103 lines
3.3 KiB
C++
103 lines
3.3 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/agc2/gain_applier.h"
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#include "api/array_view.h"
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#include "modules/audio_processing/agc2/agc2_common.h"
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#include "rtc_base/numerics/safe_minmax.h"
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namespace webrtc {
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namespace {
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// Returns true when the gain factor is so close to 1 that it would
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// not affect int16 samples.
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bool GainCloseToOne(float gain_factor) {
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return 1.f - 1.f / kMaxFloatS16Value <= gain_factor &&
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gain_factor <= 1.f + 1.f / kMaxFloatS16Value;
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}
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void ClipSignal(AudioFrameView<float> signal) {
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for (size_t k = 0; k < signal.num_channels(); ++k) {
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rtc::ArrayView<float> channel_view = signal.channel(k);
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for (auto& sample : channel_view) {
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sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
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}
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}
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}
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void ApplyGainWithRamping(float last_gain_linear,
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float gain_at_end_of_frame_linear,
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float inverse_samples_per_channel,
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AudioFrameView<float> float_frame) {
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// Do not modify the signal.
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if (last_gain_linear == gain_at_end_of_frame_linear &&
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GainCloseToOne(gain_at_end_of_frame_linear)) {
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return;
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}
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// Gain is constant and different from 1.
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if (last_gain_linear == gain_at_end_of_frame_linear) {
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for (size_t k = 0; k < float_frame.num_channels(); ++k) {
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rtc::ArrayView<float> channel_view = float_frame.channel(k);
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for (auto& sample : channel_view) {
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sample *= gain_at_end_of_frame_linear;
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}
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}
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return;
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}
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// The gain changes. We have to change slowly to avoid discontinuities.
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const float increment = (gain_at_end_of_frame_linear - last_gain_linear) *
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inverse_samples_per_channel;
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float gain = last_gain_linear;
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for (size_t i = 0; i < float_frame.samples_per_channel(); ++i) {
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for (size_t ch = 0; ch < float_frame.num_channels(); ++ch) {
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float_frame.channel(ch)[i] *= gain;
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}
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gain += increment;
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}
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}
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} // namespace
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GainApplier::GainApplier(bool hard_clip_samples, float initial_gain_factor)
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: hard_clip_samples_(hard_clip_samples),
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last_gain_factor_(initial_gain_factor),
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current_gain_factor_(initial_gain_factor) {}
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void GainApplier::ApplyGain(AudioFrameView<float> signal) {
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if (static_cast<int>(signal.samples_per_channel()) != samples_per_channel_) {
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Initialize(signal.samples_per_channel());
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}
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ApplyGainWithRamping(last_gain_factor_, current_gain_factor_,
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inverse_samples_per_channel_, signal);
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last_gain_factor_ = current_gain_factor_;
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if (hard_clip_samples_) {
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ClipSignal(signal);
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}
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}
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void GainApplier::SetGainFactor(float gain_factor) {
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RTC_DCHECK_GT(gain_factor, 0.f);
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current_gain_factor_ = gain_factor;
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}
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void GainApplier::Initialize(size_t samples_per_channel) {
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RTC_DCHECK_GT(samples_per_channel, 0);
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samples_per_channel_ = static_cast<int>(samples_per_channel);
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inverse_samples_per_channel_ = 1.f / samples_per_channel_;
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}
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} // namespace webrtc
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