1 /* 2 * Copyright 2017 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_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_ 12 #define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_ 13 14 #include <stdint.h> 15 16 #include "absl/types/optional.h" 17 #include "rtc_base/system/rtc_export.h" 18 19 namespace webrtc { 20 // This version of the stats uses Optionals, it will replace the regular 21 // AudioProcessingStatistics struct. 22 struct RTC_EXPORT AudioProcessingStats { 23 AudioProcessingStats(); 24 AudioProcessingStats(const AudioProcessingStats& other); 25 ~AudioProcessingStats(); 26 27 // The root mean square (RMS) level in dBFS (decibels from digital 28 // full-scale) of the last capture frame, after processing. It is 29 // constrained to [-127, 0]. 30 // The computation follows: https://tools.ietf.org/html/rfc6465 31 // with the intent that it can provide the RTP audio level indication. 32 // Only reported if level estimation is enabled in AudioProcessing::Config. 33 absl::optional<int> output_rms_dbfs; 34 35 // True if voice is detected in the last capture frame, after processing. 36 // It is conservative in flagging audio as speech, with low likelihood of 37 // incorrectly flagging a frame as voice. 38 // Only reported if voice detection is enabled in AudioProcessing::Config. 39 absl::optional<bool> voice_detected; 40 41 // AEC Statistics. 42 // ERL = 10log_10(P_far / P_echo) 43 absl::optional<double> echo_return_loss; 44 // ERLE = 10log_10(P_echo / P_out) 45 absl::optional<double> echo_return_loss_enhancement; 46 // Fraction of time that the AEC linear filter is divergent, in a 1-second 47 // non-overlapped aggregation window. 48 absl::optional<double> divergent_filter_fraction; 49 50 // The delay metrics consists of the delay median and standard deviation. It 51 // also consists of the fraction of delay estimates that can make the echo 52 // cancellation perform poorly. The values are aggregated until the first 53 // call to |GetStatistics()| and afterwards aggregated and updated every 54 // second. Note that if there are several clients pulling metrics from 55 // |GetStatistics()| during a session the first call from any of them will 56 // change to one second aggregation window for all. 57 absl::optional<int32_t> delay_median_ms; 58 absl::optional<int32_t> delay_standard_deviation_ms; 59 60 // Residual echo detector likelihood. 61 absl::optional<double> residual_echo_likelihood; 62 // Maximum residual echo likelihood from the last time period. 63 absl::optional<double> residual_echo_likelihood_recent_max; 64 65 // The instantaneous delay estimate produced in the AEC. The unit is in 66 // milliseconds and the value is the instantaneous value at the time of the 67 // call to |GetStatistics()|. 68 absl::optional<int32_t> delay_ms; 69 }; 70 71 } // namespace webrtc 72 73 #endif // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_ 74