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33 lines
1.3 KiB
C
33 lines
1.3 KiB
C
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/*
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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_FIXED_GAIN_CONTROLLER_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_FIXED_GAIN_CONTROLLER_H_
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#include "modules/audio_processing/agc2/limiter.h"
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#include "modules/audio_processing/include/audio_frame_view.h"
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namespace webrtc {
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class ApmDataDumper;
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class FixedGainController {
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public:
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explicit FixedGainController(ApmDataDumper* apm_data_dumper);
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FixedGainController(ApmDataDumper* apm_data_dumper,
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std::string histogram_name_prefix);
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void Process(AudioFrameView<float> signal);
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// Gain and sample rate may be changed at any time (but not
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// concurrently with any other method call).
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void SetGain(float gain_to_apply_db);
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void SetSampleRate(size_t sample_rate_hz);
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float LastAudioLevel() const;
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private:
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float gain_to_apply_ = 1.f;
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ApmDataDumper* apm_data_dumper_ = nullptr;
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Limiter limiter_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_FIXED_GAIN_CONTROLLER_H_
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