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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 "webrtc/rtc_base/array_view.h" | 14 #include "webrtc/base/array_view.h" |
15 #include "webrtc/rtc_base/optional.h" | 15 #include "webrtc/base/optional.h" |
16 #include "webrtc/typedefs.h" | 16 #include "webrtc/typedefs.h" |
17 | 17 |
18 namespace webrtc { | 18 namespace webrtc { |
19 | 19 |
20 // 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 |
21 // full-scale) of audio data. The computation follows RFC 6465: | 21 // full-scale) of audio data. The computation follows RFC 6465: |
22 // https://tools.ietf.org/html/rfc6465 | 22 // https://tools.ietf.org/html/rfc6465 |
23 // 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. |
24 // | 24 // |
25 // 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 |
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66 float sum_square_; | 66 float sum_square_; |
67 size_t sample_count_; | 67 size_t sample_count_; |
68 float max_sum_square_; | 68 float max_sum_square_; |
69 rtc::Optional<size_t> block_size_; | 69 rtc::Optional<size_t> block_size_; |
70 }; | 70 }; |
71 | 71 |
72 } // namespace webrtc | 72 } // namespace webrtc |
73 | 73 |
74 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_ | 74 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_ |
75 | 75 |
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