Use backticks not vertical bars to denote variables in comments

Bug: webrtc:12338
Change-Id: I89c8b3a328d04203177522cbdfd9e606fd4bce4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228246
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34696}

api/ice_transport_factory.h

@@ -35,9 +35,9 @@ RTC_EXPORT rtc::scoped_refptr<IceTransportInterface> CreateIceTransport(
 // without using a webrtc::PeerConnection.
 // The returned object must be accessed and destroyed on the thread that
 // created it.
-// |init.port_allocator()| is required and must outlive the created
+// `init.port_allocator()` is required and must outlive the created
 //     IceTransportInterface object.
-// |init.async_resolver_factory()| and |init.event_log()| are optional, but if
+// `init.async_resolver_factory()` and `init.event_log()` are optional, but if
 //     provided must outlive the created IceTransportInterface object.
 RTC_EXPORT rtc::scoped_refptr<IceTransportInterface> CreateIceTransport(
     IceTransportInit);

api/jsep.h

@@ -166,8 +166,8 @@ class RTC_EXPORT SessionDescriptionInterface {
   // Ownership is not transferred.
   //
   // Returns false if the session description does not have a media section
-  // that corresponds to |candidate.sdp_mid()| or
-  // |candidate.sdp_mline_index()|.
+  // that corresponds to `candidate.sdp_mid()` or
+  // `candidate.sdp_mline_index()`.
   virtual bool AddCandidate(const IceCandidateInterface* candidate) = 0;
 
   // Removes the candidates from the description, if found.

api/peer_connection_interface.h

@@ -1295,8 +1295,8 @@ class PeerConnectionObserver {
   // This is called when signaling indicates a transceiver will be receiving
   // media from the remote endpoint. This is fired during a call to
   // SetRemoteDescription. The receiving track can be accessed by:
-  // |transceiver->receiver()->track()| and its associated streams by
-  // |transceiver->receiver()->streams()|.
+  // `transceiver->receiver()->track()` and its associated streams by
+  // `transceiver->receiver()->streams()`.
   // Note: This will only be called if Unified Plan semantics are specified.
   // This behavior is specified in section 2.2.8.2.5 of the "Set the
   // RTCSessionDescription" algorithm:

api/rtp_packet_info.h

@@ -113,7 +113,7 @@ class RTC_EXPORT RtpPacketInfo {
   // capture clock offset defined in the Absolute Capture Time header extension.
   absl::optional<int64_t> local_capture_clock_offset_;
 
-  // Local |webrtc::Clock|-based timestamp of when the packet was received.
+  // Local `webrtc::Clock`-based timestamp of when the packet was received.
   Timestamp receive_time_;
 };
 

api/rtp_packet_infos.h

@@ -26,8 +26,8 @@ namespace webrtc {
 // an audio or video frame. Uses internal reference counting to make it very
 // cheap to copy.
 //
-// We should ideally just use |std::vector<RtpPacketInfo>| and have it
-// |std::move()|-ed as the per-packet information is transferred from one object
+// We should ideally just use `std::vector<RtpPacketInfo>` and have it
+// `std::move()`-ed as the per-packet information is transferred from one object
 // to another. But moving the info, instead of copying it, is not easily done
 // for the current video code.
 class RTC_EXPORT RtpPacketInfos {

api/set_local_description_observer_interface.h

@@ -21,7 +21,7 @@ namespace webrtc {
 // the observer to examine the effects of the operation without delay.
 class SetLocalDescriptionObserverInterface : public rtc::RefCountInterface {
  public:
-  // On success, |error.ok()| is true.
+  // On success, `error.ok()` is true.
   virtual void OnSetLocalDescriptionComplete(RTCError error) = 0;
 };
 

api/set_remote_description_observer_interface.h

@@ -22,7 +22,7 @@ namespace webrtc {
 // operation.
 class SetRemoteDescriptionObserverInterface : public rtc::RefCountInterface {
  public:
-  // On success, |error.ok()| is true.
+  // On success, `error.ok()` is true.
   virtual void OnSetRemoteDescriptionComplete(RTCError error) = 0;
 };
 

api/stats/rtc_stats.h

@@ -217,7 +217,7 @@ enum class NonStandardGroupId {
 
 // 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
-// are implemented by |RTCStatsMember<T>|. The value of a member may be
+// are implemented by `RTCStatsMember<T>`. The value of a member may be
 // undefined, the value can only be read if `is_defined`.
 class RTCStatsMemberInterface {
  public:
@@ -286,7 +286,7 @@ class RTCStatsMemberInterface {
 
 // Template implementation of `RTCStatsMemberInterface`.
 // The supported types are the ones described by
-// |RTCStatsMemberInterface::Type|.
+// `RTCStatsMemberInterface::Type`.
 template <typename T>
 class RTCStatsMember : public RTCStatsMemberInterface {
  public:

api/stats/rtc_stats_report.h

@@ -90,7 +90,7 @@ class RTC_EXPORT RTCStatsReport final
   // 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|.
+  // Stats iterators. Stats are ordered lexicographically on `RTCStats::id`.
   ConstIterator begin() const;
   ConstIterator end() const;
 

api/stats/rtcstats_objects.h

@@ -57,7 +57,7 @@ struct RTCDtlsTransportState {
   static const char* const kFailed;
 };
 
-// |RTCMediaStreamTrackStats::kind| is not an enum in the spec but the only
+// `RTCMediaStreamTrackStats::kind` is not an enum in the spec but the only
 // valid values are "audio" and "video".
 // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-kind
 struct RTCMediaStreamTrackKind {

api/stats_types.h

@@ -232,7 +232,7 @@ class RTC_EXPORT StatsReport {
     kStatsValueNameSrtpCipher,
     kStatsValueNameTargetDelayMs,
     kStatsValueNameTargetEncBitrate,
-    kStatsValueNameTimingFrameInfo,  // Result of |TimingFrameInfo::ToString|
+    kStatsValueNameTimingFrameInfo,  // Result of `TimingFrameInfo::ToString`
     kStatsValueNameTrackId,
     kStatsValueNameTransmitBitrate,
     kStatsValueNameTransportType,

api/task_queue/task_queue_base.h

@@ -38,7 +38,7 @@ class RTC_LOCKABLE RTC_EXPORT TaskQueueBase {
   virtual void Delete() = 0;
 
   // Schedules a task to execute. Tasks are executed in FIFO order.
-  // If |task->Run()| returns true, task is deleted on the task queue
+  // If `task->Run()` returns true, task is deleted on the task queue
   // before next QueuedTask starts executing.
   // When a TaskQueue is deleted, pending tasks will not be executed but they
   // will be deleted. The deletion of tasks may happen synchronously on the

api/video_codecs/video_encoder.h

@@ -287,7 +287,7 @@ class RTC_EXPORT VideoEncoder {
     // the last InitEncode() call.
     double framerate_fps;
     // The network bandwidth available for video. This is at least
-    // |bitrate.get_sum_bps()|, but may be higher if the application is not
+    // `bitrate.get_sum_bps()`, but may be higher if the application is not
     // network constrained.
     DataRate bandwidth_allocation;
 

api/video_codecs/video_encoder_config.h

@@ -129,7 +129,7 @@ class VideoEncoderConfig {
     // An implementation should return a std::vector<VideoStream> with the
     // wanted VideoStream settings for the given video resolution.
     // The size of the vector may not be larger than
-    // |encoder_config.number_of_streams|.
+    // `encoder_config.number_of_streams`.
     virtual std::vector<VideoStream> CreateEncoderStreams(
         int width,
         int height,

api/video_codecs/vp8_frame_buffer_controller.h

@@ -129,7 +129,7 @@ class Vp8FrameBufferController {
   // Called by the encoder before encoding a frame. Returns a set of overrides
   // 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|
+  // However, if `Vp8EncoderConfig::reset_previous_configuration_overrides`
   // is set to `true`, all previous overrides are reset.
   virtual Vp8EncoderConfig UpdateConfiguration(size_t stream_index) = 0;
 

audio/channel_receive.cc

@@ -464,7 +464,7 @@ AudioMixer::Source::AudioFrameInfo ChannelReceive::GetAudioFrameWithInfo(
     }
   }
 
-  // Fill in local capture clock offset in |audio_frame->packet_infos_|.
+  // Fill in local capture clock offset in `audio_frame->packet_infos_`.
   RtpPacketInfos::vector_type packet_infos;
   for (auto& packet_info : audio_frame->packet_infos_) {
     absl::optional<int64_t> local_capture_clock_offset;

audio/utility/audio_frame_operations.h

@@ -33,13 +33,13 @@ class AudioFrameOperations {
   // `result_frame` is empty.
   static void Add(const AudioFrame& frame_to_add, AudioFrame* result_frame);
 
-  // |frame.num_channels_| will be updated. This version checks for sufficient
+  // `frame.num_channels_` will be updated. This version checks for sufficient
   // buffer size and that `num_channels_` is mono. Use UpmixChannels
   // instead. TODO(bugs.webrtc.org/8649): remove.
   ABSL_DEPRECATED("bugs.webrtc.org/8649")
   static int MonoToStereo(AudioFrame* frame);
 
-  // |frame.num_channels_| will be updated. This version checks that
+  // `frame.num_channels_` will be updated. This version checks that
   // `num_channels_` is stereo. Use DownmixChannels
   // instead. TODO(bugs.webrtc.org/8649): remove.
   ABSL_DEPRECATED("bugs.webrtc.org/8649")
@@ -52,7 +52,7 @@ class AudioFrameOperations {
                            size_t samples_per_channel,
                            int16_t* dst_audio);
 
-  // |frame.num_channels_| will be updated. This version checks that
+  // `frame.num_channels_` will be updated. This version checks that
   // `num_channels_` is 4 channels.
   static int QuadToStereo(AudioFrame* frame);
 
@@ -66,12 +66,12 @@ class AudioFrameOperations {
                               size_t dst_channels,
                               int16_t* dst_audio);
 
-  // |frame.num_channels_| will be updated. This version checks that
+  // `frame.num_channels_` will be updated. This version checks that
   // `num_channels_` and `dst_channels` are valid and performs relevant downmix.
   // Supported channel combinations are N channels to Mono, and Quad to Stereo.
   static void DownmixChannels(size_t dst_channels, AudioFrame* frame);
 
-  // |frame.num_channels_| will be updated. This version checks that
+  // `frame.num_channels_` will be updated. This version checks that
   // `num_channels_` and `dst_channels` are valid and performs relevant
   // downmix. Supported channel combinations are Mono to N
   // channels. The single channel is replicated.

call/rtp_config.h

@@ -81,7 +81,7 @@ struct RtpConfig {
   // If rids are specified, they should correspond to the `ssrcs` vector.
   // This means that:
   // 1. rids.size() == 0 || rids.size() == ssrcs.size().
-  // 2. If rids is not empty, then |rids[i]| should use |ssrcs[i]|.
+  // 2. If rids is not empty, then `rids[i]` should use `ssrcs[i]`.
   std::vector<std::string> rids;
 
   // The value to send in the MID RTP header extension if the extension is

call/simulated_network.cc

@@ -216,8 +216,8 @@ void SimulatedNetwork::UpdateCapacityQueue(ConfigState state,
     pending_drain_bits_ -= packet.packet.size * 8;
     RTC_DCHECK(pending_drain_bits_ >= 0);
 
-    // Drop packets at an average rate of |state.config.loss_percent| with
-    // and average loss burst length of |state.config.avg_burst_loss_length|.
+    // Drop packets at an average rate of `state.config.loss_percent` with
+    // and average loss burst length of `state.config.avg_burst_loss_length`.
     if ((bursting_ && random_.Rand<double>() < state.prob_loss_bursting) ||
         (!bursting_ && random_.Rand<double>() < state.prob_start_bursting)) {
       bursting_ = true;

common_audio/vad/vad_core.c

@@ -298,8 +298,8 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
         nmk2 = nmk;
         if (!vadflag) {
           // deltaN = (x-mu)/sigma^2
-          // ngprvec[k] = |noise_probability[k]| /
-          //   (|noise_probability[0]| + |noise_probability[1]|)
+          // ngprvec[k] = `noise_probability[k]` /
+          //   (`noise_probability[0]` + `noise_probability[1]`)
 
           // (Q14 * Q11 >> 11) = Q14.
           delt = (int16_t)((ngprvec[gaussian] * deltaN[gaussian]) >> 11);
@@ -327,8 +327,8 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
         if (vadflag) {
           // Update speech mean vector:
           // `deltaS` = (x-mu)/sigma^2
-          // sgprvec[k] = |speech_probability[k]| /
-          //   (|speech_probability[0]| + |speech_probability[1]|)
+          // sgprvec[k] = `speech_probability[k]` /
+          //   (`speech_probability[0]` + `speech_probability[1]`)
 
           // (Q14 * Q11) >> 11 = Q14.
           delt = (int16_t)((sgprvec[gaussian] * deltaS[gaussian]) >> 11);
@@ -430,14 +430,14 @@ static int16_t GmmProbability(VadInstT* self, int16_t* features,
         tmp2_s16 = (int16_t)((3 * tmp_s16) >> 2);
 
         // Move Gaussian means for speech model by `tmp1_s16` and update
-        // `speech_global_mean`. Note that |self->speech_means[channel]| is
+        // `speech_global_mean`. Note that `self->speech_means[channel]` is
         // changed after the call.
         speech_global_mean = WeightedAverage(&self->speech_means[channel],
                                              tmp1_s16,
                                              &kSpeechDataWeights[channel]);
 
         // Move Gaussian means for noise model by -`tmp2_s16` and update
-        // `noise_global_mean`. Note that |self->noise_means[channel]| is
+        // `noise_global_mean`. Note that `self->noise_means[channel]` is
         // changed after the call.
         noise_global_mean = WeightedAverage(&self->noise_means[channel],
                                             -tmp2_s16,

common_audio/vad/vad_sp.h

@@ -35,7 +35,7 @@ void WebRtcVad_Downsampling(const int16_t* signal_in,
 
 // Updates and returns the smoothed feature minimum. As minimum we use the
 // median of the five smallest feature values in a 100 frames long window.
-// As long as |handle->frame_counter| is zero, that is, we haven't received any
+// As long as `handle->frame_counter` is zero, that is, we haven't received any
 // "valid" data, FindMinimum() outputs the default value of 1600.
 //
 // Inputs:

common_video/video_frame_buffer_pool.cc

@@ -20,7 +20,7 @@ namespace {
 bool HasOneRef(const rtc::scoped_refptr<VideoFrameBuffer>& buffer) {
   // Cast to rtc::RefCountedObject is safe because this function is only called
   // on locally created VideoFrameBuffers, which are either
-  // |rtc::RefCountedObject<I420Buffer>| or |rtc::RefCountedObject<NV12Buffer>|.
+  // `rtc::RefCountedObject<I420Buffer>` or `rtc::RefCountedObject<NV12Buffer>`.
   switch (buffer->type()) {
     case VideoFrameBuffer::Type::kI420: {
       return static_cast<rtc::RefCountedObject<I420Buffer>*>(buffer.get())
@@ -94,7 +94,7 @@ rtc::scoped_refptr<I420Buffer> VideoFrameBufferPool::CreateI420Buffer(
       GetExistingBuffer(width, height, VideoFrameBuffer::Type::kI420);
   if (existing_buffer) {
     // Cast is safe because the only way kI420 buffer is created is
-    // in the same function below, where |RefCountedObject<I420Buffer>| is
+    // in the same function below, where `RefCountedObject<I420Buffer>` is
     // created.
     rtc::RefCountedObject<I420Buffer>* raw_buffer =
         static_cast<rtc::RefCountedObject<I420Buffer>*>(existing_buffer.get());
@@ -125,7 +125,7 @@ rtc::scoped_refptr<NV12Buffer> VideoFrameBufferPool::CreateNV12Buffer(
       GetExistingBuffer(width, height, VideoFrameBuffer::Type::kNV12);
   if (existing_buffer) {
     // Cast is safe because the only way kI420 buffer is created is
-    // in the same function below, where |RefCountedObject<I420Buffer>| is
+    // in the same function below, where `RefCountedObject<I420Buffer>` is
     // created.
     rtc::RefCountedObject<NV12Buffer>* raw_buffer =
         static_cast<rtc::RefCountedObject<NV12Buffer>*>(existing_buffer.get());

docs/native-code/rtp-hdrext/transport-wide-cc-02/README.md

@@ -29,19 +29,19 @@ Contact <kron@google.com> or <sprang@google.com> for more info.
 Data layout of transport-wide sequence number
      1-byte header + 2 bytes of data:
 
-      0                   1                   2
+      0                   1                   2
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-     |  ID   | L=1   |transport-wide sequence number |
+     |  ID   | L=1   |transport-wide sequence number |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 Data layout of transport-wide sequence number and optional feedback request
      1-byte header + 4 bytes of data:
 
-      0                   1                   2                   3
+      0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-     |  ID   | L=3   |transport-wide sequence number |T|  seq count  |
+     |  ID   | L=3   |transport-wide sequence number |T|  seq count  |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |seq count cont.|
      +-+-+-+-+-+-+-+-+

logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc

@@ -288,7 +288,7 @@ rtclog2::IceCandidatePairEvent::IceCandidatePairEventType ConvertToProtoFormat(
 }
 
 // Copies all RTCP blocks except APP, SDES and unknown from `packet` to
-// `buffer`. `buffer` must have space for at least |packet.size()| bytes.
+// `buffer`. `buffer` must have space for at least `packet.size()` bytes.
 size_t RemoveNonAllowlistedRtcpBlocks(const rtc::Buffer& packet,
                                       uint8_t* buffer) {
   RTC_DCHECK(buffer != nullptr);

media/base/delayable.h

@@ -18,7 +18,7 @@
 namespace cricket {
 
 // Delayable is used by user code through ApplyConstraints algorithm. Its
-// methods must take precendence over similar functional in |syncable.h|.
+// methods must take precendence over similar functional in `syncable.h`.
 class Delayable {
  public:
   virtual ~Delayable() {}

media/base/video_adapter.h

@@ -86,24 +86,24 @@ class RTC_EXPORT VideoAdapter {
       const absl::optional<int>& max_fps) RTC_LOCKS_EXCLUDED(mutex_);
 
   // Requests the output frame size from `AdaptFrameResolution` to have as close
-  // as possible to |sink_wants.target_pixel_count| pixels (if set)
-  // but no more than |sink_wants.max_pixel_count|.
-  // |sink_wants.max_framerate_fps| is essentially analogous to
-  // |sink_wants.max_pixel_count|, but for framerate rather than resolution.
-  // Set |sink_wants.max_pixel_count| and/or |sink_wants.max_framerate_fps| to
+  // as possible to `sink_wants.target_pixel_count` pixels (if set)
+  // but no more than `sink_wants.max_pixel_count`.
+  // `sink_wants.max_framerate_fps` is essentially analogous to
+  // `sink_wants.max_pixel_count`, but for framerate rather than resolution.
+  // Set `sink_wants.max_pixel_count` and/or `sink_wants.max_framerate_fps` to
   // std::numeric_limit<int>::max() if no upper limit is desired.
   // The sink resolution alignment requirement is given by
-  // |sink_wants.resolution_alignment|.
+  // `sink_wants.resolution_alignment`.
   // Note: Should be called from the sink only.
   void OnSinkWants(const rtc::VideoSinkWants& sink_wants)
       RTC_LOCKS_EXCLUDED(mutex_);
 
   // Returns maximum image area, which shouldn't impose any adaptations.
-  // Can return |numeric_limits<int>::max()| if no limit is set.
+  // Can return `numeric_limits<int>::max()` if no limit is set.
   int GetTargetPixels() const;
 
   // Returns current frame-rate limit.
-  // Can return |numeric_limits<float>::infinity()| if no limit is set.
+  // Can return `numeric_limits<float>::infinity()` if no limit is set.
   float GetMaxFramerate() const;
 
  private:
@@ -124,7 +124,7 @@ class RTC_EXPORT VideoAdapter {
   const int source_resolution_alignment_;
   // The currently applied resolution alignment, as given by the requirements:
   //  - the fixed `source_resolution_alignment_`; and
-  //  - the latest |sink_wants.resolution_alignment|.
+  //  - the latest `sink_wants.resolution_alignment`.
   int resolution_alignment_ RTC_GUARDED_BY(mutex_);
 
   // The target timestamp for the next frame based on requested format.

media/engine/simulcast_encoder_adapter_unittest.cc

@@ -761,7 +761,7 @@ TEST_F(TestSimulcastEncoderAdapterFake, DoesNotLeakEncoders) {
   EXPECT_EQ(3u, helper_->factory()->encoders().size());
 
   // The adapter should destroy all encoders it has allocated. Since
-  // |helper_->factory()| is owned by `adapter_`, however, we need to rely on
+  // `helper_->factory()` is owned by `adapter_`, however, we need to rely on
   // lsan to find leaks here.
   EXPECT_EQ(0, adapter_->Release());
   adapter_.reset();

media/engine/webrtc_media_engine.cc

@@ -27,7 +27,7 @@ namespace cricket {
 
 std::unique_ptr<MediaEngineInterface> CreateMediaEngine(
     MediaEngineDependencies dependencies) {
-  // TODO(sprang): Make populating |dependencies.trials| mandatory and remove
+  // TODO(sprang): Make populating `dependencies.trials` mandatory and remove
   // these fallbacks.
   std::unique_ptr<webrtc::WebRtcKeyValueConfig> fallback_trials(
       dependencies.trials ? nullptr : new webrtc::FieldTrialBasedConfig());

media/engine/webrtc_voice_engine.cc

@@ -152,8 +152,8 @@ absl::optional<std::string> GetAudioNetworkAdaptorConfig(
     const AudioOptions& options) {
   if (options.audio_network_adaptor && *options.audio_network_adaptor &&
       options.audio_network_adaptor_config) {
-    // Turn on audio network adaptor only when |options_.audio_network_adaptor|
-    // equals true and |options_.audio_network_adaptor_config| has a value.
+    // Turn on audio network adaptor only when `options_.audio_network_adaptor`
+    // equals true and `options_.audio_network_adaptor_config` has a value.
     return options.audio_network_adaptor_config;
   }
   return absl::nullopt;
@@ -1495,10 +1495,10 @@ webrtc::RTCError WebRtcVoiceMediaChannel::SetRtpSendParameters(
   }
 
   // TODO(minyue): The following legacy actions go into
-  // |WebRtcAudioSendStream::SetRtpParameters()| which is called at the end,
+  // `WebRtcAudioSendStream::SetRtpParameters()` which is called at the end,
   // though there are two difference:
-  // 1. |WebRtcVoiceMediaChannel::SetChannelSendParameters()| only calls
-  // `SetSendCodec` while |WebRtcAudioSendStream::SetRtpParameters()| calls
+  // 1. `WebRtcVoiceMediaChannel::SetChannelSendParameters()` only calls
+  // `SetSendCodec` while `WebRtcAudioSendStream::SetRtpParameters()` calls
   // `SetSendCodecs`. The outcome should be the same.
   // 2. AudioSendStream can be recreated.
 

media/engine/webrtc_voice_engine_unittest.cc

@@ -2505,7 +2505,7 @@ TEST_P(WebRtcVoiceEngineTestFake, AudioNetworkAdaptorNotGetOverridden) {
   const int initial_num = call_.GetNumCreatedSendStreams();
   cricket::AudioOptions options;
   options.audio_network_adaptor = absl::nullopt;
-  // Unvalued |options.audio_network_adaptor|.should not reset audio network
+  // Unvalued `options.audio_network_adaptor` should not reset audio network
   // adaptor.
   SetAudioSend(kSsrcX, true, nullptr, &options);
   // AudioSendStream not expected to be recreated.

media/sctp/sctp_transport_internal.h

@@ -119,7 +119,7 @@ class SctpTransportInternal {
   // Send data down this channel (will be wrapped as SCTP packets then given to
   // usrsctp that will then post the network interface).
   // Returns true iff successful data somewhere on the send-queue/network.
-  // Uses |params.ssrc| as the SCTP sid.
+  // Uses `params.ssrc` as the SCTP sid.
   virtual bool SendData(int sid,
                         const webrtc::SendDataParams& params,
                         const rtc::CopyOnWriteBuffer& payload,

modules/audio_coding/acm2/acm_receiver.h

@@ -180,7 +180,7 @@ class AcmReceiver {
   // of NACK list are in the range of [N - `max_nack_list_size`, N).
   //
   // `max_nack_list_size` should be positive (none zero) and less than or
-  // equal to |Nack::kNackListSizeLimit|. Otherwise, No change is applied and -1
+  // equal to `Nack::kNackListSizeLimit`. Otherwise, No change is applied and -1
   // is returned. 0 is returned at success.
   //
   int EnableNack(size_t max_nack_list_size);

modules/audio_coding/acm2/audio_coding_module.cc

@@ -229,7 +229,7 @@ int32_t AudioCodingModuleImpl::Encode(
     const InputData& input_data,
     absl::optional<int64_t> absolute_capture_timestamp_ms) {
   // TODO(bugs.webrtc.org/10739): add dcheck that
-  // |audio_frame.absolute_capture_timestamp_ms()| always has a value.
+  // `audio_frame.absolute_capture_timestamp_ms()` always has a value.
   AudioEncoder::EncodedInfo encoded_info;
   uint8_t previous_pltype;
 
@@ -333,7 +333,7 @@ int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
   MutexLock lock(&acm_mutex_);
   int r = Add10MsDataInternal(audio_frame, &input_data_);
   // TODO(bugs.webrtc.org/10739): add dcheck that
-  // |audio_frame.absolute_capture_timestamp_ms()| always has a value.
+  // `audio_frame.absolute_capture_timestamp_ms()` always has a value.
   return r < 0
              ? r
              : Encode(input_data_, audio_frame.absolute_capture_timestamp_ms());

modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc

@@ -85,7 +85,7 @@ TEST(AnaBitrateControllerTest, ChangeBitrateOnTargetBitrateChanged) {
                                                       1000 /
                                                       kInitialFrameLengthMs;
   // Frame length unchanged, bitrate changes in accordance with
-  // |metrics.target_audio_bitrate_bps| and |metrics.overhead_bytes_per_packet|.
+  // `metrics.target_audio_bitrate_bps` and `metrics.overhead_bytes_per_packet`.
   UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
   CheckDecision(&controller, kInitialFrameLengthMs, kBitrateBps);
 }

modules/audio_coding/audio_network_adaptor/config.proto

@@ -169,7 +169,7 @@ message Controller {
   // Shorter distance means higher significance. The significances of
   // controllers determine their order in the processing pipeline. Controllers
   // without `scoring_point` follow their default order in
-  // |ControllerManager::controllers|.
+  // `ControllerManager::controllers`.
   optional ScoringPoint scoring_point = 1;
 
   oneof controller {

modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc

@@ -101,7 +101,7 @@ void UpdateNetworkMetrics(FecControllerPlrBasedTestStates* states,
 }
 
 // Checks that the FEC decision and `uplink_packet_loss_fraction` given by
-// |states->controller->MakeDecision| matches `expected_enable_fec` and
+// `states->controller->MakeDecision` matches `expected_enable_fec` and
 // `expected_uplink_packet_loss_fraction`, respectively.
 void CheckDecision(FecControllerPlrBasedTestStates* states,
                    bool expected_enable_fec,

modules/audio_coding/codecs/cng/webrtc_cng.cc

@@ -195,7 +195,7 @@ bool ComfortNoiseDecoder::Generate(rtc::ArrayView<int16_t> out_data,
 
   /* `lpPoly` - Coefficients in Q12.
    * `excitation` - Speech samples.
-   * |nst->dec_filtstate| - State preservation.
+   * `nst->dec_filtstate` - State preservation.
    * `out_data` - Filtered speech samples. */
   WebRtcSpl_FilterAR(lpPoly, WEBRTC_CNG_MAX_LPC_ORDER + 1, excitation,
                      num_samples, dec_filtstate_, WEBRTC_CNG_MAX_LPC_ORDER,

modules/audio_coding/codecs/isac/main/source/pitch_filter.c

@@ -140,9 +140,9 @@ static void FilterSegment(const double* in_data, PitchFilterParam* parameters,
   int j;
   double sum;
   double sum2;
-  /* Index of |parameters->buffer| where the output is written to. */
+  /* Index of `parameters->buffer` where the output is written to. */
   int pos = parameters->index + PITCH_BUFFSIZE;
-  /* Index of |parameters->buffer| where samples are read for fractional-lag
+  /* Index of `parameters->buffer` where samples are read for fractional-lag
    * computation. */
   int pos_lag = pos - parameters->lag_offset;
 
@@ -174,9 +174,9 @@ static void FilterSegment(const double* in_data, PitchFilterParam* parameters,
         /* Filter for fractional pitch. */
         sum2 = 0.0;
         for (m = PITCH_FRACORDER-1; m >= m_tmp; --m) {
-          /* |lag_index + m| is always larger than or equal to zero, see how
+          /* `lag_index + m` is always larger than or equal to zero, see how
            * m_tmp is computed. This is equivalent to assume samples outside
-           * |out_dg[j]| are zero. */
+           * `out_dg[j]` are zero. */
           sum2 += out_dg[j][lag_index + m] * parameters->interpol_coeff[m];
         }
         /* Add the contribution of differential gain change. */

modules/audio_coding/codecs/opus/audio_encoder_opus.h

@@ -139,7 +139,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
       absl::optional<int64_t> link_capacity_allocation);
 
   // TODO(minyue): remove "override" when we can deprecate
-  // |AudioEncoder::SetTargetBitrate|.
+  // `AudioEncoder::SetTargetBitrate`.
   void SetTargetBitrate(int target_bps) override;
 
   void ApplyAudioNetworkAdaptor();

modules/audio_coding/codecs/opus/opus_unittest.cc

@@ -116,7 +116,7 @@ class OpusTest
   void TestCbrEffect(bool dtx, int block_length_ms);
 
   // Prepare `speech_data_` for encoding, read from a hard-coded file.
-  // After preparation, |speech_data_.GetNextBlock()| returns a pointer to a
+  // After preparation, `speech_data_.GetNextBlock()` returns a pointer to a
   // block of `block_length_ms` milliseconds. The data is looped every
   // `loop_length_ms` milliseconds.
   void PrepareSpeechData(int block_length_ms, int loop_length_ms);

modules/audio_coding/neteq/neteq_impl_unittest.cc

@@ -510,7 +510,7 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
   EXPECT_EQ(1u, output.num_channels_);
   EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
 
-  // Verify |output.packet_infos_|.
+  // Verify `output.packet_infos_`.
   ASSERT_THAT(output.packet_infos_, SizeIs(1));
   {
     const auto& packet_info = output.packet_infos_[0];
@@ -602,7 +602,7 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
   EXPECT_EQ(1u, output.num_channels_);
   EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
 
-  // Verify |output.packet_infos_|.
+  // Verify `output.packet_infos_`.
   ASSERT_THAT(output.packet_infos_, SizeIs(1));
   {
     const auto& packet_info = output.packet_infos_[0];
@@ -648,7 +648,7 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
   // out-of-order packet should have been discarded.
   EXPECT_TRUE(packet_buffer_->Empty());
 
-  // Verify |output.packet_infos_|. Expect to only see the second packet.
+  // Verify `output.packet_infos_`. Expect to only see the second packet.
   ASSERT_THAT(output.packet_infos_, SizeIs(1));
   {
     const auto& packet_info = output.packet_infos_[0];

modules/audio_device/fine_audio_buffer.h

@@ -42,8 +42,8 @@ class FineAudioBuffer {
   bool IsReadyForPlayout() const;
   bool IsReadyForRecord() const;
 
-  // Copies audio samples into `audio_buffer` where number of requested
-  // elements is specified by |audio_buffer.size()|. The producer will always
+  // Copies audio samples into `audio_buffer` where number of requested
+  // elements is specified by `audio_buffer.size()`. The producer will always
   // fill up the audio buffer and if no audio exists, the buffer will contain
   // silence instead. The provided delay estimate in `playout_delay_ms` should
   // contain an estimate of the latency between when an audio frame is read from

modules/audio_device/win/core_audio_base_win.cc

@@ -448,7 +448,7 @@ bool CoreAudioBase::Init() {
   // - HDAudio driver
   // - kEnableLowLatencyIfSupported changed from false (default) to true.
   // TODO(henrika): IsLowLatencySupported() returns AUDCLNT_E_UNSUPPORTED_FORMAT
-  // when |sample_rate_.has_value()| returns true if rate conversion is
+  // when `sample_rate_.has_value()` returns true if rate conversion is
   // actually required (i.e., client asks for other than the default rate).
   bool low_latency_support = false;
   uint32_t min_period_in_frames = 0;

modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc

@@ -250,24 +250,24 @@ bool IsAlternativePitchStrongerThanInitial(PitchInfo last,
   RTC_DCHECK_GE(initial.period, 0);
   RTC_DCHECK_GE(alternative.period, 0);
   RTC_DCHECK_GE(period_divisor, 2);
-  // Compute a term that lowers the threshold when |alternative.period| is close
-  // to the last estimated period |last.period| - i.e., pitch tracking.
+  // Compute a term that lowers the threshold when `alternative.period` is close
+  // to the last estimated period `last.period` - i.e., pitch tracking.
   float lower_threshold_term = 0.f;
   if (std::abs(alternative.period - last.period) <= 1) {
     // The candidate pitch period is within 1 sample from the last one.
-    // Make the candidate at |alternative.period| very easy to be accepted.
+    // Make the candidate at `alternative.period` very easy to be accepted.
     lower_threshold_term = last.strength;
   } else if (std::abs(alternative.period - last.period) == 2 &&
              initial.period >
                  kInitialPitchPeriodThresholds[period_divisor - 2]) {
     // The candidate pitch period is 2 samples far from the last one and the
-    // period |initial.period| (from which |alternative.period| has been
-    // derived) is greater than a threshold. Make |alternative.period| easy to
+    // period `initial.period` (from which `alternative.period` has been
+    // derived) is greater than a threshold. Make `alternative.period` easy to
     // be accepted.
     lower_threshold_term = 0.5f * last.strength;
   }
   // Set the threshold based on the strength of the initial estimate
-  // |initial.period|. Also reduce the chance of false positives caused by a
+  // `initial.period`. Also reduce the chance of false positives caused by a
   // bias towards high frequencies (originating from short-term correlations).
   float threshold =
       std::max(0.3f, 0.7f * initial.strength - lower_threshold_term);
@@ -457,7 +457,7 @@ PitchInfo ComputeExtendedPitchPeriod48kHz(
     alternative_pitch.period = GetAlternativePitchPeriod(
         initial_pitch.period, /*multiplier=*/1, period_divisor);
     RTC_DCHECK_GE(alternative_pitch.period, kMinPitch24kHz);
-    // When looking at |alternative_pitch.period|, we also look at one of its
+    // When looking at `alternative_pitch.period`, we also look at one of its
     // sub-harmonics. `kSubHarmonicMultipliers` is used to know where to look.
     // `period_divisor` == 2 is a special case since `dual_alternative_period`
     // might be greater than the maximum pitch period.
@@ -472,7 +472,7 @@ PitchInfo ComputeExtendedPitchPeriod48kHz(
         << "The lower pitch period and the additional sub-harmonic must not "
            "coincide.";
     // Compute an auto-correlation score for the primary pitch candidate
-    // |alternative_pitch.period| by also looking at its possible sub-harmonic
+    // `alternative_pitch.period` by also looking at its possible sub-harmonic
     // `dual_alternative_period`.
     const float xy_primary_period = ComputeAutoCorrelation(
         kMaxPitch24kHz - alternative_pitch.period, pitch_buffer, vector_math);

modules/audio_processing/gain_controller2_unittest.cc

@@ -310,7 +310,7 @@ INSTANTIATE_TEST_SUITE_P(
     GainController2,
     FixedDigitalTest,
     ::testing::Values(
-        // When gain < |test::kLimiterMaxInputLevelDbFs|, the limiter will not
+        // When gain < `test::kLimiterMaxInputLevelDbFs`, the limiter will not
         // saturate the signal (at any sample rate).
         FixedDigitalTestParams(0.1f,
                                test::kLimiterMaxInputLevelDbFs - 0.01f,
@@ -320,7 +320,7 @@ INSTANTIATE_TEST_SUITE_P(
                                test::kLimiterMaxInputLevelDbFs - 0.01f,
                                48000,
                                false),
-        // When gain > |test::kLimiterMaxInputLevelDbFs|, the limiter will
+        // When gain > `test::kLimiterMaxInputLevelDbFs`, the limiter will
         // saturate the signal (at any sample rate).
         FixedDigitalTestParams(test::kLimiterMaxInputLevelDbFs + 0.01f,
                                10.f,

modules/audio_processing/include/audio_processing.h

@@ -570,8 +570,8 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
   // The int16 interfaces require:
   //   - only `NativeRate`s be used
   //   - that the input, output and reverse rates must match
-  //   - that |processing_config.output_stream()| matches
-  //     |processing_config.input_stream()|.
+  //   - that `processing_config.output_stream()` matches
+  //     `processing_config.input_stream()`.
   //
   // The float interfaces accept arbitrary rates and support differing input and
   // output layouts, but the output must have either one channel or the same

modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py

@@ -349,7 +349,7 @@ class HtmlExport(object):
 
     def _SliceDataForScoreTableCell(self, score_name, apm_config,
                                     test_data_gen, test_data_gen_params):
-        """Slices |self._scores_data_frame| to extract the data for a tab."""
+        """Slices `self._scores_data_frame` to extract the data for a tab."""
         masks = []
         masks.append(self._scores_data_frame.eval_score_name == score_name)
         masks.append(self._scores_data_frame.apm_config == apm_config)

modules/desktop_capture/desktop_region.cc

@@ -91,7 +91,7 @@ void DesktopRegion::AddRect(const DesktopRect& rect) {
     return;
 
   // Top of the part of the `rect` that hasn't been inserted yet. Increased as
-  // we iterate over the rows until it reaches |rect.bottom()|.
+  // we iterate over the rows until it reaches `rect.bottom()`.
   int top = rect.top();
 
   // Iterate over all rows that may intersect with `rect` and add new rows when
@@ -456,7 +456,7 @@ void DesktopRegion::AddSpanToRow(Row* row, int left, int right) {
 
 // static
 bool DesktopRegion::IsSpanInRow(const Row& row, const RowSpan& span) {
-  // Find the first span that starts at or after |span.left| and then check if
+  // Find the first span that starts at or after `span.left` and then check if
   // it's the same span.
   RowSpanSet::const_iterator it = std::lower_bound(
       row.spans.begin(), row.spans.end(), span.left, CompareSpanLeft);

modules/desktop_capture/win/wgc_capture_session.cc

@@ -286,7 +286,7 @@ HRESULT WgcCaptureSession::GetFrame(
   int image_width = std::min(previous_size_.Width, new_size.Width);
   int row_data_length = image_width * DesktopFrame::kBytesPerPixel;
 
-  // Make a copy of the data pointed to by |map_info.pData| so we are free to
+  // Make a copy of the data pointed to by `map_info.pData` so we are free to
   // unmap our texture.
   uint8_t* src_data = static_cast<uint8_t*>(map_info.pData);
   std::vector<uint8_t> image_data;

modules/pacing/round_robin_packet_queue.cc

@@ -175,7 +175,7 @@ std::unique_ptr<RtpPacketToSend> RoundRobinPacketQueue::Pop() {
 
   // Calculate the total amount of time spent by this packet in the queue
   // while in a non-paused state. Note that the `pause_time_sum_ms_` was
-  // subtracted from |packet.enqueue_time_ms| when the packet was pushed, and
+  // subtracted from `packet.enqueue_time_ms` when the packet was pushed, and
   // by subtracting it now we effectively remove the time spent in in the
   // queue while in a paused state.
   TimeDelta time_in_non_paused_state =

modules/rtp_rtcp/source/absolute_capture_time_interpolator.h

@@ -45,7 +45,7 @@ class AbsoluteCaptureTimeInterpolator {
                             rtc::ArrayView<const uint32_t> csrcs);
 
   // Returns a received header extension, an interpolated header extension, or
-  // |absl::nullopt| if it's not possible to interpolate a header extension.
+  // `absl::nullopt` if it's not possible to interpolate a header extension.
   absl::optional<AbsoluteCaptureTime> OnReceivePacket(
       uint32_t source,
       uint32_t rtp_timestamp,

modules/rtp_rtcp/source/absolute_capture_time_sender.h

@@ -50,7 +50,7 @@ class AbsoluteCaptureTimeSender {
   static uint32_t GetSource(uint32_t ssrc,
                             rtc::ArrayView<const uint32_t> csrcs);
 
-  // Returns a header extension to be sent, or |absl::nullopt| if the header
+  // Returns a header extension to be sent, or `absl::nullopt` if the header
   // extension shouldn't be sent.
   absl::optional<AbsoluteCaptureTime> OnSendPacket(
       uint32_t source,

modules/rtp_rtcp/source/fec_test_helper.h

@@ -113,7 +113,7 @@ class UlpfecPacketGenerator : public AugmentedPacketGenerator {
   // Creates a new RtpPacket with FEC payload and RED header. Does this by
   // creating a new fake media AugmentedPacket, clears the marker bit and adds a
   // RED header. Finally replaces the payload with the content of
-  // |packet->data|.
+  // `packet->data`.
   RtpPacketReceived BuildUlpfecRedPacket(
       const ForwardErrorCorrection::Packet& packet);
 };

modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc

@@ -138,7 +138,7 @@ class FeedbackTester {
 };
 
 // The following tests use FeedbackTester that simulates received packets as
-// specified by the parameters |received_seq[]| and |received_ts[]| (optional).
+// specified by the parameters `received_seq[]` and `received_ts[]` (optional).
 // The following is verified in these tests:
 // - Expected size of serialized packet.
 // - Expected sequence numbers and receive deltas.

modules/rtp_rtcp/source/rtcp_receiver.cc

@@ -599,7 +599,7 @@ void RTCPReceiver::HandleReportBlock(const ReportBlock& report_block,
   //
   // We can calc RTT if we send a send report and get a report block back.
 
-  // |report_block.source_ssrc()| is the SSRC identifier of the source to
+  // `report_block.source_ssrc()` is the SSRC identifier of the source to
   // which the information in this reception report block pertains.
 
   // Filter out all report blocks that are not for us.
@@ -957,7 +957,7 @@ void RTCPReceiver::HandleTmmbr(const CommonHeader& rtcp_block,
     entry->tmmbr_item = rtcp::TmmbItem(sender_ssrc, request.bitrate_bps(),
                                        request.packet_overhead());
     // FindOrCreateTmmbrInfo always sets `last_time_received_ms` to
-    // |clock_->TimeInMilliseconds()|.
+    // `clock_->TimeInMilliseconds()`.
     entry->last_updated_ms = tmmbr_info->last_time_received_ms;
 
     packet_information->packet_type_flags |= kRtcpTmmbr;

modules/rtp_rtcp/source/rtcp_receiver_unittest.cc

@@ -1335,7 +1335,7 @@ TEST(RtcpReceiverTest,
   const int64_t kUtcNowUs = 42;
 
   // The "report_block_timestamp_utc_us" is obtained from the global UTC clock
-  // (not the simulcated |mocks.clock|) and requires a scoped fake clock.
+  // (not the simulcated `mocks.clock`) and requires a scoped fake clock.
   rtc::ScopedFakeClock fake_clock;
   fake_clock.SetTime(Timestamp::Micros(kUtcNowUs));
 

modules/rtp_rtcp/source/rtcp_transceiver.h

@@ -47,7 +47,7 @@ class RtcpTransceiver : public RtcpFeedbackSenderInterface {
   void Stop(std::function<void()> on_destroyed);
 
   // Registers observer to be notified about incoming rtcp packets.
-  // Calls to observer will be done on the |config.task_queue|.
+  // Calls to observer will be done on the `config.task_queue`.
   void AddMediaReceiverRtcpObserver(uint32_t remote_ssrc,
                                     MediaReceiverRtcpObserver* observer);
   // Deregisters the observer. Might return before observer is deregistered.

modules/rtp_rtcp/source/rtcp_transceiver_impl.cc

@@ -431,7 +431,7 @@ std::vector<rtcp::ReportBlock> RtcpTransceiverImpl::CreateReportBlocks(
   if (!config_.receive_statistics)
     return {};
   // TODO(danilchap): Support sending more than
-  // |ReceiverReport::kMaxNumberOfReportBlocks| per compound rtcp packet.
+  // `ReceiverReport::kMaxNumberOfReportBlocks` per compound rtcp packet.
   std::vector<rtcp::ReportBlock> report_blocks =
       config_.receive_statistics->RtcpReportBlocks(
           rtcp::ReceiverReport::kMaxNumberOfReportBlocks);

modules/rtp_rtcp/source/rtp_sender.cc

@@ -693,7 +693,7 @@ static void CopyHeaderAndExtensionsToRtxPacket(const RtpPacketToSend& packet,
       continue;
     }
 
-    // Empty extensions should be supported, so not checking |source.empty()|.
+    // Empty extensions should be supported, so not checking `source.empty()`.
     if (!packet.HasExtension(extension)) {
       continue;
     }

modules/rtp_rtcp/source/ulpfec_generator.cc

@@ -30,12 +30,12 @@ namespace {
 constexpr size_t kRedForFecHeaderLength = 1;
 
 // This controls the maximum amount of excess overhead (actual - target)
-// allowed in order to trigger EncodeFec(), before |params_.max_fec_frames|
+// allowed in order to trigger EncodeFec(), before `params_.max_fec_frames`
 // is reached. Overhead here is defined as relative to number of media packets.
 constexpr int kMaxExcessOverhead = 50;  // Q8.
 
 // This is the minimum number of media packets required (above some protection
-// level) in order to trigger EncodeFec(), before |params_.max_fec_frames| is
+// level) in order to trigger EncodeFec(), before `params_.max_fec_frames` is
 // reached.
 constexpr size_t kMinMediaPackets = 4;
 
@@ -146,7 +146,7 @@ void UlpfecGenerator::AddPacketAndGenerateFec(const RtpPacketToSend& packet) {
 
   auto params = CurrentParams();
 
-  // Produce FEC over at most |params_.max_fec_frames| frames, or as soon as:
+  // Produce FEC over at most `params_.max_fec_frames` frames, or as soon as:
   // (1) the excess overhead (actual overhead - requested/target overhead) is
   // less than `kMaxExcessOverhead`, and
   // (2) at least `min_num_media_packets_` media packets is reached.

modules/rtp_rtcp/source/ulpfec_generator.h

@@ -83,7 +83,7 @@ class UlpfecGenerator : public VideoFecGenerator {
   // Returns true if the excess overhead (actual - target) for the FEC is below
   // the amount `kMaxExcessOverhead`. This effects the lower protection level
   // cases and low number of media packets/frame. The target overhead is given
-  // by |params_.fec_rate|, and is only achievable in the limit of large number
+  // by `params_.fec_rate`, and is only achievable in the limit of large number
   // of media packets.
   bool ExcessOverheadBelowMax() const;
 

modules/video_coding/codecs/h264/h264_decoder_impl.cc

@@ -80,9 +80,9 @@ int H264DecoderImpl::AVGetBuffer2(AVCodecContext* context,
   RTC_CHECK(context->pix_fmt == kPixelFormatDefault ||
             context->pix_fmt == kPixelFormatFullRange);
 
-  // |av_frame->width| and |av_frame->height| are set by FFmpeg. These are the
-  // actual image's dimensions and may be different from |context->width| and
-  // |context->coded_width| due to reordering.
+  // `av_frame->width` and `av_frame->height` are set by FFmpeg. These are the
+  // actual image's dimensions and may be different from `context->width` and
+  // `context->coded_width` due to reordering.
   int width = av_frame->width;
   int height = av_frame->height;
   // See `lowres`, if used the decoder scales the image by 1/2^(lowres). This
@@ -201,7 +201,7 @@ int32_t H264DecoderImpl::InitDecode(const VideoCodec* codec_settings,
   av_context_->extradata = nullptr;
   av_context_->extradata_size = 0;
 
-  // If this is ever increased, look at |av_context_->thread_safe_callbacks| and
+  // If this is ever increased, look at `av_context_->thread_safe_callbacks` and
   // make it possible to disable the thread checker in the frame buffer pool.
   av_context_->thread_count = 1;
   av_context_->thread_type = FF_THREAD_SLICE;

modules/video_coding/codecs/h264/h264_decoder_impl.h

@@ -61,7 +61,7 @@ class H264DecoderImpl : public H264Decoder {
   ~H264DecoderImpl() override;
 
   // If `codec_settings` is NULL it is ignored. If it is not NULL,
-  // |codec_settings->codecType| must be `kVideoCodecH264`.
+  // `codec_settings->codecType` must be `kVideoCodecH264`.
   int32_t InitDecode(const VideoCodec* codec_settings,
                      int32_t number_of_cores) override;
   int32_t Release() override;

modules/video_coding/codecs/h264/h264_encoder_impl.cc

@@ -89,13 +89,13 @@ VideoFrameType ConvertToVideoFrameType(EVideoFrameType type) {
 
 // Helper method used by H264EncoderImpl::Encode.
 // Copies the encoded bytes from `info` to `encoded_image`. The
-// |encoded_image->_buffer| may be deleted and reallocated if a bigger buffer is
+// `encoded_image->_buffer` may be deleted and reallocated if a bigger buffer is
 // required.
 //
 // After OpenH264 encoding, the encoded bytes are stored in `info` spread out
 // over a number of layers and "NAL units". Each NAL unit is a fragment starting
 // with the four-byte start code {0,0,0,1}. All of this data (including the
-// start codes) is copied to the |encoded_image->_buffer|.
+// start codes) is copied to the `encoded_image->_buffer`.
 static void RtpFragmentize(EncodedImage* encoded_image, SFrameBSInfo* info) {
   // Calculate minimum buffer size required to hold encoded data.
   size_t required_capacity = 0;
@@ -115,7 +115,7 @@ static void RtpFragmentize(EncodedImage* encoded_image, SFrameBSInfo* info) {
   encoded_image->SetEncodedData(buffer);
 
   // Iterate layers and NAL units, note each NAL unit as a fragment and copy
-  // the data to |encoded_image->_buffer|.
+  // the data to `encoded_image->_buffer`.
   const uint8_t start_code[4] = {0, 0, 0, 1};
   size_t frag = 0;
   encoded_image->set_size(0);
@@ -489,7 +489,7 @@ int32_t H264EncoderImpl::Encode(
     RtpFragmentize(&encoded_images_[i], &info);
 
     // Encoder can skip frames to save bandwidth in which case
-    // |encoded_images_[i]._length| == 0.
+    // `encoded_images_[i]._length` == 0.
     if (encoded_images_[i].size() > 0) {
       // Parse QP.
       h264_bitstream_parser_.ParseBitstream(encoded_images_[i]);

modules/video_coding/codecs/h264/h264_encoder_impl.h

@@ -57,7 +57,7 @@ class H264EncoderImpl : public H264Encoder {
   explicit H264EncoderImpl(const cricket::VideoCodec& codec);
   ~H264EncoderImpl() override;
 
-  // |settings.max_payload_size| is ignored.
+  // `settings.max_payload_size` is ignored.
   // The following members of `codec_settings` are used. The rest are ignored.
   // - codecType (must be kVideoCodecH264)
   // - targetBitrate

modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc

@@ -1049,7 +1049,7 @@ int LibvpxVp8Encoder::Encode(const VideoFrame& frame,
           error == WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT)) {
     ++num_tries;
     // Note we must pass 0 for `flags` field in encode call below since they are
-    // set above in |libvpx_interface_->vpx_codec_control_| function for each
+    // set above in `libvpx_interface_->vpx_codec_control_` function for each
     // encoder/spatial layer.
     error = libvpx_->codec_encode(&encoders_[0], &raw_images_[0], timestamp_,
                                   duration, 0, VPX_DL_REALTIME);

modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc

@@ -247,7 +247,7 @@ int LibvpxVp9Decoder::Decode(const EncodedImage& input_image,
                        VPX_DL_REALTIME)) {
     return WEBRTC_VIDEO_CODEC_ERROR;
   }
-  // |img->fb_priv| contains the image data, a reference counted Vp9FrameBuffer.
+  // `img->fb_priv` contains the image data, a reference counted Vp9FrameBuffer.
   // It may be released by libvpx during future vpx_codec_decode or
   // vpx_codec_destroy calls.
   img = vpx_codec_get_frame(decoder_, &iter);

modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h

@@ -226,7 +226,7 @@ class LibvpxVp9Encoder : public VP9Encoder {
   // Performance flags, ordered by `min_pixel_count`.
   const PerformanceFlags performance_flags_;
   // Caching of of `speed_configs_`, where index i maps to the resolution as
-  // specified in |codec_.spatialLayer[i]|.
+  // specified in `codec_.spatialLayer[i]`.
   std::vector<PerformanceFlags::ParameterSet>
       performance_flags_by_spatial_index_;
   void UpdatePerformanceFlags();

modules/video_coding/session_info.h

@@ -79,7 +79,7 @@ class VCMSessionInfo {
   void InformOfEmptyPacket(uint16_t seq_num);
 
   // Finds the packet of the beginning of the next VP8 partition. If
-  // none is found the returned iterator points to |packets_.end()|.
+  // none is found the returned iterator points to `packets_.end()`.
   // `it` is expected to point to the last packet of the previous partition,
   // or to the first packet of the frame. `packets_skipped` is incremented
   // for each packet found which doesn't have the beginning bit set.

p2p/base/p2p_transport_channel.h

@@ -378,7 +378,7 @@ class RTC_EXPORT P2PTransportChannel : public IceTransportInternal {
   void SetReceiving(bool receiving);
   // Clears the address and the related address fields of a local candidate to
   // avoid IP leakage. This is applicable in several scenarios as commented in
-  // |PortAllocator::SanitizeCandidate|.
+  // `PortAllocator::SanitizeCandidate`.
   Candidate SanitizeLocalCandidate(const Candidate& c) const;
   // Clears the address field of a remote candidate to avoid IP leakage. This is
   // applicable in the following scenarios:

p2p/base/transport_description_factory_unittest.cc

@@ -291,25 +291,25 @@ TEST_F(TransportDescriptionFactoryTest, TestAnswerDtlsToDtls) {
 }
 
 // Test that ice ufrag and password is changed in an updated offer and answer
-// if |TransportDescriptionOptions::ice_restart| is true.
+// if `TransportDescriptionOptions::ice_restart` is true.
 TEST_F(TransportDescriptionFactoryTest, TestIceRestart) {
   TestIceRestart(false);
 }
 
 // Test that ice ufrag and password is changed in an updated offer and answer
-// if |TransportDescriptionOptions::ice_restart| is true and DTLS is enabled.
+// if `TransportDescriptionOptions::ice_restart` is true and DTLS is enabled.
 TEST_F(TransportDescriptionFactoryTest, TestIceRestartWithDtls) {
   TestIceRestart(true);
 }
 
 // Test that ice renomination is set in an updated offer and answer
-// if |TransportDescriptionOptions::enable_ice_renomination| is true.
+// if `TransportDescriptionOptions::enable_ice_renomination` is true.
 TEST_F(TransportDescriptionFactoryTest, TestIceRenomination) {
   TestIceRenomination(false);
 }
 
 // Test that ice renomination is set in an updated offer and answer
-// if |TransportDescriptionOptions::enable_ice_renomination| is true and DTLS
+// if `TransportDescriptionOptions::enable_ice_renomination` is true and DTLS
 // is enabled.
 TEST_F(TransportDescriptionFactoryTest, TestIceRenominationWithDtls) {
   TestIceRenomination(true);

pc/dtmf_sender.h

@@ -42,7 +42,7 @@ class DtmfProviderInterface {
   // The `duration` indicates the length of the DTMF tone in ms.
   // Returns true on success and false on failure.
   virtual bool InsertDtmf(int code, int duration) = 0;
-  // Returns a |sigslot::signal0<>| signal. The signal should fire before
+  // Returns a `sigslot::signal0<>` signal. The signal should fire before
   // the provider is destroyed.
   virtual sigslot::signal0<>* GetOnDestroyedSignal() = 0;
 

pc/ice_server_parsing.cc

@@ -104,7 +104,7 @@ static bool ParsePort(const std::string& in_str, int* port) {
 // This method parses IPv6 and IPv4 literal strings, along with hostnames in
 // standard hostname:port format.
 // Consider following formats as correct.
-// |hostname:port|, |[IPV6 address]:port|, |IPv4 address|:port,
+// `hostname:port`, |[IPV6 address]:port|, |IPv4 address|:port,
 // `hostname`, |[IPv6 address]|, |IPv4 address|.
 static bool ParseHostnameAndPortFromString(const std::string& in_str,
                                            std::string* host,

pc/jsep_session_description.cc

@@ -104,7 +104,7 @@ void UpdateConnectionAddress(
     // Combining the above considerations, we use 0.0.0.0 with port 9 to
     // populate the c= and the m= lines. See `BuildMediaDescription` in
     // webrtc_sdp.cc for the SDP generation with
-    // |media_desc->connection_address()|.
+    // `media_desc->connection_address()`.
     connection_addr = rtc::SocketAddress(kDummyAddress, kDummyPort);
   }
   media_desc->set_connection_address(connection_addr);

pc/jsep_transport.h

@@ -323,7 +323,7 @@ class JsepTransport {
       RTC_GUARDED_BY(network_thread_);
 
   // This is invoked when RTCP-mux becomes active and
-  // |rtcp_dtls_transport_| is destroyed. The JsepTransportController will
+  // `rtcp_dtls_transport_` is destroyed. The JsepTransportController will
   // receive the callback and update the aggregate transport states.
   std::function<void()> rtcp_mux_active_callback_;
 

pc/media_session.cc

@@ -1755,7 +1755,7 @@ MediaSessionDescriptionFactory::CreateAnswer(
     ContentInfo& added = answer->contents().back();
     if (!added.rejected && session_options.bundle_enabled &&
         bundle_index.has_value()) {
-      // The `bundle_index` is for |media_description_options.mid|.
+      // The `bundle_index` is for `media_description_options.mid`.
       RTC_DCHECK_EQ(media_description_options.mid, added.name);
       answer_bundles[bundle_index.value()].AddContentName(added.name);
       bundle_transports[bundle_index.value()].reset(

pc/media_session_unittest.cc

@@ -2719,7 +2719,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   // offer/answer exchange plus the audio codecs only `f2_` offer, sorted in
   // preference order.
   // TODO(wu): `updated_offer` should not include the codec
-  // (i.e. |kAudioCodecs2[0]|) the other side doesn't support.
+  // (i.e. `kAudioCodecs2[0]`) the other side doesn't support.
   const AudioCodec kUpdatedAudioCodecOffer[] = {
       kAudioCodecsAnswer[0],
       kAudioCodecsAnswer[1],

pc/peer_connection_ice_unittest.cc

@@ -548,7 +548,7 @@ TEST_P(PeerConnectionIceTest,
   ASSERT_TRUE(
       caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
 
-  // |candidate.transport_name()| is empty.
+  // `candidate.transport_name()` is empty.
   cricket::Candidate candidate = CreateLocalUdpCandidate(kCalleeAddress);
   auto* audio_content = cricket::GetFirstAudioContent(
       caller->pc()->local_description()->description());
@@ -1492,7 +1492,7 @@ TEST_P(PeerConnectionIceTest, PrefersMidOverMLineIndex) {
   ASSERT_TRUE(
       caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
 
-  // |candidate.transport_name()| is empty.
+  // `candidate.transport_name()` is empty.
   cricket::Candidate candidate = CreateLocalUdpCandidate(kCalleeAddress);
   auto* audio_content = cricket::GetFirstAudioContent(
       caller->pc()->local_description()->description());

pc/peer_connection_integrationtest.cc

@@ -3194,7 +3194,7 @@ TEST_P(PeerConnectionIntegrationTest, RegatherAfterChangingIceTransportType) {
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kDefaultTimeout);
   // Note that we cannot use the metric
-  // |WebRTC.PeerConnection.CandidatePairType_UDP| in this test since this
+  // `WebRTC.PeerConnection.CandidatePairType_UDP` in this test since this
   // metric is only populated when we reach kIceConnectionComplete in the
   // current implementation.
   EXPECT_EQ(cricket::RELAY_PORT_TYPE,

pc/peer_connection_rtp_unittest.cc

@@ -58,7 +58,7 @@
 #include "test/gtest.h"
 
 // This file contains tests for RTP Media API-related behavior of
-// |webrtc::PeerConnection|, see https://w3c.github.io/webrtc-pc/#rtp-media-api.
+// `webrtc::PeerConnection`, see https://w3c.github.io/webrtc-pc/#rtp-media-api.
 
 namespace webrtc {
 
@@ -188,7 +188,7 @@ class PeerConnectionRtpTestUnifiedPlan : public PeerConnectionRtpBaseTest {
   }
 };
 
-// These tests cover |webrtc::PeerConnectionObserver| callbacks firing upon
+// These tests cover `webrtc::PeerConnectionObserver` callbacks firing upon
 // setting the remote description.
 
 TEST_P(PeerConnectionRtpTest, AddTrackWithoutStreamFiresOnAddTrack) {
@@ -1994,7 +1994,7 @@ TEST_P(PeerConnectionRtpTest, CreateTwoSendersWithSameTrack) {
 
   if (sdp_semantics_ == SdpSemantics::kPlanB) {
     // TODO(hbos): When https://crbug.com/webrtc/8734 is resolved, this should
-    // return true, and doing |callee->SetRemoteDescription()| should work.
+    // return true, and doing `callee->SetRemoteDescription()` should work.
     EXPECT_FALSE(caller->CreateOfferAndSetAsLocal());
   } else {
     EXPECT_TRUE(caller->CreateOfferAndSetAsLocal());

pc/rtc_stats_collector_unittest.cc

@@ -1466,9 +1466,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
   expected_pair.responses_received = 4321;
   expected_pair.responses_sent = 1000;
   expected_pair.consent_requests_sent = (2020 - 2000);
-  // |expected_pair.current_round_trip_time| should be undefined because the
+  // `expected_pair.current_round_trip_time` should be undefined because the
   // current RTT is not set.
-  // |expected_pair.available_[outgoing/incoming]_bitrate| should be undefined
+  // `expected_pair.available_[outgoing/incoming]_bitrate` should be undefined
   // because is is not the current pair.
 
   ASSERT_TRUE(report->Get(expected_pair.id()));
@@ -1768,7 +1768,7 @@ TEST_F(RTCStatsCollectorTest,
       IdForType<RTCMediaStreamTrackStats>(report), report->timestamp_us(),
       RTCMediaStreamTrackKind::kAudio);
   expected_remote_audio_track.track_identifier = remote_audio_track->id();
-  // |expected_remote_audio_track.media_source_id| should be undefined
+  // `expected_remote_audio_track.media_source_id` should be undefined
   // because the track is remote.
   expected_remote_audio_track.remote_source = true;
   expected_remote_audio_track.ended = false;
@@ -1920,7 +1920,7 @@ TEST_F(RTCStatsCollectorTest,
       RTCMediaStreamTrackKind::kVideo);
   expected_remote_video_track_ssrc3.track_identifier =
       remote_video_track_ssrc3->id();
-  // |expected_remote_video_track_ssrc3.media_source_id| should be undefined
+  // `expected_remote_video_track_ssrc3.media_source_id` should be undefined
   // because the track is remote.
   expected_remote_video_track_ssrc3.remote_source = true;
   expected_remote_video_track_ssrc3.ended = true;
@@ -2011,7 +2011,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
   expected_audio.header_bytes_received = 4;
   expected_audio.packets_lost = -1;
   expected_audio.packets_discarded = 7788;
-  // |expected_audio.last_packet_received_timestamp| should be undefined.
+  // `expected_audio.last_packet_received_timestamp` should be undefined.
   expected_audio.jitter = 4.5;
   expected_audio.jitter_buffer_delay = 1.0;
   expected_audio.jitter_buffer_emitted_count = 2;
@@ -2116,16 +2116,16 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
   expected_video.frames_decoded = 9;
   expected_video.key_frames_decoded = 3;
   expected_video.frames_dropped = 13;
-  // |expected_video.qp_sum| should be undefined.
+  // `expected_video.qp_sum` should be undefined.
   expected_video.total_decode_time = 9.0;
   expected_video.total_inter_frame_delay = 0.123;
   expected_video.total_squared_inter_frame_delay = 0.00456;
   expected_video.jitter = 1.199;
   expected_video.jitter_buffer_delay = 3.456;
   expected_video.jitter_buffer_emitted_count = 13;
-  // |expected_video.last_packet_received_timestamp| should be undefined.
-  // |expected_video.content_type| should be undefined.
-  // |expected_video.decoder_implementation| should be undefined.
+  // `expected_video.last_packet_received_timestamp` should be undefined.
+  // `expected_video.content_type` should be undefined.
+  // `expected_video.decoder_implementation` should be undefined.
 
   ASSERT_TRUE(report->Get(expected_video.id()));
   EXPECT_EQ(
@@ -2189,7 +2189,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
   RTCOutboundRTPStreamStats expected_audio("RTCOutboundRTPAudioStream_1",
                                            report->timestamp_us());
   expected_audio.media_source_id = "RTCAudioSource_50";
-  // |expected_audio.remote_id| should be undefined.
+  // `expected_audio.remote_id` should be undefined.
   expected_audio.ssrc = 1;
   expected_audio.media_type = "audio";
   expected_audio.kind = "audio";
@@ -2275,7 +2275,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
   RTCOutboundRTPStreamStats expected_video(stats_of_my_type[0]->id(),
                                            report->timestamp_us());
   expected_video.media_source_id = "RTCVideoSource_50";
-  // |expected_video.remote_id| should be undefined.
+  // `expected_video.remote_id` should be undefined.
   expected_video.ssrc = 1;
   expected_video.media_type = "video";
   expected_video.kind = "video";
@@ -2305,9 +2305,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
   expected_video.frames_per_second = 10.0;
   expected_video.frames_sent = 5;
   expected_video.huge_frames_sent = 2;
-  // |expected_video.content_type| should be undefined.
-  // |expected_video.qp_sum| should be undefined.
-  // |expected_video.encoder_implementation| should be undefined.
+  // `expected_video.content_type` should be undefined.
+  // `expected_video.qp_sum` should be undefined.
+  // `expected_video.encoder_implementation` should be undefined.
   ASSERT_TRUE(report->Get(expected_video.id()));
 
   EXPECT_EQ(
@@ -2889,7 +2889,7 @@ TEST_P(RTCStatsCollectorTestWithParamKind,
     report_block_data.SetReportBlock(report_block, kReportBlockTimestampUtcUs);
     report_block_data.AddRoundTripTimeSample(kRoundTripTimeSample1Ms);
     // Only the last sample should be exposed as the
-    // |RTCRemoteInboundRtpStreamStats::round_trip_time|.
+    // `RTCRemoteInboundRtpStreamStats::round_trip_time`.
     report_block_data.AddRoundTripTimeSample(kRoundTripTimeSample2Ms);
     report_block_datas.push_back(report_block_data);
   }

pc/rtp_sender.cc

@@ -538,7 +538,7 @@ void AudioRtpSender::SetSend() {
   }
 #endif
 
-  // |track_->enabled()| hops to the signaling thread, so call it before we hop
+  // `track_->enabled()` hops to the signaling thread, so call it before we hop
   // to the worker thread or else it will deadlock.
   bool track_enabled = track_->enabled();
   bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {

pc/sdp_offer_answer.cc

@@ -4091,7 +4091,7 @@ void SdpOfferAnswerHandler::UpdateLocalSenders(
 
   // Find new and active senders.
   for (const cricket::StreamParams& params : streams) {
-    // The sync_label is the MediaStream label and the |stream.id| is the
+    // The sync_label is the MediaStream label and the `stream.id` is the
     // sender id.
     const std::string& stream_id = params.first_stream_id();
     const std::string& sender_id = params.id;
@@ -4154,8 +4154,8 @@ void SdpOfferAnswerHandler::UpdateRemoteSendersList(
       break;
     }
 
-    // |params.id| is the sender id and the stream id uses the first of
-    // |params.stream_ids|. The remote description could come from a Unified
+    // `params.id` is the sender id and the stream id uses the first of
+    // `params.stream_ids`. The remote description could come from a Unified
     // Plan endpoint, with multiple or no stream_ids() signaled. Since this is
     // not supported in Plan B, we just take the first here and create the
     // default stream ID if none is specified.

pc/test/fake_rtc_certificate_generator.h

@@ -83,9 +83,9 @@ static const rtc::RTCCertificatePEM kRsaPems[] = {
 
 // ECDSA with EC_NIST_P256.
 // These PEM strings were created by generating an identity with
-// |SSLIdentity::Create| and invoking |identity->PrivateKeyToPEMString()|,
-// |identity->PublicKeyToPEMString()| and
-// |identity->certificate().ToPEMString()|.
+// `SSLIdentity::Create` and invoking `identity->PrivateKeyToPEMString()`,
+// `identity->PublicKeyToPEMString()` and
+// `identity->certificate().ToPEMString()`.
 static const rtc::RTCCertificatePEM kEcdsaPems[] = {
     rtc::RTCCertificatePEM(
         "-----BEGIN PRIVATE KEY-----\n"

pc/webrtc_session_description_factory.h

@@ -41,7 +41,7 @@ namespace webrtc {
 class WebRtcCertificateGeneratorCallback
     : public rtc::RTCCertificateGeneratorCallback {
  public:
-  // |rtc::RTCCertificateGeneratorCallback| overrides.
+  // `rtc::RTCCertificateGeneratorCallback` overrides.
   void OnSuccess(
       const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) override;
   void OnFailure() override;

rtc_base/containers/void_t.h

@@ -25,7 +25,7 @@ struct make_void {
 
 // webrtc::void_t is an implementation of std::void_t from C++17.
 //
-// We use |webrtc::void_t_internal::make_void| as a helper struct to avoid a
+// We use `webrtc::void_t_internal::make_void` as a helper struct to avoid a
 // C++14 defect:
 //   http://en.cppreference.com/w/cpp/types/void_t
 //   http://open-std.org/JTC1/SC22/WG21/docs/cwg_defects.html#1558

rtc_base/rtc_certificate.h

@@ -31,7 +31,7 @@ class SSLIdentity;
 // certificate and acts as a text representation of RTCCertificate. Certificates
 // can be serialized and deserialized to and from this format, which allows for
 // cloning and storing of certificates to disk. The PEM format is that of
-// |SSLIdentity::PrivateKeyToPEMString| and |SSLCertificate::ToPEMString|, e.g.
+// `SSLIdentity::PrivateKeyToPEMString` and `SSLCertificate::ToPEMString`, e.g.
 // the string representations used by OpenSSL.
 class RTCCertificatePEM {
  public:

rtc_base/rtc_certificate_generator.cc

@@ -51,7 +51,7 @@ scoped_refptr<RTCCertificate> RTCCertificateGenerator::GenerateCertificate(
     expires_s = std::min(expires_s, kYearInSeconds);
     // TODO(torbjorng): Stop using `time_t`, its type is unspecified. It it safe
     // to assume it can hold up to a year's worth of seconds (and more), but
-    // |SSLIdentity::Create| should stop relying on `time_t`.
+    // `SSLIdentity::Create` should stop relying on `time_t`.
     // See bugs.webrtc.org/5720.
     time_t cert_lifetime_s = static_cast<time_t>(expires_s);
     identity = SSLIdentity::Create(kIdentityName, key_params, cert_lifetime_s);

rtc_base/rtc_certificate_generator.h

@@ -23,7 +23,7 @@
 
 namespace rtc {
 
-// See |RTCCertificateGeneratorInterface::GenerateCertificateAsync|.
+// See `RTCCertificateGeneratorInterface::GenerateCertificateAsync`.
 class RTCCertificateGeneratorCallback : public RefCountInterface {
  public:
   virtual void OnSuccess(const scoped_refptr<RTCCertificate>& certificate) = 0;

rtc_base/ssl_certificate.cc

@@ -49,15 +49,15 @@ SSLCertificateStats::~SSLCertificateStats() {}
 
 std::unique_ptr<SSLCertificateStats> SSLCertificate::GetStats() const {
   // TODO(bemasc): Move this computation to a helper class that caches these
-  // values to reduce CPU use in |StatsCollector::GetStats|. This will require
-  // adding a fast |SSLCertificate::Equals| to detect certificate changes.
+  // values to reduce CPU use in `StatsCollector::GetStats`. This will require
+  // adding a fast `SSLCertificate::Equals` to detect certificate changes.
   std::string digest_algorithm;
   if (!GetSignatureDigestAlgorithm(&digest_algorithm))
     return nullptr;
 
-  // |SSLFingerprint::Create| can fail if the algorithm returned by
-  // |SSLCertificate::GetSignatureDigestAlgorithm| is not supported by the
-  // implementation of |SSLCertificate::ComputeDigest|. This currently happens
+  // `SSLFingerprint::Create` can fail if the algorithm returned by
+  // `SSLCertificate::GetSignatureDigestAlgorithm` is not supported by the
+  // implementation of `SSLCertificate::ComputeDigest`. This currently happens
   // with MD5- and SHA-224-signed certificates when linked to libNSS.
   std::unique_ptr<SSLFingerprint> ssl_fingerprint =
       SSLFingerprint::Create(digest_algorithm, *this);

rtc_base/ssl_identity_unittest.cc

@@ -65,12 +65,12 @@ const unsigned char kTestCertSha512[] = {
     0x35, 0xce, 0x26, 0x58, 0x4a, 0x33, 0x6d, 0xbc, 0xb6};
 
 // These PEM strings were created by generating an identity with
-// |SSLIdentity::Create| and invoking |identity->PrivateKeyToPEMString()|,
-// |identity->PublicKeyToPEMString()| and
-// |identity->certificate().ToPEMString()|. If the crypto library is updated,
+// `SSLIdentity::Create` and invoking `identity->PrivateKeyToPEMString()`,
+// `identity->PublicKeyToPEMString()` and
+// `identity->certificate().ToPEMString()`. If the crypto library is updated,
 // and the update changes the string form of the keys, these will have to be
 // updated too.  The fingerprint, fingerprint algorithm and base64 certificate
-// were created by calling |identity->certificate().GetStats()|.
+// were created by calling `identity->certificate().GetStats()`.
 static const char kRSA_PRIVATE_KEY_PEM[] =
     "-----BEGIN PRIVATE KEY-----\n"
     "MIICdQIBADANBgkqhkiG9w0BAQEFAASCAl8wggJbAgEAAoGBAMQPqDStRlYeDpkX\n"

rtc_base/timestamp_aligner_unittest.cc

@@ -158,7 +158,7 @@ TEST(TimestampAlignerTest, ClipToMonotonous) {
 
   // Non-monotonic translated timestamps can happen when only for
   // translated timestamps in the future. Which is tolerated if
-  // |timestamp_aligner.clip_bias_us| is large enough. Instead of
+  // `timestamp_aligner.clip_bias_us` is large enough. Instead of
   // changing that private member for this test, just add the bias to
   // `kSystemTimeUs` when calling ClipTimestamp.
   const int64_t kClipBiasUs = 100000;

sdk/android/api/org/webrtc/DataChannel.java

@@ -82,7 +82,7 @@ public class DataChannel {
     /** The data channel state has changed. */
     @CalledByNative("Observer") public void onStateChange();
     /**
-     * A data buffer was successfully received.  NOTE: |buffer.data| will be
+     * A data buffer was successfully received.  NOTE: `buffer.data` will be
      * freed once this function returns so callers who want to use the data
      * asynchronously must make sure to copy it first.
      */

sdk/objc/api/peerconnection/RTCDataChannel.h

@@ -112,7 +112,7 @@ RTC_OBJC_EXPORT
 
 /**
  * The number of bytes of application data that have been queued using
- * |sendData:| but that have not yet been transmitted to the network.
+ * `sendData:` but that have not yet been transmitted to the network.
  */
 @property(nonatomic, readonly) uint64_t bufferedAmount;
 

sdk/objc/components/audio/RTCAudioSession.h

@@ -122,7 +122,7 @@ RTC_OBJC_EXPORT
  *  WebRTC and the application layer are avoided.
  *
  *  RTCAudioSession also coordinates activation so that the audio session is
- *  activated only once. See |setActive:error:|.
+ *  activated only once. See `setActive:error:`.
  */
 RTC_OBJC_EXPORT
 @interface RTC_OBJC_TYPE (RTCAudioSession) : NSObject <RTC_OBJC_TYPE(RTCAudioSessionActivationDelegate)>

stats/rtc_stats.cc

@@ -20,7 +20,7 @@ namespace webrtc {
 
 namespace {
 
-// Produces "[a,b,c]". Works for non-vector |RTCStatsMemberInterface::Type|
+// Produces "[a,b,c]". Works for non-vector `RTCStatsMemberInterface::Type`
 // types.
 template <typename T>
 std::string VectorToString(const std::vector<T>& vector) {

system_wrappers/include/ntp_time.h

@@ -65,7 +65,7 @@ inline bool operator!=(const NtpTime& n1, const NtpTime& n2) {
 // Converts `int64_t` milliseconds to Q32.32-formatted fixed-point seconds.
 // Performs clamping if the result overflows or underflows.
 inline int64_t Int64MsToQ32x32(int64_t milliseconds) {
-  // TODO(bugs.webrtc.org/10893): Change to use |rtc::saturated_cast| once the
+  // TODO(bugs.webrtc.org/10893): Change to use `rtc::saturated_cast` once the
   // bug has been fixed.
   double result =
       std::round(milliseconds * (NtpTime::kFractionsPerSecond / 1000.0));
@@ -88,7 +88,7 @@ inline int64_t Int64MsToQ32x32(int64_t milliseconds) {
 // Converts `int64_t` milliseconds to UQ32.32-formatted fixed-point seconds.
 // Performs clamping if the result overflows or underflows.
 inline uint64_t Int64MsToUQ32x32(int64_t milliseconds) {
-  // TODO(bugs.webrtc.org/10893): Change to use |rtc::saturated_cast| once the
+  // TODO(bugs.webrtc.org/10893): Change to use `rtc::saturated_cast` once the
   // bug has been fixed.
   double result =
       std::round(milliseconds * (NtpTime::kFractionsPerSecond / 1000.0));

test/ios/google_test_runner_delegate.h

@@ -17,7 +17,7 @@
 @protocol GoogleTestRunnerDelegate
 
 // Returns YES if this delegate supports running GoogleTests via a call to
-// |runGoogleTests|.
+// `runGoogleTests`.
 @property(nonatomic, readonly, assign) BOOL supportsRunningGoogleTests;
 
 // Runs GoogleTests and returns the final exit code.

test/ios/test_support.mm

@@ -72,8 +72,8 @@ static absl::optional<std::vector<std::string>> g_metrics_to_plot;
   [_window setRootViewController:[[UIViewController alloc] init]];
 
   if (!rtc::test::ShouldRunIOSUnittestsWithXCTest()) {
-    // When running in XCTest mode, XCTest will invoke |runGoogleTest| directly.
-    // Otherwise, schedule a call to |runTests|.
+    // When running in XCTest mode, XCTest will invoke `runGoogleTest` directly.
+    // Otherwise, schedule a call to `runTests`.
     [self performSelector:@selector(runTests) withObject:nil afterDelay:0.1];
   }
 

test/pc/e2e/analyzer/video/default_video_quality_analyzer.h

@@ -297,7 +297,7 @@ class DefaultVideoQualityAnalyzer : public VideoQualityAnalyzerInterface {
     absl::optional<VideoFrame> rendered;
     // If true frame was dropped somewhere from capturing to rendering and
     // wasn't rendered on remote peer side. If `dropped` is true, `rendered`
-    // will be |absl::nullopt|.
+    // will be `absl::nullopt`.
     bool dropped;
     FrameStats frame_stats;
     OverloadReason overload_reason;