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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.
*/
#include "modules/audio_processing/agc2/gain_applier.h"
#include "api/array_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) {
return 1.f - 1.f / kMaxFloatS16Value <= gain_factor &&
gain_factor <= 1.f + 1.f / kMaxFloatS16Value;
}
void ClipSignal(AudioFrameView<float> signal) {
for (int k = 0; k < signal.num_channels(); ++k) {
rtc::ArrayView<float> channel_view = signal.channel(k);
for (auto& sample : channel_view) {
sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
}
}
}
void ApplyGainWithRamping(float last_gain_linear,
float gain_at_end_of_frame_linear,
float inverse_samples_per_channel,
AudioFrameView<float> float_frame) {
// Do not modify the signal.
if (last_gain_linear == gain_at_end_of_frame_linear &&
GainCloseToOne(gain_at_end_of_frame_linear)) {
return;
}
// Gain is constant and different from 1.
if (last_gain_linear == gain_at_end_of_frame_linear) {
for (int k = 0; k < float_frame.num_channels(); ++k) {
rtc::ArrayView<float> channel_view = float_frame.channel(k);
for (auto& sample : channel_view) {
sample *= gain_at_end_of_frame_linear;
}
}
return;
}
// The gain changes. We have to change slowly to avoid discontinuities.
const float increment = (gain_at_end_of_frame_linear - last_gain_linear) *
inverse_samples_per_channel;
float gain = last_gain_linear;
for (int i = 0; i < float_frame.samples_per_channel(); ++i) {
for (int ch = 0; ch < float_frame.num_channels(); ++ch) {
float_frame.channel(ch)[i] *= gain;
}
gain += increment;
}
}
} // namespace
GainApplier::GainApplier(bool hard_clip_samples, float initial_gain_factor)
: hard_clip_samples_(hard_clip_samples),
last_gain_factor_(initial_gain_factor),
current_gain_factor_(initial_gain_factor) {}
void GainApplier::ApplyGain(AudioFrameView<float> signal) {
if (static_cast<int>(signal.samples_per_channel()) != samples_per_channel_) {
Initialize(signal.samples_per_channel());
}
ApplyGainWithRamping(last_gain_factor_, current_gain_factor_,
inverse_samples_per_channel_, signal);
last_gain_factor_ = current_gain_factor_;
if (hard_clip_samples_) {
ClipSignal(signal);
}
}
// TODO(bugs.webrtc.org/7494): Remove once switched to gains in dB.
void GainApplier::SetGainFactor(float gain_factor) {
RTC_DCHECK_GT(gain_factor, 0.f);
current_gain_factor_ = gain_factor;
}
void GainApplier::Initialize(int samples_per_channel) {
RTC_DCHECK_GT(samples_per_channel, 0);
samples_per_channel_ = static_cast<int>(samples_per_channel);
inverse_samples_per_channel_ = 1.f / samples_per_channel_;
}
} // namespace webrtc
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