webrtc-audio-processing/webrtc/api/audio/audio_processing_statistics.h
Arun Raghavan b5c48b97f6 Bump to WebRTC M131 release
Ongoing fixes and improvements, transient suppressor is gone. Also,
dropping isac because it doesn't seem to be useful, and is just build
system deadweight now.

Upstream references:

  Version: 131.0.6778.200
  WebRTC: 79aff54b0fa9238ce3518dd9eaf9610cd6f22e82
  Chromium: 2a19506ad24af755f2a215a4c61f775393e0db42
2024-12-26 12:55:16 -05:00

69 lines
2.7 KiB
C++

/*
* 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 API_AUDIO_AUDIO_PROCESSING_STATISTICS_H_
#define API_AUDIO_AUDIO_PROCESSING_STATISTICS_H_
#include <stdint.h>
#include <optional>
#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();
// Deprecated.
// TODO(bugs.webrtc.org/11226): Remove.
// True if voice is detected in the last capture frame, after processing.
// It is conservative in flagging audio as speech, with low likelihood of
// incorrectly flagging a frame as voice.
// Only reported if voice detection is enabled in AudioProcessing::Config.
std::optional<bool> voice_detected;
// AEC Statistics.
// ERL = 10log_10(P_far / P_echo)
std::optional<double> echo_return_loss;
// ERLE = 10log_10(P_echo / P_out)
std::optional<double> echo_return_loss_enhancement;
// Fraction of time that the AEC linear filter is divergent, in a 1-second
// non-overlapped aggregation window.
std::optional<double> 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.
std::optional<int32_t> delay_median_ms;
std::optional<int32_t> delay_standard_deviation_ms;
// Residual echo detector likelihood.
std::optional<double> residual_echo_likelihood;
// Maximum residual echo likelihood from the last time period.
std::optional<double> 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()`.
std::optional<int32_t> delay_ms;
};
} // namespace webrtc
#endif // API_AUDIO_AUDIO_PROCESSING_STATISTICS_H_