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/*
* Copyright (c) 2014 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/intelligibility/intelligibility_utils.h"
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include <algorithm>
#include <limits>
#include "rtc_base/numerics/safe_minmax.h"
namespace webrtc {
namespace intelligibility {
namespace {
const float kMinFactor = 0.01f;
const float kMaxFactor = 100.f;
// Return |current| changed towards |target|, with the relative change being at
// most |limit|.
float UpdateFactor(float target, float current, float limit) {
const float gain = target / (current + std::numeric_limits<float>::epsilon());
const float clamped_gain = rtc::SafeClamp(gain, 1 - limit, 1 + limit);
return rtc::SafeClamp(current * clamped_gain, kMinFactor, kMaxFactor);
}
} // namespace
template<typename T>
PowerEstimator<T>::PowerEstimator(size_t num_freqs, float decay)
: power_(num_freqs, 0.f), decay_(decay) {}
template<typename T>
void PowerEstimator<T>::Step(const T* data) {
for (size_t i = 0; i < power_.size(); ++i) {
power_[i] = decay_ * power_[i] +
(1.f - decay_) * std::abs(data[i]) * std::abs(data[i]);
}
}
template class PowerEstimator<float>;
template class PowerEstimator<std::complex<float>>;
GainApplier::GainApplier(size_t freqs, float relative_change_limit)
: num_freqs_(freqs),
relative_change_limit_(relative_change_limit),
target_(freqs, 1.f),
current_(freqs, 1.f) {}
GainApplier::~GainApplier() {}
void GainApplier::Apply(const std::complex<float>* in_block,
std::complex<float>* out_block) {
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
for (size_t i = 0; i < num_freqs_; ++i) {
current_[i] = UpdateFactor(target_[i], current_[i], relative_change_limit_);
out_block[i] = sqrtf(fabsf(current_[i])) * in_block[i];
}
}
DelayBuffer::DelayBuffer(size_t delay, size_t num_channels)
: buffer_(num_channels, std::vector<float>(delay, 0.f)), read_index_(0u) {}
DelayBuffer::~DelayBuffer() {}
void DelayBuffer::Delay(float* const* data, size_t length) {
size_t sample_index = read_index_;
for (size_t i = 0u; i < buffer_.size(); ++i) {
sample_index = read_index_;
for (size_t j = 0u; j < length; ++j) {
float swap = data[i][j];
data[i][j] = buffer_[i][sample_index];
buffer_[i][sample_index] = swap;
if (++sample_index == buffer_.size()) {
sample_index = 0u;
}
}
}
read_index_ = sample_index;
}
} // namespace intelligibility
} // namespace webrtc