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/*
* Copyright (c) 2016 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/level_controller/gain_applier.h"
#include <algorithm>
#include "api/array_view.h"
#include "rtc_base/checks.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
namespace webrtc {
namespace {
const float kMaxSampleValue = 32767.f;
const float kMinSampleValue = -32767.f;
int CountSaturations(rtc::ArrayView<const float> in) {
return std::count_if(in.begin(), in.end(), [](const float& v) {
return v >= kMaxSampleValue || v <= kMinSampleValue;
});
}
int CountSaturations(const AudioBuffer& audio) {
int num_saturations = 0;
for (size_t k = 0; k < audio.num_channels(); ++k) {
num_saturations += CountSaturations(rtc::ArrayView<const float>(
audio.channels_const_f()[k], audio.num_frames()));
}
return num_saturations;
}
void LimitToAllowedRange(rtc::ArrayView<float> x) {
for (auto& v : x) {
v = std::max(kMinSampleValue, v);
v = std::min(kMaxSampleValue, v);
}
}
void LimitToAllowedRange(AudioBuffer* audio) {
for (size_t k = 0; k < audio->num_channels(); ++k) {
LimitToAllowedRange(
rtc::ArrayView<float>(audio->channels_f()[k], audio->num_frames()));
}
}
float ApplyIncreasingGain(float new_gain,
float old_gain,
float step_size,
rtc::ArrayView<float> x) {
RTC_DCHECK_LT(0.f, step_size);
float gain = old_gain;
for (auto& v : x) {
gain = std::min(new_gain, gain + step_size);
v *= gain;
}
return gain;
}
float ApplyDecreasingGain(float new_gain,
float old_gain,
float step_size,
rtc::ArrayView<float> x) {
RTC_DCHECK_GT(0.f, step_size);
float gain = old_gain;
for (auto& v : x) {
gain = std::max(new_gain, gain + step_size);
v *= gain;
}
return gain;
}
float ApplyConstantGain(float gain, rtc::ArrayView<float> x) {
for (auto& v : x) {
v *= gain;
}
return gain;
}
float ApplyGain(float new_gain,
float old_gain,
float increase_step_size,
float decrease_step_size,
rtc::ArrayView<float> x) {
RTC_DCHECK_LT(0.f, increase_step_size);
RTC_DCHECK_GT(0.f, decrease_step_size);
if (new_gain == old_gain) {
return ApplyConstantGain(new_gain, x);
} else if (new_gain > old_gain) {
return ApplyIncreasingGain(new_gain, old_gain, increase_step_size, x);
} else {
return ApplyDecreasingGain(new_gain, old_gain, decrease_step_size, x);
}
}
} // namespace
GainApplier::GainApplier(ApmDataDumper* data_dumper)
: data_dumper_(data_dumper) {}
void GainApplier::Initialize(int sample_rate_hz) {
RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
sample_rate_hz == AudioProcessing::kSampleRate48kHz);
const float kGainIncreaseStepSize48kHz = 0.0001f;
const float kGainDecreaseStepSize48kHz = -0.01f;
const float kGainSaturatedDecreaseStepSize48kHz = -0.05f;
last_frame_was_saturated_ = false;
old_gain_ = 1.f;
gain_increase_step_size_ =
kGainIncreaseStepSize48kHz *
(static_cast<float>(AudioProcessing::kSampleRate48kHz) / sample_rate_hz);
gain_normal_decrease_step_size_ =
kGainDecreaseStepSize48kHz *
(static_cast<float>(AudioProcessing::kSampleRate48kHz) / sample_rate_hz);
gain_saturated_decrease_step_size_ =
kGainSaturatedDecreaseStepSize48kHz *
(static_cast<float>(AudioProcessing::kSampleRate48kHz) / sample_rate_hz);
}
int GainApplier::Process(float new_gain, AudioBuffer* audio) {
RTC_CHECK_NE(0.f, gain_increase_step_size_);
RTC_CHECK_NE(0.f, gain_normal_decrease_step_size_);
RTC_CHECK_NE(0.f, gain_saturated_decrease_step_size_);
int num_saturations = 0;
if (new_gain != 1.f) {
float last_applied_gain = 1.f;
float gain_decrease_step_size = last_frame_was_saturated_
? gain_saturated_decrease_step_size_
: gain_normal_decrease_step_size_;
for (size_t k = 0; k < audio->num_channels(); ++k) {
last_applied_gain = ApplyGain(
new_gain, old_gain_, gain_increase_step_size_,
gain_decrease_step_size,
rtc::ArrayView<float>(audio->channels_f()[k], audio->num_frames()));
}
num_saturations = CountSaturations(*audio);
LimitToAllowedRange(audio);
old_gain_ = last_applied_gain;
}
data_dumper_->DumpRaw("lc_last_applied_gain", 1, &old_gain_);
return num_saturations;
}
} // namespace webrtc
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