chromium/third_party/webrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.h

/*
 *  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_COMPUTE_INTERPOLATED_GAIN_CURVE_H_
#define MODULES_AUDIO_PROCESSING_AGC2_COMPUTE_INTERPOLATED_GAIN_CURVE_H_

#include <array>

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

namespace webrtc {

namespace test {

// Parameters for interpolated gain curve using under-approximation to
// avoid saturation.
//
// The saturation gain is defined in order to let hard-clipping occur for
// those samples having a level that falls in the saturation region. It is an
// upper bound of the actual gain to apply - i.e., that returned by the
// limiter.

// Knee and beyond-knee regions approximation parameters.
// The gain curve is approximated as a piece-wise linear function.
// `approx_params_x_` are the boundaries between adjacent linear pieces,
// `approx_params_m_` and `approx_params_q_` are the slope and the y-intercept
// values of each piece.
struct InterpolatedParameters { … };

InterpolatedParameters ComputeInterpolatedGainCurveApproximationParams();
}  // namespace test
}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AGC2_COMPUTE_INTERPOLATED_GAIN_CURVE_H_