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0b489303d2
Bug: webrtc:12338 Change-Id: I85bff694dd2ead83c939c4d1945eff82e1296001 No-Presubmit: True Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227161 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34690}
64 lines
2.1 KiB
C++
64 lines
2.1 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
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#include <string>
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#include <vector>
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#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
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#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
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#include "modules/audio_processing/include/audio_frame_view.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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class ApmDataDumper;
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class Limiter {
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public:
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Limiter(size_t sample_rate_hz,
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ApmDataDumper* apm_data_dumper,
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const std::string& histogram_name_prefix);
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Limiter(const Limiter& limiter) = delete;
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Limiter& operator=(const Limiter& limiter) = delete;
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~Limiter();
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// Applies limiter and hard-clipping to `signal`.
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void Process(AudioFrameView<float> signal);
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InterpolatedGainCurve::Stats GetGainCurveStats() const;
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// Supported rates must be
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// * supported by FixedDigitalLevelEstimator
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// * below kMaximalNumberOfSamplesPerChannel*1000/kFrameDurationMs
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// so that samples_per_channel fit in the
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// per_sample_scaling_factors_ array.
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void SetSampleRate(size_t sample_rate_hz);
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// Resets the internal state.
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void Reset();
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float LastAudioLevel() const;
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private:
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const InterpolatedGainCurve interp_gain_curve_;
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FixedDigitalLevelEstimator level_estimator_;
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ApmDataDumper* const apm_data_dumper_ = nullptr;
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// Work array containing the sub-frame scaling factors to be interpolated.
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std::array<float, kSubFramesInFrame + 1> scaling_factors_ = {};
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std::array<float, kMaximalNumberOfSamplesPerChannel>
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per_sample_scaling_factors_ = {};
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float last_scaling_factor_ = 1.f;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
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