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/peer_connection_interface.h b/api/peer_connection_interface.h
index 5499b7d..b9350ac 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -235,9 +235,9 @@
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 @@
// 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 @@
};
// 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
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 @@
// 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 @@
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 @@
// 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 @@
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 @@
// 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 @@
// 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 @@
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 @@
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 @@
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 @@
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 @@
// 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>