Files
platform-external-webrtc/modules/audio_processing/transient/moving_moments.cc
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

53 lines
1.4 KiB
C++

/*
* Copyright (c) 2013 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/transient/moving_moments.h"
#include <math.h>
#include <string.h>
#include "rtc_base/checks.h"
namespace webrtc {
MovingMoments::MovingMoments(size_t length)
: length_(length),
queue_(),
sum_(0.0),
sum_of_squares_(0.0) {
RTC_DCHECK_GT(length, 0);
for (size_t i = 0; i < length; ++i) {
queue_.push(0.0);
}
}
MovingMoments::~MovingMoments() {}
void MovingMoments::CalculateMoments(const float* in, size_t in_length,
float* first, float* second) {
RTC_DCHECK(in);
RTC_DCHECK_GT(in_length, 0);
RTC_DCHECK(first);
RTC_DCHECK(second);
for (size_t i = 0; i < in_length; ++i) {
const float old_value = queue_.front();
queue_.pop();
queue_.push(in[i]);
sum_ += in[i] - old_value;
sum_of_squares_ += in[i] * in[i] - old_value * old_value;
first[i] = sum_ / length_;
second[i] = sum_of_squares_ / length_;
}
}
} // namespace webrtc