/* * Copyright 2017 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_ #define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_ #include #include "absl/types/optional.h" #include "rtc_base/system/rtc_export.h" namespace webrtc { // This version of the stats uses Optionals, it will replace the regular // AudioProcessingStatistics struct. struct RTC_EXPORT AudioProcessingStats { AudioProcessingStats(); AudioProcessingStats(const AudioProcessingStats& other); ~AudioProcessingStats(); // AEC Statistics. // ERL = 10log_10(P_far / P_echo) absl::optional echo_return_loss; // ERLE = 10log_10(P_echo / P_out) absl::optional echo_return_loss_enhancement; // Fraction of time that the AEC linear filter is divergent, in a 1-second // non-overlapped aggregation window. absl::optional divergent_filter_fraction; // The delay metrics consists of the delay median and standard deviation. It // also consists of the fraction of delay estimates that can make the echo // cancellation perform poorly. The values are aggregated until the first // call to |GetStatistics()| and afterwards aggregated and updated every // second. Note that if there are several clients pulling metrics from // |GetStatistics()| during a session the first call from any of them will // change to one second aggregation window for all. absl::optional delay_median_ms; absl::optional delay_standard_deviation_ms; // Residual echo detector likelihood. absl::optional residual_echo_likelihood; // Maximum residual echo likelihood from the last time period. absl::optional residual_echo_likelihood_recent_max; // The instantaneous delay estimate produced in the AEC. The unit is in // milliseconds and the value is the instantaneous value at the time of the // call to |GetStatistics()|. absl::optional delay_ms; }; } // namespace webrtc #endif // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_