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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7
webrtc/modules/audio_processing/agc/agc.h
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@ -13,6 +13,7 @@
#include <memory>
#include "api/array_view.h"
#include "modules/audio_processing/vad/voice_activity_detector.h"
namespace webrtc {
@ -24,13 +25,13 @@ class Agc {
Agc();
virtual ~Agc();
// |audio| must be mono; in a multi-channel stream, provide the first (usually
// `audio` must be mono; in a multi-channel stream, provide the first (usually
// left) channel.
virtual void Process(const int16_t* audio, size_t length, int sample_rate_hz);
virtual void Process(rtc::ArrayView<const int16_t> audio);
// Retrieves the difference between the target RMS level and the current
// signal RMS level in dB. Returns true if an update is available and false
// otherwise, in which case |error| should be ignored and no action taken.
// otherwise, in which case `error` should be ignored and no action taken.
virtual bool GetRmsErrorDb(int* error);
virtual void Reset();