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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_APPLIER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
#include "modules/audio_processing/agc2/gain_applier.h"
#include "modules/audio_processing/agc2/vad_with_level.h"
#include "modules/audio_processing/include/audio_frame_view.h"
namespace webrtc {
class ApmDataDumper;
// Part of the adaptive digital controller that applies a digital adaptive gain.
// The gain is updated towards a target. The logic decides when gain updates are
// allowed, it controls the adaptation speed and caps the target based on the
// estimated noise level and the speech level estimate confidence.
class AdaptiveDigitalGainApplier {
public:
// Information about a frame to process.
struct FrameInfo {
float input_level_dbfs; // Estimated speech plus noise level.
float input_noise_level_dbfs; // Estimated noise level.
VadLevelAnalyzer::Result vad_result;
float limiter_envelope_dbfs; // Envelope level from the limiter.
bool estimate_is_confident;
};
// Ctor.
// `adjacent_speech_frames_threshold` indicates how many speech frames are
// required before a gain increase is allowed. `max_gain_change_db_per_second`
// limits the adaptation speed (uniformly operated across frames).
// `max_output_noise_level_dbfs` limits the output noise level.
AdaptiveDigitalGainApplier(ApmDataDumper* apm_data_dumper,
int adjacent_speech_frames_threshold,
float max_gain_change_db_per_second,
float max_output_noise_level_dbfs);
AdaptiveDigitalGainApplier(const AdaptiveDigitalGainApplier&) = delete;
AdaptiveDigitalGainApplier& operator=(const AdaptiveDigitalGainApplier&) =
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 int adjacent_speech_frames_threshold_;
const float max_gain_change_db_per_10ms_;
const float max_output_noise_level_dbfs_;
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_APPLIER_H_
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