chromium/third_party/webrtc/modules/audio_processing/agc2/gain_applier.cc

/*
 *  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.
 */

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

#include "api/audio/audio_view.h"
#include "modules/audio_processing/agc2/agc2_common.h"
#include "rtc_base/numerics/safe_minmax.h"

namespace webrtc {
namespace {

// Returns true when the gain factor is so close to 1 that it would
// not affect int16 samples.
bool GainCloseToOne(float gain_factor) { … }

void ClipSignal(DeinterleavedView<float> signal) { … }

void ApplyGainWithRamping(float last_gain_linear,
                          float gain_at_end_of_frame_linear,
                          float inverse_samples_per_channel,
                          DeinterleavedView<float> float_frame) { … }

}  // namespace

GainApplier::GainApplier(bool hard_clip_samples, float initial_gain_factor)
    : … { … }

void GainApplier::ApplyGain(DeinterleavedView<float> signal) { … }

// TODO(bugs.webrtc.org/7494): Remove once switched to gains in dB.
void GainApplier::SetGainFactor(float gain_factor) { … }

void GainApplier::Initialize(int samples_per_channel) { … }

}  // namespace webrtc