mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-12 09:09:38 +01:00
5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
65 lines
2.1 KiB
C++
65 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/constructormagic.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_
|