dce40cf804
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}
60 lines
1.9 KiB
C++
60 lines
1.9 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.
|
|
*/
|
|
|
|
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
|
|
#define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
|
|
|
|
#include <cstddef>
|
|
|
|
#include "webrtc/typedefs.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Computes the root mean square (RMS) level in dBFs (decibels from digital
|
|
// full-scale) of audio data. The computation follows RFC 6465:
|
|
// https://tools.ietf.org/html/rfc6465
|
|
// with the intent that it can provide the RTP audio level indication.
|
|
//
|
|
// The expected approach is to provide constant-sized chunks of audio to
|
|
// Process(). When enough chunks have been accumulated to form a packet, call
|
|
// RMS() to get the audio level indicator for the RTP header.
|
|
class RMSLevel {
|
|
public:
|
|
static const int kMinLevel = 127;
|
|
|
|
RMSLevel();
|
|
~RMSLevel();
|
|
|
|
// Can be called to reset internal states, but is not required during normal
|
|
// operation.
|
|
void Reset();
|
|
|
|
// Pass each chunk of audio to Process() to accumulate the level.
|
|
void Process(const int16_t* data, size_t length);
|
|
|
|
// If all samples with the given |length| have a magnitude of zero, this is
|
|
// a shortcut to avoid some computation.
|
|
void ProcessMuted(size_t length);
|
|
|
|
// Computes the RMS level over all data passed to Process() since the last
|
|
// call to RMS(). The returned value is positive but should be interpreted as
|
|
// negative as per the RFC. It is constrained to [0, 127].
|
|
int RMS();
|
|
|
|
private:
|
|
float sum_square_;
|
|
size_t sample_count_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
|
|
|