fixed_digital_level_estimator.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_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_

#define MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_

#include <array>

#include <vector>

#include "modules/audio_processing/agc2/agc2_common.h"

#include "modules/audio_processing/include/audio_frame_view.h"

namespace webrtc {

class ApmDataDumper;

// Produces a smooth signal level estimate from an input audio

// stream. The estimate smoothing is done through exponential

// filtering.

class FixedDigitalLevelEstimator {

 public:

  // `samples_per_channel` is expected to be derived from this formula:

  //   sample_rate_hz * kFrameDurationMs / 1000

  // or, for a 10ms duration:

  //   sample_rate_hz / 100

  // I.e. the number of samples for 10ms of the given sample rate. The

  // expectation is that samples per channel is divisible by

  // kSubFramesInSample. For kFrameDurationMs=10 and

  // kSubFramesInSample=20, this means that the original sample rate has to be

  // divisible by 2000 and therefore `samples_per_channel` by 20.

  FixedDigitalLevelEstimator(size_t samples_per_channel,

                             ApmDataDumper* apm_data_dumper);

  FixedDigitalLevelEstimator(const FixedDigitalLevelEstimator&) = delete;

  FixedDigitalLevelEstimator& operator=(const FixedDigitalLevelEstimator&) =

      delete;

  // The input is assumed to be in FloatS16 format. Scaled input will

  // produce similarly scaled output. A frame of with kFrameDurationMs

  // ms of audio produces a level estimates in the same scale. The

  // level estimate contains kSubFramesInFrame values.

  std::array<float, kSubFramesInFrame> ComputeLevel(

      DeinterleavedView<const float> float_frame);

  // Rate may be changed at any time (but not concurrently) from the

  // value passed to the constructor. The class is not thread safe.

  void SetSamplesPerChannel(size_t samples_per_channel);

  // Resets the level estimator internal state.

  void Reset();

  float LastAudioLevel() const { return filter_state_level_; }

 private:

  void CheckParameterCombination();

  ApmDataDumper* const apm_data_dumper_ = nullptr;

  float filter_state_level_;

  int samples_in_frame_;

  int samples_in_sub_frame_;

};

}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_