Files
platform-external-webrtc/webrtc/modules/audio_processing/rms_level.cc
Peter Kasting dce40cf804 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 21:52:45 +00:00

62 lines
1.4 KiB
C++

/*
* 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 "webrtc/modules/audio_processing/rms_level.h"
#include <assert.h>
#include <math.h>
namespace webrtc {
static const float kMaxSquaredLevel = 32768 * 32768;
RMSLevel::RMSLevel()
: sum_square_(0),
sample_count_(0) {}
RMSLevel::~RMSLevel() {}
void RMSLevel::Reset() {
sum_square_ = 0;
sample_count_ = 0;
}
void RMSLevel::Process(const int16_t* data, size_t length) {
for (size_t i = 0; i < length; ++i) {
sum_square_ += data[i] * data[i];
}
sample_count_ += length;
}
void RMSLevel::ProcessMuted(size_t length) {
sample_count_ += length;
}
int RMSLevel::RMS() {
if (sample_count_ == 0 || sum_square_ == 0) {
Reset();
return kMinLevel;
}
// Normalize by the max level.
float rms = sum_square_ / (sample_count_ * kMaxSquaredLevel);
// 20log_10(x^0.5) = 10log_10(x)
rms = 10 * log10(rms);
assert(rms <= 0);
if (rms < -kMinLevel)
rms = -kMinLevel;
rms = -rms;
Reset();
return static_cast<int>(rms + 0.5);
}
} // namespace webrtc