mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00
980c4601e1
- Bug fix: the desired initial gain quickly dropped to 0 dB hence starting a call with a too low level - New tuning to make AGC2 more robust to VAD mistakes - Smarter max gain increase speed: to deal with an increased threshold of adjacent speech frames, the gain applier temporarily allows a faster gain increase to deal with a longer time spent waiting for enough speech frames in a row to be observed - Saturation protector isolated from `AdaptiveModeLevelEstimator` to simplify the unit tests for the latter (non bit-exact change) - AGC2 adaptive digital config: unnecessary params deprecated - Code readability improvements - Data dumps clean-up and better naming Bug: webrtc:7494 Change-Id: I4e36059bdf2566cc2a7e1a7e95b7430ba9ae9844 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215140 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33736}
59 lines
1.8 KiB
C++
59 lines
1.8 KiB
C++
/*
|
|
* Copyright (c) 2021 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_SATURATION_PROTECTOR_BUFFER_H_
|
|
#define MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_BUFFER_H_
|
|
|
|
#include <array>
|
|
|
|
#include "absl/types/optional.h"
|
|
#include "modules/audio_processing/agc2/agc2_common.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Ring buffer for the saturation protector which only supports (i) push back
|
|
// and (ii) read oldest item.
|
|
class SaturationProtectorBuffer {
|
|
public:
|
|
SaturationProtectorBuffer();
|
|
~SaturationProtectorBuffer();
|
|
|
|
bool operator==(const SaturationProtectorBuffer& b) const;
|
|
inline bool operator!=(const SaturationProtectorBuffer& b) const {
|
|
return !(*this == b);
|
|
}
|
|
|
|
// Maximum number of values that the buffer can contain.
|
|
int Capacity() const;
|
|
|
|
// Number of values in the buffer.
|
|
int Size() const;
|
|
|
|
void Reset();
|
|
|
|
// Pushes back `v`. If the buffer is full, the oldest value is replaced.
|
|
void PushBack(float v);
|
|
|
|
// Returns the oldest item in the buffer. Returns an empty value if the
|
|
// buffer is empty.
|
|
absl::optional<float> Front() const;
|
|
|
|
private:
|
|
int FrontIndex() const;
|
|
// `buffer_` has `size_` elements (up to the size of `buffer_`) and `next_` is
|
|
// the position where the next new value is written in `buffer_`.
|
|
std::array<float, kSaturationProtectorBufferSize> buffer_;
|
|
int next_ = 0;
|
|
int size_ = 0;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_BUFFER_H_
|