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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
60 lines
2.4 KiB
C++
60 lines
2.4 KiB
C++
/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
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#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
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#include <stdint.h>
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#include "absl/types/optional.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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// This version of the stats uses Optionals, it will replace the regular
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// AudioProcessingStatistics struct.
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struct RTC_EXPORT AudioProcessingStats {
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AudioProcessingStats();
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AudioProcessingStats(const AudioProcessingStats& other);
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~AudioProcessingStats();
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// AEC Statistics.
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// ERL = 10log_10(P_far / P_echo)
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absl::optional<double> echo_return_loss;
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// ERLE = 10log_10(P_echo / P_out)
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absl::optional<double> echo_return_loss_enhancement;
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// Fraction of time that the AEC linear filter is divergent, in a 1-second
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// non-overlapped aggregation window.
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absl::optional<double> divergent_filter_fraction;
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// The delay metrics consists of the delay median and standard deviation. It
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// also consists of the fraction of delay estimates that can make the echo
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// cancellation perform poorly. The values are aggregated until the first
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// call to |GetStatistics()| and afterwards aggregated and updated every
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// second. Note that if there are several clients pulling metrics from
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// |GetStatistics()| during a session the first call from any of them will
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// change to one second aggregation window for all.
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absl::optional<int32_t> delay_median_ms;
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absl::optional<int32_t> delay_standard_deviation_ms;
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// Residual echo detector likelihood.
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absl::optional<double> residual_echo_likelihood;
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// Maximum residual echo likelihood from the last time period.
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absl::optional<double> residual_echo_likelihood_recent_max;
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// The instantaneous delay estimate produced in the AEC. The unit is in
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// milliseconds and the value is the instantaneous value at the time of the
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// call to |GetStatistics()|.
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absl::optional<int32_t> delay_ms;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
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