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/*
* Copyright (c) 2021 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/capture_levels_adjuster/audio_samples_scaler.h"
#include <algorithm>
#include "api/array_view.h"
#include "modules/audio_processing/audio_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_minmax.h"
namespace webrtc {
AudioSamplesScaler::AudioSamplesScaler(float initial_gain)
: previous_gain_(initial_gain), target_gain_(initial_gain) {}
void AudioSamplesScaler::Process(AudioBuffer& audio_buffer) {
if (static_cast<int>(audio_buffer.num_frames()) != samples_per_channel_) {
// Update the members depending on audio-buffer length if needed.
RTC_DCHECK_GT(audio_buffer.num_frames(), 0);
samples_per_channel_ = static_cast<int>(audio_buffer.num_frames());
one_by_samples_per_channel_ = 1.f / samples_per_channel_;
}
if (target_gain_ == 1.f && previous_gain_ == target_gain_) {
// If only a gain of 1 is to be applied, do an early return without applying
// any gain.
return;
}
float gain = previous_gain_;
if (previous_gain_ == target_gain_) {
// Apply a non-changing gain.
for (size_t channel = 0; channel < audio_buffer.num_channels(); ++channel) {
rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
samples_per_channel_);
for (float& sample : channel_view) {
sample *= gain;
}
}
} else {
const float increment =
(target_gain_ - previous_gain_) * one_by_samples_per_channel_;
if (increment > 0.f) {
// Apply an increasing gain.
for (size_t channel = 0; channel < audio_buffer.num_channels();
++channel) {
gain = previous_gain_;
rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
samples_per_channel_);
for (float& sample : channel_view) {
gain = std::min(gain + increment, target_gain_);
sample *= gain;
}
}
} else {
// Apply a decreasing gain.
for (size_t channel = 0; channel < audio_buffer.num_channels();
++channel) {
gain = previous_gain_;
rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
samples_per_channel_);
for (float& sample : channel_view) {
gain = std::max(gain + increment, target_gain_);
sample *= gain;
}
}
}
}
previous_gain_ = target_gain_;
// Saturate the samples to be in the S16 range.
for (size_t channel = 0; channel < audio_buffer.num_channels(); ++channel) {
rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
samples_per_channel_);
for (float& sample : channel_view) {
constexpr float kMinFloatS16Value = -32768.f;
constexpr float kMaxFloatS16Value = 32767.f;
sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
}
}
}
} // namespace webrtc