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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6
webrtc/modules/audio_processing/gain_control_impl.h
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@ -15,9 +15,9 @@
#include <stdint.h>
#include <memory>
#include <optional>
#include <vector>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/audio_processing/agc/gain_control.h"
@ -81,8 +81,8 @@ class GainControlImpl : public GainControl {
std::vector<std::unique_ptr<MonoAgcState>> mono_agcs_;
std::vector<int> capture_levels_;
absl::optional<size_t> num_proc_channels_;
absl::optional<int> sample_rate_hz_;
std::optional<size_t> num_proc_channels_;
std::optional<int> sample_rate_hz_;
static int instance_counter_;
};