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