webrtc/modules/audio_processing/include/audio_processing_statistics.h
Mirko Bonadei 99eea42fc1 Reland "Reland "Export symbols needed by the Chromium component build (part 1).""
This reverts commit b49520bfc0.

Reason for revert: Problem fixed in https://chromium-review.googlesource.com/c/chromium/src/+/1261398.

Original change's description:
> Revert "Reland "Export symbols needed by the Chromium component build (part 1).""
> 
> This reverts commit 588f4642d1.
> 
> Reason for revert: Breaks WebRTC Chromium FYI Win Builder (dbg).
> lld-link: error: undefined symbol: "__declspec(dllimport) __thiscall webrtc::Config::Config(void)" (__imp_??0Config@webrtc@@QAE@XZ)
> [...]
> 
> Original change's description:
> > Reland "Export symbols needed by the Chromium component build (part 1)."
> > 
> > This reverts commit 2ea9af2275.
> > 
> > Reason for revert: The problem will be fixed by
> > https://chromium-review.googlesource.com/c/chromium/src/+/1261122.
> > 
> > Original change's description:
> > > Revert "Export symbols needed by the Chromium component build (part 1)."
> > > 
> > > This reverts commit 9e24dcff16.
> > > 
> > > 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}
> > 
> > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
> > 
> > Change-Id: I83bbc7f550fc23e823c4d055e0a6f60c828960dd
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:9419
> > Reviewed-on: https://webrtc-review.googlesource.com/c/103740
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24980}
> 
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
> 
> Change-Id: I4b7cfe492f2c8eeda5c8ac52520e0cfc95ade9b0
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9419
> Reviewed-on: https://webrtc-review.googlesource.com/c/103801
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24983}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:9419
Change-Id: Id986a0a03cdc2818690337784396882af067f7fa
Reviewed-on: https://webrtc-review.googlesource.com/c/104602
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25049}
2018-10-08 12:54:06 +00:00

57 lines
2.4 KiB
C++

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