webrtc/modules/audio_processing/agc2/limiter.h
Steve Anton 10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00

64 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_LIMITER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
#include <string>
#include <vector>
#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
class ApmDataDumper;
class Limiter {
public:
Limiter(size_t sample_rate_hz,
ApmDataDumper* apm_data_dumper,
std::string histogram_name_prefix);
Limiter(const Limiter& limiter) = delete;
Limiter& operator=(const Limiter& limiter) = delete;
~Limiter();
// Applies limiter and hard-clipping to |signal|.
void Process(AudioFrameView<float> signal);
InterpolatedGainCurve::Stats GetGainCurveStats() const;
// Supported rates must be
// * supported by FixedDigitalLevelEstimator
// * below kMaximalNumberOfSamplesPerChannel*1000/kFrameDurationMs
// so that samples_per_channel fit in the
// per_sample_scaling_factors_ array.
void SetSampleRate(size_t sample_rate_hz);
// Resets the internal state.
void Reset();
float LastAudioLevel() const;
private:
const InterpolatedGainCurve interp_gain_curve_;
FixedDigitalLevelEstimator level_estimator_;
ApmDataDumper* const apm_data_dumper_ = nullptr;
// Work array containing the sub-frame scaling factors to be interpolated.
std::array<float, kSubFramesInFrame + 1> scaling_factors_ = {};
std::array<float, kMaximalNumberOfSamplesPerChannel>
per_sample_scaling_factors_ = {};
float last_scaling_factor_ = 1.f;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_