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}

api/audio/audio_frame.cc

@@ -52,7 +52,7 @@ void AudioFrame::Reset() {
 }
 
 void AudioFrame::ResetWithoutMuting() {
-  // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
+  // TODO(wu): Zero is a valid value for `timestamp_`. We should initialize
   // to an invalid value, or add a new member to indicate invalidity.
   timestamp_ = 0;
   elapsed_time_ms_ = -1;

api/audio/audio_frame.h

@@ -139,7 +139,7 @@ class AudioFrame {
   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 @@ class AudioFrame {
   //   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 @@ class AudioFrame {
 
   // 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_;
 

api/audio/audio_frame_processor.h

@@ -28,12 +28,12 @@ class AudioFrameProcessor {
 
   // Processes the frame received from WebRTC, is called by WebRTC off the
   // realtime audio capturing path. AudioFrameProcessor must reply with
-  // processed frames by calling |sink_callback| if it was provided in SetSink()
-  // call. |sink_callback| can be called in the context of Process().
+  // processed frames by calling `sink_callback` if it was provided in SetSink()
+  // call. `sink_callback` can be called in the context of Process().
   virtual void Process(std::unique_ptr<AudioFrame> frame) = 0;
 
   // Atomically replaces the current sink with the new one. Before the
-  // first call to this function, or if the provided |sink_callback| is nullptr,
+  // first call to this function, or if the provided `sink_callback` is nullptr,
   // processed frames are simply discarded.
   virtual void SetSink(OnAudioFrameCallback sink_callback) = 0;
 };

api/audio/audio_mixer.h

@@ -35,9 +35,9 @@ class AudioMixer : public rtc::RefCountInterface {
       kError,   // The audio_frame will not be used.
     };
 
-    // Overwrites |audio_frame|. The data_ field is overwritten with
+    // Overwrites `audio_frame`. The data_ field is overwritten with
     // 10 ms of new audio (either 1 or 2 interleaved channels) at
-    // |sample_rate_hz|. All fields in |audio_frame| must be updated.
+    // `sample_rate_hz`. All fields in `audio_frame` must be updated.
     virtual AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
                                                  AudioFrame* audio_frame) = 0;
 
@@ -66,7 +66,7 @@ class AudioMixer : public rtc::RefCountInterface {
   // should mix at a rate that doesn't cause quality loss of the
   // sources' audio. The mixing rate is one of the rates listed in
   // AudioProcessing::NativeRate. All fields in
-  // |audio_frame_for_mixing| must be updated.
+  // `audio_frame_for_mixing` must be updated.
   virtual void Mix(size_t number_of_channels,
                    AudioFrame* audio_frame_for_mixing) = 0;