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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.
33 lines
1.3 KiB
C++
33 lines
1.3 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_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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