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/*
* 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_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
#include <vector>
#include "modules/audio_processing/agc2/gain_applier.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "modules/audio_processing/include/audio_processing.h"
namespace webrtc {
class ApmDataDumper;
// Selects the target digital gain, decides when and how quickly to adapt to the
// target and applies the current gain to 10 ms frames.
class AdaptiveDigitalGainController {
public:
// Information about a frame to process.
struct FrameInfo {
float speech_probability; // Probability of speech in the [0, 1] range.
float speech_level_dbfs; // Estimated speech level (dBFS).
bool speech_level_reliable; // True with reliable speech level estimation.
float noise_rms_dbfs; // Estimated noise RMS level (dBFS).
float headroom_db; // Headroom (dB).
// TODO(bugs.webrtc.org/7494): Remove `limiter_envelope_dbfs`.
float limiter_envelope_dbfs; // Envelope level from the limiter (dBFS).
};
AdaptiveDigitalGainController(
ApmDataDumper* apm_data_dumper,
const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
int adjacent_speech_frames_threshold);
AdaptiveDigitalGainController(const AdaptiveDigitalGainController&) = delete;
AdaptiveDigitalGainController& operator=(
const AdaptiveDigitalGainController&) = delete;
// Analyzes `info`, updates the digital gain and applies it to a 10 ms
// `frame`. Supports any sample rate supported by APM.
void Process(const FrameInfo& info, AudioFrameView<float> frame);
private:
ApmDataDumper* const apm_data_dumper_;
GainApplier gain_applier_;
const AudioProcessing::Config::GainController2::AdaptiveDigital config_;
const int adjacent_speech_frames_threshold_;
const float max_gain_change_db_per_10ms_;
int calls_since_last_gain_log_;
int frames_to_gain_increase_allowed_;
float last_gain_db_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
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