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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/fixed_digital_level_estimator.h"
#include <algorithm>
#include <cmath>
#include "api/array_view.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
constexpr float kInitialFilterStateLevel = 0.f;
} // namespace
FixedDigitalLevelEstimator::FixedDigitalLevelEstimator(
size_t sample_rate_hz,
ApmDataDumper* apm_data_dumper)
: apm_data_dumper_(apm_data_dumper),
filter_state_level_(kInitialFilterStateLevel) {
SetSampleRate(sample_rate_hz);
CheckParameterCombination();
RTC_DCHECK(apm_data_dumper_);
apm_data_dumper_->DumpRaw("agc2_level_estimator_samplerate", sample_rate_hz);
}
void FixedDigitalLevelEstimator::CheckParameterCombination() {
RTC_DCHECK_GT(samples_in_frame_, 0);
RTC_DCHECK_LE(kSubFramesInFrame, samples_in_frame_);
RTC_DCHECK_EQ(samples_in_frame_ % kSubFramesInFrame, 0);
RTC_DCHECK_GT(samples_in_sub_frame_, 1);
}
std::array<float, kSubFramesInFrame> FixedDigitalLevelEstimator::ComputeLevel(
const AudioFrameView<const float>& float_frame) {
RTC_DCHECK_GT(float_frame.num_channels(), 0);
RTC_DCHECK_EQ(float_frame.samples_per_channel(), samples_in_frame_);
// Compute max envelope without smoothing.
std::array<float, kSubFramesInFrame> envelope{};
for (size_t channel_idx = 0; channel_idx < float_frame.num_channels();
++channel_idx) {
const auto channel = float_frame.channel(channel_idx);
for (size_t sub_frame = 0; sub_frame < kSubFramesInFrame; ++sub_frame) {
for (size_t sample_in_sub_frame = 0;
sample_in_sub_frame < samples_in_sub_frame_; ++sample_in_sub_frame) {
envelope[sub_frame] =
std::max(envelope[sub_frame],
std::abs(channel[sub_frame * samples_in_sub_frame_ +
sample_in_sub_frame]));
}
}
}
// Make sure envelope increases happen one step earlier so that the
// corresponding *gain decrease* doesn't miss a sudden signal
// increase due to interpolation.
for (size_t sub_frame = 0; sub_frame < kSubFramesInFrame - 1; ++sub_frame) {
if (envelope[sub_frame] < envelope[sub_frame + 1]) {
envelope[sub_frame] = envelope[sub_frame + 1];
}
}
// Add attack / decay smoothing.
for (size_t sub_frame = 0; sub_frame < kSubFramesInFrame; ++sub_frame) {
const float envelope_value = envelope[sub_frame];
if (envelope_value > filter_state_level_) {
envelope[sub_frame] = envelope_value * (1 - kAttackFilterConstant) +
filter_state_level_ * kAttackFilterConstant;
} else {
envelope[sub_frame] = envelope_value * (1 - kDecayFilterConstant) +
filter_state_level_ * kDecayFilterConstant;
}
filter_state_level_ = envelope[sub_frame];
// Dump data for debug.
RTC_DCHECK(apm_data_dumper_);
const auto channel = float_frame.channel(0);
apm_data_dumper_->DumpRaw("agc2_level_estimator_samples",
samples_in_sub_frame_,
&channel[sub_frame * samples_in_sub_frame_]);
apm_data_dumper_->DumpRaw("agc2_level_estimator_level",
envelope[sub_frame]);
}
return envelope;
}
void FixedDigitalLevelEstimator::SetSampleRate(size_t sample_rate_hz) {
samples_in_frame_ = rtc::CheckedDivExact(sample_rate_hz * kFrameDurationMs,
static_cast<size_t>(1000));
samples_in_sub_frame_ =
rtc::CheckedDivExact(samples_in_frame_, kSubFramesInFrame);
CheckParameterCombination();
}
void FixedDigitalLevelEstimator::Reset() {
filter_state_level_ = kInitialFilterStateLevel;
}
} // namespace webrtc
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