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Use backticks not vertical bars to denote variables in comments for /modules/audio_processing

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Bug: webrtc:12338
Change-Id: I85bff694dd2ead83c939c4d1945eff82e1296001
No-Presubmit: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227161
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34690}
This commit is contained in:
Artem Titov
2021-07-28 20:50:03 +02:00
committed by WebRTC LUCI CQ
parent dc6801c618
commit 0b489303d2
102 changed files with 483 additions and 483 deletions
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6
modules/audio_processing/include/audio_processing_statistics.h
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@ -50,9 +50,9 @@ struct RTC_EXPORT AudioProcessingStats {
// 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
// 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
// `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;
@ -64,7 +64,7 @@ struct RTC_EXPORT AudioProcessingStats {
// 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()|.
// call to `GetStatistics()`.
absl::optional<int32_t> delay_ms;
};