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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.
*/
#ifndef MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
#define MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
#include <stddef.h>
#include <stdint.h>
#include <optional>
#include "api/array_view.h"
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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
// Analyze(). When enough chunks have been accumulated to form a packet, call
// Average() to get the audio level indicator for the RTP header.
class RmsLevel {
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public:
struct Levels {
int average;
int peak;
};
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enum : int { kMinLevelDb = 127, kInaudibleButNotMuted = 126 };
RmsLevel();
~RmsLevel();
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// Can be called to reset internal states, but is not required during normal
// operation.
void Reset();
// Pass each chunk of audio to Analyze() to accumulate the level.
void Analyze(rtc::ArrayView<const int16_t> data);
void Analyze(rtc::ArrayView<const float> data);
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// If all samples with the given `length` have a magnitude of zero, this is
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// a shortcut to avoid some computation.
void AnalyzeMuted(size_t length);
// Computes the RMS level over all data passed to Analyze() since the last
// call to Average(). The returned value is positive but should be interpreted
// as negative as per the RFC. It is constrained to [0, 127]. Resets the
// internal state to start a new measurement period.
int Average();
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// Like Average() above, but also returns the RMS peak value. Resets the
// internal state to start a new measurement period.
Levels AverageAndPeak();
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private:
// Compares `block_size` with `block_size_`. If they are different, calls
// Reset() and stores the new size.
void CheckBlockSize(size_t block_size);
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float sum_square_;
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}
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size_t sample_count_;
float max_sum_square_;
std::optional<size_t> block_size_;
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};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_