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

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Bug: webrtc:12338
Change-Id: Ib97b2c3d64dbd895f261ffa76a2e885bd934a87f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34554}
This commit is contained in:
Artem Titov
2021-07-25 21:50:14 +02:00
committed by WebRTC LUCI CQ
parent 7750d802a5
commit 0e61fdd27c
94 changed files with 455 additions and 455 deletions
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6
api/rtp_receiver_interface.h
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@ -54,7 +54,7 @@ class RTC_EXPORT RtpReceiverInterface : public rtc::RefCountInterface {
// TODO(https://bugs.webrtc.org/907849) remove default implementation
virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const;
// The list of streams that |track| is associated with. This is the same as
// The list of streams that `track` is associated with. This is the same as
// the [[AssociatedRemoteMediaStreams]] internal slot in the spec.
// https://w3c.github.io/webrtc-pc/#dfn-associatedremotemediastreams
// TODO(hbos): Make pure virtual as soon as Chromium's mock implements this.
@ -84,8 +84,8 @@ class RTC_EXPORT RtpReceiverInterface : public rtc::RefCountInterface {
virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
// Sets the jitter buffer minimum delay until media playout. Actual observed
// delay may differ depending on the congestion control. |delay_seconds| is a
// positive value including 0.0 measured in seconds. |nullopt| means default
// delay may differ depending on the congestion control. `delay_seconds` is a
// positive value including 0.0 measured in seconds. `nullopt` means default
// value must be used.
virtual void SetJitterBufferMinimumDelay(
absl::optional<double> delay_seconds) = 0;