1 /* 2 * Copyright (c) 2019 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 MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_H_ 12 #define MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_H_ 13 14 #include "modules/audio_processing/audio_buffer.h" 15 #include "modules/audio_processing/rms_level.h" 16 17 namespace webrtc { 18 19 // An estimation component used to retrieve level metrics. 20 class LevelEstimator { 21 public: 22 LevelEstimator(); 23 ~LevelEstimator(); 24 25 LevelEstimator(LevelEstimator&) = delete; 26 LevelEstimator& operator=(LevelEstimator&) = delete; 27 28 void ProcessStream(const AudioBuffer& audio); 29 30 // Returns the root mean square (RMS) level in dBFs (decibels from digital 31 // full-scale), or alternately dBov. It is computed over all primary stream 32 // frames since the last call to RMS(). The returned value is positive but 33 // should be interpreted as negative. It is constrained to [0, 127]. 34 // 35 // The computation follows: https://tools.ietf.org/html/rfc6465 36 // with the intent that it can provide the RTP audio level indication. 37 // 38 // Frames passed to ProcessStream() with an |_energy| of zero are considered 39 // to have been muted. The RMS of the frame will be interpreted as -127. RMS()40 int RMS() { return rms_.Average(); } 41 42 private: 43 RmsLevel rms_; 44 }; 45 } // namespace webrtc 46 47 #endif // MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_H_ 48