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}

1 files changed, 43 insertions, 43 deletions

diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index 5499b7d87c..b9350ac9e1 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -235,9 +235,9 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     std::string username;
     std::string password;
     TlsCertPolicy tls_cert_policy = kTlsCertPolicySecure;
-    // If the URIs in |urls| only contain IP addresses, this field can be used
+    // If the URIs in `urls` only contain IP addresses, this field can be used
     // to indicate the hostname, which may be necessary for TLS (using the SNI
-    // extension). If |urls| itself contains the hostname, this isn't
+    // extension). If `urls` itself contains the hostname, this isn't
     // necessary.
     std::string hostname;
     // List of protocols to be used in the TLS ALPN extension.
@@ -526,7 +526,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     // re-determining was removed in ICEbis (ICE v2).
     bool redetermine_role_on_ice_restart = true;
 
-    // This flag is only effective when |continual_gathering_policy| is
+    // This flag is only effective when `continual_gathering_policy` is
     // GATHER_CONTINUALLY.
     //
     // If true, after the ICE transport type is changed such that new types of
@@ -712,8 +712,8 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
   };
 
   // Used by GetStats to decide which stats to include in the stats reports.
-  // |kStatsOutputLevelStandard| includes the standard stats for Javascript API;
-  // |kStatsOutputLevelDebug| includes both the standard stats and additional
+  // `kStatsOutputLevelStandard` includes the standard stats for Javascript API;
+  // `kStatsOutputLevelDebug` includes both the standard stats and additional
   // stats for debugging purposes.
   enum StatsOutputLevel {
     kStatsOutputLevelStandard,
@@ -754,10 +754,10 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
 
   // Add a new MediaStreamTrack to be sent on this PeerConnection, and return
   // the newly created RtpSender. The RtpSender will be associated with the
-  // streams specified in the |stream_ids| list.
+  // streams specified in the `stream_ids` list.
   //
   // Errors:
-  // - INVALID_PARAMETER: |track| is null, has a kind other than audio or video,
+  // - INVALID_PARAMETER: `track` is null, has a kind other than audio or video,
   //       or a sender already exists for the track.
   // - INVALID_STATE: The PeerConnection is closed.
   virtual RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
@@ -774,7 +774,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
   // corresponding RtpTransceiver direction as no longer sending.
   //
   // Errors:
-  // - INVALID_PARAMETER: |sender| is null or (Plan B only) the sender is not
+  // - INVALID_PARAMETER: `sender` is null or (Plan B only) the sender is not
   //       associated with this PeerConnection.
   // - INVALID_STATE: PeerConnection is closed.
   // TODO(bugs.webrtc.org/9534): Rename to RemoveTrack once the other signature
@@ -786,7 +786,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
   // transceivers. Adding a transceiver will cause future calls to CreateOffer
   // to add a media description for the corresponding transceiver.
   //
-  // The initial value of |mid| in the returned transceiver is null. Setting a
+  // The initial value of `mid` in the returned transceiver is null. Setting a
   // new session description may change it to a non-null value.
   //
   // https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
@@ -805,7 +805,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
   // of the transceiver (and sender/receiver) will be derived from the kind of
   // the track.
   // Errors:
-  // - INVALID_PARAMETER: |track| is null.
+  // - INVALID_PARAMETER: `track` is null.
   virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
   AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track) = 0;
   virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
@@ -815,7 +815,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
   // Adds a transceiver with the given kind. Can either be MEDIA_TYPE_AUDIO or
   // MEDIA_TYPE_VIDEO.
   // Errors:
-  // - INVALID_PARAMETER: |media_type| is not MEDIA_TYPE_AUDIO or
+  // - INVALID_PARAMETER: `media_type` is not MEDIA_TYPE_AUDIO or
   //                      MEDIA_TYPE_VIDEO.
   virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
   AddTransceiver(cricket::MediaType media_type) = 0;
@@ -830,9 +830,9 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
   // The standard way to do this would be through "addTransceiver", but we
   // don't support that API yet.
   //
-  // |kind| must be "audio" or "video".
+  // `kind` must be "audio" or "video".
   //
-  // |stream_id| is used to populate the msid attribute; if empty, one will
+  // `stream_id` is used to populate the msid attribute; if empty, one will
   // be generated automatically.
   //
   // This method is not supported with kUnifiedPlan semantics. Please use
@@ -986,7 +986,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
   // returned by CreateOffer() or CreateAnswer() or else the operation should
   // fail. Our implementation however allows some amount of "SDP munging", but
   // please note that this is HIGHLY DISCOURAGED. If you do not intent to munge
-  // SDP, the method below that doesn't take |desc| as an argument will create
+  // SDP, the method below that doesn't take `desc` as an argument will create
   // the offer or answer for you.
   //
   // The observer is invoked as soon as the operation completes, which could be
@@ -1044,10 +1044,10 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
 
   virtual PeerConnectionInterface::RTCConfiguration GetConfiguration() = 0;
 
-  // Sets the PeerConnection's global configuration to |config|.
+  // Sets the PeerConnection's global configuration to `config`.
   //
-  // The members of |config| that may be changed are |type|, |servers|,
-  // |ice_candidate_pool_size| and |prune_turn_ports| (though the candidate
+  // The members of `config` that may be changed are `type`, `servers`,
+  // `ice_candidate_pool_size` and `prune_turn_ports` (though the candidate
   // pool size can't be changed after the first call to SetLocalDescription).
   // Note that this means the BUNDLE and RTCP-multiplexing policies cannot be
   // changed with this method.
@@ -1055,14 +1055,14 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
   // Any changes to STUN/TURN servers or ICE candidate policy will affect the
   // next gathering phase, and cause the next call to createOffer to generate
   // new ICE credentials, as described in JSEP. This also occurs when
-  // |prune_turn_ports| changes, for the same reasoning.
+  // `prune_turn_ports` changes, for the same reasoning.
   //
-  // If an error occurs, returns false and populates |error| if non-null:
-  // - INVALID_MODIFICATION if |config| contains a modified parameter other
+  // If an error occurs, returns false and populates `error` if non-null:
+  // - INVALID_MODIFICATION if `config` contains a modified parameter other
   //   than one of the parameters listed above.
-  // - INVALID_RANGE if |ice_candidate_pool_size| is out of range.
+  // - INVALID_RANGE if `ice_candidate_pool_size` is out of range.
   // - SYNTAX_ERROR if parsing an ICE server URL failed.
-  // - INVALID_PARAMETER if a TURN server is missing |username| or |password|.
+  // - INVALID_PARAMETER if a TURN server is missing `username` or `password`.
   // - INTERNAL_ERROR if an unexpected error occurred.
   //
   // TODO(nisse): Make this pure virtual once all Chrome subclasses of
@@ -1071,9 +1071,9 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
       const PeerConnectionInterface::RTCConfiguration& config);
 
   // Provides a remote candidate to the ICE Agent.
-  // A copy of the |candidate| will be created and added to the remote
+  // A copy of the `candidate` will be created and added to the remote
   // description. So the caller of this method still has the ownership of the
-  // |candidate|.
+  // `candidate`.
   // TODO(hbos): The spec mandates chaining this operation onto the operations
   // chain; deprecate and remove this version in favor of the callback-based
   // signature.
@@ -1096,13 +1096,13 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
   // this PeerConnection. Other limitations might affect these limits and
   // are respected (for example "b=AS" in SDP).
   //
-  // Setting |current_bitrate_bps| will reset the current bitrate estimate
+  // Setting `current_bitrate_bps` will reset the current bitrate estimate
   // to the provided value.
   virtual RTCError SetBitrate(const BitrateSettings& bitrate) = 0;
 
   // Enable/disable playout of received audio streams. Enabled by default. Note
   // that even if playout is enabled, streams will only be played out if the
-  // appropriate SDP is also applied. Setting |playout| to false will stop
+  // appropriate SDP is also applied. Setting `playout` to false will stop
   // playout of the underlying audio device but starts a task which will poll
   // for audio data every 10ms to ensure that audio processing happens and the
   // audio statistics are updated.
@@ -1157,13 +1157,13 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
   virtual void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) {}
 
   // Start RtcEventLog using an existing output-sink. Takes ownership of
-  // |output| and passes it on to Call, which will take the ownership. If the
+  // `output` and passes it on to Call, which will take the ownership. If the
   // operation fails the output will be closed and deallocated. The event log
-  // will send serialized events to the output object every |output_period_ms|.
+  // will send serialized events to the output object every `output_period_ms`.
   // Applications using the event log should generally make their own trade-off
   // regarding the output period. A long period is generally more efficient,
   // with potential drawbacks being more bursty thread usage, and more events
-  // lost in case the application crashes. If the |output_period_ms| argument is
+  // lost in case the application crashes. If the `output_period_ms` argument is
   // omitted, webrtc selects a default deemed to be workable in most cases.
   virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
                                 int64_t output_period_ms) = 0;
@@ -1222,7 +1222,7 @@ class PeerConnectionObserver {
   // Used to fire spec-compliant onnegotiationneeded events, which should only
   // fire when the Operations Chain is empty. The observer is responsible for
   // queuing a task (e.g. Chromium: jump to main thread) to maybe fire the
-  // event. The event identified using |event_id| must only fire if
+  // event. The event identified using `event_id` must only fire if
   // PeerConnection::ShouldFireNegotiationNeededEvent() returns true since it is
   // possible for the event to become invalidated by operations subsequently
   // chained.
@@ -1256,7 +1256,7 @@ class PeerConnectionObserver {
 
   // Gathering of an ICE candidate failed.
   // See https://w3c.github.io/webrtc-pc/#event-icecandidateerror
-  // |host_candidate| is a stringified socket address.
+  // `host_candidate` is a stringified socket address.
   virtual void OnIceCandidateError(const std::string& host_candidate,
                                    const std::string& url,
                                    int error_code,
@@ -1393,7 +1393,7 @@ struct RTC_EXPORT PeerConnectionFactoryDependencies final {
       network_state_predictor_factory;
   std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
   // This will only be used if CreatePeerConnection is called without a
-  // |port_allocator|, causing the default allocator and network manager to be
+  // `port_allocator`, causing the default allocator and network manager to be
   // used.
   std::unique_ptr<rtc::NetworkMonitorFactory> network_monitor_factory;
   std::unique_ptr<NetEqFactory> neteq_factory;
@@ -1467,12 +1467,12 @@ class RTC_EXPORT PeerConnectionFactoryInterface
       const PeerConnectionInterface::RTCConfiguration& configuration,
       PeerConnectionDependencies dependencies);
 
-  // Deprecated; |allocator| and |cert_generator| may be null, in which case
+  // Deprecated; `allocator` and `cert_generator` may be null, in which case
   // default implementations will be used.
   //
-  // |observer| must not be null.
+  // `observer` must not be null.
   //
-  // Note that this method does not take ownership of |observer|; it's the
+  // Note that this method does not take ownership of `observer`; it's the
   // responsibility of the caller to delete it. It can be safely deleted after
   // Close has been called on the returned PeerConnection, which ensures no
   // more observer callbacks will be invoked.
@@ -1483,13 +1483,13 @@ class RTC_EXPORT PeerConnectionFactoryInterface
       std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
       PeerConnectionObserver* observer);
 
-  // Returns the capabilities of an RTP sender of type |kind|.
+  // Returns the capabilities of an RTP sender of type `kind`.
   // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
   // TODO(orphis): Make pure virtual when all subclasses implement it.
   virtual RtpCapabilities GetRtpSenderCapabilities(
       cricket::MediaType kind) const;
 
-  // Returns the capabilities of an RTP receiver of type |kind|.
+  // Returns the capabilities of an RTP receiver of type `kind`.
   // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
   // TODO(orphis): Make pure virtual when all subclasses implement it.
   virtual RtpCapabilities GetRtpReceiverCapabilities(
@@ -1499,22 +1499,22 @@ class RTC_EXPORT PeerConnectionFactoryInterface
       const std::string& stream_id) = 0;
 
   // Creates an AudioSourceInterface.
-  // |options| decides audio processing settings.
+  // `options` decides audio processing settings.
   virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
       const cricket::AudioOptions& options) = 0;
 
-  // Creates a new local VideoTrack. The same |source| can be used in several
+  // Creates a new local VideoTrack. The same `source` can be used in several
   // tracks.
   virtual rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
       const std::string& label,
       VideoTrackSourceInterface* source) = 0;
 
-  // Creates an new AudioTrack. At the moment |source| can be null.
+  // Creates an new AudioTrack. At the moment `source` can be null.
   virtual rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
       const std::string& label,
       AudioSourceInterface* source) = 0;
 
-  // Starts AEC dump using existing file. Takes ownership of |file| and passes
+  // Starts AEC dump using existing file. Takes ownership of `file` and passes
   // it on to VoiceEngine (via other objects) immediately, which will take
   // the ownerhip. If the operation fails, the file will be closed.
   // A maximum file size in bytes can be specified. When the file size limit is
@@ -1549,8 +1549,8 @@ class RTC_EXPORT PeerConnectionFactoryInterface
 // video-specific interfaces, and omit the corresponding modules from its
 // build.
 //
-// If |network_thread| or |worker_thread| are null, the PeerConnectionFactory
-// will create the necessary thread internally. If |signaling_thread| is null,
+// If `network_thread` or `worker_thread` are null, the PeerConnectionFactory
+// will create the necessary thread internally. If `signaling_thread` is null,
 // the PeerConnectionFactory will use the thread on which this method is called
 // as the signaling thread, wrapping it in an rtc::Thread object if needed.
 RTC_EXPORT rtc::scoped_refptr<PeerConnectionFactoryInterface>