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4db667be74
This adds a second (!) VoiceDetection instance in APM, activated via webrtc::AudioProcessing::Config and which reports its values in the webrtc::AudioProcessingStats struct. The alternative is to reuse the existing instance, but that would require adding a proxy interface returned by AudioProcessing::voice_detection() to update the internal config of AudioProcessingImpl when calling voice_detection()->Enable(). Complexity-wise, no reasonable client will enable both interfaces simultaneously, so the footprint is negligible. Bug: webrtc:9947 Change-Id: I7d8e28b9bf06abab8f9c6822424bdb9d803b987d Reviewed-on: https://webrtc-review.googlesource.com/c/115243 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26101}
73 lines
3.1 KiB
C++
73 lines
3.1 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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// The root mean square (RMS) level in dBFS (decibels from digital
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// full-scale) of the last capture frame, after processing. It is
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// constrained to [-127, 0].
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// The computation follows: https://tools.ietf.org/html/rfc6465
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// with the intent that it can provide the RTP audio level indication.
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// Only reported if level estimation is enabled in AudioProcessing::Config.
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absl::optional<int> output_rms_dbfs;
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// True if voice is detected in the last capture frame, after processing.
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// It is conservative in flagging audio as speech, with low likelihood of
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// incorrectly flagging a frame as voice.
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// Only reported if voice detection is enabled in AudioProcessing::Config.
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absl::optional<bool> voice_detected;
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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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