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/adaptation/resource.h

@@ -57,7 +57,7 @@ class RTC_EXPORT Resource : public rtc::RefCountInterface {
   ~Resource() override;
 
   virtual std::string Name() const = 0;
-  // The |listener| may be informed of resource usage measurements on any task
+  // The `listener` may be informed of resource usage measurements on any task
   // queue, but not after this method is invoked with the null argument.
   virtual void SetResourceListener(ResourceListener* listener) = 0;
 };

api/async_dns_resolver.h

@@ -41,10 +41,10 @@ namespace webrtc {
 class AsyncDnsResolverResult {
  public:
   virtual ~AsyncDnsResolverResult() = default;
-  // Returns true iff the address from |Start| was successfully resolved.
+  // Returns true iff the address from `Start` was successfully resolved.
-  // If the address was successfully resolved, sets |addr| to a copy of the
+  // If the address was successfully resolved, sets `addr` to a copy of the
-  // address from |Start| with the IP address set to the top most resolved
+  // address from `Start` with the IP address set to the top most resolved
-  // address of |family| (|addr| will have both hostname and the resolved ip).
+  // address of `family` (`addr` will have both hostname and the resolved ip).
   virtual bool GetResolvedAddress(int family,
                                   rtc::SocketAddress* addr) const = 0;
   // Returns error from resolver.
@@ -55,7 +55,7 @@ class RTC_EXPORT AsyncDnsResolverInterface {
  public:
   virtual ~AsyncDnsResolverInterface() = default;
 
-  // Start address resolution of the hostname in |addr|.
+  // Start address resolution of the hostname in `addr`.
   virtual void Start(const rtc::SocketAddress& addr,
                      std::function<void()> callback) = 0;
   virtual const AsyncDnsResolverResult& result() const = 0;

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()
+  // processed frames by calling `sink_callback` if it was provided in SetSink()
-  // call. |sink_callback| can be called in the context of Process().
+  // 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;
 

api/audio_codecs/audio_decoder.h

@@ -53,8 +53,8 @@ class AudioDecoder {
     // Returns true if this packet contains DTX.
     virtual bool IsDtxPacket() const;
 
-    // Decodes this frame of audio and writes the result in |decoded|.
+    // Decodes this frame of audio and writes the result in `decoded`.
-    // |decoded| must be large enough to store as many samples as indicated by a
+    // `decoded` must be large enough to store as many samples as indicated by a
     // call to Duration() . On success, returns an absl::optional containing the
     // total number of samples across all channels, as well as whether the
     // decoder produced comfort noise or speech. On failure, returns an empty
@@ -85,8 +85,8 @@ class AudioDecoder {
   // Let the decoder parse this payload and prepare zero or more decodable
   // frames. Each frame must be between 10 ms and 120 ms long. The caller must
   // ensure that the AudioDecoder object outlives any frame objects returned by
-  // this call. The decoder is free to swap or move the data from the |payload|
+  // this call. The decoder is free to swap or move the data from the `payload`
-  // buffer. |timestamp| is the input timestamp, in samples, corresponding to
+  // buffer. `timestamp` is the input timestamp, in samples, corresponding to
   // the start of the payload.
   virtual std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
                                                 uint32_t timestamp);
@@ -95,12 +95,12 @@ class AudioDecoder {
   // obsolete; callers should call ParsePayload instead. For now, subclasses
   // must still implement DecodeInternal.
 
-  // Decodes |encode_len| bytes from |encoded| and writes the result in
+  // Decodes `encode_len` bytes from `encoded` and writes the result in
-  // |decoded|. The maximum bytes allowed to be written into |decoded| is
+  // `decoded`. The maximum bytes allowed to be written into `decoded` is
-  // |max_decoded_bytes|. Returns the total number of samples across all
+  // `max_decoded_bytes`. Returns the total number of samples across all
-  // channels. If the decoder produced comfort noise, |speech_type|
+  // channels. If the decoder produced comfort noise, `speech_type`
   // is set to kComfortNoise, otherwise it is kSpeech. The desired output
-  // sample rate is provided in |sample_rate_hz|, which must be valid for the
+  // sample rate is provided in `sample_rate_hz`, which must be valid for the
   // codec at hand.
   int Decode(const uint8_t* encoded,
              size_t encoded_len,
@@ -123,11 +123,11 @@ class AudioDecoder {
 
   // Calls the packet-loss concealment of the decoder to update the state after
   // one or several lost packets. The caller has to make sure that the
-  // memory allocated in |decoded| should accommodate |num_frames| frames.
+  // memory allocated in `decoded` should accommodate `num_frames` frames.
   virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
 
   // Asks the decoder to generate packet-loss concealment and append it to the
-  // end of |concealment_audio|. The concealment audio should be in
+  // end of `concealment_audio`. The concealment audio should be in
   // channel-interleaved format, with as many channels as the last decoded
   // packet produced. The implementation must produce at least
   // requested_samples_per_channel, or nothing at all. This is a signal to the
@@ -146,19 +146,19 @@ class AudioDecoder {
   // Returns the last error code from the decoder.
   virtual int ErrorCode();
 
-  // Returns the duration in samples-per-channel of the payload in |encoded|
+  // Returns the duration in samples-per-channel of the payload in `encoded`
-  // which is |encoded_len| bytes long. Returns kNotImplemented if no duration
+  // which is `encoded_len` bytes long. Returns kNotImplemented if no duration
   // estimate is available, or -1 in case of an error.
   virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
 
   // Returns the duration in samples-per-channel of the redandant payload in
-  // |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no
+  // `encoded` which is `encoded_len` bytes long. Returns kNotImplemented if no
   // duration estimate is available, or -1 in case of an error.
   virtual int PacketDurationRedundant(const uint8_t* encoded,
                                       size_t encoded_len) const;
 
   // Detects whether a packet has forward error correction. The packet is
-  // comprised of the samples in |encoded| which is |encoded_len| bytes long.
+  // comprised of the samples in `encoded` which is `encoded_len` bytes long.
   // Returns true if the packet has FEC and false otherwise.
   virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
 

api/audio_codecs/audio_decoder_factory_template.h

@@ -89,8 +89,8 @@ class AudioDecoderFactoryT : public AudioDecoderFactory {
 // Each decoder type is given as a template argument to the function; it should
 // be a struct with the following static member functions:
 //
-//   // Converts |audio_format| to a ConfigType instance. Returns an empty
+//   // Converts `audio_format` to a ConfigType instance. Returns an empty
-//   // optional if |audio_format| doesn't correctly specify a decoder of our
+//   // optional if `audio_format` doesn't correctly specify a decoder of our
 //   // type.
 //   absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
 //

api/audio_codecs/audio_encoder.h

@@ -95,13 +95,13 @@ class AudioEncoder {
 
   // This is the main struct for auxiliary encoding information. Each encoded
   // packet should be accompanied by one EncodedInfo struct, containing the
-  // total number of |encoded_bytes|, the |encoded_timestamp| and the
+  // total number of `encoded_bytes`, the `encoded_timestamp` and the
-  // |payload_type|. If the packet contains redundant encodings, the |redundant|
+  // `payload_type`. If the packet contains redundant encodings, the `redundant`
   // vector will be populated with EncodedInfoLeaf structs. Each struct in the
   // vector represents one encoding; the order of structs in the vector is the
   // same as the order in which the actual payloads are written to the byte
   // stream. When EncoderInfoLeaf structs are present in the vector, the main
-  // struct's |encoded_bytes| will be the sum of all the |encoded_bytes| in the
+  // struct's `encoded_bytes` will be the sum of all the `encoded_bytes` in the
   // vector.
   struct EncodedInfo : public EncodedInfoLeaf {
     EncodedInfo();
@@ -143,7 +143,7 @@ class AudioEncoder {
 
   // Accepts one 10 ms block of input audio (i.e., SampleRateHz() / 100 *
   // NumChannels() samples). Multi-channel audio must be sample-interleaved.
-  // The encoder appends zero or more bytes of output to |encoded| and returns
+  // The encoder appends zero or more bytes of output to `encoded` and returns
   // additional encoding information.  Encode() checks some preconditions, calls
   // EncodeImpl() which does the actual work, and then checks some
   // postconditions.
@@ -205,7 +205,7 @@ class AudioEncoder {
   virtual void DisableAudioNetworkAdaptor();
 
   // Provides uplink packet loss fraction to this encoder to allow it to adapt.
-  // |uplink_packet_loss_fraction| is in the range [0.0, 1.0].
+  // `uplink_packet_loss_fraction` is in the range [0.0, 1.0].
   virtual void OnReceivedUplinkPacketLossFraction(
       float uplink_packet_loss_fraction);
 

api/audio_codecs/audio_encoder_factory_template.h

@@ -103,8 +103,8 @@ class AudioEncoderFactoryT : public AudioEncoderFactory {
 // Each encoder type is given as a template argument to the function; it should
 // be a struct with the following static member functions:
 //
-//   // Converts |audio_format| to a ConfigType instance. Returns an empty
+//   // Converts `audio_format` to a ConfigType instance. Returns an empty
-//   // optional if |audio_format| doesn't correctly specify an encoder of our
+//   // optional if `audio_format` doesn't correctly specify an encoder of our
 //   // type.
 //   absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
 //

api/audio_codecs/audio_format.h

@@ -39,7 +39,7 @@ struct RTC_EXPORT SdpAudioFormat {
                  Parameters&& param);
   ~SdpAudioFormat();
 
-  // Returns true if this format is compatible with |o|. In SDP terminology:
+  // Returns true if this format is compatible with `o`. In SDP terminology:
   // would it represent the same codec between an offer and an answer? As
   // opposed to operator==, this method disregards codec parameters.
   bool Matches(const SdpAudioFormat& o) const;

api/audio_codecs/opus/audio_encoder_opus_config.h

@@ -49,10 +49,10 @@ struct RTC_EXPORT AudioEncoderOpusConfig {
   bool cbr_enabled;
   int max_playback_rate_hz;
 
-  // |complexity| is used when the bitrate goes above
+  // `complexity` is used when the bitrate goes above
-  // |complexity_threshold_bps| + |complexity_threshold_window_bps|;
+  // `complexity_threshold_bps` + `complexity_threshold_window_bps`;
-  // |low_rate_complexity| is used when the bitrate falls below
+  // `low_rate_complexity` is used when the bitrate falls below
-  // |complexity_threshold_bps| - |complexity_threshold_window_bps|. In the
+  // `complexity_threshold_bps` - `complexity_threshold_window_bps`. In the
   // interval in the middle, we keep using the most recent of the two
   // complexity settings.
   int complexity;

api/call/bitrate_allocation.h

@@ -32,7 +32,7 @@ struct BitrateAllocationUpdate {
   double packet_loss_ratio = 0;
   // Predicted round trip time.
   TimeDelta round_trip_time = TimeDelta::PlusInfinity();
-  // |bwe_period| is deprecated, use |stable_target_bitrate| allocation instead.
+  // `bwe_period` is deprecated, use `stable_target_bitrate` allocation instead.
   TimeDelta bwe_period = TimeDelta::PlusInfinity();
   // Congestion window pushback bitrate reduction fraction. Used in
   // VideoStreamEncoder to reduce the bitrate by the given fraction

api/candidate.cc

@@ -92,7 +92,7 @@ uint32_t Candidate::GetPriority(uint32_t type_preference,
   //            (2^8)*(local preference) +
   //            (2^0)*(256 - component ID)
 
-  // |local_preference| length is 2 bytes, 0-65535 inclusive.
+  // `local_preference` length is 2 bytes, 0-65535 inclusive.
   // In our implemenation we will partion local_preference into
   //              0                 1
   //       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5

api/candidate.h

@@ -112,7 +112,7 @@ class RTC_EXPORT Candidate {
   uint32_t generation() const { return generation_; }
   void set_generation(uint32_t generation) { generation_ = generation; }
 
-  // |network_cost| measures the cost/penalty of using this candidate. A network
+  // `network_cost` measures the cost/penalty of using this candidate. A network
   // cost of 0 indicates this candidate can be used freely. A value of
   // rtc::kNetworkCostMax indicates it should be used only as the last resort.
   void set_network_cost(uint16_t network_cost) {
@@ -167,9 +167,9 @@ class RTC_EXPORT Candidate {
   bool operator!=(const Candidate& o) const;
 
   // Returns a sanitized copy configured by the given booleans. If
-  // |use_host_address| is true, the returned copy has its IP removed from
+  // `use_host_address` is true, the returned copy has its IP removed from
-  // |address()|, which leads |address()| to be a hostname address. If
+  // `address()`, which leads `address()` to be a hostname address. If
-  // |filter_related_address|, the returned copy has its related address reset
+  // `filter_related_address`, the returned copy has its related address reset
   // to the wildcard address (i.e. 0.0.0.0 for IPv4 and :: for IPv6). Note that
   // setting both booleans to false returns an identical copy to the original
   // candidate.

api/data_channel_interface.h

@@ -42,14 +42,14 @@ struct DataChannelInit {
   // The max period of time in milliseconds in which retransmissions will be
   // sent. After this time, no more retransmissions will be sent.
   //
-  // Cannot be set along with |maxRetransmits|.
+  // Cannot be set along with `maxRetransmits`.
-  // This is called |maxPacketLifeTime| in the WebRTC JS API.
+  // This is called `maxPacketLifeTime` in the WebRTC JS API.
   // Negative values are ignored, and positive values are clamped to [0-65535]
   absl::optional<int> maxRetransmitTime;
 
   // The max number of retransmissions.
   //
-  // Cannot be set along with |maxRetransmitTime|.
+  // Cannot be set along with `maxRetransmitTime`.
   // Negative values are ignored, and positive values are clamped to [0-65535]
   absl::optional<int> maxRetransmits;
 
@@ -57,7 +57,7 @@ struct DataChannelInit {
   std::string protocol;
 
   // True if the channel has been externally negotiated and we do not send an
-  // in-band signalling in the form of an "open" message. If this is true, |id|
+  // in-band signalling in the form of an "open" message. If this is true, `id`
   // below must be set; otherwise it should be unset and will be negotiated
   // in-band.
   bool negotiated = false;
@@ -70,7 +70,7 @@ struct DataChannelInit {
 };
 
 // At the JavaScript level, data can be passed in as a string or a blob, so
-// this structure's |binary| flag tells whether the data should be interpreted
+// this structure's `binary` flag tells whether the data should be interpreted
 // as binary or text.
 struct DataBuffer {
   DataBuffer(const rtc::CopyOnWriteBuffer& data, bool binary)
@@ -180,7 +180,7 @@ class RTC_EXPORT DataChannelInterface : public rtc::RefCountInterface {
   // https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.7
   virtual void Close() = 0;
 
-  // Sends |data| to the remote peer. If the data can't be sent at the SCTP
+  // Sends `data` to the remote peer. If the data can't be sent at the SCTP
   // level (due to congestion control), it's buffered at the data channel level,
   // up to a maximum of 16MB. If Send is called while this buffer is full, the
   // data channel will be closed abruptly.

api/dtmf_sender_interface.h

@@ -23,8 +23,8 @@ namespace webrtc {
 // DtmfSender.
 class DtmfSenderObserverInterface {
  public:
-  // Triggered when DTMF |tone| is sent.
+  // Triggered when DTMF `tone` is sent.
-  // If |tone| is empty that means the DtmfSender has sent out all the given
+  // If `tone` is empty that means the DtmfSender has sent out all the given
   // tones.
   // The callback includes the state of the tone buffer at the time when
   // the tone finished playing.
@@ -58,7 +58,7 @@ class DtmfSenderInterface : public rtc::RefCountInterface {
   // able to send packets, and a "telephone-event" codec must be negotiated.
   virtual bool CanInsertDtmf() = 0;
 
-  // Queues a task that sends the DTMF |tones|. The |tones| parameter is treated
+  // Queues a task that sends the DTMF `tones`. The `tones` parameter is treated
   // as a series of characters. The characters 0 through 9, A through D, #, and
   // * generate the associated DTMF tones. The characters a to d are equivalent
   // to A to D. The character ',' indicates a delay of 2 seconds before
@@ -66,18 +66,18 @@ class DtmfSenderInterface : public rtc::RefCountInterface {
   //
   // Unrecognized characters are ignored.
   //
-  // The |duration| parameter indicates the duration in ms to use for each
+  // The `duration` parameter indicates the duration in ms to use for each
-  // character passed in the |tones| parameter. The duration cannot be more
+  // character passed in the `tones` parameter. The duration cannot be more
   // than 6000 or less than 70.
   //
-  // The |inter_tone_gap| parameter indicates the gap between tones in ms. The
+  // The `inter_tone_gap` parameter indicates the gap between tones in ms. The
-  // |inter_tone_gap| must be at least 50 ms but should be as short as
+  // `inter_tone_gap` must be at least 50 ms but should be as short as
   // possible.
   //
-  // The |comma_delay| parameter indicates the delay after the ','
+  // The `comma_delay` parameter indicates the delay after the ','
-  // character. InsertDtmf specifies |comma_delay| as an argument
+  // character. InsertDtmf specifies `comma_delay` as an argument
   // with a default value of 2 seconds as per the WebRTC spec. This parameter
-  // allows users to comply with legacy WebRTC clients. The |comma_delay|
+  // allows users to comply with legacy WebRTC clients. The `comma_delay`
   // must be at least 50 ms.
   //
   // If InsertDtmf is called on the same object while an existing task for this

api/fec_controller.h

@@ -38,7 +38,7 @@ class VCMProtectionCallback {
 // FecController calculates how much of the allocated network
 // capacity that can be used by an encoder and how much that
 // is needed for redundant packets such as FEC and NACK. It uses an
-// implementation of |VCMProtectionCallback| to set new FEC parameters and get
+// implementation of `VCMProtectionCallback` to set new FEC parameters and get
 // the bitrate currently used for FEC and NACK.
 // Usage:
 // Setup by calling SetProtectionMethod and SetEncodingData.

api/frame_transformer_interface.h

@@ -30,7 +30,7 @@ class TransformableFrameInterface {
   // method call.
   virtual rtc::ArrayView<const uint8_t> GetData() const = 0;
 
-  // Copies |data| into the owned frame payload data.
+  // Copies `data` into the owned frame payload data.
   virtual void SetData(rtc::ArrayView<const uint8_t> data) = 0;
 
   virtual uint32_t GetTimestamp() const = 0;
@@ -78,7 +78,7 @@ class TransformedFrameCallback : public rtc::RefCountInterface {
 // the TransformedFrameCallback interface (see above).
 class FrameTransformerInterface : public rtc::RefCountInterface {
  public:
-  // Transforms |frame| using the implementing class' processing logic.
+  // Transforms `frame` using the implementing class' processing logic.
   virtual void Transform(
       std::unique_ptr<TransformableFrameInterface> transformable_frame) = 0;
 

api/jsep.h

@@ -73,7 +73,7 @@ class RTC_EXPORT IceCandidateInterface {
 
 // Creates a IceCandidateInterface based on SDP string.
 // Returns null if the sdp string can't be parsed.
-// |error| may be null.
+// `error` may be null.
 RTC_EXPORT IceCandidateInterface* CreateIceCandidate(const std::string& sdp_mid,
                                                      int sdp_mline_index,
                                                      const std::string& sdp,
@@ -91,7 +91,7 @@ class IceCandidateCollection {
  public:
   virtual ~IceCandidateCollection() {}
   virtual size_t count() const = 0;
-  // Returns true if an equivalent |candidate| exist in the collection.
+  // Returns true if an equivalent `candidate` exist in the collection.
   virtual bool HasCandidate(const IceCandidateInterface* candidate) const = 0;
   virtual const IceCandidateInterface* at(size_t index) const = 0;
 };
@@ -158,7 +158,7 @@ class RTC_EXPORT SessionDescriptionInterface {
   virtual SdpType GetType() const;
 
   // kOffer/kPrAnswer/kAnswer
-  // TODO(steveanton): Remove this in favor of |GetType| that returns SdpType.
+  // TODO(steveanton): Remove this in favor of `GetType` that returns SdpType.
   virtual std::string type() const = 0;
 
   // Adds the specified candidate to the description.
@@ -190,7 +190,7 @@ class RTC_EXPORT SessionDescriptionInterface {
 
 // Creates a SessionDescriptionInterface based on the SDP string and the type.
 // Returns null if the sdp string can't be parsed or the type is unsupported.
-// |error| may be null.
+// `error` may be null.
 // TODO(steveanton): This function is deprecated. Please use the functions below
 // which take an SdpType enum instead. Remove this once it is no longer used.
 RTC_EXPORT SessionDescriptionInterface* CreateSessionDescription(
@@ -200,8 +200,8 @@ RTC_EXPORT SessionDescriptionInterface* CreateSessionDescription(
 
 // Creates a SessionDescriptionInterface based on the SDP string and the type.
 // Returns null if the SDP string cannot be parsed.
-// If using the signature with |error_out|, details of the parsing error may be
+// If using the signature with `error_out`, details of the parsing error may be
-// written to |error_out| if it is not null.
+// written to `error_out` if it is not null.
 RTC_EXPORT std::unique_ptr<SessionDescriptionInterface>
 CreateSessionDescription(SdpType type, const std::string& sdp);
 RTC_EXPORT std::unique_ptr<SessionDescriptionInterface>
@@ -221,7 +221,7 @@ std::unique_ptr<SessionDescriptionInterface> CreateSessionDescription(
 class RTC_EXPORT CreateSessionDescriptionObserver
     : public rtc::RefCountInterface {
  public:
-  // This callback transfers the ownership of the |desc|.
+  // This callback transfers the ownership of the `desc`.
   // TODO(deadbeef): Make this take an std::unique_ptr<> to avoid confusion
   // around ownership.
   virtual void OnSuccess(SessionDescriptionInterface* desc) = 0;

api/jsep_ice_candidate.h

@@ -37,7 +37,7 @@ class RTC_EXPORT JsepIceCandidate : public IceCandidateInterface {
   JsepIceCandidate(const JsepIceCandidate&) = delete;
   JsepIceCandidate& operator=(const JsepIceCandidate&) = delete;
   ~JsepIceCandidate() override;
-  // |err| may be null.
+  // `err` may be null.
   bool Initialize(const std::string& sdp, SdpParseError* err);
   void SetCandidate(const cricket::Candidate& candidate) {
     candidate_ = candidate;

api/jsep_session_description.h

@@ -43,7 +43,7 @@ class JsepSessionDescription : public SessionDescriptionInterface {
       absl::string_view session_version);
   virtual ~JsepSessionDescription();
 
-  // Takes ownership of |description|.
+  // Takes ownership of `description`.
   bool Initialize(std::unique_ptr<cricket::SessionDescription> description,
                   const std::string& session_id,
                   const std::string& session_version);

api/media_stream_interface.h

@@ -200,7 +200,7 @@ class AudioTrackSinkInterface {
     RTC_NOTREACHED() << "This method must be overridden, or not used.";
   }
 
-  // In this method, |absolute_capture_timestamp_ms|, when available, is
+  // In this method, `absolute_capture_timestamp_ms`, when available, is
   // supposed to deliver the timestamp when this audio frame was originally
   // captured. This timestamp MUST be based on the same clock as
   // rtc::TimeMillis().
@@ -240,7 +240,7 @@ class RTC_EXPORT AudioSourceInterface : public MediaSourceInterface {
   // TODO(deadbeef): Makes all the interfaces pure virtual after they're
   // implemented in chromium.
 
-  // Sets the volume of the source. |volume| is in  the range of [0, 10].
+  // Sets the volume of the source. `volume` is in  the range of [0, 10].
   // TODO(tommi): This method should be on the track and ideally volume should
   // be applied in the track in a way that does not affect clones of the track.
   virtual void SetVolume(double volume) {}
@@ -268,7 +268,7 @@ class AudioProcessorInterface : public rtc::RefCountInterface {
     AudioProcessingStats apm_statistics;
   };
 
-  // Get audio processor statistics. The |has_remote_tracks| argument should be
+  // Get audio processor statistics. The `has_remote_tracks` argument should be
   // set if there are active remote tracks (this would usually be true during
   // a call). If there are no remote tracks some of the stats will not be set by
   // the AudioProcessor, because they only make sense if there is at least one

api/neteq/neteq.h

@@ -183,7 +183,7 @@ class NetEq {
     SdpAudioFormat sdp_format;
   };
 
-  // Creates a new NetEq object, with parameters set in |config|. The |config|
+  // Creates a new NetEq object, with parameters set in `config`. The `config`
   // object will only have to be valid for the duration of the call to this
   // method.
   static NetEq* Create(
@@ -205,15 +205,15 @@ class NetEq {
   virtual void InsertEmptyPacket(const RTPHeader& rtp_header) = 0;
 
   // Instructs NetEq to deliver 10 ms of audio data. The data is written to
-  // |audio_frame|. All data in |audio_frame| is wiped; |data_|, |speech_type_|,
+  // `audio_frame`. All data in `audio_frame` is wiped; `data_`, `speech_type_`,
-  // |num_channels_|, |sample_rate_hz_|, |samples_per_channel_|, and
+  // `num_channels_`, `sample_rate_hz_`, `samples_per_channel_`, and
-  // |vad_activity_| are updated upon success. If an error is returned, some
+  // `vad_activity_` are updated upon success. If an error is returned, some
   // fields may not have been updated, or may contain inconsistent values.
-  // If muted state is enabled (through Config::enable_muted_state), |muted|
+  // If muted state is enabled (through Config::enable_muted_state), `muted`
   // may be set to true after a prolonged expand period. When this happens, the
-  // |data_| in |audio_frame| is not written, but should be interpreted as being
+  // `data_` in `audio_frame` is not written, but should be interpreted as being
   // all zeros. For testing purposes, an override can be supplied in the
-  // |action_override| argument, which will cause NetEq to take this action
+  // `action_override` argument, which will cause NetEq to take this action
   // next, instead of the action it would normally choose. An optional output
   // argument for fetching the current sample rate can be provided, which
   // will return the same value as last_output_sample_rate_hz() but will avoid
@@ -228,12 +228,12 @@ class NetEq {
   // Replaces the current set of decoders with the given one.
   virtual void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) = 0;
 
-  // Associates |rtp_payload_type| with the given codec, which NetEq will
+  // Associates `rtp_payload_type` with the given codec, which NetEq will
   // instantiate when it needs it. Returns true iff successful.
   virtual bool RegisterPayloadType(int rtp_payload_type,
                                    const SdpAudioFormat& audio_format) = 0;
 
-  // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
+  // Removes `rtp_payload_type` from the codec database. Returns 0 on success,
   // -1 on failure. Removing a payload type that is not registered is ok and
   // will not result in an error.
   virtual int RemovePayloadType(uint8_t rtp_payload_type) = 0;
@@ -250,12 +250,12 @@ class NetEq {
   // Sets a maximum delay in milliseconds for packet buffer. The latency will
   // not exceed the given value, even required delay (given the channel
   // conditions) is higher. Calling this method has the same effect as setting
-  // the |max_delay_ms| value in the NetEq::Config struct.
+  // the `max_delay_ms` value in the NetEq::Config struct.
   virtual bool SetMaximumDelay(int delay_ms) = 0;
 
   // Sets a base minimum delay in milliseconds for packet buffer. The minimum
-  // delay which is set via |SetMinimumDelay| can't be lower than base minimum
+  // delay which is set via `SetMinimumDelay` can't be lower than base minimum
-  // delay. Calling this method is similar to setting the |min_delay_ms| value
+  // delay. Calling this method is similar to setting the `min_delay_ms` value
   // in the NetEq::Config struct. Returns true if the base minimum is
   // successfully applied, otherwise false is returned.
   virtual bool SetBaseMinimumDelayMs(int delay_ms) = 0;
@@ -272,7 +272,7 @@ class NetEq {
   // The packet buffer part of the delay is not updated during DTX/CNG periods.
   virtual int FilteredCurrentDelayMs() const = 0;
 
-  // Writes the current network statistics to |stats|. The statistics are reset
+  // Writes the current network statistics to `stats`. The statistics are reset
   // after the call.
   virtual int NetworkStatistics(NetEqNetworkStatistics* stats) = 0;
 

api/neteq/neteq_controller.h

@@ -115,13 +115,13 @@ class NetEqController {
   virtual void SoftReset() = 0;
 
   // Given info about the latest received packet, and current jitter buffer
-  // status, returns the operation. |target_timestamp| and |expand_mutefactor|
+  // status, returns the operation. `target_timestamp` and `expand_mutefactor`
-  // are provided for reference. |last_packet_samples| is the number of samples
+  // are provided for reference. `last_packet_samples` is the number of samples
   // obtained from the last decoded frame. If there is a packet available, it
-  // should be supplied in |packet|. The mode resulting from the last call to
+  // should be supplied in `packet`. The mode resulting from the last call to
-  // NetEqImpl::GetAudio is supplied in |last_mode|. If there is a DTMF event to
+  // NetEqImpl::GetAudio is supplied in `last_mode`. If there is a DTMF event to
-  // play, |play_dtmf| should be set to true. The output variable
+  // play, `play_dtmf` should be set to true. The output variable
-  // |reset_decoder| will be set to true if a reset is required; otherwise it is
+  // `reset_decoder` will be set to true if a reset is required; otherwise it is
   // left unchanged (i.e., it can remain true if it was true before the call).
   virtual NetEq::Operation GetDecision(const NetEqStatus& status,
                                        bool* reset_decoder) = 0;
@@ -144,11 +144,11 @@ class NetEqController {
   virtual bool SetBaseMinimumDelay(int delay_ms) = 0;
   virtual int GetBaseMinimumDelay() const = 0;
 
-  // These methods test the |cng_state_| for different conditions.
+  // These methods test the `cng_state_` for different conditions.
   virtual bool CngRfc3389On() const = 0;
   virtual bool CngOff() const = 0;
 
-  // Resets the |cng_state_| to kCngOff.
+  // Resets the `cng_state_` to kCngOff.
   virtual void SetCngOff() = 0;
 
   // Reports back to DecisionLogic whether the decision to do expand remains or
@@ -157,7 +157,7 @@ class NetEqController {
   // sync buffer.
   virtual void ExpandDecision(NetEq::Operation operation) = 0;
 
-  // Adds |value| to |sample_memory_|.
+  // Adds `value` to `sample_memory_`.
   virtual void AddSampleMemory(int32_t value) = 0;
 
   // Returns the target buffer level in ms.

api/neteq/neteq_controller_factory.h

@@ -23,7 +23,7 @@ class NetEqControllerFactory {
  public:
   virtual ~NetEqControllerFactory() = default;
 
-  // Creates a new NetEqController object, with parameters set in |config|.
+  // Creates a new NetEqController object, with parameters set in `config`.
   virtual std::unique_ptr<NetEqController> CreateNetEqController(
       const NetEqController::Config& config) const = 0;
 };

api/neteq/neteq_factory.h

@@ -24,7 +24,7 @@ class NetEqFactory {
  public:
   virtual ~NetEqFactory() = default;
 
-  // Creates a new NetEq object, with parameters set in |config|. The |config|
+  // Creates a new NetEq object, with parameters set in `config`. The `config`
   // object will only have to be valid for the duration of the call to this
   // method.
   virtual std::unique_ptr<NetEq> CreateNetEq(

api/numerics/samples_stats_counter.h

@@ -82,7 +82,7 @@ class SamplesStatsCounter {
   // additions were done. This function may not be called if there are no
   // samples.
   //
-  // |percentile| has to be in [0; 1]. 0 percentile is the min in the array and
+  // `percentile` has to be in [0; 1]. 0 percentile is the min in the array and
   // 1 percentile is the max in the array.
   double GetPercentile(double percentile);
   // Returns array view with all samples added into counter. There are no
@@ -105,14 +105,14 @@ class SamplesStatsCounter {
   bool sorted_ = false;
 };
 
-// Multiply all sample values on |value| and return new SamplesStatsCounter
+// Multiply all sample values on `value` and return new SamplesStatsCounter
 // with resulted samples. Doesn't change origin SamplesStatsCounter.
 SamplesStatsCounter operator*(const SamplesStatsCounter& counter, double value);
 inline SamplesStatsCounter operator*(double value,
                                      const SamplesStatsCounter& counter) {
   return counter * value;
 }
-// Divide all sample values on |value| and return new SamplesStatsCounter with
+// Divide all sample values on `value` and return new SamplesStatsCounter with
 // resulted samples. Doesn't change origin SamplesStatsCounter.
 SamplesStatsCounter operator/(const SamplesStatsCounter& counter, double value);
 

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;
+  // `kStatsOutputLevelStandard` includes the standard stats for Javascript API;
-  // |kStatsOutputLevelDebug| includes both the standard stats and additional
+  // `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|,
+  // The members of `config` that may be changed are `type`, `servers`,
-  // |ice_candidate_pool_size| and |prune_turn_ports| (though the candidate
+  // `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:
+  // If an error occurs, returns false and populates `error` if non-null:
-  // - INVALID_MODIFICATION if |config| contains a modified parameter other
+  // - 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
+// If `network_thread` or `worker_thread` are null, the PeerConnectionFactory
-// will create the necessary thread internally. If |signaling_thread| is null,
+// 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>

api/rtc_error.h

@@ -176,7 +176,7 @@ inline std::ostream& operator<<(  // no-presubmit-check TODO(webrtc:8982)
 #endif  // WEBRTC_UNIT_TEST
 
 // Helper macro that can be used by implementations to create an error with a
-// message and log it. |message| should be a string literal or movable
+// message and log it. `message` should be a string literal or movable
 // std::string.
 #define LOG_AND_RETURN_ERROR_EX(type, message, severity)           \
   {                                                                \

api/rtc_event_log/rtc_event_log.h

@@ -42,7 +42,7 @@ class RtcEventLog {
   // which it would be permissible to read and/or modify it.
   virtual void StopLogging() = 0;
 
-  // Stops logging to file and calls |callback| when the file has been closed.
+  // Stops logging to file and calls `callback` when the file has been closed.
   // Note that it is not safe to call any other members, including the
   // destructor, until the callback has been called.
   // TODO(srte): Remove default implementation when it's safe to do so.

api/rtc_event_log_output.h

@@ -29,7 +29,7 @@ class RtcEventLogOutput {
   // Write encoded events to an output. Returns true if the output was
   // successfully written in its entirety. Otherwise, no guarantee is given
   // about how much data was written, if any. The output sink becomes inactive
-  // after the first time |false| is returned. Write() may not be called on
+  // after the first time `false` is returned. Write() may not be called on
   // an inactive output sink.
   virtual bool Write(const std::string& output) = 0;
 

api/rtp_packet_info.h

@@ -23,9 +23,9 @@
 namespace webrtc {
 
 //
-// Structure to hold information about a received |RtpPacket|. It is primarily
+// Structure to hold information about a received `RtpPacket`. It is primarily
 // used to carry per-packet information from when a packet is received until
-// the information is passed to |SourceTracker|.
+// the information is passed to `SourceTracker`.
 //
 class RTC_EXPORT RtpPacketInfo {
  public:
@@ -102,8 +102,8 @@ class RTC_EXPORT RtpPacketInfo {
 
   // Fields from the Absolute Capture Time header extension:
   // http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
-  // To not be confused with |local_capture_clock_offset_|, the
+  // To not be confused with `local_capture_clock_offset_`, the
-  // |estimated_capture_clock_offset| in |absolute_capture_time_| should
+  // `estimated_capture_clock_offset` in `absolute_capture_time_` should
   // represent the clock offset between a remote sender and the capturer, and
   // thus equals to the corresponding values in the received RTP packets,
   // subjected to possible interpolations.

api/rtp_parameters.h

@@ -126,7 +126,7 @@ struct RTC_EXPORT RtpCodecCapability {
   RtpCodecCapability();
   ~RtpCodecCapability();
 
-  // Build MIME "type/subtype" string from |name| and |kind|.
+  // Build MIME "type/subtype" string from `name` and `kind`.
   std::string mime_type() const { return MediaTypeToString(kind) + "/" + name; }
 
   // Used to identify the codec. Equivalent to MIME subtype.
@@ -537,7 +537,7 @@ struct RTC_EXPORT RtpCodecParameters {
   RtpCodecParameters(const RtpCodecParameters&);
   ~RtpCodecParameters();
 
-  // Build MIME "type/subtype" string from |name| and |kind|.
+  // Build MIME "type/subtype" string from `name` and `kind`.
   std::string mime_type() const { return MediaTypeToString(kind) + "/" + name; }
 
   // Used to identify the codec. Equivalent to MIME subtype.
@@ -562,7 +562,7 @@ struct RTC_EXPORT RtpCodecParameters {
   absl::optional<int> num_channels;
 
   // The maximum packetization time to be used by an RtpSender.
-  // If |ptime| is also set, this will be ignored.
+  // If `ptime` is also set, this will be ignored.
   // TODO(deadbeef): Not implemented.
   absl::optional<int> max_ptime;
 
@@ -607,7 +607,7 @@ struct RTC_EXPORT RtpCapabilities {
 
   // Supported Forward Error Correction (FEC) mechanisms. Note that the RED,
   // ulpfec and flexfec codecs used by these mechanisms will still appear in
-  // |codecs|.
+  // `codecs`.
   std::vector<FecMechanism> fec;
 
   bool operator==(const RtpCapabilities& o) const {

api/rtp_receiver_interface.h

@@ -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
+  // delay may differ depending on the congestion control. `delay_seconds` is a
-  // positive value including 0.0 measured in seconds. |nullopt| means default
+  // positive value including 0.0 measured in seconds. `nullopt` means default
   // value must be used.
   virtual void SetJitterBufferMinimumDelay(
       absl::optional<double> delay_seconds) = 0;

api/scoped_refptr.h

@@ -24,13 +24,13 @@
 //   void some_function() {
 //     scoped_refptr<MyFoo> foo = new MyFoo();
 //     foo->Method(param);
-//     // |foo| is released when this function returns
+//     // `foo` is released when this function returns
 //   }
 //
 //   void some_other_function() {
 //     scoped_refptr<MyFoo> foo = new MyFoo();
 //     ...
-//     foo = nullptr;  // explicitly releases |foo|
+//     foo = nullptr;  // explicitly releases `foo`
 //     ...
 //     if (foo)
 //       foo->Method(param);
@@ -45,10 +45,10 @@
 //     scoped_refptr<MyFoo> b;
 //
 //     b.swap(a);
-//     // now, |b| references the MyFoo object, and |a| references null.
+//     // now, `b` references the MyFoo object, and `a` references null.
 //   }
 //
-// To make both |a| and |b| in the above example reference the same MyFoo
+// To make both `a` and `b` in the above example reference the same MyFoo
 // object, simply use the assignment operator:
 //
 //   {
@@ -56,7 +56,7 @@
 //     scoped_refptr<MyFoo> b;
 //
 //     b = a;
-//     // now, |a| and |b| each own a reference to the same MyFoo object.
+//     // now, `a` and `b` each own a reference to the same MyFoo object.
 //   }
 //
 

api/stats/rtc_stats.h

@@ -35,7 +35,7 @@ class RTCStatsMemberInterface;
 //   static const char kType[];
 // It is used as a unique class identifier and a string representation of the
 // class type, see https://w3c.github.io/webrtc-stats/#rtcstatstype-str*.
-// Use the |WEBRTC_RTCSTATS_IMPL| macro when implementing subclasses, see macro
+// Use the `WEBRTC_RTCSTATS_IMPL` macro when implementing subclasses, see macro
 // for details.
 //
 // Derived classes list their dictionary members, RTCStatsMember<T>, as public
@@ -47,7 +47,7 @@ class RTCStatsMemberInterface;
 // foo.baz->push_back("hello world");
 // uint32_t x = *foo.bar;
 //
-// Pointers to all the members are available with |Members|, allowing iteration:
+// Pointers to all the members are available with `Members`, allowing iteration:
 //
 // for (const RTCStatsMemberInterface* member : foo.Members()) {
 //   printf("%s = %s\n", member->name(), member->ValueToString().c_str());
@@ -65,11 +65,11 @@ class RTC_EXPORT RTCStats {
   const std::string& id() const { return id_; }
   // Time relative to the UNIX epoch (Jan 1, 1970, UTC), in microseconds.
   int64_t timestamp_us() const { return timestamp_us_; }
-  // Returns the static member variable |kType| of the implementing class.
+  // Returns the static member variable `kType` of the implementing class.
   virtual const char* type() const = 0;
-  // Returns a vector of pointers to all the |RTCStatsMemberInterface| members
+  // Returns a vector of pointers to all the `RTCStatsMemberInterface` members
   // of this class. This allows for iteration of members. For a given class,
-  // |Members| always returns the same members in the same order.
+  // `Members` always returns the same members in the same order.
   std::vector<const RTCStatsMemberInterface*> Members() const;
   // Checks if the two stats objects are of the same type and have the same
   // member values. Timestamps are not compared. These operators are exposed for
@@ -81,8 +81,8 @@ class RTC_EXPORT RTCStats {
   // object, listing all of its members (names and values).
   std::string ToJson() const;
 
-  // Downcasts the stats object to an |RTCStats| subclass |T|. DCHECKs that the
+  // Downcasts the stats object to an `RTCStats` subclass `T`. DCHECKs that the
-  // object is of type |T|.
+  // object is of type `T`.
   template <typename T>
   const T& cast_to() const {
     RTC_DCHECK_EQ(type(), T::kType);
@@ -90,8 +90,8 @@ class RTC_EXPORT RTCStats {
   }
 
  protected:
-  // Gets a vector of all members of this |RTCStats| object, including members
+  // Gets a vector of all members of this `RTCStats` object, including members
-  // derived from parent classes. |additional_capacity| is how many more members
+  // derived from parent classes. `additional_capacity` is how many more members
   // shall be reserved in the vector (so that subclasses can allocate a vector
   // with room for both parent and child members without it having to resize).
   virtual std::vector<const RTCStatsMemberInterface*>
@@ -101,21 +101,21 @@ class RTC_EXPORT RTCStats {
   int64_t timestamp_us_;
 };
 
-// All |RTCStats| classes should use these macros.
+// All `RTCStats` classes should use these macros.
-// |WEBRTC_RTCSTATS_DECL| is placed in a public section of the class definition.
+// `WEBRTC_RTCSTATS_DECL` is placed in a public section of the class definition.
-// |WEBRTC_RTCSTATS_IMPL| is placed outside the class definition (in a .cc).
+// `WEBRTC_RTCSTATS_IMPL` is placed outside the class definition (in a .cc).
 //
-// These macros declare (in _DECL) and define (in _IMPL) the static |kType| and
+// These macros declare (in _DECL) and define (in _IMPL) the static `kType` and
-// overrides methods as required by subclasses of |RTCStats|: |copy|, |type| and
+// overrides methods as required by subclasses of `RTCStats`: `copy`, `type` and
-// |MembersOfThisObjectAndAncestors|. The |...| argument is a list of addresses
+// `MembersOfThisObjectAndAncestors`. The |...| argument is a list of addresses
 // to each member defined in the implementing class. The list must have at least
 // one member.
 //
 // (Since class names need to be known to implement these methods this cannot be
-// part of the base |RTCStats|. While these methods could be implemented using
+// part of the base `RTCStats`. While these methods could be implemented using
 // templates, that would only work for immediate subclasses. Subclasses of
 // subclasses also have to override these methods, resulting in boilerplate
-// code. Using a macro avoids this and works for any |RTCStats| class, including
+// code. Using a macro avoids this and works for any `RTCStats` class, including
 // grandchildren.)
 //
 // Sample usage:
@@ -215,10 +215,10 @@ enum class NonStandardGroupId {
   kRtcStatsRelativePacketArrivalDelay,
 };
 
-// Interface for |RTCStats| members, which have a name and a value of a type
+// Interface for `RTCStats` members, which have a name and a value of a type
-// defined in a subclass. Only the types listed in |Type| are supported, these
+// defined in a subclass. Only the types listed in `Type` are supported, these
 // are implemented by |RTCStatsMember<T>|. The value of a member may be
-// undefined, the value can only be read if |is_defined|.
+// undefined, the value can only be read if `is_defined`.
 class RTCStatsMemberInterface {
  public:
   // Member value types.
@@ -284,7 +284,7 @@ class RTCStatsMemberInterface {
   bool is_defined_;
 };
 
-// Template implementation of |RTCStatsMemberInterface|.
+// Template implementation of `RTCStatsMemberInterface`.
 // The supported types are the ones described by
 // |RTCStatsMemberInterface::Type|.
 template <typename T>

api/stats/rtc_stats_report.h

@@ -30,7 +30,7 @@
 namespace webrtc {
 
 // A collection of stats.
-// This is accessible as a map from |RTCStats::id| to |RTCStats|.
+// This is accessible as a map from `RTCStats::id` to `RTCStats`.
 class RTC_EXPORT RTCStatsReport final
     : public rtc::RefCountedNonVirtual<RTCStatsReport> {
  public:
@@ -71,8 +71,8 @@ class RTC_EXPORT RTCStatsReport final
   const RTCStats* Get(const std::string& id) const;
   size_t size() const { return stats_.size(); }
 
-  // Gets the stat object of type |T| by ID, where |T| is any class descending
+  // Gets the stat object of type `T` by ID, where `T` is any class descending
-  // from |RTCStats|.
+  // from `RTCStats`.
   // Returns null if there is no stats object for the given ID or it is the
   // wrong type.
   template <typename T>
@@ -85,17 +85,17 @@ class RTC_EXPORT RTCStatsReport final
   }
 
   // Removes the stats object from the report, returning ownership of it or null
-  // if there is no object with |id|.
+  // if there is no object with `id`.
   std::unique_ptr<const RTCStats> Take(const std::string& id);
-  // Takes ownership of all the stats in |other|, leaving it empty.
+  // Takes ownership of all the stats in `other`, leaving it empty.
   void TakeMembersFrom(rtc::scoped_refptr<RTCStatsReport> other);
 
   // Stats iterators. Stats are ordered lexicographically on |RTCStats::id|.
   ConstIterator begin() const;
   ConstIterator end() const;
 
-  // Gets the subset of stats that are of type |T|, where |T| is any class
+  // Gets the subset of stats that are of type `T`, where `T` is any class
-  // descending from |RTCStats|.
+  // descending from `RTCStats`.
   template <typename T>
   std::vector<const T*> GetStatsOfType() const {
     std::vector<const T*> stats_of_type;

api/stats/rtcstats_objects.h

@@ -197,7 +197,7 @@ class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats {
 };
 
 // https://w3c.github.io/webrtc-stats/#icecandidate-dict*
-// TODO(hbos): |RTCStatsCollector| only collects candidates that are part of
+// TODO(hbos): `RTCStatsCollector` only collects candidates that are part of
 // ice candidate pairs, but there could be candidates not paired with anything.
 // crbug.com/632723
 // TODO(qingsi): Add the stats of STUN binding requests (keepalives) and collect
@@ -221,7 +221,7 @@ class RTC_EXPORT RTCIceCandidateStats : public RTCStats {
   // TODO(hbos): Support enum types? "RTCStatsMember<RTCIceCandidateType>"?
   RTCStatsMember<std::string> candidate_type;
   RTCStatsMember<int32_t> priority;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/632723
+  // TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/632723
   RTCStatsMember<std::string> url;
 
  protected:
@@ -232,8 +232,8 @@ class RTC_EXPORT RTCIceCandidateStats : public RTCStats {
 };
 
 // In the spec both local and remote varieties are of type RTCIceCandidateStats.
-// But here we define them as subclasses of |RTCIceCandidateStats| because the
+// But here we define them as subclasses of `RTCIceCandidateStats` because the
-// |kType| need to be different ("RTCStatsType type") in the local/remote case.
+// `kType` need to be different ("RTCStatsType type") in the local/remote case.
 // https://w3c.github.io/webrtc-stats/#rtcstatstype-str*
 // This forces us to have to override copy() and type().
 class RTC_EXPORT RTCLocalIceCandidateStats final : public RTCIceCandidateStats {
@@ -289,28 +289,28 @@ class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
   RTCStatsMember<std::string> media_source_id;
   RTCStatsMember<bool> remote_source;
   RTCStatsMember<bool> ended;
-  // TODO(hbos): |RTCStatsCollector| does not return stats for detached tracks.
+  // TODO(hbos): `RTCStatsCollector` does not return stats for detached tracks.
   // crbug.com/659137
   RTCStatsMember<bool> detached;
-  // See |RTCMediaStreamTrackKind| for valid values.
+  // See `RTCMediaStreamTrackKind` for valid values.
   RTCStatsMember<std::string> kind;
   RTCStatsMember<double> jitter_buffer_delay;
   RTCStatsMember<uint64_t> jitter_buffer_emitted_count;
   // Video-only members
   RTCStatsMember<uint32_t> frame_width;
   RTCStatsMember<uint32_t> frame_height;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
+  // TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
   RTCStatsMember<double> frames_per_second;
   RTCStatsMember<uint32_t> frames_sent;
   RTCStatsMember<uint32_t> huge_frames_sent;
   RTCStatsMember<uint32_t> frames_received;
   RTCStatsMember<uint32_t> frames_decoded;
   RTCStatsMember<uint32_t> frames_dropped;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
+  // TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
   RTCStatsMember<uint32_t> frames_corrupted;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
+  // TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
   RTCStatsMember<uint32_t> partial_frames_lost;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
+  // TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
   RTCStatsMember<uint32_t> full_frames_lost;
   // Audio-only members
   RTCStatsMember<double> audio_level;         // Receive-only

api/stats_types.cc

@@ -835,7 +835,7 @@ StatsReport* StatsCollection::ReplaceOrAddNew(const StatsReport::Id& id) {
   return InsertNew(id);
 }
 
-// Looks for a report with the given |id|.  If one is not found, null
+// Looks for a report with the given `id`.  If one is not found, null
 // will be returned.
 StatsReport* StatsCollection::Find(const StatsReport::Id& id) {
   RTC_DCHECK(thread_checker_.IsCurrent());

api/stats_types.h

@@ -39,58 +39,58 @@ class RTC_EXPORT StatsReport {
 
   enum StatsType {
     // StatsReport types.
-    // A StatsReport of |type| = "googSession" contains overall information
+    // A StatsReport of `type` = "googSession" contains overall information
     // about the thing libjingle calls a session (which may contain one
     // or more RTP sessions.
     kStatsReportTypeSession,
 
-    // A StatsReport of |type| = "googTransport" contains information
+    // A StatsReport of `type` = "googTransport" contains information
     // about a libjingle "transport".
     kStatsReportTypeTransport,
 
-    // A StatsReport of |type| = "googComponent" contains information
+    // A StatsReport of `type` = "googComponent" contains information
     // about a libjingle "channel" (typically, RTP or RTCP for a transport).
     // This is intended to be the same thing as an ICE "Component".
     kStatsReportTypeComponent,
 
-    // A StatsReport of |type| = "googCandidatePair" contains information
+    // A StatsReport of `type` = "googCandidatePair" contains information
     // about a libjingle "connection" - a single source/destination port pair.
     // This is intended to be the same thing as an ICE "candidate pair".
     kStatsReportTypeCandidatePair,
 
-    // A StatsReport of |type| = "VideoBWE" is statistics for video Bandwidth
+    // A StatsReport of `type` = "VideoBWE" is statistics for video Bandwidth
-    // Estimation, which is global per-session.  The |id| field is "bweforvideo"
+    // Estimation, which is global per-session.  The `id` field is "bweforvideo"
     // (will probably change in the future).
     kStatsReportTypeBwe,
 
-    // A StatsReport of |type| = "ssrc" is statistics for a specific rtp stream.
+    // A StatsReport of `type` = "ssrc" is statistics for a specific rtp stream.
-    // The |id| field is the SSRC in decimal form of the rtp stream.
+    // The `id` field is the SSRC in decimal form of the rtp stream.
     kStatsReportTypeSsrc,
 
-    // A StatsReport of |type| = "remoteSsrc" is statistics for a specific
+    // A StatsReport of `type` = "remoteSsrc" is statistics for a specific
     // rtp stream, generated by the remote end of the connection.
     kStatsReportTypeRemoteSsrc,
 
-    // A StatsReport of |type| = "googTrack" is statistics for a specific media
+    // A StatsReport of `type` = "googTrack" is statistics for a specific media
-    // track. The |id| field is the track id.
+    // track. The `id` field is the track id.
     kStatsReportTypeTrack,
 
-    // A StatsReport of |type| = "localcandidate" or "remotecandidate" is
+    // A StatsReport of `type` = "localcandidate" or "remotecandidate" is
     // attributes on a specific ICE Candidate. It links to its connection pair
     // by candidate id. The string value is taken from
     // http://w3c.github.io/webrtc-stats/#rtcstatstype-enum*.
     kStatsReportTypeIceLocalCandidate,
     kStatsReportTypeIceRemoteCandidate,
 
-    // A StatsReport of |type| = "googCertificate" contains an SSL certificate
+    // A StatsReport of `type` = "googCertificate" contains an SSL certificate
-    // transmitted by one of the endpoints of this connection.  The |id| is
+    // transmitted by one of the endpoints of this connection.  The `id` is
     // controlled by the fingerprint, and is used to identify the certificate in
     // the Channel stats (as "googLocalCertificateId" or
     // "googRemoteCertificateId") and in any child certificates (as
     // "googIssuerId").
     kStatsReportTypeCertificate,
 
-    // A StatsReport of |type| = "datachannel" with statistics for a
+    // A StatsReport of `type` = "datachannel" with statistics for a
     // particular DataChannel.
     kStatsReportTypeDataChannel,
   };
@@ -331,7 +331,7 @@ class RTC_EXPORT StatsReport {
     bool bool_val() const;
     const Id& id_val() const;
 
-    // Returns the string representation of |name|.
+    // Returns the string representation of `name`.
     const char* display_name() const;
 
     // Converts the native value to a string representation of the value.
@@ -339,7 +339,7 @@ class RTC_EXPORT StatsReport {
 
     Type type() const { return type_; }
 
-    // TODO(tommi): Move |name| and |display_name| out of the Value struct.
+    // TODO(tommi): Move `name` and `display_name` out of the Value struct.
     const StatsValueName name;
 
    private:
@@ -364,7 +364,7 @@ class RTC_EXPORT StatsReport {
   typedef rtc::scoped_refptr<Value> ValuePtr;
   typedef std::map<StatsValueName, ValuePtr> Values;
 
-  // Ownership of |id| is passed to |this|.
+  // Ownership of `id` is passed to `this`.
   explicit StatsReport(const Id& id);
   ~StatsReport();
 
@@ -434,13 +434,13 @@ class StatsCollection {
   const_iterator end() const;
   size_t size() const;
 
-  // Creates a new report object with |id| that does not already
+  // Creates a new report object with `id` that does not already
   // exist in the list of reports.
   StatsReport* InsertNew(const StatsReport::Id& id);
   StatsReport* FindOrAddNew(const StatsReport::Id& id);
   StatsReport* ReplaceOrAddNew(const StatsReport::Id& id);
 
-  // Looks for a report with the given |id|.  If one is not found, null
+  // Looks for a report with the given `id`.  If one is not found, null
   // will be returned.
   StatsReport* Find(const StatsReport::Id& id);
 

api/task_queue/queued_task.h

@@ -20,9 +20,9 @@ class QueuedTask {
   virtual ~QueuedTask() = default;
 
   // Main routine that will run when the task is executed on the desired queue.
-  // The task should return |true| to indicate that it should be deleted or
+  // The task should return `true` to indicate that it should be deleted or
-  // |false| to indicate that the queue should consider ownership of the task
+  // `false` to indicate that the queue should consider ownership of the task
-  // having been transferred.  Returning |false| can be useful if a task has
+  // having been transferred.  Returning `false` can be useful if a task has
   // re-posted itself to a different queue or is otherwise being re-used.
   virtual bool Run() = 0;
 };

api/task_queue/task_queue_test.cc

@@ -37,7 +37,7 @@ TEST_P(TaskQueueTest, PostAndCheckCurrent) {
   rtc::Event event;
   auto queue = CreateTaskQueue(factory, "PostAndCheckCurrent");
 
-  // We're not running a task, so |queue| shouldn't be current.
+  // We're not running a task, so `queue` shouldn't be current.
   // Note that because rtc::Thread also supports the TQ interface and
   // TestMainImpl::Init wraps the main test thread (bugs.webrtc.org/9714), that
   // means that TaskQueueBase::Current() will still return a valid value.
@@ -190,7 +190,7 @@ TEST_P(TaskQueueTest, PostAndReuse) {
 }
 
 TEST_P(TaskQueueTest, PostALot) {
-  // Waits until DecrementCount called |count| times. Thread safe.
+  // Waits until DecrementCount called `count` times. Thread safe.
   class BlockingCounter {
    public:
     explicit BlockingCounter(int initial_count) : count_(initial_count) {}

api/test/audio_quality_analyzer_interface.h

@@ -25,9 +25,9 @@ class AudioQualityAnalyzerInterface : public StatsObserverInterface {
   ~AudioQualityAnalyzerInterface() override = default;
 
   // Will be called by the framework before the test.
-  // |test_case_name| is name of test case, that should be used to report all
+  // `test_case_name` is name of test case, that should be used to report all
   // audio metrics.
-  // |analyzer_helper| is a pointer to a class that will allow track_id to
+  // `analyzer_helper` is a pointer to a class that will allow track_id to
   // stream_id matching. The caller is responsible for ensuring the
   // AnalyzerHelper outlives the instance of the AudioQualityAnalyzerInterface.
   virtual void Start(std::string test_case_name,

api/test/audioproc_float.h

@@ -23,12 +23,12 @@ namespace test {
 // utility can be used to simulate the audioprocessing module using a recording
 // (either an AEC dump or wav files), and generate the output as a wav file.
 // Any audio_processing object specified in the input is used for the
-// simulation. The optional |audio_processing| object provides the
+// simulation. The optional `audio_processing` object provides the
 // AudioProcessing instance that is used during the simulation. Note that when
 // the audio_processing object is specified all functionality that relies on
 // using the AudioProcessingBuilder is deactivated, since the AudioProcessing
 // object is already created and the builder is not used in the simulation. It
-// is needed to pass the command line flags as |argc| and |argv|, so these can
+// is needed to pass the command line flags as `argc` and `argv`, so these can
 // be interpreted properly by the utility. To see a list of all supported
 // command line flags, run the executable with the '--help' flag.
 int AudioprocFloat(rtc::scoped_refptr<AudioProcessing> audio_processing,
@@ -38,10 +38,10 @@ int AudioprocFloat(rtc::scoped_refptr<AudioProcessing> audio_processing,
 // This is an interface for the audio processing simulation utility. This
 // utility can be used to simulate the audioprocessing module using a recording
 // (either an AEC dump or wav files), and generate the output as a wav file.
-// The |ap_builder| object will be used to create the AudioProcessing instance
+// The `ap_builder` object will be used to create the AudioProcessing instance
-// that is used during the simulation. The |ap_builder| supports setting of
+// that is used during the simulation. The `ap_builder` supports setting of
 // injectable components, which will be passed on to the created AudioProcessing
-// instance. It is needed to pass the command line flags as |argc| and |argv|,
+// instance. It is needed to pass the command line flags as `argc` and `argv`,
 // so these can be interpreted properly by the utility.
 // To get a fully-working audioproc_f utility, all that is needed is to write a
 // main function, create an AudioProcessingBuilder, optionally set custom
@@ -56,9 +56,9 @@ int AudioprocFloat(std::unique_ptr<AudioProcessingBuilder> ap_builder,
 // Interface for the audio processing simulation utility, which is similar to
 // the one above, but which adds the option of receiving the input as a string
 // and returning the output as an array. The first three arguments fulfill the
-// same purpose as above. Pass the |input_aecdump| to provide the content of an
+// same purpose as above. Pass the `input_aecdump` to provide the content of an
 // AEC dump file as a string. After the simulation is completed,
-// |processed_capture_samples| will contain the the samples processed on the
+// `processed_capture_samples` will contain the the samples processed on the
 // capture side.
 int AudioprocFloat(std::unique_ptr<AudioProcessingBuilder> ap_builder,
                    int argc,

api/test/create_frame_generator.h

@@ -24,8 +24,8 @@ namespace test {
 
 // Creates a frame generator that produces frames with small squares that
 // move randomly towards the lower right corner.
-// |type| has the default value FrameGeneratorInterface::OutputType::I420.
+// `type` has the default value FrameGeneratorInterface::OutputType::I420.
-// |num_squares| has the default value 10.
+// `num_squares` has the default value 10.
 std::unique_ptr<FrameGeneratorInterface> CreateSquareFrameGenerator(
     int width,
     int height,
@@ -66,7 +66,7 @@ CreateScrollingInputFromYuvFilesFrameGenerator(
 
 // Creates a frame generator that produces randomly generated slides. It fills
 // the frames with randomly sized and colored squares.
-// |frame_repeat_count| determines how many times each slide is shown.
+// `frame_repeat_count` determines how many times each slide is shown.
 std::unique_ptr<FrameGeneratorInterface>
 CreateSlideFrameGenerator(int width, int height, int frame_repeat_count);
 

api/test/create_peer_connection_quality_test_frame_generator.cc

@@ -30,7 +30,7 @@ void ValidateScreenShareConfig(const VideoConfig& video_config,
                                const ScreenShareConfig& screen_share_config) {
   if (screen_share_config.slides_yuv_file_names.empty()) {
     if (screen_share_config.scrolling_params) {
-      // If we have scrolling params, then its |source_width| and |source_heigh|
+      // If we have scrolling params, then its `source_width` and `source_heigh`
       // will be used as width and height of video input, so we have to validate
       // it against width and height of default input.
       RTC_CHECK_EQ(screen_share_config.scrolling_params->source_width,

api/test/create_peer_connection_quality_test_frame_generator.h

@@ -21,7 +21,7 @@ namespace webrtc {
 namespace webrtc_pc_e2e {
 
 // Creates a frame generator that produces frames with small squares that move
-// randomly towards the lower right corner. |type| has the default value
+// randomly towards the lower right corner. `type` has the default value
 // FrameGeneratorInterface::OutputType::I420. video_config specifies frame
 // weight and height.
 std::unique_ptr<test::FrameGeneratorInterface> CreateSquareFrameGenerator(

api/test/create_peerconnection_quality_test_fixture.h

@@ -25,10 +25,10 @@ namespace webrtc_pc_e2e {
 
 // Create test fixture to establish test call between Alice and Bob.
 // During the test Alice will be caller and Bob will answer the call.
-// |test_case_name| is a name of test case, that will be used for all metrics
+// `test_case_name` is a name of test case, that will be used for all metrics
 // reporting.
-// |time_controller| is used to manage all rtc::Thread's and TaskQueue
+// `time_controller` is used to manage all rtc::Thread's and TaskQueue
-// instances. Instance of |time_controller| have to outlive created fixture.
+// instances. Instance of `time_controller` have to outlive created fixture.
 // Returns a non-null PeerConnectionE2EQualityTestFixture instance.
 std::unique_ptr<PeerConnectionE2EQualityTestFixture>
 CreatePeerConnectionE2EQualityTestFixture(

api/test/create_time_controller.h

@@ -17,7 +17,7 @@
 
 namespace webrtc {
 
-// Creates a time coltroller that wraps |alarm|.
+// Creates a time coltroller that wraps `alarm`.
 std::unique_ptr<TimeController> CreateTimeController(
     ControlledAlarmClock* alarm);
 

api/test/network_emulation/cross_traffic.h

@@ -27,12 +27,12 @@ class CrossTrafficRoute {
  public:
   virtual ~CrossTrafficRoute() = default;
 
-  // Triggers sending of dummy packets with size |packet_size| bytes.
+  // Triggers sending of dummy packets with size `packet_size` bytes.
   virtual void TriggerPacketBurst(size_t num_packets, size_t packet_size) = 0;
   // Sends a packet over the nodes. The content of the packet is unspecified;
   // only the size metter for the emulation purposes.
   virtual void SendPacket(size_t packet_size) = 0;
-  // Sends a packet over the nodes and runs |action| when it has been delivered.
+  // Sends a packet over the nodes and runs `action` when it has been delivered.
   virtual void NetworkDelayedAction(size_t packet_size,
                                     std::function<void()> action) = 0;
 };

api/test/network_emulation/network_emulation_interfaces.h

@@ -204,9 +204,9 @@ class EmulatedNetworkStats {
 class EmulatedEndpoint : public EmulatedNetworkReceiverInterface {
  public:
   // Send packet into network.
-  // |from| will be used to set source address for the packet in destination
+  // `from` will be used to set source address for the packet in destination
   // socket.
-  // |to| will be used for routing verification and picking right socket by port
+  // `to` will be used for routing verification and picking right socket by port
   // on destination endpoint.
   virtual void SendPacket(const rtc::SocketAddress& from,
                           const rtc::SocketAddress& to,
@@ -214,12 +214,12 @@ class EmulatedEndpoint : public EmulatedNetworkReceiverInterface {
                           uint16_t application_overhead = 0) = 0;
 
   // Binds receiver to this endpoint to send and receive data.
-  // |desired_port| is a port that should be used. If it is equal to 0,
+  // `desired_port` is a port that should be used. If it is equal to 0,
   // endpoint will pick the first available port starting from
-  // |kFirstEphemeralPort|.
+  // `kFirstEphemeralPort`.
   //
   // Returns the port, that should be used (it will be equals to desired, if
-  // |desired_port| != 0 and is free or will be the one, selected by endpoint)
+  // `desired_port` != 0 and is free or will be the one, selected by endpoint)
   // or absl::nullopt if desired_port in used. Also fails if there are no more
   // free ports to bind to.
   //
@@ -256,7 +256,7 @@ class EmulatedEndpoint : public EmulatedNetworkReceiverInterface {
 // they are guranteed to be delivered eventually, even on lossy networks.
 class TcpMessageRoute {
  public:
-  // Sends a TCP message of the given |size| over the route, |on_received| is
+  // Sends a TCP message of the given `size` over the route, `on_received` is
   // called when the message has been delivered. Note that the connection
   // parameters are reset iff there's no currently pending message on the route.
   virtual void SendMessage(size_t size, std::function<void()> on_received) = 0;

api/test/network_emulation_manager.h

@@ -130,7 +130,7 @@ class EmulatedNetworkManagerInterface {
   virtual std::vector<EmulatedEndpoint*> endpoints() const = 0;
 
   // Passes summarized network stats for endpoints for this manager into
-  // specified |stats_callback|. Callback will be executed on network emulation
+  // specified `stats_callback`. Callback will be executed on network emulation
   // internal task queue.
   virtual void GetStats(
       std::function<void(std::unique_ptr<EmulatedNetworkStats>)> stats_callback)
@@ -180,13 +180,13 @@ class NetworkEmulationManager {
 
   // Creates an emulated network node, which represents single network in
   // the emulated network layer. Uses default implementation on network behavior
-  // which can be configured with |config|. |random_seed| can be provided to
+  // which can be configured with `config`. `random_seed` can be provided to
   // alter randomization behavior.
   virtual EmulatedNetworkNode* CreateEmulatedNode(
       BuiltInNetworkBehaviorConfig config,
       uint64_t random_seed = 1) = 0;
   // Creates an emulated network node, which represents single network in
-  // the emulated network layer. |network_behavior| determines how created node
+  // the emulated network layer. `network_behavior` determines how created node
   // will forward incoming packets to the next receiver.
   virtual EmulatedNetworkNode* CreateEmulatedNode(
       std::unique_ptr<NetworkBehaviorInterface> network_behavior) = 0;
@@ -205,8 +205,8 @@ class NetworkEmulationManager {
 
   // Creates a route between endpoints going through specified network nodes.
   // This route is single direction only and describe how traffic that was
-  // sent by network interface |from| have to be delivered to the network
+  // sent by network interface `from` have to be delivered to the network
-  // interface |to|. Return object can be used to remove created route. The
+  // interface `to`. Return object can be used to remove created route. The
   // route must contains at least one network node inside it.
   //
   // Assume that E{0-9} are endpoints and N{0-9} are network nodes, then
@@ -228,7 +228,7 @@ class NetworkEmulationManager {
       const std::vector<EmulatedNetworkNode*>& via_nodes,
       EmulatedEndpoint* to) = 0;
 
-  // Creates a route over the given |via_nodes| creating the required endpoints
+  // Creates a route over the given `via_nodes` creating the required endpoints
   // in the process. The returned EmulatedRoute pointer can be used in other
   // calls as a transport route for message or cross traffic.
   virtual EmulatedRoute* CreateRoute(
@@ -239,7 +239,7 @@ class NetworkEmulationManager {
   // packet's destination IP.
   //
   // This route is single direction only and describe how traffic that was
-  // sent by network interface |from| have to be delivered in case if routing
+  // sent by network interface `from` have to be delivered in case if routing
   // was unspecified. Return object can be used to remove created route. The
   // route must contains at least one network node inside it.
   //
@@ -269,29 +269,29 @@ class NetworkEmulationManager {
   // packets being dropped.
   virtual void ClearRoute(EmulatedRoute* route) = 0;
 
-  // Creates a simulated TCP connection using |send_route| for traffic and
+  // Creates a simulated TCP connection using `send_route` for traffic and
-  // |ret_route| for feedback. This can be used to emulate HTTP cross traffic
+  // `ret_route` for feedback. This can be used to emulate HTTP cross traffic
   // and to implement realistic reliable signaling over lossy networks.
   // TODO(srte): Handle clearing of the routes involved.
   virtual TcpMessageRoute* CreateTcpRoute(EmulatedRoute* send_route,
                                           EmulatedRoute* ret_route) = 0;
 
-  // Creates a route over the given |via_nodes|. Returns an object that can be
+  // Creates a route over the given `via_nodes`. Returns an object that can be
   // used to emulate network load with cross traffic over the created route.
   virtual CrossTrafficRoute* CreateCrossTrafficRoute(
       const std::vector<EmulatedNetworkNode*>& via_nodes) = 0;
 
-  // Starts generating cross traffic using given |generator|. Takes ownership
+  // Starts generating cross traffic using given `generator`. Takes ownership
   // over the generator.
   virtual CrossTrafficGenerator* StartCrossTraffic(
       std::unique_ptr<CrossTrafficGenerator> generator) = 0;
 
-  // Stops generating cross traffic that was started using given |generator|.
+  // Stops generating cross traffic that was started using given `generator`.
-  // The |generator| shouldn't be used after and the reference may be invalid.
+  // The `generator` shouldn't be used after and the reference may be invalid.
   virtual void StopCrossTraffic(CrossTrafficGenerator* generator) = 0;
 
   // Creates EmulatedNetworkManagerInterface which can be used then to inject
-  // network emulation layer into PeerConnection. |endpoints| - are available
+  // network emulation layer into PeerConnection. `endpoints` - are available
   // network interfaces for PeerConnection. If endpoint is enabled, it will be
   // immediately available for PeerConnection, otherwise user will be able to
   // enable endpoint later to make it available for PeerConnection.
@@ -299,8 +299,8 @@ class NetworkEmulationManager {
   CreateEmulatedNetworkManagerInterface(
       const std::vector<EmulatedEndpoint*>& endpoints) = 0;
 
-  // Passes summarized network stats for specified |endpoints| into specified
+  // Passes summarized network stats for specified `endpoints` into specified
-  // |stats_callback|. Callback will be executed on network emulation
+  // `stats_callback`. Callback will be executed on network emulation
   // internal task queue.
   virtual void GetStats(
       rtc::ArrayView<EmulatedEndpoint* const> endpoints,

api/test/peerconnection_quality_test_fixture.h

@@ -67,17 +67,17 @@ class PeerConnectionE2EQualityTestFixture {
   // bottom right corner of the picture.
   //
   // In such case source dimensions must be greater or equal to the sliding
-  // window dimensions. So |source_width| and |source_height| are the dimensions
+  // window dimensions. So `source_width` and `source_height` are the dimensions
-  // of the source frame, while |VideoConfig::width| and |VideoConfig::height|
+  // of the source frame, while `VideoConfig::width` and `VideoConfig::height`
   // are the dimensions of the sliding window.
   //
-  // Because |source_width| and |source_height| are dimensions of the source
+  // Because `source_width` and `source_height` are dimensions of the source
   // frame, they have to be width and height of videos from
-  // |ScreenShareConfig::slides_yuv_file_names|.
+  // `ScreenShareConfig::slides_yuv_file_names`.
   //
   // Because scrolling have to be done on single slide it also requires, that
-  // |duration| must be less or equal to
+  // `duration` must be less or equal to
-  // |ScreenShareConfig::slide_change_interval|.
+  // `ScreenShareConfig::slide_change_interval`.
   struct ScrollingParams {
     ScrollingParams(TimeDelta duration,
                     size_t source_width,
@@ -110,16 +110,16 @@ class PeerConnectionE2EQualityTestFixture {
     // will be applied in such case.
     bool generate_slides = false;
     // If present scrolling will be applied. Please read extra requirement on
-    // |slides_yuv_file_names| for scrolling.
+    // `slides_yuv_file_names` for scrolling.
     absl::optional<ScrollingParams> scrolling_params;
     // Contains list of yuv files with slides.
     //
     // If empty, default set of slides will be used. In such case
-    // |VideoConfig::width| must be equal to |kDefaultSlidesWidth| and
+    // `VideoConfig::width` must be equal to `kDefaultSlidesWidth` and
-    // |VideoConfig::height| must be equal to |kDefaultSlidesHeight| or if
+    // `VideoConfig::height` must be equal to `kDefaultSlidesHeight` or if
-    // |scrolling_params| are specified, then |ScrollingParams::source_width|
+    // `scrolling_params` are specified, then `ScrollingParams::source_width`
-    // must be equal to |kDefaultSlidesWidth| and
+    // must be equal to `kDefaultSlidesWidth` and
-    // |ScrollingParams::source_height| must be equal to |kDefaultSlidesHeight|.
+    // `ScrollingParams::source_height` must be equal to `kDefaultSlidesHeight`.
     std::vector<std::string> slides_yuv_file_names;
   };
 
@@ -128,7 +128,7 @@ class PeerConnectionE2EQualityTestFixture {
   // SVC support is limited:
   // During SVC testing there is no SFU, so framework will try to emulate SFU
   // behavior in regular p2p call. Because of it there are such limitations:
-  //  * if |target_spatial_index| is not equal to the highest spatial layer
+  //  * if `target_spatial_index` is not equal to the highest spatial layer
   //    then no packet/frame drops are allowed.
   //
   //    If there will be any drops, that will affect requested layer, then
@@ -154,11 +154,11 @@ class PeerConnectionE2EQualityTestFixture {
     // Specifies spatial index of the video stream to analyze.
     // There are 2 cases:
     // 1. simulcast encoder is used:
-    //    in such case |target_spatial_index| will specify the index of
+    //    in such case `target_spatial_index` will specify the index of
     //    simulcast stream, that should be analyzed. Other streams will be
     //    dropped.
     // 2. SVC encoder is used:
-    //    in such case |target_spatial_index| will specify the top interesting
+    //    in such case `target_spatial_index` will specify the top interesting
     //    spatial layer and all layers below, including target one will be
     //    processed. All layers above target one will be dropped.
     // If not specified than whatever stream will be received will be analyzed.
@@ -166,8 +166,8 @@ class PeerConnectionE2EQualityTestFixture {
     // network.
     absl::optional<int> target_spatial_index;
 
-    // Encoding parameters per simulcast layer. If not empty, |encoding_params|
+    // Encoding parameters per simulcast layer. If not empty, `encoding_params`
-    // size have to be equal to |simulcast_streams_count|. Will be used to set
+    // size have to be equal to `simulcast_streams_count`. Will be used to set
     // transceiver send encoding params for simulcast layers. Applicable only
     // for codecs that support simulcast (ex. Vp8) and will be ignored
     // otherwise. RtpEncodingParameters::rid may be changed by fixture
@@ -220,7 +220,7 @@ class PeerConnectionE2EQualityTestFixture {
     // was captured during the test for this video stream on sender side.
     // It is useful when generator is used as input.
     absl::optional<std::string> input_dump_file_name;
-    // Used only if |input_dump_file_name| is set. Specifies the module for the
+    // Used only if `input_dump_file_name` is set. Specifies the module for the
     // video frames to be dumped. Modulo equals X means every Xth frame will be
     // written to the dump file. The value must be greater than 0.
     int input_dump_sampling_modulo = 1;
@@ -229,7 +229,7 @@ class PeerConnectionE2EQualityTestFixture {
     // output files will be appended with indexes. The produced files contains
     // what was rendered for this video stream on receiver side.
     absl::optional<std::string> output_dump_file_name;
-    // Used only if |output_dump_file_name| is set. Specifies the module for the
+    // Used only if `output_dump_file_name` is set. Specifies the module for the
     // video frames to be dumped. Modulo equals X means every Xth frame will be
     // written to the dump file. The value must be greater than 0.
     int output_dump_sampling_modulo = 1;
@@ -282,9 +282,9 @@ class PeerConnectionE2EQualityTestFixture {
     std::string name = cricket::kVp8CodecName;
     // Map of parameters, that have to be specified on SDP codec. Each parameter
     // is described by key and value. Codec parameters will match the specified
-    // map if and only if for each key from |required_params| there will be
+    // map if and only if for each key from `required_params` there will be
     // a parameter with name equal to this key and parameter value will be equal
-    // to the value from |required_params| for this key.
+    // to the value from `required_params` for this key.
     // If empty then only name will be used to match the codec.
     std::map<std::string, std::string> required_params;
   };
@@ -351,7 +351,7 @@ class PeerConnectionE2EQualityTestFixture {
         CapturingDeviceIndex capturing_device_index) = 0;
     // Set the list of video codecs used by the peer during the test. These
     // codecs will be negotiated in SDP during offer/answer exchange. The order
-    // of these codecs during negotiation will be the same as in |video_codecs|.
+    // of these codecs during negotiation will be the same as in `video_codecs`.
     // Codecs have to be available in codecs list provided by peer connection to
     // be negotiated. If some of specified codecs won't be found, the test will
     // crash.
@@ -416,9 +416,9 @@ class PeerConnectionE2EQualityTestFixture {
 
     // Invoked by framework after peer connection factory and peer connection
     // itself will be created but before offer/answer exchange will be started.
-    // |test_case_name| is name of test case, that should be used to report all
+    // `test_case_name` is name of test case, that should be used to report all
     // metrics.
-    // |reporter_helper| is a pointer to a class that will allow track_id to
+    // `reporter_helper` is a pointer to a class that will allow track_id to
     // stream_id matching. The caller is responsible for ensuring the
     // TrackIdStreamInfoMap will be valid from Start() to
     // StopAndReportResults().
@@ -433,14 +433,14 @@ class PeerConnectionE2EQualityTestFixture {
   virtual ~PeerConnectionE2EQualityTestFixture() = default;
 
   // Add activity that will be executed on the best effort at least after
-  // |target_time_since_start| after call will be set up (after offer/answer
+  // `target_time_since_start` after call will be set up (after offer/answer
   // exchange, ICE gathering will be done and ICE candidates will passed to
-  // remote side). |func| param is amount of time spent from the call set up.
+  // remote side). `func` param is amount of time spent from the call set up.
   virtual void ExecuteAt(TimeDelta target_time_since_start,
                          std::function<void(TimeDelta)> func) = 0;
-  // Add activity that will be executed every |interval| with first execution
+  // Add activity that will be executed every `interval` with first execution
-  // on the best effort at least after |initial_delay_since_start| after call
+  // on the best effort at least after `initial_delay_since_start` after call
-  // will be set up (after all participants will be connected). |func| param is
+  // will be set up (after all participants will be connected). `func` param is
   // amount of time spent from the call set up.
   virtual void ExecuteEvery(TimeDelta initial_delay_since_start,
                             TimeDelta interval,
@@ -452,15 +452,15 @@ class PeerConnectionE2EQualityTestFixture {
 
   // Add a new peer to the call and return an object through which caller
   // can configure peer's behavior.
-  // |network_thread| will be used as network thread for peer's peer connection
+  // `network_thread` will be used as network thread for peer's peer connection
-  // |network_manager| will be used to provide network interfaces for peer's
+  // `network_manager` will be used to provide network interfaces for peer's
   // peer connection.
-  // |configurer| function will be used to configure peer in the call.
+  // `configurer` function will be used to configure peer in the call.
   virtual void AddPeer(rtc::Thread* network_thread,
                        rtc::NetworkManager* network_manager,
                        rtc::FunctionView<void(PeerConfigurer*)> configurer) = 0;
   // Runs the media quality test, which includes setting up the call with
-  // configured participants, running it according to provided |run_params| and
+  // configured participants, running it according to provided `run_params` and
   // terminating it properly at the end. During call duration media quality
   // metrics are gathered, which are then reported to stdout and (if configured)
   // to the json/protobuf output file through the WebRTC perf test results

api/test/stats_observer_interface.h

@@ -23,7 +23,7 @@ class StatsObserverInterface {
   virtual ~StatsObserverInterface() = default;
 
   // Method called when stats reports are available for the PeerConnection
-  // identified by |pc_label|.
+  // identified by `pc_label`.
   virtual void OnStatsReports(
       absl::string_view pc_label,
       const rtc::scoped_refptr<const RTCStatsReport>& report) = 0;

api/test/time_controller.h

@@ -44,7 +44,7 @@ class TimeController {
   // Creates a process thread.
   virtual std::unique_ptr<ProcessThread> CreateProcessThread(
       const char* thread_name) = 0;
-  // Creates an rtc::Thread instance. If |socket_server| is nullptr, a default
+  // Creates an rtc::Thread instance. If `socket_server` is nullptr, a default
   // noop socket server is created.
   // Returned thread is not null and started.
   virtual std::unique_ptr<rtc::Thread> CreateThread(
@@ -55,12 +55,12 @@ class TimeController {
   // thread.
   virtual rtc::Thread* GetMainThread() = 0;
   // Allow task queues and process threads created by this instance to execute
-  // for the given |duration|.
+  // for the given `duration`.
   virtual void AdvanceTime(TimeDelta duration) = 0;
 
   // Waits until condition() == true, polling condition() in small time
   // intervals.
-  // Returns true if condition() was evaluated to true before |max_duration|
+  // Returns true if condition() was evaluated to true before `max_duration`
   // elapsed and false otherwise.
   bool Wait(const std::function<bool()>& condition,
             TimeDelta max_duration = TimeDelta::Seconds(5));
@@ -75,17 +75,17 @@ class ControlledAlarmClock {
   // Gets a clock that tells the alarm clock's notion of time.
   virtual Clock* GetClock() = 0;
 
-  // Schedules the alarm to fire at |deadline|.
+  // Schedules the alarm to fire at `deadline`.
-  // An alarm clock only supports one deadline. Calls to |ScheduleAlarmAt| with
+  // An alarm clock only supports one deadline. Calls to `ScheduleAlarmAt` with
   // an earlier deadline will reset the alarm to fire earlier.Calls to
-  // |ScheduleAlarmAt| with a later deadline are ignored. Returns true if the
+  // `ScheduleAlarmAt` with a later deadline are ignored. Returns true if the
   // deadline changed, false otherwise.
   virtual bool ScheduleAlarmAt(Timestamp deadline) = 0;
 
   // Sets the callback that should be run when the alarm fires.
   virtual void SetCallback(std::function<void()> callback) = 0;
 
-  // Waits for |duration| to pass, according to the alarm clock.
+  // Waits for `duration` to pass, according to the alarm clock.
   virtual void Sleep(TimeDelta duration) = 0;
 };
 

api/test/track_id_stream_info_map.h

@@ -16,7 +16,7 @@
 namespace webrtc {
 namespace webrtc_pc_e2e {
 
-// Instances of |TrackIdStreamInfoMap| provide bookkeeping capabilities that
+// Instances of `TrackIdStreamInfoMap` provide bookkeeping capabilities that
 // are useful to associate stats reports track_ids to the remote stream info.
 class TrackIdStreamInfoMap {
  public:
@@ -26,12 +26,12 @@ class TrackIdStreamInfoMap {
   // StatsObserverInterface::OnStatsReports is invoked.
 
   // Returns a reference to a stream label owned by the TrackIdStreamInfoMap.
-  // Precondition: |track_id| must be already mapped to stream label.
+  // Precondition: `track_id` must be already mapped to stream label.
   virtual absl::string_view GetStreamLabelFromTrackId(
       absl::string_view track_id) const = 0;
 
   // Returns a reference to a sync group name owned by the TrackIdStreamInfoMap.
-  // Precondition: |track_id| must be already mapped to sync group.
+  // Precondition: `track_id` must be already mapped to sync group.
   virtual absl::string_view GetSyncGroupLabelFromTrackId(
       absl::string_view track_id) const = 0;
 };

api/test/video_quality_analyzer_interface.h

@@ -72,9 +72,9 @@ class VideoQualityAnalyzerInterface : public StatsObserverInterface {
   ~VideoQualityAnalyzerInterface() override = default;
 
   // Will be called by framework before test.
-  // |test_case_name| is name of test case, that should be used to report all
+  // `test_case_name` is name of test case, that should be used to report all
   // video metrics.
-  // |threads_count| is number of threads that analyzer can use for heavy
+  // `threads_count` is number of threads that analyzer can use for heavy
   // calculations. Analyzer can perform simple calculations on the calling
   // thread in each method, but should remember, that it is the same thread,
   // that is used in video pipeline.
@@ -83,57 +83,57 @@ class VideoQualityAnalyzerInterface : public StatsObserverInterface {
                      int max_threads_count) {}
 
   // Will be called when frame was generated from the input stream.
-  // |peer_name| is name of the peer on which side frame was captured.
+  // `peer_name` is name of the peer on which side frame was captured.
   // Returns frame id, that will be set by framework to the frame.
   virtual uint16_t OnFrameCaptured(absl::string_view peer_name,
                                    const std::string& stream_label,
                                    const VideoFrame& frame) = 0;
   // Will be called before calling the encoder.
-  // |peer_name| is name of the peer on which side frame came to encoder.
+  // `peer_name` is name of the peer on which side frame came to encoder.
   virtual void OnFramePreEncode(absl::string_view peer_name,
                                 const VideoFrame& frame) {}
   // Will be called for each EncodedImage received from encoder. Single
   // VideoFrame can produce multiple EncodedImages. Each encoded image will
   // have id from VideoFrame.
-  // |peer_name| is name of the peer on which side frame was encoded.
+  // `peer_name` is name of the peer on which side frame was encoded.
   virtual void OnFrameEncoded(absl::string_view peer_name,
                               uint16_t frame_id,
                               const EncodedImage& encoded_image,
                               const EncoderStats& stats) {}
   // Will be called for each frame dropped by encoder.
-  // |peer_name| is name of the peer on which side frame drop was detected.
+  // `peer_name` is name of the peer on which side frame drop was detected.
   virtual void OnFrameDropped(absl::string_view peer_name,
                               EncodedImageCallback::DropReason reason) {}
   // Will be called before calling the decoder.
-  // |peer_name| is name of the peer on which side frame was received.
+  // `peer_name` is name of the peer on which side frame was received.
   virtual void OnFramePreDecode(absl::string_view peer_name,
                                 uint16_t frame_id,
                                 const EncodedImage& encoded_image) {}
   // Will be called after decoding the frame.
-  // |peer_name| is name of the peer on which side frame was decoded.
+  // `peer_name` is name of the peer on which side frame was decoded.
   virtual void OnFrameDecoded(absl::string_view peer_name,
                               const VideoFrame& frame,
                               const DecoderStats& stats) {}
   // Will be called when frame will be obtained from PeerConnection stack.
-  // |peer_name| is name of the peer on which side frame was rendered.
+  // `peer_name` is name of the peer on which side frame was rendered.
   virtual void OnFrameRendered(absl::string_view peer_name,
                                const VideoFrame& frame) {}
   // Will be called if encoder return not WEBRTC_VIDEO_CODEC_OK.
   // All available codes are listed in
   // modules/video_coding/include/video_error_codes.h
-  // |peer_name| is name of the peer on which side error acquired.
+  // `peer_name` is name of the peer on which side error acquired.
   virtual void OnEncoderError(absl::string_view peer_name,
                               const VideoFrame& frame,
                               int32_t error_code) {}
   // Will be called if decoder return not WEBRTC_VIDEO_CODEC_OK.
   // All available codes are listed in
   // modules/video_coding/include/video_error_codes.h
-  // |peer_name| is name of the peer on which side error acquired.
+  // `peer_name` is name of the peer on which side error acquired.
   virtual void OnDecoderError(absl::string_view peer_name,
                               uint16_t frame_id,
                               int32_t error_code) {}
   // Will be called every time new stats reports are available for the
-  // Peer Connection identified by |pc_label|.
+  // Peer Connection identified by `pc_label`.
   void OnStatsReports(
       absl::string_view pc_label,
       const rtc::scoped_refptr<const RTCStatsReport>& report) override {}

api/test/video_quality_test_fixture.h

@@ -98,7 +98,7 @@ class VideoQualityTestFixtureInterface {
       InterLayerPredMode inter_layer_pred = InterLayerPredMode::kOn;
       // If empty, bitrates are generated in VP9Impl automatically.
       std::vector<SpatialLayer> spatial_layers;
-      // If set, default parameters will be used instead of |streams|.
+      // If set, default parameters will be used instead of `streams`.
       bool infer_streams = false;
     } ss[2];
     struct Logging {

api/transport/bitrate_settings.h

@@ -18,7 +18,7 @@
 
 namespace webrtc {
 
-// Configuration of send bitrate. The |start_bitrate_bps| value is
+// Configuration of send bitrate. The `start_bitrate_bps` value is
 // used for multiple purposes, both as a prior in the bandwidth
 // estimator, and for initial configuration of the encoder. We may
 // want to create separate apis for those, and use a smaller struct

api/transport/data_channel_transport_interface.h

@@ -48,14 +48,14 @@ struct SendDataParams {
   // retransmitted by the transport before it is dropped.
   // Setting this value to zero disables retransmission.
   // Valid values are in the range [0-UINT16_MAX].
-  // |max_rtx_count| and |max_rtx_ms| may not be set simultaneously.
+  // `max_rtx_count` and `max_rtx_ms` may not be set simultaneously.
   absl::optional<int> max_rtx_count;
 
   // If set, the maximum number of milliseconds for which the transport
   // may retransmit this message before it is dropped.
   // Setting this value to zero disables retransmission.
   // Valid values are in the range [0-UINT16_MAX].
-  // |max_rtx_count| and |max_rtx_ms| may not be set simultaneously.
+  // `max_rtx_count` and `max_rtx_ms` may not be set simultaneously.
   absl::optional<int> max_rtx_ms;
 };
 
@@ -96,18 +96,18 @@ class DataChannelTransportInterface {
  public:
   virtual ~DataChannelTransportInterface() = default;
 
-  // Opens a data |channel_id| for sending.  May return an error if the
+  // Opens a data `channel_id` for sending.  May return an error if the
-  // specified |channel_id| is unusable.  Must be called before |SendData|.
+  // specified `channel_id` is unusable.  Must be called before `SendData`.
   virtual RTCError OpenChannel(int channel_id) = 0;
 
   // Sends a data buffer to the remote endpoint using the given send parameters.
-  // |buffer| may not be larger than 256 KiB. Returns an error if the send
+  // `buffer` may not be larger than 256 KiB. Returns an error if the send
   // fails.
   virtual RTCError SendData(int channel_id,
                             const SendDataParams& params,
                             const rtc::CopyOnWriteBuffer& buffer) = 0;
 
-  // Closes |channel_id| gracefully.  Returns an error if |channel_id| is not
+  // Closes `channel_id` gracefully.  Returns an error if `channel_id` is not
   // open.  Data sent after the closing procedure begins will not be
   // transmitted. The channel becomes closed after pending data is transmitted.
   virtual RTCError CloseChannel(int channel_id) = 0;

api/transport/sctp_transport_factory_interface.h

@@ -32,7 +32,7 @@ class SctpTransportFactoryInterface {
  public:
   virtual ~SctpTransportFactoryInterface() = default;
 
-  // Create an SCTP transport using |channel| for the underlying transport.
+  // Create an SCTP transport using `channel` for the underlying transport.
   virtual std::unique_ptr<cricket::SctpTransportInternal> CreateSctpTransport(
       rtc::PacketTransportInternal* channel) = 0;
 };

api/transport/stun.h

@@ -254,11 +254,11 @@ class StunMessage {
   // This is used for testing.
   void SetStunMagicCookie(uint32_t val);
 
-  // Contruct a copy of |this|.
+  // Contruct a copy of `this`.
   std::unique_ptr<StunMessage> Clone() const;
 
-  // Check if the attributes of this StunMessage equals those of |other|
+  // Check if the attributes of this StunMessage equals those of `other`
-  // for all attributes that |attribute_type_mask| return true
+  // for all attributes that `attribute_type_mask` return true
   bool EqualAttributes(const StunMessage* other,
                        std::function<bool(int type)> attribute_type_mask) const;
 
@@ -570,11 +570,11 @@ class StunUInt16ListAttribute : public StunAttribute {
 std::string StunMethodToString(int msg_type);
 
 // Returns the (successful) response type for the given request type.
-// Returns -1 if |request_type| is not a valid request type.
+// Returns -1 if `request_type` is not a valid request type.
 int GetStunSuccessResponseType(int request_type);
 
 // Returns the error response type for the given request type.
-// Returns -1 if |request_type| is not a valid request type.
+// Returns -1 if `request_type` is not a valid request type.
 int GetStunErrorResponseType(int request_type);
 
 // Returns whether a given message is a request type.
@@ -595,13 +595,13 @@ bool ComputeStunCredentialHash(const std::string& username,
                                const std::string& password,
                                std::string* hash);
 
-// Make a copy af |attribute| and return a new StunAttribute.
+// Make a copy af `attribute` and return a new StunAttribute.
 //   This is useful if you don't care about what kind of attribute you
 //   are handling.
 //
 // The implementation copies by calling Write() followed by Read().
 //
-// If |tmp_buffer| is supplied this buffer will be used, otherwise
+// If `tmp_buffer` is supplied this buffer will be used, otherwise
 // a buffer will created in the method.
 std::unique_ptr<StunAttribute> CopyStunAttribute(
     const StunAttribute& attribute,

api/turn_customizer.h

@@ -29,7 +29,7 @@ class TurnCustomizer {
       cricket::StunMessage* message) = 0;
 
   // TURN can send data using channel data messages or Send indication.
-  // This method should return false if |data| should be sent using
+  // This method should return false if `data` should be sent using
   // a Send indication instead of a ChannelData message, even if a
   // channel is bound.
   virtual bool AllowChannelData(cricket::PortInterface* port,

api/video/color_space.cc

@@ -12,7 +12,7 @@
 
 namespace webrtc {
 namespace {
-// Try to convert |enum_value| into the enum class T. |enum_bitmask| is created
+// Try to convert `enum_value` into the enum class T. `enum_bitmask` is created
 // by the funciton below. Returns true if conversion was successful, false
 // otherwise.
 template <typename T>
@@ -43,7 +43,7 @@ constexpr int MakeMask(const int index, const int length, T (&values)[N]) {
 }
 
 // Create a bitmask where each bit corresponds to one potential enum value.
-// |values| should be an array listing all possible enum values. The bit is set
+// `values` should be an array listing all possible enum values. The bit is set
 // to one if the corresponding enum exists. Only works for enums with values
 // less than 64.
 template <typename T, size_t N>

api/video/encoded_frame.h

@@ -46,7 +46,7 @@ class EncodedFrame : public webrtc::VCMEncodedFrame {
   int64_t Id() const { return id_; }
 
   // TODO(philipel): Add simple modify/access functions to prevent adding too
-  // many |references|.
+  // many `references`.
   size_t num_references = 0;
   int64_t references[kMaxFrameReferences];
   // Is this subframe the last one in the superframe (In RTP stream that would

api/video/encoded_image.h

@@ -98,7 +98,7 @@ class RTC_EXPORT EncodedImage {
   }
 
   // These methods can be used to set/get size of subframe with spatial index
-  // |spatial_index| on encoded frames that consist of multiple spatial layers.
+  // `spatial_index` on encoded frames that consist of multiple spatial layers.
   absl::optional<size_t> SpatialLayerFrameSize(int spatial_index) const;
   void SetSpatialLayerFrameSize(int spatial_index, size_t size_bytes);
 
@@ -195,7 +195,7 @@ class RTC_EXPORT EncodedImage {
   // carries the webrtc::VideoFrame id field from the sender to the receiver.
   absl::optional<uint16_t> video_frame_tracking_id_;
   // Information about packets used to assemble this video 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
   RtpPacketInfos packet_infos_;

api/video/i010_buffer.h

@@ -34,7 +34,7 @@ class I010Buffer : public I010BufferInterface {
   // Convert and put I420 buffer into a new buffer.
   static rtc::scoped_refptr<I010Buffer> Copy(const I420BufferInterface& buffer);
 
-  // Return a rotated copy of |src|.
+  // Return a rotated copy of `src`.
   static rtc::scoped_refptr<I010Buffer> Rotate(const I010BufferInterface& src,
                                                VideoRotation rotation);
 
@@ -55,15 +55,15 @@ class I010Buffer : public I010BufferInterface {
   uint16_t* MutableDataU();
   uint16_t* MutableDataV();
 
-  // Scale the cropped area of |src| to the size of |this| buffer, and
+  // Scale the cropped area of `src` to the size of `this` buffer, and
-  // write the result into |this|.
+  // write the result into `this`.
   void CropAndScaleFrom(const I010BufferInterface& src,
                         int offset_x,
                         int offset_y,
                         int crop_width,
                         int crop_height);
 
-  // Scale all of |src| to the size of |this| buffer, with no cropping.
+  // Scale all of `src` to the size of `this` buffer, with no cropping.
   void ScaleFrom(const I010BufferInterface& src);
 
   // Pastes whole picture to canvas at (offset_row, offset_col).

api/video/i420_buffer.h

@@ -49,7 +49,7 @@ class RTC_EXPORT I420Buffer : public I420BufferInterface {
                                              const uint8_t* data_v,
                                              int stride_v);
 
-  // Returns a rotated copy of |src|.
+  // Returns a rotated copy of `src`.
   static rtc::scoped_refptr<I420Buffer> Rotate(const I420BufferInterface& src,
                                                VideoRotation rotation);
   // Deprecated.
@@ -83,8 +83,8 @@ class RTC_EXPORT I420Buffer : public I420BufferInterface {
   uint8_t* MutableDataU();
   uint8_t* MutableDataV();
 
-  // Scale the cropped area of |src| to the size of |this| buffer, and
+  // Scale the cropped area of `src` to the size of `this` buffer, and
-  // write the result into |this|.
+  // write the result into `this`.
   void CropAndScaleFrom(const I420BufferInterface& src,
                         int offset_x,
                         int offset_y,
@@ -95,7 +95,7 @@ class RTC_EXPORT I420Buffer : public I420BufferInterface {
   // aspect ratio without distorting the image.
   void CropAndScaleFrom(const I420BufferInterface& src);
 
-  // Scale all of |src| to the size of |this| buffer, with no cropping.
+  // Scale all of `src` to the size of `this` buffer, with no cropping.
   void ScaleFrom(const I420BufferInterface& src);
 
   // Pastes whole picture to canvas at (offset_row, offset_col).

api/video/nv12_buffer.h

@@ -56,8 +56,8 @@ class RTC_EXPORT NV12Buffer : public NV12BufferInterface {
   // are resolved in a better way. Or in the mean time, use SetBlack.
   void InitializeData();
 
-  // Scale the cropped area of |src| to the size of |this| buffer, and
+  // Scale the cropped area of `src` to the size of `this` buffer, and
-  // write the result into |this|.
+  // write the result into `this`.
   void CropAndScaleFrom(const NV12BufferInterface& src,
                         int offset_x,
                         int offset_y,

api/video/video_bitrate_allocation.h

@@ -50,8 +50,8 @@ class RTC_EXPORT VideoBitrateAllocation {
   // Get the sum of all the temporal layer for a specific spatial layer.
   uint32_t GetSpatialLayerSum(size_t spatial_index) const;
 
-  // Sum of bitrates of temporal layers, from layer 0 to |temporal_index|
+  // Sum of bitrates of temporal layers, from layer 0 to `temporal_index`
-  // inclusive, of specified spatial layer |spatial_index|. Bitrates of lower
+  // inclusive, of specified spatial layer `spatial_index`. Bitrates of lower
   // spatial layers are not included.
   uint32_t GetTemporalLayerSum(size_t spatial_index,
                                size_t temporal_index) const;

api/video/video_frame.h

@@ -272,7 +272,7 @@ class RTC_EXPORT VideoFrame {
   // update_rect() will return a rectangle corresponding to the entire frame.
   absl::optional<UpdateRect> update_rect_;
   // Information about packets used to assemble this video 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
   RtpPacketInfos packet_infos_;

api/video/video_frame_buffer.h

@@ -84,8 +84,8 @@ class RTC_EXPORT VideoFrameBuffer : public rtc::RefCountInterface {
   // A format specific scale function. Default implementation works by
   // converting to I420. But more efficient implementations may override it,
   // especially for kNative.
-  // First, the image is cropped to |crop_width| and |crop_height| and then
+  // First, the image is cropped to `crop_width` and `crop_height` and then
-  // scaled to |scaled_width| and |scaled_height|.
+  // scaled to `scaled_width` and `scaled_height`.
   virtual rtc::scoped_refptr<VideoFrameBuffer> CropAndScale(int offset_x,
                                                             int offset_y,
                                                             int crop_width,

api/video/video_source_interface.h

@@ -54,7 +54,7 @@ struct RTC_EXPORT VideoSinkWants {
   int max_framerate_fps = std::numeric_limits<int>::max();
 
   // Tells the source that the sink wants width and height of the video frames
-  // to be divisible by |resolution_alignment|.
+  // to be divisible by `resolution_alignment`.
   // For example: With I420, this value would be a multiple of 2.
   // Note that this field is unrelated to any horizontal or vertical stride
   // requirements the encoder has on the incoming video frame buffers.
@@ -71,13 +71,13 @@ struct RTC_EXPORT VideoSinkWants {
   // to scaleResolutionDownBy or turning off simulcast or SVC layers.
   //
   // For example, we may capture at 720p and due to adaptation (e.g. applying
-  // |max_pixel_count| constraints) create webrtc::VideoFrames of size 480p, but
+  // `max_pixel_count` constraints) create webrtc::VideoFrames of size 480p, but
   // if we do scaleResolutionDownBy:2 then the only resolution we end up
   // encoding is 240p. In this case we still need to provide webrtc::VideoFrames
   // of size 480p but we can optimize internal buffers for 240p, avoiding
   // downsampling to 480p if possible.
   //
-  // Note that the |resolutions| can change while frames are in flight and
+  // Note that the `resolutions` can change while frames are in flight and
   // should only be used as a hint when constructing the webrtc::VideoFrame.
   std::vector<FrameSize> resolutions;
 };

api/video/video_stream_decoder_create.h

@@ -20,7 +20,7 @@
 #include "api/video_codecs/sdp_video_format.h"
 
 namespace webrtc {
-// The |decoder_settings| parameter is a map between:
+// The `decoder_settings` parameter is a map between:
 // <payload type> -->  <<video format>, <number of cores>>.
 // The video format is used when instantiating a decoder, and
 // the number of cores is used when initializing the decoder.

api/video/video_stream_encoder_interface.h

@@ -68,9 +68,9 @@ class VideoStreamEncoderInterface : public rtc::VideoSinkInterface<VideoFrame> {
   GetAdaptationResources() = 0;
 
   // Sets the source that will provide video frames to the VideoStreamEncoder's
-  // OnFrame method. |degradation_preference| control whether or not resolution
+  // OnFrame method. `degradation_preference` control whether or not resolution
   // or frame rate may be reduced. The VideoStreamEncoder registers itself with
-  // |source|, and signals adaptation decisions to the source in the form of
+  // `source`, and signals adaptation decisions to the source in the form of
   // VideoSinkWants.
   // TODO(nisse): When adaptation logic is extracted from this class,
   // it no longer needs to know the source.
@@ -78,8 +78,8 @@ class VideoStreamEncoderInterface : public rtc::VideoSinkInterface<VideoFrame> {
       rtc::VideoSourceInterface<VideoFrame>* source,
       const DegradationPreference& degradation_preference) = 0;
 
-  // Sets the |sink| that gets the encoded frames. |rotation_applied| means
+  // Sets the `sink` that gets the encoded frames. `rotation_applied` means
-  // that the source must support rotation. Only set |rotation_applied| if the
+  // that the source must support rotation. Only set `rotation_applied` if the
   // remote side does not support the rotation extension.
   virtual void SetSink(EncoderSink* sink, bool rotation_applied) = 0;
 
@@ -102,13 +102,13 @@ class VideoStreamEncoderInterface : public rtc::VideoSinkInterface<VideoFrame> {
   virtual void OnLossNotification(
       const VideoEncoder::LossNotification& loss_notification) = 0;
 
-  // Set the currently estimated network properties. A |target_bitrate|
+  // Set the currently estimated network properties. A `target_bitrate`
   // of zero pauses the encoder.
-  // |stable_target_bitrate| is a filtered version of |target_bitrate|. It  is
+  // `stable_target_bitrate` is a filtered version of `target_bitrate`. It  is
   // always less or equal to it. It can be used to avoid rapid changes of
   // expensive encoding settings, such as resolution.
-  // |link_allocation| is the bandwidth available for this video stream on the
+  // `link_allocation` is the bandwidth available for this video stream on the
-  // network link. It is always at least |target_bitrate| but may be higher
+  // network link. It is always at least `target_bitrate` but may be higher
   // if we are not network constrained.
   virtual void OnBitrateUpdated(DataRate target_bitrate,
                                 DataRate stable_target_bitrate,
@@ -122,8 +122,8 @@ class VideoStreamEncoderInterface : public rtc::VideoSinkInterface<VideoFrame> {
   virtual void SetFecControllerOverride(
       FecControllerOverride* fec_controller_override) = 0;
 
-  // Creates and configures an encoder with the given |config|. The
+  // Creates and configures an encoder with the given `config`. The
-  // |max_data_payload_length| is used to support single NAL unit
+  // `max_data_payload_length` is used to support single NAL unit
   // packetization for H.264.
   virtual void ConfigureEncoder(VideoEncoderConfig config,
                                 size_t max_data_payload_length) = 0;

api/video/video_stream_encoder_observer.h

@@ -101,7 +101,7 @@ class VideoStreamEncoderObserver : public CpuOveruseMetricsObserver {
       const VideoBitrateAllocation& allocation) {}
 
   // Informes observer if an internal encoder scaler has reduced video
-  // resolution or not. |is_scaled| is a flag indicating if the video is scaled
+  // resolution or not. `is_scaled` is a flag indicating if the video is scaled
   // down.
   virtual void OnEncoderInternalScalerUpdate(bool is_scaled) {}
 

api/video/video_timing.h

@@ -55,7 +55,7 @@ struct TimingFrameInfo {
   // synchronized, -1 otherwise.
   int64_t EndToEndDelay() const;
 
-  // Returns true if current frame took longer to process than |other| frame.
+  // Returns true if current frame took longer to process than `other` frame.
   // If other frame's clocks are not synchronized, current frame is always
   // preferred.
   bool IsLongerThan(const TimingFrameInfo& other) const;

api/video_codecs/test/video_decoder_software_fallback_wrapper_unittest.cc

@@ -75,7 +75,7 @@ class VideoDecoderSoftwareFallbackWrapperTest : public ::testing::Test {
     int reset_count_ = 0;
   };
   test::ScopedFieldTrials override_field_trials_;
-  // |fake_decoder_| is owned and released by |fallback_wrapper_|.
+  // `fake_decoder_` is owned and released by `fallback_wrapper_`.
   CountingFakeDecoder* fake_decoder_;
   std::unique_ptr<VideoDecoder> fallback_wrapper_;
 };

api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc

@@ -172,7 +172,7 @@ class VideoEncoderSoftwareFallbackWrapperTestBase : public ::testing::Test {
 
   test::ScopedFieldTrials override_field_trials_;
   FakeEncodedImageCallback callback_;
-  // |fake_encoder_| is owned and released by |fallback_wrapper_|.
+  // `fake_encoder_` is owned and released by `fallback_wrapper_`.
   CountingFakeEncoder* fake_encoder_;
   CountingFakeEncoder* fake_sw_encoder_;
   bool wrapper_initialized_;

api/video_codecs/video_decoder_factory.h

@@ -40,7 +40,7 @@ class RTC_EXPORT VideoDecoderFactory {
   // power efficient, which is currently interpreted as if there is support for
   // hardware acceleration.
   // See https://w3c.github.io/webrtc-svc/#scalabilitymodes* for a specification
-  // of valid values for |scalability_mode|.
+  // of valid values for `scalability_mode`.
   // NOTE: QueryCodecSupport is currently an experimental feature that is
   // subject to change without notice.
   virtual CodecSupport QueryCodecSupport(

api/video_codecs/video_encoder.h

@@ -167,7 +167,7 @@ class RTC_EXPORT VideoEncoder {
     ScalingSettings scaling_settings;
 
     // The width and height of the incoming video frames should be divisible
-    // by |requested_resolution_alignment|. If they are not, the encoder may
+    // by `requested_resolution_alignment`. If they are not, the encoder may
     // drop the incoming frame.
     // For example: With I420, this value would be a multiple of 2.
     // Note that this field is unrelated to any horizontal or vertical stride
@@ -175,12 +175,12 @@ class RTC_EXPORT VideoEncoder {
     int requested_resolution_alignment;
 
     // Same as above but if true, each simulcast layer should also be divisible
-    // by |requested_resolution_alignment|.
+    // by `requested_resolution_alignment`.
-    // Note that scale factors |scale_resolution_down_by| may be adjusted so a
+    // Note that scale factors `scale_resolution_down_by` may be adjusted so a
     // common multiple is not too large to avoid largely cropped frames and
     // possibly with an aspect ratio far from the original.
     // Warning: large values of scale_resolution_down_by could be changed
-    // considerably, especially if |requested_resolution_alignment| is large.
+    // considerably, especially if `requested_resolution_alignment` is large.
     bool apply_alignment_to_all_simulcast_layers;
 
     // If true, encoder supports working with a native handle (e.g. texture
@@ -215,7 +215,7 @@ class RTC_EXPORT VideoEncoder {
     bool has_internal_source;
 
     // For each spatial layer (simulcast stream or SVC layer), represented as an
-    // element in |fps_allocation| a vector indicates how many temporal layers
+    // element in `fps_allocation` a vector indicates how many temporal layers
     // the encoder is using for that spatial layer.
     // For each spatial/temporal layer pair, the frame rate fraction is given as
     // an 8bit unsigned integer where 0 = 0% and 255 = 100%.
@@ -243,8 +243,8 @@ class RTC_EXPORT VideoEncoder {
     // Recommended bitrate limits for different resolutions.
     std::vector<ResolutionBitrateLimits> resolution_bitrate_limits;
 
-    // Obtains the limits from |resolution_bitrate_limits| that best matches the
+    // Obtains the limits from `resolution_bitrate_limits` that best matches the
-    // |frame_size_pixels|.
+    // `frame_size_pixels`.
     absl::optional<ResolutionBitrateLimits>
     GetEncoderBitrateLimitsForResolution(int frame_size_pixels) const;
 
@@ -279,7 +279,7 @@ class RTC_EXPORT VideoEncoder {
     VideoBitrateAllocation bitrate;
     // Target framerate, in fps. A value <= 0.0 is invalid and should be
     // interpreted as framerate target not available. In this case the encoder
-    // should fall back to the max framerate specified in |codec_settings| of
+    // should fall back to the max framerate specified in `codec_settings` of
     // the last InitEncode() call.
     double framerate_fps;
     // The network bandwidth available for video. This is at least
@@ -299,15 +299,15 @@ class RTC_EXPORT VideoEncoder {
     uint32_t timestamp_of_last_received;
     // Describes whether the dependencies of the last received frame were
     // all decodable.
-    // |false| if some dependencies were undecodable, |true| if all dependencies
+    // `false` if some dependencies were undecodable, `true` if all dependencies
-    // were decodable, and |nullopt| if the dependencies are unknown.
+    // were decodable, and `nullopt` if the dependencies are unknown.
     absl::optional<bool> dependencies_of_last_received_decodable;
     // Describes whether the received frame was decodable.
-    // |false| if some dependency was undecodable or if some packet belonging
+    // `false` if some dependency was undecodable or if some packet belonging
     // to the last received frame was missed.
-    // |true| if all dependencies were decodable and all packets belonging
+    // `true` if all dependencies were decodable and all packets belonging
     // to the last received frame were received.
-    // |nullopt| if no packet belonging to the last frame was missed, but the
+    // `nullopt` if no packet belonging to the last frame was missed, but the
     // last packet in the frame was not yet received.
     absl::optional<bool> last_received_decodable;
   };

api/video_codecs/video_encoder_config.h

@@ -24,7 +24,7 @@
 
 namespace webrtc {
 
-// The |VideoStream| struct describes a simulcast layer, or "stream".
+// The `VideoStream` struct describes a simulcast layer, or "stream".
 struct VideoStream {
   VideoStream();
   ~VideoStream();
@@ -46,7 +46,7 @@ struct VideoStream {
   int max_bitrate_bps;
 
   // Scaling factor applied to the stream size.
-  // |width| and |height| values are already scaled down.
+  // `width` and `height` values are already scaled down.
   double scale_resolution_down_by;
 
   // Maximum Quantization Parameter to use when encoding the stream.
@@ -171,7 +171,7 @@ class VideoEncoderConfig {
   // The simulcast layer's configurations set by the application for this video
   // sender. These are modified by the video_stream_factory before being passed
   // down to lower layers for the video encoding.
-  // |simulcast_layers| is also used for configuring non-simulcast (when there
+  // `simulcast_layers` is also used for configuring non-simulcast (when there
   // is a single VideoStream).
   std::vector<VideoStream> simulcast_layers;
 

api/video_codecs/video_encoder_factory.h

@@ -29,7 +29,7 @@ class VideoEncoderFactory {
  public:
   // TODO(magjed): Try to get rid of this struct.
   struct CodecInfo {
-    // |has_internal_source| is true if encoders created by this factory of the
+    // `has_internal_source` is true if encoders created by this factory of the
     // given codec will use internal camera sources, meaning that they don't
     // require/expect frames to be delivered via webrtc::VideoEncoder::Encode.
     // This flag is used as the internal_source parameter to
@@ -88,7 +88,7 @@ class VideoEncoderFactory {
   // power efficient, which is currently interpreted as if there is support for
   // hardware acceleration.
   // See https://w3c.github.io/webrtc-svc/#scalabilitymodes* for a specification
-  // of valid values for |scalability_mode|.
+  // of valid values for `scalability_mode`.
   // NOTE: QueryCodecSupport is currently an experimental feature that is
   // subject to change without notice.
   virtual CodecSupport QueryCodecSupport(

api/video_codecs/video_encoder_software_fallback_wrapper.cc

@@ -39,8 +39,8 @@ namespace {
 // If forced fallback is allowed, either:
 //
 // 1) The forced fallback is requested if the resolution is less than or equal
-//    to |max_pixels_|. The resolution is allowed to be scaled down to
+//    to `max_pixels_`. The resolution is allowed to be scaled down to
-//    |min_pixels_|.
+//    `min_pixels_`.
 //
 // 2) The forced fallback is requested if temporal support is preferred and the
 //    SW fallback supports temporal layers while the HW encoder does not.
@@ -274,8 +274,8 @@ bool VideoEncoderSoftwareFallbackWrapper::InitFallbackEncoder(bool is_forced) {
 void VideoEncoderSoftwareFallbackWrapper::SetFecControllerOverride(
     FecControllerOverride* fec_controller_override) {
   // It is important that only one of those would ever interact with the
-  // |fec_controller_override| at a given time. This is the responsibility
+  // `fec_controller_override` at a given time. This is the responsibility
-  // of |this| to maintain.
+  // of `this` to maintain.
 
   fec_controller_override_ = fec_controller_override;
   current_encoder()->SetFecControllerOverride(fec_controller_override);

api/video_codecs/video_encoder_software_fallback_wrapper.h

@@ -32,7 +32,7 @@ CreateVideoEncoderSoftwareFallbackWrapper(
     bool prefer_temporal_support);
 
 // Default fallback for call-sites not yet updated with
-// |prefer_temporal_support|.
+// `prefer_temporal_support`.
 // TODO(sprang): Remove when usage is gone.
 RTC_EXPORT inline std::unique_ptr<VideoEncoder>
 CreateVideoEncoderSoftwareFallbackWrapper(

api/video_codecs/vp8_frame_buffer_controller.h

@@ -66,7 +66,7 @@ struct Vp8EncoderConfig {
     // Number of active temporal layers. Set to 0 if not used.
     uint32_t ts_number_layers;
 
-    // Arrays of length |ts_number_layers|, indicating (cumulative) target
+    // Arrays of length `ts_number_layers`, indicating (cumulative) target
     // bitrate and rate decimator (e.g. 4 if every 4th frame is in the given
     // layer) for each active temporal layer, starting with temporal id 0.
     std::array<uint32_t, kMaxLayers> ts_target_bitrate;
@@ -75,7 +75,7 @@ struct Vp8EncoderConfig {
     // The periodicity of the temporal pattern. Set to 0 if not used.
     uint32_t ts_periodicity;
 
-    // Array of length |ts_periodicity| indicating the sequence of temporal id's
+    // Array of length `ts_periodicity` indicating the sequence of temporal id's
     // to assign to incoming frames.
     std::array<uint32_t, kMaxPeriodicity> ts_layer_id;
   };
@@ -106,7 +106,7 @@ class Vp8FrameBufferController {
   // The limits are suggestion-only; the controller is allowed to exceed them.
   virtual void SetQpLimits(size_t stream_index, int min_qp, int max_qp) = 0;
 
-  // Number of streamed controlled by |this|.
+  // Number of streamed controlled by `this`.
   virtual size_t StreamCount() const = 0;
 
   // If this method returns true, the encoder is free to drop frames for
@@ -121,7 +121,7 @@ class Vp8FrameBufferController {
   virtual bool SupportsEncoderFrameDropping(size_t stream_index) const = 0;
 
   // New target bitrate for a stream (each entry in
-  // |bitrates_bps| is for another temporal layer).
+  // `bitrates_bps` is for another temporal layer).
   virtual void OnRatesUpdated(size_t stream_index,
                               const std::vector<uint32_t>& bitrates_bps,
                               int framerate_fps) = 0;
@@ -130,7 +130,7 @@ class Vp8FrameBufferController {
   // the controller wishes to enact in the encoder's configuration.
   // If a value is not overridden, previous overrides are still in effect.
   // However, if |Vp8EncoderConfig::reset_previous_configuration_overrides|
-  // is set to |true|, all previous overrides are reset.
+  // is set to `true`, all previous overrides are reset.
   virtual Vp8EncoderConfig UpdateConfiguration(size_t stream_index) = 0;
 
   // Returns the recommended VP8 encode flags needed.
@@ -142,13 +142,13 @@ class Vp8FrameBufferController {
   virtual Vp8FrameConfig NextFrameConfig(size_t stream_index,
                                          uint32_t rtp_timestamp) = 0;
 
-  // Called after the encode step is done. |rtp_timestamp| must match the
+  // Called after the encode step is done. `rtp_timestamp` must match the
   // parameter use in the NextFrameConfig() call.
-  // |is_keyframe| must be true iff the encoder decided to encode this frame as
+  // `is_keyframe` must be true iff the encoder decided to encode this frame as
   // a keyframe.
-  // If |info| is not null, the encoder may update |info| with codec specific
+  // If `info` is not null, the encoder may update `info` with codec specific
-  // data such as temporal id. |qp| should indicate the frame-level QP this
+  // data such as temporal id. `qp` should indicate the frame-level QP this
-  // frame was encoded at. If the encoder does not support extracting this, |qp|
+  // frame was encoded at. If the encoder does not support extracting this, `qp`
   // should be set to 0.
   virtual void OnEncodeDone(size_t stream_index,
                             uint32_t rtp_timestamp,
@@ -161,7 +161,7 @@ class Vp8FrameBufferController {
   virtual void OnFrameDropped(size_t stream_index, uint32_t rtp_timestamp) = 0;
 
   // Called by the encoder when the packet loss rate changes.
-  // |packet_loss_rate| runs between 0.0 (no loss) and 1.0 (everything lost).
+  // `packet_loss_rate` runs between 0.0 (no loss) and 1.0 (everything lost).
   virtual void OnPacketLossRateUpdate(float packet_loss_rate) = 0;
 
   // Called by the encoder when the round trip time changes.

api/video_track_source_proxy_factory.h

@@ -15,9 +15,9 @@
 
 namespace webrtc {
 
-// Creates a proxy source for |source| which makes sure the real
+// Creates a proxy source for `source` which makes sure the real
 // VideoTrackSourceInterface implementation is destroyed on the signaling thread
-// and marshals calls to |worker_thread| and |signaling_thread|.
+// and marshals calls to `worker_thread` and `signaling_thread`.
 rtc::scoped_refptr<VideoTrackSourceInterface> RTC_EXPORT
 CreateVideoTrackSourceProxy(rtc::Thread* signaling_thread,
                             rtc::Thread* worker_thread,

api/voip/voip_base.h

@@ -56,53 +56,53 @@ class VoipBase {
   // Creates a channel.
   // Each channel handle maps into one audio media session where each has
   // its own separate module for send/receive rtp packet with one peer.
-  // Caller must set |transport|, webrtc::Transport callback pointer to
+  // Caller must set `transport`, webrtc::Transport callback pointer to
   // receive rtp/rtcp packets from corresponding media session in VoIP engine.
   // VoipEngine framework expects applications to handle network I/O directly
   // and injection for incoming RTP from remote endpoint is handled via
-  // VoipNetwork interface. |local_ssrc| is optional and when local_ssrc is not
+  // VoipNetwork interface. `local_ssrc` is optional and when local_ssrc is not
   // set, some random value will be used by voip engine.
   // Returns a ChannelId created for caller to handle subsequent Channel
   // operations.
   virtual ChannelId CreateChannel(Transport* transport,
                                   absl::optional<uint32_t> local_ssrc) = 0;
 
-  // Releases |channel_id| that no longer has any use.
+  // Releases `channel_id` that no longer has any use.
   // Returns following VoipResult;
-  //  kOk - |channel_id| is released.
+  //  kOk - `channel_id` is released.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   //  kInternal - Fails to stop audio output device.
   virtual VoipResult ReleaseChannel(ChannelId channel_id) = 0;
 
-  // Starts sending on |channel_id|. This starts microphone if not started yet.
+  // Starts sending on `channel_id`. This starts microphone if not started yet.
   // Returns following VoipResult;
   //  kOk - Channel successfully started to send.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   //  kFailedPrecondition - Missing prerequisite on VoipCodec::SetSendCodec.
   //  kInternal - initialization has failed on selected microphone.
   virtual VoipResult StartSend(ChannelId channel_id) = 0;
 
-  // Stops sending on |channel_id|. If this is the last active channel, it will
+  // Stops sending on `channel_id`. If this is the last active channel, it will
   // stop microphone input from underlying audio platform layer.
   // Returns following VoipResult;
   //  kOk - Channel successfully stopped to send.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   //  kInternal - Failed to stop the active microphone device.
   virtual VoipResult StopSend(ChannelId channel_id) = 0;
 
-  // Starts playing on speaker device for |channel_id|.
+  // Starts playing on speaker device for `channel_id`.
   // This will start underlying platform speaker device if not started.
   // Returns following VoipResult;
   //  kOk - Channel successfully started to play out.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   //  kFailedPrecondition - Missing prerequisite on VoipCodec::SetReceiveCodecs.
   //  kInternal - Failed to initializate the selected speaker device.
   virtual VoipResult StartPlayout(ChannelId channel_id) = 0;
 
-  // Stops playing on speaker device for |channel_id|.
+  // Stops playing on speaker device for `channel_id`.
   // Returns following VoipResult;
   //  kOk - Channel successfully stopped t play out.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   virtual VoipResult StopPlayout(ChannelId channel_id) = 0;
 
  protected:

api/voip/voip_codec.h

@@ -31,7 +31,7 @@ class VoipCodec {
   // Set encoder type here along with its payload type to use.
   // Returns following VoipResult;
   //  kOk - sending codec is set as provided.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   virtual VoipResult SetSendCodec(ChannelId channel_id,
                                   int payload_type,
                                   const SdpAudioFormat& encoder_spec) = 0;
@@ -42,7 +42,7 @@ class VoipCodec {
   // direction.
   // Returns following VoipResult;
   //  kOk - receiving codecs are set as provided.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   virtual VoipResult SetReceiveCodecs(
       ChannelId channel_id,
       const std::map<int, SdpAudioFormat>& decoder_specs) = 0;

api/voip/voip_dtmf.h

@@ -45,20 +45,20 @@ class VoipDtmf {
   // type has been negotiated with remote.
   // Returns following VoipResult;
   //  kOk - telephone event type is registered as provided.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   virtual VoipResult RegisterTelephoneEventType(ChannelId channel_id,
                                                 int rtp_payload_type,
                                                 int sample_rate_hz) = 0;
 
   // Send DTMF named event as specified by
   // https://tools.ietf.org/html/rfc4733#section-3.2
-  // |duration_ms| specifies the duration of DTMF packets that will be emitted
+  // `duration_ms` specifies the duration of DTMF packets that will be emitted
   // in place of real RTP packets instead.
   // Must be called after RegisterTelephoneEventType and VoipBase::StartSend
   // have been called.
   // Returns following VoipResult;
   //  kOk - requested DTMF event is successfully scheduled.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   //  kFailedPrecondition - Missing prerequisite on RegisterTelephoneEventType
   //   or sending state.
   virtual VoipResult SendDtmfEvent(ChannelId channel_id,

api/voip/voip_network.h

@@ -24,7 +24,7 @@ class VoipNetwork {
   // The data received from the network including RTP header is passed here.
   // Returns following VoipResult;
   //  kOk - received RTP packet is processed.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   virtual VoipResult ReceivedRTPPacket(
       ChannelId channel_id,
       rtc::ArrayView<const uint8_t> rtp_packet) = 0;
@@ -32,7 +32,7 @@ class VoipNetwork {
   // The data received from the network including RTCP header is passed here.
   // Returns following VoipResult;
   //  kOk - received RTCP packet is processed.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   virtual VoipResult ReceivedRTCPPacket(
       ChannelId channel_id,
       rtc::ArrayView<const uint8_t> rtcp_packet) = 0;

api/voip/voip_statistics.h

@@ -75,17 +75,17 @@ struct ChannelStatistics {
 // the jitter buffer (NetEq) performance.
 class VoipStatistics {
  public:
-  // Gets the audio ingress statistics by |ingress_stats| reference.
+  // Gets the audio ingress statistics by `ingress_stats` reference.
   // Returns following VoipResult;
   //  kOk - successfully set provided IngressStatistics reference.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   virtual VoipResult GetIngressStatistics(ChannelId channel_id,
                                           IngressStatistics& ingress_stats) = 0;
 
-  // Gets the channel statistics by |channel_stats| reference.
+  // Gets the channel statistics by `channel_stats` reference.
   // Returns following VoipResult;
   //  kOk - successfully set provided ChannelStatistics reference.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   virtual VoipResult GetChannelStatistics(ChannelId channel_id,
                                           ChannelStatistics& channel_stats) = 0;
 

api/voip/voip_volume_control.h

@@ -37,21 +37,21 @@ class VoipVolumeControl {
   // mute doesn't affect audio input level and energy values as input sample is
   // silenced after the measurement.
   // Returns following VoipResult;
-  //  kOk - input source muted or unmuted as provided by |enable|.
+  //  kOk - input source muted or unmuted as provided by `enable`.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   virtual VoipResult SetInputMuted(ChannelId channel_id, bool enable) = 0;
 
-  // Gets the microphone volume info via |volume_info| reference.
+  // Gets the microphone volume info via `volume_info` reference.
   // Returns following VoipResult;
   //  kOk - successfully set provided input volume info.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   virtual VoipResult GetInputVolumeInfo(ChannelId channel_id,
                                         VolumeInfo& volume_info) = 0;
 
-  // Gets the speaker volume info via |volume_info| reference.
+  // Gets the speaker volume info via `volume_info` reference.
   // Returns following VoipResult;
   //  kOk - successfully set provided output volume info.
-  //  kInvalidArgument - |channel_id| is invalid.
+  //  kInvalidArgument - `channel_id` is invalid.
   virtual VoipResult GetOutputVolumeInfo(ChannelId channel_id,
                                          VolumeInfo& volume_info) = 0;