adaptive_digital_gain_controller.h

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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 "api/audio/audio_processing.h"

#include "api/audio/audio_view.h"

#include "modules/audio_processing/agc2/gain_applier.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, DeinterleavedView<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_