52 lines
1.5 KiB
C
52 lines
1.5 KiB
C
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/*
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* Copyright (c) 2012 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_AGC_AGC_H_
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#define MODULES_AUDIO_PROCESSING_AGC_AGC_H_
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#include <memory>
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#include "modules/audio_processing/vad/voice_activity_detector.h"
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namespace webrtc {
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class LoudnessHistogram;
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class Agc {
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public:
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Agc();
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virtual ~Agc();
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// |audio| must be mono; in a multi-channel stream, provide the first (usually
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// left) channel.
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virtual void Process(const int16_t* audio, size_t length, int sample_rate_hz);
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// Retrieves the difference between the target RMS level and the current
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// signal RMS level in dB. Returns true if an update is available and false
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// otherwise, in which case |error| should be ignored and no action taken.
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virtual bool GetRmsErrorDb(int* error);
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virtual void Reset();
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virtual int set_target_level_dbfs(int level);
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virtual int target_level_dbfs() const;
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virtual float voice_probability() const;
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private:
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double target_level_loudness_;
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int target_level_dbfs_;
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std::unique_ptr<LoudnessHistogram> histogram_;
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std::unique_ptr<LoudnessHistogram> inactive_histogram_;
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VoiceActivityDetector vad_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC_AGC_H_
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