity_at_init_;