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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_ |
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_ | 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_ |
13 | 13 |
| 14 #include <cstddef> |
| 15 |
14 #include "webrtc/typedefs.h" | 16 #include "webrtc/typedefs.h" |
15 | 17 |
16 namespace webrtc { | 18 namespace webrtc { |
17 | 19 |
18 // Computes the root mean square (RMS) level in dBFs (decibels from digital | 20 // Computes the root mean square (RMS) level in dBFs (decibels from digital |
19 // full-scale) of audio data. The computation follows RFC 6465: | 21 // full-scale) of audio data. The computation follows RFC 6465: |
20 // https://tools.ietf.org/html/rfc6465 | 22 // https://tools.ietf.org/html/rfc6465 |
21 // with the intent that it can provide the RTP audio level indication. | 23 // with the intent that it can provide the RTP audio level indication. |
22 // | 24 // |
23 // The expected approach is to provide constant-sized chunks of audio to | 25 // The expected approach is to provide constant-sized chunks of audio to |
24 // Process(). When enough chunks have been accumulated to form a packet, call | 26 // Process(). When enough chunks have been accumulated to form a packet, call |
25 // RMS() to get the audio level indicator for the RTP header. | 27 // RMS() to get the audio level indicator for the RTP header. |
26 class RMSLevel { | 28 class RMSLevel { |
27 public: | 29 public: |
28 static const int kMinLevel = 127; | 30 static const int kMinLevel = 127; |
29 | 31 |
30 RMSLevel(); | 32 RMSLevel(); |
31 ~RMSLevel(); | 33 ~RMSLevel(); |
32 | 34 |
33 // Can be called to reset internal states, but is not required during normal | 35 // Can be called to reset internal states, but is not required during normal |
34 // operation. | 36 // operation. |
35 void Reset(); | 37 void Reset(); |
36 | 38 |
37 // Pass each chunk of audio to Process() to accumulate the level. | 39 // Pass each chunk of audio to Process() to accumulate the level. |
38 void Process(const int16_t* data, int length); | 40 void Process(const int16_t* data, size_t length); |
39 | 41 |
40 // If all samples with the given |length| have a magnitude of zero, this is | 42 // If all samples with the given |length| have a magnitude of zero, this is |
41 // a shortcut to avoid some computation. | 43 // a shortcut to avoid some computation. |
42 void ProcessMuted(int length); | 44 void ProcessMuted(size_t length); |
43 | 45 |
44 // Computes the RMS level over all data passed to Process() since the last | 46 // Computes the RMS level over all data passed to Process() since the last |
45 // call to RMS(). The returned value is positive but should be interpreted as | 47 // call to RMS(). The returned value is positive but should be interpreted as |
46 // negative as per the RFC. It is constrained to [0, 127]. | 48 // negative as per the RFC. It is constrained to [0, 127]. |
47 int RMS(); | 49 int RMS(); |
48 | 50 |
49 private: | 51 private: |
50 float sum_square_; | 52 float sum_square_; |
51 int sample_count_; | 53 size_t sample_count_; |
52 }; | 54 }; |
53 | 55 |
54 } // namespace webrtc | 56 } // namespace webrtc |
55 | 57 |
56 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_ | 58 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_ |
57 | 59 |
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