diff --git a/api/ice_transport_factory.h b/api/ice_transport_factory.h index a9fd04ee7b..2268ea5e12 100644 --- a/api/ice_transport_factory.h +++ b/api/ice_transport_factory.h @@ -35,9 +35,9 @@ RTC_EXPORT rtc::scoped_refptr 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 CreateIceTransport( IceTransportInit); diff --git a/api/jsep.h b/api/jsep.h index 3348d7b239..d2aa57c784 100644 --- a/api/jsep.h +++ b/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. diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h index b9350ac9e1..a3c420f80b 100644 --- a/api/peer_connection_interface.h +++ b/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: diff --git a/api/rtp_packet_info.h b/api/rtp_packet_info.h index 13d3a3922a..bc9839f479 100644 --- a/api/rtp_packet_info.h +++ b/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 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_; }; diff --git a/api/rtp_packet_infos.h b/api/rtp_packet_infos.h index d63646414d..2ca3174037 100644 --- a/api/rtp_packet_infos.h +++ b/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| and have it -// |std::move()|-ed as the per-packet information is transferred from one object +// We should ideally just use `std::vector` 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 { diff --git a/api/set_local_description_observer_interface.h b/api/set_local_description_observer_interface.h index 90d000cd81..8e7b6258d3 100644 --- a/api/set_local_description_observer_interface.h +++ b/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; }; diff --git a/api/set_remote_description_observer_interface.h b/api/set_remote_description_observer_interface.h index 178255564a..d1c075309f 100644 --- a/api/set_remote_description_observer_interface.h +++ b/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; }; diff --git a/api/stats/rtc_stats.h b/api/stats/rtc_stats.h index 8ad39b4e23..a5fae52c29 100644 --- a/api/stats/rtc_stats.h +++ b/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|. The value of a member may be +// are implemented by `RTCStatsMember`. 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 class RTCStatsMember : public RTCStatsMemberInterface { public: diff --git a/api/stats/rtc_stats_report.h b/api/stats/rtc_stats_report.h index a26db86c77..2ced422370 100644 --- a/api/stats/rtc_stats_report.h +++ b/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 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; diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h index b18ef97a60..8a6327ed4c 100644 --- a/api/stats/rtcstats_objects.h +++ b/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 { diff --git a/api/stats_types.h b/api/stats_types.h index 6745d14836..9a03db3e40 100644 --- a/api/stats_types.h +++ b/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, diff --git a/api/task_queue/task_queue_base.h b/api/task_queue/task_queue_base.h index 88419edd8f..d8af6e67db 100644 --- a/api/task_queue/task_queue_base.h +++ b/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 diff --git a/api/video_codecs/video_encoder.h b/api/video_codecs/video_encoder.h index 2bdf8d015d..3035dd7209 100644 --- a/api/video_codecs/video_encoder.h +++ b/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; diff --git a/api/video_codecs/video_encoder_config.h b/api/video_codecs/video_encoder_config.h index 5440f1f435..cfda2ad7cf 100644 --- a/api/video_codecs/video_encoder_config.h +++ b/api/video_codecs/video_encoder_config.h @@ -129,7 +129,7 @@ class VideoEncoderConfig { // An implementation should return a std::vector 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 CreateEncoderStreams( int width, int height, diff --git a/api/video_codecs/vp8_frame_buffer_controller.h b/api/video_codecs/vp8_frame_buffer_controller.h index 852008f3e3..fc494f7293 100644 --- a/api/video_codecs/vp8_frame_buffer_controller.h +++ b/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; diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc index 3ca3b51bb1..b741a8c1ca 100644 --- a/audio/channel_receive.cc +++ b/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 local_capture_clock_offset; diff --git a/audio/utility/audio_frame_operations.h b/audio/utility/audio_frame_operations.h index 7e954dfde9..2a5f29f4f5 100644 --- a/audio/utility/audio_frame_operations.h +++ b/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. diff --git a/call/rtp_config.h b/call/rtp_config.h index ae5ae3b85f..c3b5b4a255 100644 --- a/call/rtp_config.h +++ b/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 rids; // The value to send in the MID RTP header extension if the extension is diff --git a/call/simulated_network.cc b/call/simulated_network.cc index f8a5bd893d..fc34fda914 100644 --- a/call/simulated_network.cc +++ b/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() < state.prob_loss_bursting) || (!bursting_ && random_.Rand() < state.prob_start_bursting)) { bursting_ = true; diff --git a/common_audio/vad/vad_core.c b/common_audio/vad/vad_core.c index d62d5ffb08..0872449a7c 100644 --- a/common_audio/vad/vad_core.c +++ b/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, diff --git a/common_audio/vad/vad_sp.h b/common_audio/vad/vad_sp.h index 37ee19f9f4..89138c57cf 100644 --- a/common_audio/vad/vad_sp.h +++ b/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: diff --git a/common_video/video_frame_buffer_pool.cc b/common_video/video_frame_buffer_pool.cc index d225370a4d..a450bd1e4b 100644 --- a/common_video/video_frame_buffer_pool.cc +++ b/common_video/video_frame_buffer_pool.cc @@ -20,7 +20,7 @@ namespace { bool HasOneRef(const rtc::scoped_refptr& buffer) { // Cast to rtc::RefCountedObject is safe because this function is only called // on locally created VideoFrameBuffers, which are either - // |rtc::RefCountedObject| or |rtc::RefCountedObject|. + // `rtc::RefCountedObject` or `rtc::RefCountedObject`. switch (buffer->type()) { case VideoFrameBuffer::Type::kI420: { return static_cast*>(buffer.get()) @@ -94,7 +94,7 @@ rtc::scoped_refptr 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| is + // in the same function below, where `RefCountedObject` is // created. rtc::RefCountedObject* raw_buffer = static_cast*>(existing_buffer.get()); @@ -125,7 +125,7 @@ rtc::scoped_refptr 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| is + // in the same function below, where `RefCountedObject` is // created. rtc::RefCountedObject* raw_buffer = static_cast*>(existing_buffer.get()); diff --git a/docs/native-code/rtp-hdrext/transport-wide-cc-02/README.md b/docs/native-code/rtp-hdrext/transport-wide-cc-02/README.md index 20b1d51dd2..8dc82612c7 100644 --- a/docs/native-code/rtp-hdrext/transport-wide-cc-02/README.md +++ b/docs/native-code/rtp-hdrext/transport-wide-cc-02/README.md @@ -29,19 +29,19 @@ Contact or 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.| +-+-+-+-+-+-+-+-+ diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc index 4b9be76d22..236aea7754 100644 --- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc +++ b/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); diff --git a/media/base/delayable.h b/media/base/delayable.h index 90ce5d7089..f0344c5dec 100644 --- a/media/base/delayable.h +++ b/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() {} diff --git a/media/base/video_adapter.h b/media/base/video_adapter.h index 76fefabf81..0493323a83 100644 --- a/media/base/video_adapter.h +++ b/media/base/video_adapter.h @@ -86,24 +86,24 @@ class RTC_EXPORT VideoAdapter { const absl::optional& 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::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::max()| if no limit is set. + // Can return `numeric_limits::max()` if no limit is set. int GetTargetPixels() const; // Returns current frame-rate limit. - // Can return |numeric_limits::infinity()| if no limit is set. + // Can return `numeric_limits::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. diff --git a/media/engine/simulcast_encoder_adapter_unittest.cc b/media/engine/simulcast_encoder_adapter_unittest.cc index 5a2bf8e7e3..5f3e54f3aa 100644 --- a/media/engine/simulcast_encoder_adapter_unittest.cc +++ b/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(); diff --git a/media/engine/webrtc_media_engine.cc b/media/engine/webrtc_media_engine.cc index 7ac666ec9e..6ce52e4c8e 100644 --- a/media/engine/webrtc_media_engine.cc +++ b/media/engine/webrtc_media_engine.cc @@ -27,7 +27,7 @@ namespace cricket { std::unique_ptr 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 fallback_trials( dependencies.trials ? nullptr : new webrtc::FieldTrialBasedConfig()); diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc index e9ffb21d05..cbc6abf506 100644 --- a/media/engine/webrtc_voice_engine.cc +++ b/media/engine/webrtc_voice_engine.cc @@ -152,8 +152,8 @@ absl::optional 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. diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc index 4b2742c8d1..1fd2480e85 100644 --- a/media/engine/webrtc_voice_engine_unittest.cc +++ b/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. diff --git a/media/sctp/sctp_transport_internal.h b/media/sctp/sctp_transport_internal.h index e44efb507b..93a59b9dc7 100644 --- a/media/sctp/sctp_transport_internal.h +++ b/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, diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h index 9963603713..18b662aed0 100644 --- a/modules/audio_coding/acm2/acm_receiver.h +++ b/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); diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc index d6291395a3..8ba1b9f3de 100644 --- a/modules/audio_coding/acm2/audio_coding_module.cc +++ b/modules/audio_coding/acm2/audio_coding_module.cc @@ -229,7 +229,7 @@ int32_t AudioCodingModuleImpl::Encode( const InputData& input_data, absl::optional 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()); diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc index 76f52ade80..3155f198a4 100644 --- a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc +++ b/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); } diff --git a/modules/audio_coding/audio_network_adaptor/config.proto b/modules/audio_coding/audio_network_adaptor/config.proto index 4f8b2c7e19..63b220ddf0 100644 --- a/modules/audio_coding/audio_network_adaptor/config.proto +++ b/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 { diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc index 355431ac1c..743b087163 100644 --- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc +++ b/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, diff --git a/modules/audio_coding/codecs/cng/webrtc_cng.cc b/modules/audio_coding/codecs/cng/webrtc_cng.cc index bfe77c74df..48f1b8c296 100644 --- a/modules/audio_coding/codecs/cng/webrtc_cng.cc +++ b/modules/audio_coding/codecs/cng/webrtc_cng.cc @@ -195,7 +195,7 @@ bool ComfortNoiseDecoder::Generate(rtc::ArrayView 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, diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_filter.c b/modules/audio_coding/codecs/isac/main/source/pitch_filter.c index 899d8423c4..bf03dfff2e 100644 --- a/modules/audio_coding/codecs/isac/main/source/pitch_filter.c +++ b/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. */ diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/modules/audio_coding/codecs/opus/audio_encoder_opus.h index ab954feba7..c7ee4f4523 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_opus.h +++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.h @@ -139,7 +139,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder { absl::optional link_capacity_allocation); // TODO(minyue): remove "override" when we can deprecate - // |AudioEncoder::SetTargetBitrate|. + // `AudioEncoder::SetTargetBitrate`. void SetTargetBitrate(int target_bps) override; void ApplyAudioNetworkAdaptor(); diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc index b507a32706..b40d73805f 100644 --- a/modules/audio_coding/codecs/opus/opus_unittest.cc +++ b/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); diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc index 875e62c92b..b0fee47ade 100644 --- a/modules/audio_coding/neteq/neteq_impl_unittest.cc +++ b/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]; diff --git a/modules/audio_device/fine_audio_buffer.h b/modules/audio_device/fine_audio_buffer.h index 99f282c1c1..a6c3042bb2 100644 --- a/modules/audio_device/fine_audio_buffer.h +++ b/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 diff --git a/modules/audio_device/win/core_audio_base_win.cc b/modules/audio_device/win/core_audio_base_win.cc index 12c51463a3..c42c091ed2 100644 --- a/modules/audio_device/win/core_audio_base_win.cc +++ b/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; diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc index 4000e33dc5..e8c912518d 100644 --- a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc +++ b/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); diff --git a/modules/audio_processing/gain_controller2_unittest.cc b/modules/audio_processing/gain_controller2_unittest.cc index b1ab00e937..8f65a89cde 100644 --- a/modules/audio_processing/gain_controller2_unittest.cc +++ b/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, diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h index 047776b2a0..6761ed4a45 100644 --- a/modules/audio_processing/include/audio_processing.h +++ b/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 diff --git a/modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py b/modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py index 0affbed162..fe3a6c7cb9 100644 --- a/modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py +++ b/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) diff --git a/modules/desktop_capture/desktop_region.cc b/modules/desktop_capture/desktop_region.cc index d4e71798c4..2c87c11eb3 100644 --- a/modules/desktop_capture/desktop_region.cc +++ b/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); diff --git a/modules/desktop_capture/win/wgc_capture_session.cc b/modules/desktop_capture/win/wgc_capture_session.cc index 5caaaeab28..22dbf90204 100644 --- a/modules/desktop_capture/win/wgc_capture_session.cc +++ b/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(map_info.pData); std::vector image_data; diff --git a/modules/pacing/round_robin_packet_queue.cc b/modules/pacing/round_robin_packet_queue.cc index 1feb5a96c8..ef37e5256b 100644 --- a/modules/pacing/round_robin_packet_queue.cc +++ b/modules/pacing/round_robin_packet_queue.cc @@ -175,7 +175,7 @@ std::unique_ptr 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 = diff --git a/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h b/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h index a59e2b4469..f5ec820dd5 100644 --- a/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h +++ b/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h @@ -45,7 +45,7 @@ class AbsoluteCaptureTimeInterpolator { rtc::ArrayView 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 OnReceivePacket( uint32_t source, uint32_t rtp_timestamp, diff --git a/modules/rtp_rtcp/source/absolute_capture_time_sender.h b/modules/rtp_rtcp/source/absolute_capture_time_sender.h index 3deff3d67d..be5a77d5e1 100644 --- a/modules/rtp_rtcp/source/absolute_capture_time_sender.h +++ b/modules/rtp_rtcp/source/absolute_capture_time_sender.h @@ -50,7 +50,7 @@ class AbsoluteCaptureTimeSender { static uint32_t GetSource(uint32_t ssrc, rtc::ArrayView 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 OnSendPacket( uint32_t source, diff --git a/modules/rtp_rtcp/source/fec_test_helper.h b/modules/rtp_rtcp/source/fec_test_helper.h index 7a24ecf39f..92e09fd44f 100644 --- a/modules/rtp_rtcp/source/fec_test_helper.h +++ b/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); }; diff --git a/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc index 3f99b038f6..abaa0786a3 100644 --- a/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc +++ b/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. diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc index 762255c303..32f442a7f7 100644 --- a/modules/rtp_rtcp/source/rtcp_receiver.cc +++ b/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; diff --git a/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc index e61ae648c3..585d6980e6 100644 --- a/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc +++ b/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)); diff --git a/modules/rtp_rtcp/source/rtcp_transceiver.h b/modules/rtp_rtcp/source/rtcp_transceiver.h index 862d4be815..20fda94a85 100644 --- a/modules/rtp_rtcp/source/rtcp_transceiver.h +++ b/modules/rtp_rtcp/source/rtcp_transceiver.h @@ -47,7 +47,7 @@ class RtcpTransceiver : public RtcpFeedbackSenderInterface { void Stop(std::function 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. diff --git a/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc b/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc index 5753ffd692..0f29b4dcd0 100644 --- a/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc +++ b/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc @@ -431,7 +431,7 @@ std::vector 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 report_blocks = config_.receive_statistics->RtcpReportBlocks( rtcp::ReceiverReport::kMaxNumberOfReportBlocks); diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc index ccc72a6948..707973f2a6 100644 --- a/modules/rtp_rtcp/source/rtp_sender.cc +++ b/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; } diff --git a/modules/rtp_rtcp/source/ulpfec_generator.cc b/modules/rtp_rtcp/source/ulpfec_generator.cc index 2d585d7516..20402fc4d3 100644 --- a/modules/rtp_rtcp/source/ulpfec_generator.cc +++ b/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. diff --git a/modules/rtp_rtcp/source/ulpfec_generator.h b/modules/rtp_rtcp/source/ulpfec_generator.h index c992458169..88a8b459e6 100644 --- a/modules/rtp_rtcp/source/ulpfec_generator.h +++ b/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; diff --git a/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/modules/video_coding/codecs/h264/h264_decoder_impl.cc index e6969872fe..11d36b7d19 100644 --- a/modules/video_coding/codecs/h264/h264_decoder_impl.cc +++ b/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; diff --git a/modules/video_coding/codecs/h264/h264_decoder_impl.h b/modules/video_coding/codecs/h264/h264_decoder_impl.h index 6ba4eb77bb..2c90a40cb1 100644 --- a/modules/video_coding/codecs/h264/h264_decoder_impl.h +++ b/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; diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl.cc b/modules/video_coding/codecs/h264/h264_encoder_impl.cc index 4e78d1ea99..887aa58098 100644 --- a/modules/video_coding/codecs/h264/h264_encoder_impl.cc +++ b/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]); diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl.h b/modules/video_coding/codecs/h264/h264_encoder_impl.h index b96de10774..1163464421 100644 --- a/modules/video_coding/codecs/h264/h264_encoder_impl.h +++ b/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 diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc index 712a83354c..1f70569526 100644 --- a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc +++ b/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); diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc b/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc index 8d8cb95daf..0f8ade33e9 100644 --- a/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc +++ b/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); diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h index 7c87d58259..826e8d667f 100644 --- a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h +++ b/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 performance_flags_by_spatial_index_; void UpdatePerformanceFlags(); diff --git a/modules/video_coding/session_info.h b/modules/video_coding/session_info.h index dc27a64352..846352a8ae 100644 --- a/modules/video_coding/session_info.h +++ b/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. diff --git a/p2p/base/p2p_transport_channel.h b/p2p/base/p2p_transport_channel.h index 025cac2d24..28248e7948 100644 --- a/p2p/base/p2p_transport_channel.h +++ b/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: diff --git a/p2p/base/transport_description_factory_unittest.cc b/p2p/base/transport_description_factory_unittest.cc index 08efe1263a..01120a89e8 100644 --- a/p2p/base/transport_description_factory_unittest.cc +++ b/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); diff --git a/pc/dtmf_sender.h b/pc/dtmf_sender.h index 5f200545e9..a208b100d4 100644 --- a/pc/dtmf_sender.h +++ b/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; diff --git a/pc/ice_server_parsing.cc b/pc/ice_server_parsing.cc index c1c855758a..a38e28cd66 100644 --- a/pc/ice_server_parsing.cc +++ b/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, diff --git a/pc/jsep_session_description.cc b/pc/jsep_session_description.cc index 4c1a4e7571..57ccf7ca6e 100644 --- a/pc/jsep_session_description.cc +++ b/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); diff --git a/pc/jsep_transport.h b/pc/jsep_transport.h index 5593122958..e3e929bfd2 100644 --- a/pc/jsep_transport.h +++ b/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 rtcp_mux_active_callback_; diff --git a/pc/media_session.cc b/pc/media_session.cc index b66d7f6c60..4bbb87782a 100644 --- a/pc/media_session.cc +++ b/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( diff --git a/pc/media_session_unittest.cc b/pc/media_session_unittest.cc index fa08f40784..a02b4c1415 100644 --- a/pc/media_session_unittest.cc +++ b/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], diff --git a/pc/peer_connection_ice_unittest.cc b/pc/peer_connection_ice_unittest.cc index 8726afb3b0..a27d174c98 100644 --- a/pc/peer_connection_ice_unittest.cc +++ b/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()); diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc index 0ab0e4d5f0..31652ac4b4 100644 --- a/pc/peer_connection_integrationtest.cc +++ b/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, diff --git a/pc/peer_connection_rtp_unittest.cc b/pc/peer_connection_rtp_unittest.cc index 2822854a2d..715546b2ea 100644 --- a/pc/peer_connection_rtp_unittest.cc +++ b/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()); diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc index 3fc8b8e8c1..8c4f0a6623 100644 --- a/pc/rtc_stats_collector_unittest.cc +++ b/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(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); } diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc index 9883945734..d4286371be 100644 --- a/pc/rtp_sender.cc +++ b/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(RTC_FROM_HERE, [&] { diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc index 22bf377d7c..c167f35f9a 100644 --- a/pc/sdp_offer_answer.cc +++ b/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. diff --git a/pc/test/fake_rtc_certificate_generator.h b/pc/test/fake_rtc_certificate_generator.h index b591c4c4ab..fc931ad661 100644 --- a/pc/test/fake_rtc_certificate_generator.h +++ b/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" diff --git a/pc/webrtc_session_description_factory.h b/pc/webrtc_session_description_factory.h index d0b3ad75ec..8e80fb556d 100644 --- a/pc/webrtc_session_description_factory.h +++ b/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& certificate) override; void OnFailure() override; diff --git a/rtc_base/containers/void_t.h b/rtc_base/containers/void_t.h index 62c57d4bec..149fc70c11 100644 --- a/rtc_base/containers/void_t.h +++ b/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 diff --git a/rtc_base/rtc_certificate.h b/rtc_base/rtc_certificate.h index 882c735036..0102c4f98c 100644 --- a/rtc_base/rtc_certificate.h +++ b/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: diff --git a/rtc_base/rtc_certificate_generator.cc b/rtc_base/rtc_certificate_generator.cc index 3a597814ae..16ff23c740 100644 --- a/rtc_base/rtc_certificate_generator.cc +++ b/rtc_base/rtc_certificate_generator.cc @@ -51,7 +51,7 @@ scoped_refptr 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(expires_s); identity = SSLIdentity::Create(kIdentityName, key_params, cert_lifetime_s); diff --git a/rtc_base/rtc_certificate_generator.h b/rtc_base/rtc_certificate_generator.h index ee68b27969..065b8b5002 100644 --- a/rtc_base/rtc_certificate_generator.h +++ b/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& certificate) = 0; diff --git a/rtc_base/ssl_certificate.cc b/rtc_base/ssl_certificate.cc index 3f12c79215..ed42998353 100644 --- a/rtc_base/ssl_certificate.cc +++ b/rtc_base/ssl_certificate.cc @@ -49,15 +49,15 @@ SSLCertificateStats::~SSLCertificateStats() {} std::unique_ptr 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 ssl_fingerprint = SSLFingerprint::Create(digest_algorithm, *this); diff --git a/rtc_base/ssl_identity_unittest.cc b/rtc_base/ssl_identity_unittest.cc index a8be7963b5..53f4a2ad10 100644 --- a/rtc_base/ssl_identity_unittest.cc +++ b/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" diff --git a/rtc_base/timestamp_aligner_unittest.cc b/rtc_base/timestamp_aligner_unittest.cc index 0a050ffd7f..ca91b62625 100644 --- a/rtc_base/timestamp_aligner_unittest.cc +++ b/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; diff --git a/sdk/android/api/org/webrtc/DataChannel.java b/sdk/android/api/org/webrtc/DataChannel.java index 804915d762..b9301f1faa 100644 --- a/sdk/android/api/org/webrtc/DataChannel.java +++ b/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. */ diff --git a/sdk/objc/api/peerconnection/RTCDataChannel.h b/sdk/objc/api/peerconnection/RTCDataChannel.h index 6f4ef37f8f..89eb58bc3f 100644 --- a/sdk/objc/api/peerconnection/RTCDataChannel.h +++ b/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; diff --git a/sdk/objc/components/audio/RTCAudioSession.h b/sdk/objc/components/audio/RTCAudioSession.h index 59250fee55..58811553b3 100644 --- a/sdk/objc/components/audio/RTCAudioSession.h +++ b/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 diff --git a/stats/rtc_stats.cc b/stats/rtc_stats.cc index 4895edc738..e6eb51e55c 100644 --- a/stats/rtc_stats.cc +++ b/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 std::string VectorToString(const std::vector& vector) { diff --git a/system_wrappers/include/ntp_time.h b/system_wrappers/include/ntp_time.h index cb58018bfa..b912bc8a0c 100644 --- a/system_wrappers/include/ntp_time.h +++ b/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)); diff --git a/test/ios/google_test_runner_delegate.h b/test/ios/google_test_runner_delegate.h index f0bcfe98d1..bb3493aea6 100644 --- a/test/ios/google_test_runner_delegate.h +++ b/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. diff --git a/test/ios/test_support.mm b/test/ios/test_support.mm index 3da896fa08..24cbcc7939 100644 --- a/test/ios/test_support.mm +++ b/test/ios/test_support.mm @@ -72,8 +72,8 @@ static absl::optional> 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]; } diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h index f5f3920a61..1461dd7375 100644 --- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h +++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h @@ -297,7 +297,7 @@ class DefaultVideoQualityAnalyzer : public VideoQualityAnalyzerInterface { absl::optional 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; diff --git a/test/pc/e2e/peer_connection_quality_test.cc b/test/pc/e2e/peer_connection_quality_test.cc index 72af279663..53e6220b7b 100644 --- a/test/pc/e2e/peer_connection_quality_test.cc +++ b/test/pc/e2e/peer_connection_quality_test.cc @@ -445,12 +445,12 @@ void PeerConnectionE2EQualityTest::SetupCallOnSignalingThread( RtpTransceiverInit transceiver_params; if (video_config.simulcast_config) { transceiver_params.direction = RtpTransceiverDirection::kSendOnly; - // Because simulcast enabled |alice_->params()->video_codecs| has only 1 + // Because simulcast enabled `alice_->params()->video_codecs` has only 1 // element. if (alice_->params()->video_codecs[0].name == cricket::kVp8CodecName) { // For Vp8 simulcast we need to add as many RtpEncodingParameters to the // track as many simulcast streams requested. If they specified in - // |video_config.simulcast_config| it should be copied from there. + // `video_config.simulcast_config` it should be copied from there. for (int i = 0; i < video_config.simulcast_config->simulcast_streams_count; ++i) { RtpEncodingParameters enc_params; diff --git a/test/pc/e2e/sdp/sdp_changer.cc b/test/pc/e2e/sdp/sdp_changer.cc index b3e5fc25a9..b3ee3b78bb 100644 --- a/test/pc/e2e/sdp/sdp_changer.cc +++ b/test/pc/e2e/sdp/sdp_changer.cc @@ -185,7 +185,7 @@ LocalAndRemoteSdp SignalingInterceptor::PatchOffer( } if (!params_.stream_label_to_simulcast_streams_count.empty()) { - // Because simulcast enabled |params_.video_codecs| has only 1 element. + // Because simulcast enabled `params_.video_codecs` has only 1 element. if (first_codec.name == cricket::kVp8CodecName) { return PatchVp8Offer(std::move(offer)); } @@ -378,7 +378,7 @@ LocalAndRemoteSdp SignalingInterceptor::PatchAnswer( } if (!params_.stream_label_to_simulcast_streams_count.empty()) { - // Because simulcast enabled |params_.video_codecs| has only 1 element. + // Because simulcast enabled `params_.video_codecs` has only 1 element. if (first_codec.name == cricket::kVp8CodecName) { return PatchVp8Answer(std::move(answer)); } diff --git a/test/pc/e2e/test_peer_factory.cc b/test/pc/e2e/test_peer_factory.cc index 5ba9b44141..5683de4072 100644 --- a/test/pc/e2e/test_peer_factory.cc +++ b/test/pc/e2e/test_peer_factory.cc @@ -73,7 +73,7 @@ void SetMandatoryEntities(InjectableComponents* components, // Returns mapping from stream label to optional spatial index. // If we have stream label "Foo" and mapping contains -// 1. |absl::nullopt| means "Foo" isn't simulcast/SVC stream +// 1. `absl::nullopt` means "Foo" isn't simulcast/SVC stream // 2. `kAnalyzeAnySpatialStream` means all simulcast/SVC streams are required // 3. Concrete value means that particular simulcast/SVC stream have to be // analyzed. diff --git a/video/alignment_adjuster.h b/video/alignment_adjuster.h index 4c4e15518d..ea2a9a0bef 100644 --- a/video/alignment_adjuster.h +++ b/video/alignment_adjuster.h @@ -19,10 +19,10 @@ namespace webrtc { class AlignmentAdjuster { public: // Returns the resolution alignment requested by the encoder (i.e - // |EncoderInfo::requested_resolution_alignment| which ensures that delivered + // `EncoderInfo::requested_resolution_alignment` which ensures that delivered // frames to the encoder are divisible by this alignment). // - // If |EncoderInfo::apply_alignment_to_all_simulcast_layers| is enabled, the + // If `EncoderInfo::apply_alignment_to_all_simulcast_layers` is enabled, the // alignment will be adjusted to ensure that each simulcast layer also is // divisible by `requested_resolution_alignment`. The configured scale factors // `scale_resolution_down_by` may be adjusted to a common multiple to limit diff --git a/video/frame_encode_metadata_writer_unittest.cc b/video/frame_encode_metadata_writer_unittest.cc index 631dded2bc..8b60a8c4c7 100644 --- a/video/frame_encode_metadata_writer_unittest.cc +++ b/video/frame_encode_metadata_writer_unittest.cc @@ -64,7 +64,7 @@ bool IsTimingFrame(const EncodedImage& image) { // Emulates `num_frames` on `num_streams` frames with capture timestamps // increased by 1 from 0. Size of each frame is between // `min_frame_size` and `max_frame_size`, outliers are counted relatevely to -// |average_frame_sizes[]| for each stream. +// `average_frame_sizes[]` for each stream. std::vector> GetTimingFrames( const int64_t delay_ms, const size_t min_frame_size, diff --git a/video/receive_statistics_proxy.h b/video/receive_statistics_proxy.h index 4efc0f60ee..1e5189dc2d 100644 --- a/video/receive_statistics_proxy.h +++ b/video/receive_statistics_proxy.h @@ -158,7 +158,7 @@ class ReceiveStatisticsProxy : public VCMReceiveStatisticsCallback, rtc::SampleCounter qp_sample_ RTC_GUARDED_BY(mutex_); int num_bad_states_ RTC_GUARDED_BY(mutex_); int num_certain_states_ RTC_GUARDED_BY(mutex_); - // Note: The |stats_.rtp_stats| member is not used or populated by this class. + // Note: The `stats_.rtp_stats` member is not used or populated by this class. mutable VideoReceiveStream::Stats stats_ RTC_GUARDED_BY(mutex_); RateStatistics decode_fps_estimator_ RTC_GUARDED_BY(mutex_); RateStatistics renders_fps_estimator_ RTC_GUARDED_BY(mutex_); diff --git a/video/receive_statistics_proxy2.h b/video/receive_statistics_proxy2.h index 7797d93217..2dda15e240 100644 --- a/video/receive_statistics_proxy2.h +++ b/video/receive_statistics_proxy2.h @@ -163,7 +163,7 @@ class ReceiveStatisticsProxy : public VCMReceiveStatisticsCallback, rtc::SampleCounter qp_sample_ RTC_GUARDED_BY(main_thread_); int num_bad_states_ RTC_GUARDED_BY(main_thread_); int num_certain_states_ RTC_GUARDED_BY(main_thread_); - // Note: The |stats_.rtp_stats| member is not used or populated by this class. + // Note: The `stats_.rtp_stats` member is not used or populated by this class. mutable VideoReceiveStream::Stats stats_ RTC_GUARDED_BY(main_thread_); // Same as stats_.ssrc, but const (no lock required). const uint32_t remote_ssrc_; diff --git a/video/send_statistics_proxy.cc b/video/send_statistics_proxy.cc index efdf03fa14..a50008c9be 100644 --- a/video/send_statistics_proxy.cc +++ b/video/send_statistics_proxy.cc @@ -951,7 +951,7 @@ void SendStatisticsProxy::OnSendEncodedImage( encode_frame_rate = 1.0; double target_frame_size_bytes = stats_.target_media_bitrate_bps / (8.0 * encode_frame_rate); - // |stats_.target_media_bitrate_bps| is set in + // `stats_.target_media_bitrate_bps` is set in // SendStatisticsProxy::OnSetEncoderTargetRate. stats_.total_encoded_bytes_target += round(target_frame_size_bytes); if (codec_info) { @@ -1196,7 +1196,7 @@ void SendStatisticsProxy::UpdateAdaptationStats() { stats_.quality_limitation_reason = quality_limitation_reason_tracker_.current_reason(); - // |stats_.quality_limitation_durations_ms| depends on the current time + // `stats_.quality_limitation_durations_ms` depends on the current time // when it is polled; it is updated in SendStatisticsProxy::GetStats(). } diff --git a/video/video_analyzer.h b/video/video_analyzer.h index bb08fbc5cd..c121370043 100644 --- a/video/video_analyzer.h +++ b/video/video_analyzer.h @@ -157,12 +157,12 @@ class VideoAnalyzer : public PacketReceiver, void OnFrame(const VideoFrame& video_frame) RTC_LOCKS_EXCLUDED(lock_) override; - // Called when |send_stream_.SetSource()| is called. + // Called when `send_stream_.SetSource()` is called. void AddOrUpdateSink(rtc::VideoSinkInterface* sink, const rtc::VideoSinkWants& wants) RTC_LOCKS_EXCLUDED(lock_) override; - // Called by `send_stream_` when |send_stream_.SetSource()| is called. + // Called by `send_stream_` when `send_stream_.SetSource()` is called. void RemoveSink(rtc::VideoSinkInterface* sink) RTC_LOCKS_EXCLUDED(lock_) override; diff --git a/video/video_receive_stream.cc b/video/video_receive_stream.cc index da5701c20b..2ddaa5bfbc 100644 --- a/video/video_receive_stream.cc +++ b/video/video_receive_stream.cc @@ -508,7 +508,7 @@ void VideoReceiveStream::OnFrame(const VideoFrame& video_frame) { double estimated_freq_khz; // TODO(bugs.webrtc.org/10739): we should set local capture clock offset for - // |video_frame.packet_infos|. But VideoFrame is const qualified here. + // `video_frame.packet_infos`. But VideoFrame is const qualified here. // TODO(tommi): GetStreamSyncOffsetInMs grabs three locks. One inside the // function itself, another in GetChannel() and a third in diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc index ce1eb7e395..ba3f4a2f71 100644 --- a/video/video_receive_stream2.cc +++ b/video/video_receive_stream2.cc @@ -563,7 +563,7 @@ void VideoReceiveStream2::OnFrame(const VideoFrame& video_frame) { VideoFrameMetaData frame_meta(video_frame, clock_->CurrentTime()); // TODO(bugs.webrtc.org/10739): we should set local capture clock offset for - // |video_frame.packet_infos|. But VideoFrame is const qualified here. + // `video_frame.packet_infos`. But VideoFrame is const qualified here. call_->worker_thread()->PostTask( ToQueuedTask(task_safety_, [frame_meta, this]() { diff --git a/video/video_send_stream_impl.h b/video/video_send_stream_impl.h index 5ee4d19a2e..a29f186af2 100644 --- a/video/video_send_stream_impl.h +++ b/video/video_send_stream_impl.h @@ -107,7 +107,7 @@ class VideoSendStreamImpl : public webrtc::BitrateAllocatorObserver, VideoLayersAllocation allocation) override; // Implements EncodedImageCallback. The implementation routes encoded frames - // to the `payload_router_` and |config.pre_encode_callback| if set. + // to the `payload_router_` and `config.pre_encode_callback` if set. // Called on an arbitrary encoder callback thread. EncodedImageCallback::Result OnEncodedImage( const EncodedImage& encoded_image, diff --git a/video/video_source_sink_controller.cc b/video/video_source_sink_controller.cc index 4cd12d8a27..6955e3b1e7 100644 --- a/video/video_source_sink_controller.cc +++ b/video/video_source_sink_controller.cc @@ -157,7 +157,7 @@ rtc::VideoSinkWants VideoSourceSinkController::CurrentSettingsToSinkWants() const { rtc::VideoSinkWants wants; wants.rotation_applied = rotation_applied_; - // |wants.black_frames| is not used, it always has its default value false. + // `wants.black_frames` is not used, it always has its default value false. wants.max_pixel_count = rtc::dchecked_cast(restrictions_.max_pixels_per_frame().value_or( std::numeric_limits::max())); diff --git a/video/video_stream_encoder.cc b/video/video_stream_encoder.cc index b56edbed45..be611fad63 100644 --- a/video/video_stream_encoder.cc +++ b/video/video_stream_encoder.cc @@ -1038,7 +1038,7 @@ void VideoStreamEncoder::ReconfigureEncoder() { // The resolutions that we're actually encoding with. std::vector encoder_resolutions; - // TODO(hbos): For the case of SVC, also make use of |codec.spatialLayers|. + // TODO(hbos): For the case of SVC, also make use of `codec.spatialLayers`. // For now, SVC layers are handled by the VP9 encoder. for (const auto& simulcastStream : codec.simulcastStream) { if (!simulcastStream.active) @@ -1344,7 +1344,7 @@ bool VideoStreamEncoder::EncoderPaused() const { // Pause video if paused by caller or as long as the network is down or the // pacer queue has grown too large in buffered mode. // If the pacer queue has grown too large or the network is down, - // |last_encoder_rate_settings_->encoder_target| will be 0. + // `last_encoder_rate_settings_->encoder_target` will be 0. return !last_encoder_rate_settings_ || last_encoder_rate_settings_->encoder_target == DataRate::Zero(); }