2014-05-05 18:22:21 +00:00
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/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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2017-09-15 06:47:31 +02:00
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#ifndef MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
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#define MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
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2014-05-14 19:00:59 +00:00
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2018-10-23 12:03:01 +02:00
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#include <stddef.h>
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#include <stdint.h>
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2024-08-29 13:00:40 +00:00
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#include <optional>
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2017-09-15 06:47:31 +02:00
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#include "api/array_view.h"
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2014-05-05 18:22:21 +00:00
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namespace webrtc {
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// Computes the root mean square (RMS) level in dBFs (decibels from digital
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// full-scale) of audio data. The computation follows RFC 6465:
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// https://tools.ietf.org/html/rfc6465
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// with the intent that it can provide the RTP audio level indication.
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//
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// The expected approach is to provide constant-sized chunks of audio to
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2016-11-29 04:26:24 -08:00
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// Analyze(). When enough chunks have been accumulated to form a packet, call
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// Average() to get the audio level indicator for the RTP header.
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class RmsLevel {
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2014-05-05 18:22:21 +00:00
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public:
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2016-11-29 04:26:24 -08:00
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struct Levels {
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int average;
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int peak;
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};
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2014-05-05 18:22:21 +00:00
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2022-05-06 08:58:38 +02:00
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enum : int { kMinLevelDb = 127, kInaudibleButNotMuted = 126 };
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2016-11-29 08:09:09 -08:00
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2016-11-29 04:26:24 -08:00
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RmsLevel();
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~RmsLevel();
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2014-05-05 18:22:21 +00:00
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// Can be called to reset internal states, but is not required during normal
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// operation.
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void Reset();
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2016-11-29 04:26:24 -08:00
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// Pass each chunk of audio to Analyze() to accumulate the level.
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void Analyze(rtc::ArrayView<const int16_t> data);
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2019-08-20 09:19:21 +02:00
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void Analyze(rtc::ArrayView<const float> data);
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2014-05-05 18:22:21 +00:00
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2021-07-28 20:50:03 +02:00
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// If all samples with the given `length` have a magnitude of zero, this is
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2014-05-05 18:22:21 +00:00
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// a shortcut to avoid some computation.
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2016-11-29 04:26:24 -08:00
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void AnalyzeMuted(size_t length);
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// Computes the RMS level over all data passed to Analyze() since the last
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// call to Average(). The returned value is positive but should be interpreted
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// as negative as per the RFC. It is constrained to [0, 127]. Resets the
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// internal state to start a new measurement period.
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int Average();
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2014-05-05 18:22:21 +00:00
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2016-11-29 04:26:24 -08:00
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// Like Average() above, but also returns the RMS peak value. Resets the
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// internal state to start a new measurement period.
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Levels AverageAndPeak();
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2014-05-05 18:22:21 +00:00
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private:
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2021-07-28 20:50:03 +02:00
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// Compares `block_size` with `block_size_`. If they are different, calls
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2016-11-29 04:26:24 -08:00
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// Reset() and stores the new size.
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void CheckBlockSize(size_t block_size);
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2014-05-05 18:22:21 +00:00
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float sum_square_;
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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
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size_t sample_count_;
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2016-11-29 04:26:24 -08:00
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float max_sum_square_;
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2024-08-29 13:00:40 +00:00
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std::optional<size_t> block_size_;
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2014-05-05 18:22:21 +00:00
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};
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} // namespace webrtc
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2014-05-14 19:00:59 +00:00
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2017-09-15 06:47:31 +02:00
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#endif // MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
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