webrtc/modules/audio_processing/agc2/saturation_protector.h
Alessio Bazzica 56f63c3e7e AGC2 AdaptiveModeLevelEstimator min consecutive speech frames (2/3)
This is the second CL needed to add a new `AdaptiveModeLevelEstimator`
feature that makes AGC2 more robus to VAD mistakes: the level estimator
discards estimation updates when too few consecutive speech frames are
observed.

In this CL, the `SaturationProtector` class has been replaced by a
struct that define the state and two functions to change it.
This is done in order to use the saturation protector state in
`AdaptiveModeLevelEstimator::State` and will allow to add a
temporary state in `AdaptiveModeLevelEstimator` (see the child CL).

Tested: Bit-exactness verified with audioproc_f

Bug: webrtc:7494
Change-Id: Ic5ecd1e174010656ed20664ef7b7e5798ebb7978
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185041
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32226}
2020-09-29 11:02:10 +00:00

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2.9 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_SATURATION_PROTECTOR_H_
#define MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
#include <array>
#include "absl/types/optional.h"
#include "modules/audio_processing/agc2/agc2_common.h"
#include "rtc_base/numerics/safe_compare.h"
namespace webrtc {
namespace saturation_protector_impl {
// Ring buffer which only supports (i) push back and (ii) read oldest item.
class RingBuffer {
public:
bool operator==(const RingBuffer& b) const;
inline bool operator!=(const RingBuffer& b) const { return !(*this == b); }
// Maximum number of values that the buffer can contain.
int Capacity() const { return buffer_.size(); }
// Number of values in the buffer.
int Size() const { return size_; }
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:
inline int FrontIndex() const {
return rtc::SafeEq(size_, buffer_.size()) ? next_ : 0;
}
// `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, kPeakEnveloperBufferSize> buffer_;
int next_ = 0;
int size_ = 0;
};
} // namespace saturation_protector_impl
// Saturation protector state. Exposed publicly for check-pointing and restore
// ops.
struct SaturationProtectorState {
bool operator==(const SaturationProtectorState& s) const;
inline bool operator!=(const SaturationProtectorState& s) const {
return !(*this == s);
}
float margin_db; // Recommended margin.
saturation_protector_impl::RingBuffer peak_delay_buffer;
float max_peaks_dbfs;
int time_since_push_ms; // Time since the last ring buffer push operation.
};
// Resets the saturation protector state.
void ResetSaturationProtectorState(float initial_margin_db,
SaturationProtectorState& state);
// Updates `state` by analyzing the estimated speech level `speech_level_dbfs`
// and the peak power `speech_peak_dbfs` for an observed frame which is
// reliably classified as "speech". `state` must not be modified without calling
// this function.
void UpdateSaturationProtectorState(float speech_peak_dbfs,
float speech_level_dbfs,
SaturationProtectorState& state);
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_