2ea9af2275
This reverts commit 9e24dcff167c4eb3555bf0ce6eaba090c10fbe53. Reason for revert: Breaks chromium.webrtc.fyi bots. Original change's description: > Export symbols needed by the Chromium component build (part 1). > > This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h) > to mark WebRTC symbols as visible from a shared library, this doesn't > mean these symbols are part of the public API (please continue to refer > to [1] for info about what is considered public WebRTC API). > > [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md > > Bug: webrtc:9419 > Change-Id: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99 > Reviewed-on: https://webrtc-review.googlesource.com/c/103505 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24969} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org Change-Id: I01f6e18f0d2c0f0309cdaa6c943c3927e1f1f49f No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9419 Reviewed-on: https://webrtc-review.googlesource.com/c/103720 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24974}
57 lines
2.3 KiB
C++
57 lines
2.3 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 "absl/types/optional.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 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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