Use backticks not vertical bars to denote variables in comments for /api
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}
diff --git a/api/audio/audio_frame.h b/api/audio/audio_frame.h
index 78539f5..726b9a9 100644
--- a/api/audio/audio_frame.h
+++ b/api/audio/audio_frame.h
@@ -139,7 +139,7 @@
int64_t profile_timestamp_ms_ = 0;
// Information about packets used to assemble this audio frame. This is needed
- // by |SourceTracker| when the frame is delivered to the RTCRtpReceiver's
+ // by `SourceTracker` when the frame is delivered to the RTCRtpReceiver's
// MediaStreamTrack, in order to implement getContributingSources(). See:
// https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getcontributingsources
//
@@ -149,7 +149,7 @@
// sync buffer is the small sample-holding buffer located after the audio
// decoder and before where samples are assembled into output frames.
//
- // |RtpPacketInfos| may also be empty if the audio samples did not come from
+ // `RtpPacketInfos` may also be empty if the audio samples did not come from
// RTP packets. E.g. if the audio were locally generated by packet loss
// concealment, comfort noise generation, etc.
RtpPacketInfos packet_infos_;
@@ -165,7 +165,7 @@
// Absolute capture timestamp when this audio frame was originally captured.
// This is only valid for audio frames captured on this machine. The absolute
- // capture timestamp of a received frame is found in |packet_infos_|.
+ // capture timestamp of a received frame is found in `packet_infos_`.
// This timestamp MUST be based on the same clock as rtc::TimeMillis().
absl::optional<int64_t> absolute_capture_timestamp_ms_;