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This reverts commita751f167c6. Reason for revert: dependency in a downstream project removed Original change's description: > Revert "Remove unused APM voice activity detection sub-module" > > This reverts commitb4e06d032e. > > Reason for revert: breaking downstream projects > > Original change's description: > > Remove unused APM voice activity detection sub-module > > > > API changes: > > - webrtc::AudioProcessing::Config::VoiceDetection removed > > - webrtc::AudioProcessingStats::voice_detected deprecated > > - cricket::AudioOptions::typing_detection deprecated > > - webrtc::StatsReport::StatsValueName:: > > kStatsValueNameTypingNoiseState deprecated > > > > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0 > > > > Bug: webrtc:11226,webrtc:11292 > > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Reviewed-by: Björn Terelius <terelius@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35975} > > TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11226,webrtc:11292 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35977} # Not skipping CQ checks because this is a reland. Bug: webrtc:11226,webrtc:11292 Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35984}
67 lines
2.8 KiB
C++
67 lines
2.8 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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// Deprecated.
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// TODO(bugs.webrtc.org/11226): Remove.
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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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