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