1 /* 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 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_ 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_ 13 14 #include <cstddef> 15 16 #include "webrtc/typedefs.h" 17 18 namespace webrtc { 19 20 // Computes the root mean square (RMS) level in dBFs (decibels from digital 21 // full-scale) of audio data. The computation follows RFC 6465: 22 // https://tools.ietf.org/html/rfc6465 23 // with the intent that it can provide the RTP audio level indication. 24 // 25 // The expected approach is to provide constant-sized chunks of audio to 26 // Process(). When enough chunks have been accumulated to form a packet, call 27 // RMS() to get the audio level indicator for the RTP header. 28 class RMSLevel { 29 public: 30 static const int kMinLevel = 127; 31 32 RMSLevel(); 33 ~RMSLevel(); 34 35 // Can be called to reset internal states, but is not required during normal 36 // operation. 37 void Reset(); 38 39 // Pass each chunk of audio to Process() to accumulate the level. 40 void Process(const int16_t* data, size_t length); 41 42 // If all samples with the given |length| have a magnitude of zero, this is 43 // a shortcut to avoid some computation. 44 void ProcessMuted(size_t length); 45 46 // Computes the RMS level over all data passed to Process() since the last 47 // call to RMS(). The returned value is positive but should be interpreted as 48 // negative as per the RFC. It is constrained to [0, 127]. 49 int RMS(); 50 51 private: 52 float sum_square_; 53 size_t sample_count_; 54 }; 55 56 } // namespace webrtc 57 58 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_ 59 60
