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/agc2/adaptive_digital_gain_controller.h
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@ -13,9 +13,9 @@
#include <vector>
#include "api/audio/audio_processing.h"
#include "api/audio/audio_view.h"
#include "modules/audio_processing/agc2/gain_applier.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "modules/audio_processing/include/audio_processing.h"
namespace webrtc {
@ -46,7 +46,7 @@ class AdaptiveDigitalGainController {
// Analyzes `info`, updates the digital gain and applies it to a 10 ms
// `frame`. Supports any sample rate supported by APM.
void Process(const FrameInfo& info, AudioFrameView<float> frame);
void Process(const FrameInfo& info, DeinterleavedView<float> frame);
private:
ApmDataDumper* const apm_data_dumper_;