In preparation for a coming refactoring CL, the (fixed) extra saturation margin is now applied into `AdaptiveModeLevelEstimator`. This CL also improves the unit tests by hard-coding its saturation params instead of reading them from a field trial. This reduces the chances of making the test flaky if a default value changes. Tested: Bit-exactness verified with audioproc_f Bug: webrtc:7494 Change-Id: I6765def9887a2f4e55b04d929af754cfecbb1626 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184927 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32172}
73 lines
2.1 KiB
C++
73 lines
2.1 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"
|
|
|
|
namespace webrtc {
|
|
|
|
class ApmDataDumper;
|
|
|
|
class SaturationProtector {
|
|
public:
|
|
explicit SaturationProtector(ApmDataDumper* apm_data_dumper);
|
|
SaturationProtector(ApmDataDumper* apm_data_dumper,
|
|
float initial_saturation_margin_db);
|
|
|
|
void Reset();
|
|
|
|
// Updates the margin 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".
|
|
void UpdateMargin(float speech_peak_dbfs, float speech_level_dbfs);
|
|
|
|
// Returns latest computed margin.
|
|
float margin_db() const { return margin_db_; }
|
|
|
|
void DebugDumpEstimate() const;
|
|
|
|
private:
|
|
// Ring buffer which only supports (i) push back and (ii) read oldest item.
|
|
class RingBuffer {
|
|
public:
|
|
void Reset();
|
|
// Pushes back `v`. If the buffer is full, the oldest item 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:
|
|
std::array<float, kPeakEnveloperBufferSize> buffer_;
|
|
int next_ = 0;
|
|
int size_ = 0;
|
|
};
|
|
|
|
float GetDelayedPeakDbfs() const;
|
|
|
|
ApmDataDumper* apm_data_dumper_;
|
|
// Parameters.
|
|
const float initial_saturation_margin_db_;
|
|
// State.
|
|
float margin_db_;
|
|
RingBuffer peak_delay_buffer_;
|
|
float max_peaks_dbfs_;
|
|
int time_since_push_ms_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
|