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andrew@webrtc.org382c0c22014-05-05 18:22:21 +00001/*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#ifndef MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
12#define MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
andrew@webrtc.org21299d42014-05-14 19:00:59 +000013
Danil Chapovalovdb9f7ab2018-06-19 10:50:11 +020014#include "absl/types/optional.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020015#include "api/array_view.h"
andrew@webrtc.org382c0c22014-05-05 18:22:21 +000016
17namespace webrtc {
18
19// Computes the root mean square (RMS) level in dBFs (decibels from digital
20// full-scale) of audio data. The computation follows RFC 6465:
21// https://tools.ietf.org/html/rfc6465
22// with the intent that it can provide the RTP audio level indication.
23//
24// The expected approach is to provide constant-sized chunks of audio to
henrik.lundin50499422016-11-29 04:26:24 -080025// Analyze(). When enough chunks have been accumulated to form a packet, call
26// Average() to get the audio level indicator for the RTP header.
27class RmsLevel {
andrew@webrtc.org382c0c22014-05-05 18:22:21 +000028 public:
henrik.lundin50499422016-11-29 04:26:24 -080029 struct Levels {
30 int average;
31 int peak;
32 };
andrew@webrtc.org382c0c22014-05-05 18:22:21 +000033
henrik.lundin290d43a2016-11-29 08:09:09 -080034 static constexpr int kMinLevelDb = 127;
35
henrik.lundin50499422016-11-29 04:26:24 -080036 RmsLevel();
37 ~RmsLevel();
andrew@webrtc.org382c0c22014-05-05 18:22:21 +000038
39 // Can be called to reset internal states, but is not required during normal
40 // operation.
41 void Reset();
42
henrik.lundin50499422016-11-29 04:26:24 -080043 // Pass each chunk of audio to Analyze() to accumulate the level.
44 void Analyze(rtc::ArrayView<const int16_t> data);
andrew@webrtc.org382c0c22014-05-05 18:22:21 +000045
46 // If all samples with the given |length| have a magnitude of zero, this is
47 // a shortcut to avoid some computation.
henrik.lundin50499422016-11-29 04:26:24 -080048 void AnalyzeMuted(size_t length);
andrew@webrtc.org382c0c22014-05-05 18:22:21 +000049
henrik.lundin50499422016-11-29 04:26:24 -080050 // Computes the RMS level over all data passed to Analyze() since the last
51 // call to Average(). The returned value is positive but should be interpreted
52 // as negative as per the RFC. It is constrained to [0, 127]. Resets the
53 // internal state to start a new measurement period.
54 int Average();
55
56 // Like Average() above, but also returns the RMS peak value. Resets the
57 // internal state to start a new measurement period.
58 Levels AverageAndPeak();
andrew@webrtc.org382c0c22014-05-05 18:22:21 +000059
60 private:
henrik.lundin50499422016-11-29 04:26:24 -080061 // Compares |block_size| with |block_size_|. If they are different, calls
62 // Reset() and stores the new size.
63 void CheckBlockSize(size_t block_size);
64
andrew@webrtc.org382c0c22014-05-05 18:22:21 +000065 float sum_square_;
Peter Kastingdce40cf2015-08-24 14:52:23 -070066 size_t sample_count_;
henrik.lundin50499422016-11-29 04:26:24 -080067 float max_sum_square_;
Danil Chapovalovdb9f7ab2018-06-19 10:50:11 +020068 absl::optional<size_t> block_size_;
andrew@webrtc.org382c0c22014-05-05 18:22:21 +000069};
70
71} // namespace webrtc
andrew@webrtc.org21299d42014-05-14 19:00:59 +000072
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020073#endif // MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_