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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/limiter.h"
#include <algorithm>
#include <array>
#include <cmath>
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/safe_minmax.h"
namespace webrtc {
namespace {
// This constant affects the way scaling factors are interpolated for the first
// sub-frame of a frame. Only in the case in which the first sub-frame has an
// estimated level which is greater than the that of the previous analyzed
// sub-frame, linear interpolation is replaced with a power function which
// reduces the chances of over-shooting (and hence saturation), however reducing
// the fixed gain effectiveness.
constexpr float kAttackFirstSubframeInterpolationPower = 8.0f;
void InterpolateFirstSubframe(float last_factor,
float current_factor,
rtc::ArrayView<float> subframe) {
const int n = rtc::dchecked_cast<int>(subframe.size());
constexpr float p = kAttackFirstSubframeInterpolationPower;
for (int i = 0; i < n; ++i) {
subframe[i] = std::pow(1.f - i / n, p) * (last_factor - current_factor) +
current_factor;
}
}
void ComputePerSampleSubframeFactors(
const std::array<float, kSubFramesInFrame + 1>& scaling_factors,
int samples_per_channel,
rtc::ArrayView<float> per_sample_scaling_factors) {
const int num_subframes = scaling_factors.size() - 1;
const int subframe_size =
rtc::CheckedDivExact(samples_per_channel, num_subframes);
// Handle first sub-frame differently in case of attack.
const bool is_attack = scaling_factors[0] > scaling_factors[1];
if (is_attack) {
InterpolateFirstSubframe(
scaling_factors[0], scaling_factors[1],
rtc::ArrayView<float>(
per_sample_scaling_factors.subview(0, subframe_size)));
}
for (int i = is_attack ? 1 : 0; i < num_subframes; ++i) {
const int subframe_start = i * subframe_size;
const float scaling_start = scaling_factors[i];
const float scaling_end = scaling_factors[i + 1];
const float scaling_diff = (scaling_end - scaling_start) / subframe_size;
for (int j = 0; j < subframe_size; ++j) {
per_sample_scaling_factors[subframe_start + j] =
scaling_start + scaling_diff * j;
}
}
}
void ScaleSamples(rtc::ArrayView<const float> per_sample_scaling_factors,
AudioFrameView<float> signal) {
const int samples_per_channel = signal.samples_per_channel();
RTC_DCHECK_EQ(samples_per_channel, per_sample_scaling_factors.size());
for (int i = 0; i < signal.num_channels(); ++i) {
rtc::ArrayView<float> channel = signal.channel(i);
for (int j = 0; j < samples_per_channel; ++j) {
channel[j] = rtc::SafeClamp(channel[j] * per_sample_scaling_factors[j],
kMinFloatS16Value, kMaxFloatS16Value);
}
}
}
void CheckLimiterSampleRate(int sample_rate_hz) {
// Check that per_sample_scaling_factors_ is large enough.
RTC_DCHECK_LE(sample_rate_hz,
kMaximalNumberOfSamplesPerChannel * 1000 / kFrameDurationMs);
}
} // namespace
Limiter::Limiter(int sample_rate_hz,
ApmDataDumper* apm_data_dumper,
absl::string_view histogram_name)
: interp_gain_curve_(apm_data_dumper, histogram_name),
level_estimator_(sample_rate_hz, apm_data_dumper),
apm_data_dumper_(apm_data_dumper) {
CheckLimiterSampleRate(sample_rate_hz);
}
Limiter::~Limiter() = default;
void Limiter::Process(AudioFrameView<float> signal) {
const std::array<float, kSubFramesInFrame> level_estimate =
level_estimator_.ComputeLevel(signal);
RTC_DCHECK_EQ(level_estimate.size() + 1, scaling_factors_.size());
scaling_factors_[0] = last_scaling_factor_;
std::transform(level_estimate.begin(), level_estimate.end(),
scaling_factors_.begin() + 1, [this](float x) {
return interp_gain_curve_.LookUpGainToApply(x);
});
const int samples_per_channel = signal.samples_per_channel();
RTC_DCHECK_LE(samples_per_channel, kMaximalNumberOfSamplesPerChannel);
auto per_sample_scaling_factors = rtc::ArrayView<float>(
&per_sample_scaling_factors_[0], samples_per_channel);
ComputePerSampleSubframeFactors(scaling_factors_, samples_per_channel,
per_sample_scaling_factors);
ScaleSamples(per_sample_scaling_factors, signal);
last_scaling_factor_ = scaling_factors_.back();
// Dump data for debug.
apm_data_dumper_->DumpRaw("agc2_limiter_last_scaling_factor",
last_scaling_factor_);
apm_data_dumper_->DumpRaw(
"agc2_limiter_region",
static_cast<int>(interp_gain_curve_.get_stats().region));
}
InterpolatedGainCurve::Stats Limiter::GetGainCurveStats() const {
return interp_gain_curve_.get_stats();
}
void Limiter::SetSampleRate(int sample_rate_hz) {
CheckLimiterSampleRate(sample_rate_hz);
level_estimator_.SetSampleRate(sample_rate_hz);
}
void Limiter::Reset() {
level_estimator_.Reset();
}
float Limiter::LastAudioLevel() const {
return level_estimator_.LastAudioLevel();
}
} // namespace webrtc
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