Arun Raghavan b5c48b97f6 Bump to WebRTC M131 release
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
2024-12-26 12:55:16 -05:00

51 lines
1.6 KiB
C++

/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
#include <stddef.h>
#include "api/audio/audio_view.h"
#include "modules/audio_processing/include/audio_frame_view.h"
namespace webrtc {
class GainApplier {
public:
GainApplier(bool hard_clip_samples, float initial_gain_factor);
void ApplyGain(DeinterleavedView<float> signal);
void SetGainFactor(float gain_factor);
float GetGainFactor() const { return current_gain_factor_; }
[[deprecated("Use DeinterleavedView<> version")]] void ApplyGain(
AudioFrameView<float> signal) {
ApplyGain(signal.view());
}
private:
void Initialize(int samples_per_channel);
// Whether to clip samples after gain is applied. If 'true', result
// will fit in FloatS16 range.
const bool hard_clip_samples_;
float last_gain_factor_;
// If this value is not equal to 'last_gain_factor', gain will be
// ramped from 'last_gain_factor_' to this value during the next
// 'ApplyGain'.
float current_gain_factor_;
int samples_per_channel_ = -1;
float inverse_samples_per_channel_ = -1.f;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_