Use backticks not vertical bars to denote variables in comments for /media Bug: webrtc:12338 Change-Id: Ia800a4017ede1f647b36f809ef3c5b37a2616fdd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226949 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34567}
diff --git a/media/base/adapted_video_track_source.h b/media/base/adapted_video_track_source.h index 59ae036..d40baef 100644 --- a/media/base/adapted_video_track_source.h +++ b/media/base/adapted_video_track_source.h
@@ -38,7 +38,7 @@ ~AdaptedVideoTrackSource() override; protected: - // Allows derived classes to initialize |video_adapter_| with a custom + // Allows derived classes to initialize `video_adapter_` with a custom // alignment. explicit AdaptedVideoTrackSource(int required_alignment); // Checks the apply_rotation() flag. If the frame needs rotation, and it is a
diff --git a/media/base/codec.cc b/media/base/codec.cc index cb6913e..a116184 100644 --- a/media/base/codec.cc +++ b/media/base/codec.cc
@@ -81,7 +81,7 @@ return; } if (Has(param)) { - // Param already in |this|. + // Param already in `this`. return; } params_.push_back(param);
diff --git a/media/base/codec.h b/media/base/codec.h index c7c99bf..29c54a8 100644 --- a/media/base/codec.h +++ b/media/base/codec.h
@@ -78,7 +78,7 @@ bool Matches(const Codec& codec) const; bool MatchesCapability(const webrtc::RtpCodecCapability& capability) const; - // Find the parameter for |name| and write the value to |out|. + // Find the parameter for `name` and write the value to `out`. bool GetParam(const std::string& name, std::string* out) const; bool GetParam(const std::string& name, int* out) const; @@ -92,8 +92,8 @@ bool HasFeedbackParam(const FeedbackParam& param) const; void AddFeedbackParam(const FeedbackParam& param); - // Filter |this| feedbacks params such that only those shared by both |this| - // and |other| are kept. + // Filter `this` feedbacks params such that only those shared by both `this` + // and `other` are kept. void IntersectFeedbackParams(const Codec& other); virtual webrtc::RtpCodecParameters ToCodecParameters() const; @@ -176,7 +176,7 @@ bool operator!=(const VideoCodec& c) const { return !(*this == c); } - // Return packetization which both |local_codec| and |remote_codec| support. + // Return packetization which both `local_codec` and `remote_codec` support. static absl::optional<std::string> IntersectPacketization( const VideoCodec& local_codec, const VideoCodec& remote_codec); @@ -202,7 +202,7 @@ void SetDefaultParameters(); }; -// Get the codec setting associated with |payload_type|. If there +// Get the codec setting associated with `payload_type`. If there // is no codec associated with that payload type it returns nullptr. template <class Codec> const Codec* FindCodecById(const std::vector<Codec>& codecs, int payload_type) { @@ -218,7 +218,7 @@ bool HasRemb(const Codec& codec); bool HasRrtr(const Codec& codec); bool HasTransportCc(const Codec& codec); -// Returns the first codec in |supported_codecs| that matches |codec|, or +// Returns the first codec in `supported_codecs` that matches `codec`, or // nullptr if no codec matches. const VideoCodec* FindMatchingCodec( const std::vector<VideoCodec>& supported_codecs,
diff --git a/media/base/media_channel.cc b/media/base/media_channel.cc index 01b043b..11953c2 100644 --- a/media/base/media_channel.cc +++ b/media/base/media_channel.cc
@@ -116,7 +116,7 @@ } // This is the DSCP value used for both RTP and RTCP channels if DSCP is -// enabled. It can be changed at any time via |SetPreferredDscp|. +// enabled. It can be changed at any time via `SetPreferredDscp`. rtc::DiffServCodePoint MediaChannel::PreferredDscp() const { RTC_DCHECK_RUN_ON(network_thread_); return preferred_dscp_;
diff --git a/media/base/media_channel.h b/media/base/media_channel.h index c6bbc07..6467a44 100644 --- a/media/base/media_channel.h +++ b/media/base/media_channel.h
@@ -278,7 +278,7 @@ bool DscpEnabled() const; // This is the DSCP value used for both RTP and RTCP channels if DSCP is - // enabled. It can be changed at any time via |SetPreferredDscp|. + // enabled. It can be changed at any time via `SetPreferredDscp`. rtc::DiffServCodePoint PreferredDscp() const; void SetPreferredDscp(rtc::DiffServCodePoint new_dscp); @@ -655,7 +655,7 @@ int64_t bucket_delay = 0; }; -// Maps from payload type to |RtpCodecParameters|. +// Maps from payload type to `RtpCodecParameters`. typedef std::map<int, webrtc::RtpCodecParameters> RtpCodecParametersMap; struct VoiceMediaInfo { @@ -778,7 +778,7 @@ cricket::MediaType media_type() const override; virtual bool SetSendParameters(const AudioSendParameters& params) = 0; virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0; - // Get the receive parameters for the incoming stream identified by |ssrc|. + // Get the receive parameters for the incoming stream identified by `ssrc`. virtual webrtc::RtpParameters GetRtpReceiveParameters( uint32_t ssrc) const = 0; // Retrieve the receive parameters for the default receive @@ -799,9 +799,9 @@ virtual bool SetDefaultOutputVolume(double volume) = 0; // Returns if the telephone-event has been negotiated. virtual bool CanInsertDtmf() = 0; - // Send a DTMF |event|. The DTMF out-of-band signal will be used. - // The |ssrc| should be either 0 or a valid send stream ssrc. - // The valid value for the |event| are 0 to 15 which corresponding to + // Send a DTMF `event`. The DTMF out-of-band signal will be used. + // The `ssrc` should be either 0 or a valid send stream ssrc. + // The valid value for the `event` are 0 to 15 which corresponding to // DTMF event 0-9, *, #, A-D. virtual bool InsertDtmf(uint32_t ssrc, int event, int duration) = 0; // Gets quality stats for the channel. @@ -850,7 +850,7 @@ cricket::MediaType media_type() const override; virtual bool SetSendParameters(const VideoSendParameters& params) = 0; virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0; - // Get the receive parameters for the incoming stream identified by |ssrc|. + // Get the receive parameters for the incoming stream identified by `ssrc`. virtual webrtc::RtpParameters GetRtpReceiveParameters( uint32_t ssrc) const = 0; // Retrieve the receive parameters for the default receive @@ -861,7 +861,7 @@ // Starts or stops transmission (and potentially capture) of local video. virtual bool SetSend(bool send) = 0; // Configure stream for sending and register a source. - // The |ssrc| must correspond to a registered send stream. + // The `ssrc` must correspond to a registered send stream. virtual bool SetVideoSend( uint32_t ssrc, const VideoOptions* options, @@ -883,13 +883,13 @@ virtual void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) = 0; // Gets quality stats for the channel. virtual bool GetStats(VideoMediaInfo* info) = 0; - // Set recordable encoded frame callback for |ssrc| + // Set recordable encoded frame callback for `ssrc` virtual void SetRecordableEncodedFrameCallback( uint32_t ssrc, std::function<void(const webrtc::RecordableEncodedFrame&)> callback) = 0; - // Clear recordable encoded frame callback for |ssrc| + // Clear recordable encoded frame callback for `ssrc` virtual void ClearRecordableEncodedFrameCallback(uint32_t ssrc) = 0; - // Cause generation of a keyframe for |ssrc| + // Cause generation of a keyframe for `ssrc` virtual void GenerateKeyFrame(uint32_t ssrc) = 0; virtual std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const = 0;
diff --git a/media/base/media_constants.h b/media/base/media_constants.h index bf7f0c3..1f471e7 100644 --- a/media/base/media_constants.h +++ b/media/base/media_constants.h
@@ -67,7 +67,7 @@ extern const char kParamValueTrue[]; // Parameters are stored as parameter/value pairs. For parameters who do not -// have a value, |kParamValueEmpty| should be used as value. +// have a value, `kParamValueEmpty` should be used as value. extern const char kParamValueEmpty[]; // opus parameters.
diff --git a/media/base/rtp_utils.cc b/media/base/rtp_utils.cc index e796482..97a3c4c 100644 --- a/media/base/rtp_utils.cc +++ b/media/base/rtp_utils.cc
@@ -69,7 +69,7 @@ extension_data[2] = static_cast<uint8_t>(send_time); } -// Assumes |length| is actual packet length + tag length. Updates HMAC at end of +// Assumes `length` is actual packet length + tag length. Updates HMAC at end of // the RTP packet. void UpdateRtpAuthTag(uint8_t* rtp, size_t length, @@ -359,7 +359,7 @@ RTC_DCHECK(data); RTC_DCHECK(length); - // if there is no valid |rtp_sendtime_extension_id| and |srtp_auth_key| in + // if there is no valid `rtp_sendtime_extension_id` and `srtp_auth_key` in // PacketOptions, nothing to be updated in this packet. if (packet_time_params.rtp_sendtime_extension_id == -1 && packet_time_params.srtp_auth_key.empty()) {
diff --git a/media/base/rtp_utils.h b/media/base/rtp_utils.h index e10403c..a501fd7 100644 --- a/media/base/rtp_utils.h +++ b/media/base/rtp_utils.h
@@ -50,10 +50,10 @@ // True if |payload type| is 0-127. bool IsValidRtpPayloadType(int payload_type); -// True if |size| is appropriate for the indicated packet type. +// True if `size` is appropriate for the indicated packet type. bool IsValidRtpPacketSize(RtpPacketType packet_type, size_t size); -// Returns "RTCP", "RTP" or "Unknown" according to |packet_type|. +// Returns "RTCP", "RTP" or "Unknown" according to `packet_type`. absl::string_view RtpPacketTypeToString(RtpPacketType packet_type); // Verifies that a packet has a valid RTP header. @@ -67,7 +67,7 @@ int extension_id, uint64_t time_us); -// Applies specified |options| to the packet. It updates the absolute send time +// Applies specified `options` to the packet. It updates the absolute send time // extension header if it is present present then updates HMAC. bool RTC_EXPORT ApplyPacketOptions(uint8_t* data,
diff --git a/media/base/rtp_utils_unittest.cc b/media/base/rtp_utils_unittest.cc index 543babe..a594f94 100644 --- a/media/base/rtp_utils_unittest.cc +++ b/media/base/rtp_utils_unittest.cc
@@ -67,9 +67,9 @@ }; // Index of AbsSendTimeExtn data in message -// |kRtpMsgWithOneByteAbsSendTimeExtension|. +// `kRtpMsgWithOneByteAbsSendTimeExtension`. static const int kAstIndexInOneByteRtpMsg = 21; -// and in message |kRtpMsgWithTwoByteAbsSendTimeExtension|. +// and in message `kRtpMsgWithTwoByteAbsSendTimeExtension`. static const int kAstIndexInTwoByteRtpMsg = 21; static const rtc::ArrayView<const char> kPcmuFrameArrayView =
diff --git a/media/base/sdp_video_format_utils.h b/media/base/sdp_video_format_utils.h index 6671c18..80c1e4d 100644 --- a/media/base/sdp_video_format_utils.h +++ b/media/base/sdp_video_format_utils.h
@@ -17,18 +17,18 @@ namespace webrtc { // Generate codec parameters that will be used as answer in an SDP negotiation // based on local supported parameters and remote offered parameters. Both -// |local_supported_params|, |remote_offered_params|, and |answer_params| +// `local_supported_params`, `remote_offered_params`, and `answer_params` // represent sendrecv media descriptions, i.e they are a mix of both encode and -// decode capabilities. In theory, when the profile in |local_supported_params| -// represent a strict superset of the profile in |remote_offered_params|, we -// could limit the profile in |answer_params| to the profile in -// |remote_offered_params|. However, to simplify the code, each supported H264 +// decode capabilities. In theory, when the profile in `local_supported_params` +// represent a strict superset of the profile in `remote_offered_params`, we +// could limit the profile in `answer_params` to the profile in +// `remote_offered_params`. However, to simplify the code, each supported H264 // profile should be listed explicitly in the list of local supported codecs, // even if they are redundant. Then each local codec in the list should be // tested one at a time against the remote codec, and only when the profiles are // equal should this function be called. Therefore, this function does not need -// to handle profile intersection, and the profile of |local_supported_params| -// and |remote_offered_params| must be equal before calling this function. The +// to handle profile intersection, and the profile of `local_supported_params` +// and `remote_offered_params` must be equal before calling this function. The // parameters that are used when negotiating are the level part of // profile-level-id and level-asymmetry-allowed. void H264GenerateProfileLevelIdForAnswer(
diff --git a/media/base/test_utils.h b/media/base/test_utils.h index 46783a1..22bda4f 100644 --- a/media/base/test_utils.h +++ b/media/base/test_utils.h
@@ -35,7 +35,7 @@ } #define MAKE_VECTOR(a) cricket::MakeVector(a, arraysize(a)) -// Checks whether |codecs| contains |codec|; checks using Codec::Matches(). +// Checks whether `codecs` contains `codec`; checks using Codec::Matches(). template <class C> bool ContainsMatchingCodec(const std::vector<C>& codecs, const C& codec) { typename std::vector<C>::const_iterator it; @@ -47,11 +47,11 @@ return false; } -// Create Simulcast StreamParams with given |ssrcs| and |cname|. +// Create Simulcast StreamParams with given `ssrcs` and `cname`. cricket::StreamParams CreateSimStreamParams(const std::string& cname, const std::vector<uint32_t>& ssrcs); -// Create Simulcast stream with given |ssrcs| and |rtx_ssrcs|. -// The number of |rtx_ssrcs| must match number of |ssrcs|. +// Create Simulcast stream with given `ssrcs` and `rtx_ssrcs`. +// The number of `rtx_ssrcs` must match number of `ssrcs`. cricket::StreamParams CreateSimWithRtxStreamParams( const std::string& cname, const std::vector<uint32_t>& ssrcs,
diff --git a/media/base/video_adapter.cc b/media/base/video_adapter.cc index ddcf4ca..7a213a2 100644 --- a/media/base/video_adapter.cc +++ b/media/base/video_adapter.cc
@@ -36,14 +36,14 @@ } // Determines number of output pixels if both width and height of an input of - // |input_pixels| pixels is scaled with the fraction numerator / denominator. + // `input_pixels` pixels is scaled with the fraction numerator / denominator. int scale_pixel_count(int input_pixels) { return (numerator * numerator * input_pixels) / (denominator * denominator); } }; -// Round |value_to_round| to a multiple of |multiple|. Prefer rounding upwards, -// but never more than |max_value|. +// Round `value_to_round` to a multiple of `multiple`. Prefer rounding upwards, +// but never more than `max_value`. int roundUp(int value_to_round, int multiple, int max_value) { const int rounded_value = (value_to_round + multiple - 1) / multiple * multiple; @@ -51,8 +51,8 @@ : (max_value / multiple * multiple); } -// Generates a scale factor that makes |input_pixels| close to |target_pixels|, -// but no higher than |max_pixels|. +// Generates a scale factor that makes `input_pixels` close to `target_pixels`, +// but no higher than `max_pixels`. Fraction FindScale(int input_width, int input_height, int target_pixels, @@ -73,7 +73,7 @@ Fraction best_scale = Fraction{1, 1}; if (variable_start_scale_factor) { - // Start scaling down by 2/3 depending on |input_width| and |input_height|. + // Start scaling down by 2/3 depending on `input_width` and `input_height`. if (input_width % 3 == 0 && input_height % 3 == 0) { // 2/3 (then alternates 3/4, 2/3, 3/4,...). current_scale = Fraction{6, 6}; @@ -152,7 +152,7 @@ if (max_fps <= 0) return false; - // If |max_framerate_request_| is not set, it will default to maxint, which + // If `max_framerate_request_` is not set, it will default to maxint, which // will lead to a frame_interval_ns rounded to 0. int64_t frame_interval_ns = rtc::kNumNanosecsPerSec / max_fps; if (frame_interval_ns <= 0) { @@ -356,7 +356,7 @@ float VideoAdapter::GetMaxFramerate() const { webrtc::MutexLock lock(&mutex_); - // Minimum of |max_fps_| and |max_framerate_request_| is used to throttle + // Minimum of `max_fps_` and `max_framerate_request_` is used to throttle // frame-rate. int framerate = std::min(max_framerate_request_, max_fps_.value_or(max_framerate_request_));
diff --git a/media/base/video_adapter.h b/media/base/video_adapter.h index 3ed5895..76fefab 100644 --- a/media/base/video_adapter.h +++ b/media/base/video_adapter.h
@@ -33,7 +33,7 @@ public: VideoAdapter(); // The source requests output frames whose width and height are divisible - // by |source_resolution_alignment|. + // by `source_resolution_alignment`. explicit VideoAdapter(int source_resolution_alignment); virtual ~VideoAdapter(); @@ -52,7 +52,7 @@ // DEPRECATED. Please use OnOutputFormatRequest below. // TODO(asapersson): Remove this once it is no longer used. // Requests the output frame size and frame interval from - // |AdaptFrameResolution| to not be larger than |format|. Also, the input + // `AdaptFrameResolution` to not be larger than `format`. Also, the input // frame size will be cropped to match the requested aspect ratio. The // requested aspect ratio is orientation agnostic and will be adjusted to // maintain the input orientation, so it doesn't matter if e.g. 1280x720 or @@ -61,13 +61,13 @@ void OnOutputFormatRequest(const absl::optional<VideoFormat>& format) RTC_LOCKS_EXCLUDED(mutex_); - // Requests output frame size and frame interval from |AdaptFrameResolution|. - // |target_aspect_ratio|: The input frame size will be cropped to match the + // Requests output frame size and frame interval from `AdaptFrameResolution`. + // `target_aspect_ratio`: The input frame size will be cropped to match the // requested aspect ratio. The aspect ratio is orientation agnostic and will // be adjusted to maintain the input orientation (i.e. it doesn't matter if // e.g. <1280,720> or <720,1280> is requested). - // |max_pixel_count|: The maximum output frame size. - // |max_fps|: The maximum output framerate. + // `max_pixel_count`: The maximum output frame size. + // `max_fps`: The maximum output framerate. // Note: Should be called from the source only. void OnOutputFormatRequest( const absl::optional<std::pair<int, int>>& target_aspect_ratio, @@ -85,7 +85,7 @@ const absl::optional<int>& max_portrait_pixel_count, const absl::optional<int>& max_fps) RTC_LOCKS_EXCLUDED(mutex_); - // Requests the output frame size from |AdaptFrameResolution| to have as close + // 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 @@ -123,7 +123,7 @@ // The fixed source resolution alignment requirement. const int source_resolution_alignment_; // The currently applied resolution alignment, as given by the requirements: - // - the fixed |source_resolution_alignment_|; and + // - the fixed `source_resolution_alignment_`; and // - the latest |sink_wants.resolution_alignment|. int resolution_alignment_ RTC_GUARDED_BY(mutex_);
diff --git a/media/base/video_broadcaster.cc b/media/base/video_broadcaster.cc index 3c20eca..1b55786 100644 --- a/media/base/video_broadcaster.cc +++ b/media/base/video_broadcaster.cc
@@ -30,7 +30,7 @@ RTC_DCHECK(sink != nullptr); webrtc::MutexLock lock(&sinks_and_wants_lock_); if (!FindSinkPair(sink)) { - // |Sink| is a new sink, which didn't receive previous frame. + // `Sink` is a new sink, which didn't receive previous frame. previous_frame_sent_to_all_sinks_ = false; } VideoSourceBase::AddOrUpdateSink(sink, wants);
diff --git a/media/base/video_common.h b/media/base/video_common.h index e7ad22f..f27e008 100644 --- a/media/base/video_common.h +++ b/media/base/video_common.h
@@ -213,10 +213,10 @@ std::string ToString() const; }; -// Returns the largest positive integer that divides both |a| and |b|. +// Returns the largest positive integer that divides both `a` and `b`. int GreatestCommonDivisor(int a, int b); -// Returns the smallest positive integer that is divisible by both |a| and |b|. +// Returns the smallest positive integer that is divisible by both `a` and `b`. int LeastCommonMultiple(int a, int b); } // namespace cricket
diff --git a/media/engine/multiplex_codec_factory.h b/media/engine/multiplex_codec_factory.h index ea57149..a4272a2 100644 --- a/media/engine/multiplex_codec_factory.h +++ b/media/engine/multiplex_codec_factory.h
@@ -42,7 +42,7 @@ // - Select "multiplex" codec in SDP negotiation. class RTC_EXPORT MultiplexEncoderFactory : public VideoEncoderFactory { public: - // |supports_augmenting_data| defines if the encoder would support augmenting + // `supports_augmenting_data` defines if the encoder would support augmenting // data. If set, the encoder expects to receive video frame buffers of type // AugmentedVideoFrameBuffer. MultiplexEncoderFactory(std::unique_ptr<VideoEncoderFactory> factory, @@ -59,7 +59,7 @@ class RTC_EXPORT MultiplexDecoderFactory : public VideoDecoderFactory { public: - // |supports_augmenting_data| defines if the decoder would support augmenting + // `supports_augmenting_data` defines if the decoder would support augmenting // data. If set, the decoder is expected to output video frame buffers of type // AugmentedVideoFrameBuffer. MultiplexDecoderFactory(std::unique_ptr<VideoDecoderFactory> factory,
diff --git a/media/engine/payload_type_mapper.h b/media/engine/payload_type_mapper.h index d8ab4a4..1d5cd71 100644 --- a/media/engine/payload_type_mapper.h +++ b/media/engine/payload_type_mapper.h
@@ -27,12 +27,12 @@ PayloadTypeMapper(); ~PayloadTypeMapper(); - // Finds the current payload type for |format| or assigns a new one, if no + // Finds the current payload type for `format` or assigns a new one, if no // current mapping exists. Will return an empty value if it was unable to // create a mapping, i.e. if all dynamic payload type ids have been used up. absl::optional<int> GetMappingFor(const webrtc::SdpAudioFormat& format); - // Finds the current payload type for |format|, if any. Returns an empty value + // Finds the current payload type for `format`, if any. Returns an empty value // if no payload type mapping exists for the format. absl::optional<int> FindMappingFor( const webrtc::SdpAudioFormat& format) const;
diff --git a/media/engine/simulcast.cc b/media/engine/simulcast.cc index ebc6a24..6d65dd2 100644 --- a/media/engine/simulcast.cc +++ b/media/engine/simulcast.cc
@@ -71,16 +71,16 @@ int width; int height; // The maximum number of simulcast layers can be used for - // resolutions at |widthxheight| for legacy applications. + // resolutions at `widthxheight` for legacy applications. size_t max_layers; - // The maximum bitrate for encoding stream at |widthxheight|, when we are + // The maximum bitrate for encoding stream at `widthxheight`, when we are // not sending the next higher spatial stream. webrtc::DataRate max_bitrate; - // The target bitrate for encoding stream at |widthxheight|, when this layer + // The target bitrate for encoding stream at `widthxheight`, when this layer // is not the highest layer (i.e., when we are sending another higher spatial // stream). webrtc::DataRate target_bitrate; - // The minimum bitrate needed for encoding stream at |widthxheight|. + // The minimum bitrate needed for encoding stream at `widthxheight`. webrtc::DataRate min_bitrate; }; @@ -210,7 +210,7 @@ const float rate = (total_pixels_up - total_pixels) / static_cast<float>(total_pixels_up - total_pixels_down); - // Use upper resolution if |rate| is below the configured threshold. + // Use upper resolution if `rate` is below the configured threshold. size_t max_layers = (rate < max_roundup_rate.value_or(kDefaultMaxRoundupRate)) ? formats[index - 1].max_layers : formats[index].max_layers; @@ -296,7 +296,7 @@ "Disabled")) { // Max layers from one higher resolution in kSimulcastFormats will be used // if the ratio (pixels_up - pixels) / (pixels_up - pixels_down) is less - // than configured |max_ratio|. pixels_down is the selected index in + // than configured `max_ratio`. pixels_down is the selected index in // kSimulcastFormats based on pixels. webrtc::FieldTrialOptional<double> max_ratio("max_ratio"); webrtc::ParseFieldTrial({&max_ratio}, @@ -369,8 +369,8 @@ // 1|. width = NormalizeSimulcastSize(width, layer_count); height = NormalizeSimulcastSize(height, layer_count); - // Add simulcast streams, from highest resolution (|s| = num_simulcast_layers - // -1) to lowest resolution at |s| = 0. + // Add simulcast streams, from highest resolution (`s` = num_simulcast_layers + // -1) to lowest resolution at `s` = 0. for (size_t s = layer_count - 1;; --s) { layers[s].width = width; layers[s].height = height;
diff --git a/media/engine/simulcast.h b/media/engine/simulcast.h index 5defa52..aa8c394 100644 --- a/media/engine/simulcast.h +++ b/media/engine/simulcast.h
@@ -21,12 +21,12 @@ namespace cricket { -// Gets the total maximum bitrate for the |streams|. +// Gets the total maximum bitrate for the `streams`. webrtc::DataRate GetTotalMaxBitrate( const std::vector<webrtc::VideoStream>& streams); -// Adds any bitrate of |max_bitrate| that is above the total maximum bitrate for -// the |layers| to the highest quality layer. +// Adds any bitrate of `max_bitrate` that is above the total maximum bitrate for +// the `layers` to the highest quality layer. void BoostMaxSimulcastLayer(webrtc::DataRate max_bitrate, std::vector<webrtc::VideoStream>* layers);
diff --git a/media/engine/simulcast_encoder_adapter.cc b/media/engine/simulcast_encoder_adapter.cc index 116f987..f7be8b3 100644 --- a/media/engine/simulcast_encoder_adapter.cc +++ b/media/engine/simulcast_encoder_adapter.cc
@@ -287,7 +287,7 @@ RTC_DCHECK_RUN_ON(&encoder_queue_); while (!stream_contexts_.empty()) { - // Move the encoder instances and put it on the |cached_encoder_contexts_| + // Move the encoder instances and put it on the `cached_encoder_contexts_` // where it may possibly be reused from (ordering does not matter). cached_encoder_contexts_.push_front( std::move(stream_contexts_.back()).ReleaseEncoderContext()); @@ -415,7 +415,7 @@ } // Intercept frame encode complete callback only for upper streams, where - // we need to set a correct stream index. Set |parent| to nullptr for the + // we need to set a correct stream index. Set `parent` to nullptr for the // lowest stream to bypass the callback. SimulcastEncoderAdapter* parent = stream_idx > 0 ? this : nullptr; @@ -699,8 +699,8 @@ is_lowest_quality_stream && prefer_temporal_support_on_base_layer_; - // Toggling of |prefer_temporal_support| requires encoder recreation. Find - // and reuse encoder with desired |prefer_temporal_support|. Otherwise, if + // Toggling of `prefer_temporal_support` requires encoder recreation. Find + // and reuse encoder with desired `prefer_temporal_support`. Otherwise, if // there is no such encoder in the cache, create a new instance. auto encoder_context_iter = std::find_if(cached_encoder_contexts_.begin(), @@ -769,7 +769,7 @@ codec_params.VP8()->numberOfTemporalLayers = stream_params.numberOfTemporalLayers; if (!is_highest_quality_stream) { - // For resolutions below CIF, set the codec |complexity| parameter to + // For resolutions below CIF, set the codec `complexity` parameter to // kComplexityHigher, which maps to cpu_used = -4. int pixels_per_frame = codec_params.width * codec_params.height; if (pixels_per_frame < 352 * 288) {
diff --git a/media/engine/simulcast_encoder_adapter.h b/media/engine/simulcast_encoder_adapter.h index 07e3ccd..1d2200b 100644 --- a/media/engine/simulcast_encoder_adapter.h +++ b/media/engine/simulcast_encoder_adapter.h
@@ -43,8 +43,8 @@ // TODO(bugs.webrtc.org/11000): Remove when downstream usage is gone. SimulcastEncoderAdapter(VideoEncoderFactory* primarty_factory, const SdpVideoFormat& format); - // |primary_factory| produces the first-choice encoders to use. - // |fallback_factory|, if non-null, is used to create fallback encoder that + // `primary_factory` produces the first-choice encoders to use. + // `fallback_factory`, if non-null, is used to create fallback encoder that // will be used if InitEncode() fails for the primary encoder. SimulcastEncoderAdapter(VideoEncoderFactory* primary_factory, VideoEncoderFactory* fallback_factory, @@ -147,7 +147,7 @@ void DestroyStoredEncoders(); // This method creates encoder. May reuse previously created encoders from - // |cached_encoder_contexts_|. It's const because it's used from + // `cached_encoder_contexts_`. It's const because it's used from // const GetEncoderInfo(). std::unique_ptr<EncoderContext> FetchOrCreateEncoderContext( bool is_lowest_quality_stream) const; @@ -182,7 +182,7 @@ // Store previously created and released encoders , so they don't have to be // recreated. Remaining encoders are destroyed by the destructor. - // Marked as |mutable| becuase we may need to temporarily create encoder in + // Marked as `mutable` becuase we may need to temporarily create encoder in // GetEncoderInfo(), which is const. mutable std::list<std::unique_ptr<EncoderContext>> cached_encoder_contexts_;
diff --git a/media/engine/simulcast_encoder_adapter_unittest.cc b/media/engine/simulcast_encoder_adapter_unittest.cc index 48e005f..c946b60 100644 --- a/media/engine/simulcast_encoder_adapter_unittest.cc +++ b/media/engine/simulcast_encoder_adapter_unittest.cc
@@ -186,7 +186,7 @@ int32_t init_encode_return_value_ = 0; std::vector<MockVideoEncoder*> encoders_; std::vector<const char*> encoder_names_; - // Keep number of entries in sync with |kMaxSimulcastStreams|. + // Keep number of entries in sync with `kMaxSimulcastStreams`. std::vector<int> requested_resolution_alignments_ = {1, 1, 1}; bool supports_simulcast_ = false; }; @@ -387,7 +387,7 @@ video_format_(video_format) {} // Can only be called once as the SimulcastEncoderAdapter will take the - // ownership of |factory_|. + // ownership of `factory_`. VideoEncoder* CreateMockEncoderAdapter() { return new SimulcastEncoderAdapter(primary_factory_.get(), fallback_factory_.get(), video_format_); @@ -433,8 +433,8 @@ void ReSetUp() { if (adapter_) { adapter_->Release(); - // |helper_| owns factories which |adapter_| needs to destroy encoders. - // Release |adapter_| before |helper_| (released in SetUp()). + // `helper_` owns factories which `adapter_` needs to destroy encoders. + // Release `adapter_` before `helper_` (released in SetUp()). adapter_.reset(); } SetUp(); @@ -755,7 +755,7 @@ 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/unhandled_packets_buffer.cc b/media/engine/unhandled_packets_buffer.cc index ebc841e..cb6f0ec 100644 --- a/media/engine/unhandled_packets_buffer.cc +++ b/media/engine/unhandled_packets_buffer.cc
@@ -35,7 +35,7 @@ insert_pos_ = (insert_pos_ + 1) % kMaxStashedPackets; } -// Backfill |consumer| with all stored packet related |ssrcs|. +// Backfill `consumer` with all stored packet related `ssrcs`. void UnhandledPacketsBuffer::BackfillPackets( rtc::ArrayView<const uint32_t> ssrcs, std::function<void(uint32_t, int64_t, rtc::CopyOnWriteBuffer)> consumer) {
diff --git a/media/engine/unhandled_packets_buffer.h b/media/engine/unhandled_packets_buffer.h index ef03588..63a6195 100644 --- a/media/engine/unhandled_packets_buffer.h +++ b/media/engine/unhandled_packets_buffer.h
@@ -35,7 +35,7 @@ int64_t packet_time_us, rtc::CopyOnWriteBuffer packet); - // Feed all packets with |ssrcs| into |consumer|. + // Feed all packets with `ssrcs` into `consumer`. void BackfillPackets( rtc::ArrayView<const uint32_t> ssrcs, std::function<void(uint32_t, int64_t, rtc::CopyOnWriteBuffer)> consumer);
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc index 017ce53..d42047a 100644 --- a/media/engine/webrtc_video_engine.cc +++ b/media/engine/webrtc_video_engine.cc
@@ -448,7 +448,7 @@ webrtc::VideoSendStream::StreamStats& rtp_substream = rtp_substreams[media_ssrc]; - // We only merge |rtp_stats|. All other metrics are not applicable for RTX + // We only merge `rtp_stats`. All other metrics are not applicable for RTX // and FlexFEC. // TODO(hbos): kRtx and kFlexfec stats should use a separate struct to make // it clear what is or is not applicable. @@ -1543,7 +1543,7 @@ flexfec_config->protected_media_ssrcs = {ssrc}; flexfec_config->rtp.local_ssrc = config->rtp.local_ssrc; flexfec_config->rtcp_mode = config->rtp.rtcp_mode; - // TODO(brandtr): We should be spec-compliant and set |transport_cc| here + // TODO(brandtr): We should be spec-compliant and set `transport_cc` here // based on the rtcp-fb for the FlexFEC codec, not the media codec. flexfec_config->rtp.transport_cc = config->rtp.transport_cc; flexfec_config->rtp.extensions = config->rtp.extensions; @@ -1573,7 +1573,7 @@ last_unsignalled_ssrc_creation_time_ms_ = absl::nullopt; // Delete any created default streams. This is needed to avoid SSRC collisions - // in Call's RtpDemuxer, in the case that |this| has created a default video + // in Call's RtpDemuxer, in the case that `this` has created a default video // receiver, and then some other WebRtcVideoChannel gets the SSRC signaled // in the corresponding Unified Plan "m=" section. auto it = receive_streams_.begin(); @@ -2179,7 +2179,7 @@ WebRtcVideoChannel::WebRtcVideoSendStream::GetDegradationPreference() const { // Do not adapt resolution for screen content as this will likely // result in blurry and unreadable text. - // |this| acts like a VideoSource to make sure SinkWants are handled on the + // `this` acts like a VideoSource to make sure SinkWants are handled on the // correct thread. if (!enable_cpu_overuse_detection_) { return webrtc::DegradationPreference::DISABLED; @@ -2263,7 +2263,7 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::SetSendParameters( const ChangedSendParameters& params) { RTC_DCHECK_RUN_ON(&thread_checker_); - // |recreate_stream| means construction-time parameters have changed and the + // `recreate_stream` means construction-time parameters have changed and the // sending stream needs to be reset with the new config. bool recreate_stream = false; if (params.rtcp_mode) { @@ -2552,7 +2552,7 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::ReconfigureEncoder() { RTC_DCHECK_RUN_ON(&thread_checker_); if (!stream_) { - // The webrtc::VideoSendStream |stream_| has not yet been created but other + // The webrtc::VideoSendStream `stream_` has not yet been created but other // parameters has changed. return; } @@ -2632,8 +2632,8 @@ common_info.aggregated_framerate_sent = stats.encode_frame_rate; common_info.aggregated_huge_frames_sent = stats.huge_frames_sent; - // If we don't have any substreams, get the remaining metrics from |stats|. - // Otherwise, these values are obtained from |sub_stream| below. + // If we don't have any substreams, get the remaining metrics from `stats`. + // Otherwise, these values are obtained from `sub_stream` below. if (stats.substreams.empty()) { for (uint32_t ssrc : parameters_.config.rtp.ssrcs) { common_info.add_ssrc(ssrc); @@ -2998,7 +2998,7 @@ config_.rtp.nack.rtp_history_ms = nack_history_ms; config_.rtp.transport_cc = transport_cc_enabled; config_.rtp.rtcp_mode = rtcp_mode; - // TODO(brandtr): We should be spec-compliant and set |transport_cc| here + // TODO(brandtr): We should be spec-compliant and set `transport_cc` here // based on the rtcp-fb for the FlexFEC codec, not the media codec. flexfec_config_.rtp.transport_cc = config_.rtp.transport_cc; flexfec_config_.rtcp_mode = config_.rtp.rtcp_mode; @@ -3298,7 +3298,7 @@ std::vector<VideoCodecSettings> video_codecs; std::map<int, VideoCodec::CodecType> payload_codec_type; - // |rtx_mapping| maps video payload type to rtx payload type. + // `rtx_mapping` maps video payload type to rtx payload type. std::map<int, int> rtx_mapping; std::map<int, int> rtx_time_mapping;
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h index a67a010..8b3a7f4 100644 --- a/media/engine/webrtc_video_engine.h +++ b/media/engine/webrtc_video_engine.h
@@ -218,7 +218,7 @@ std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override; - // Take the buffered packets for |ssrcs| and feed them into DeliverPacket. + // Take the buffered packets for `ssrcs` and feed them into DeliverPacket. // This method does nothing unless unknown_ssrc_packet_buffer_ is configured. void BackfillBufferedPackets(rtc::ArrayView<const uint32_t> ssrcs); @@ -258,12 +258,12 @@ VideoCodecSettings(); // Checks if all members of |*this| are equal to the corresponding members - // of |other|. + // of `other`. bool operator==(const VideoCodecSettings& other) const; bool operator!=(const VideoCodecSettings& other) const; - // Checks if all members of |a|, except |flexfec_payload_type|, are equal - // to the corresponding members of |b|. + // Checks if all members of `a`, except `flexfec_payload_type`, are equal + // to the corresponding members of `b`. static bool EqualsDisregardingFlexfec(const VideoCodecSettings& a, const VideoCodecSettings& b); @@ -290,7 +290,7 @@ // These optionals are unset if not changed. absl::optional<std::vector<VideoCodecSettings>> codec_settings; absl::optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions; - // Keep track of the FlexFEC payload type separately from |codec_settings|. + // Keep track of the FlexFEC payload type separately from `codec_settings`. // This allows us to recreate the FlexfecReceiveStream separately from the // VideoReceiveStream when the FlexFEC payload type is changed. absl::optional<int> flexfec_payload_type; @@ -389,8 +389,8 @@ const VideoCodec& codec) const; void ReconfigureEncoder(); - // Calls Start or Stop according to whether or not |sending_| is true, - // and whether or not the encoding in |rtp_parameters_| is active. + // Calls Start or Stop according to whether or not `sending_` is true, + // and whether or not the encoding in `rtp_parameters_` is active. void UpdateSendState(); webrtc::DegradationPreference GetDegradationPreference() const @@ -494,7 +494,7 @@ webrtc::Call* const call_; const StreamParams stream_params_; - // Both |stream_| and |flexfec_stream_| are managed by |this|. They are + // Both `stream_` and `flexfec_stream_` are managed by `this`. They are // destroyed by calling call_->DestroyVideoReceiveStream and // call_->DestroyFlexfecReceiveStream, respectively. webrtc::VideoReceiveStream* stream_; @@ -577,8 +577,8 @@ // criteria because the streams live on the worker thread and the demuxer // lives on the network thread. Because packets are posted from the network // thread to the worker thread, they can still be in-flight when streams are - // reconfgured. This can happen when |demuxer_criteria_id_| and - // |demuxer_criteria_completed_id_| don't match. During this time, we do not + // reconfgured. This can happen when `demuxer_criteria_id_` and + // `demuxer_criteria_completed_id_` don't match. During this time, we do not // want to create unsignalled receive streams and should instead drop the // packets. E.g: // * If RemoveRecvStream(old_ssrc) was recently called, there may be packets
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc index 97764f8..7c1bf6e 100644 --- a/media/engine/webrtc_video_engine_unittest.cc +++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -127,8 +127,8 @@ cricket::kRtcpFbParamCcm, cricket::kRtcpFbCcmParamFir))); } -// Return true if any codec in |codecs| is an RTX codec with associated payload -// type |payload_type|. +// Return true if any codec in `codecs` is an RTX codec with associated payload +// type `payload_type`. bool HasRtxCodec(const std::vector<cricket::VideoCodec>& codecs, int payload_type) { for (const cricket::VideoCodec& codec : codecs) { @@ -1102,7 +1102,7 @@ // Tests when GetSources is called with non-existing ssrc, it will return an // empty list of RtpSource without crashing. TEST_F(WebRtcVideoEngineTest, GetSourcesWithNonExistingSsrc) { - // Setup an recv stream with |kSsrc|. + // Setup an recv stream with `kSsrc`. AddSupportedVideoCodecType("VP8"); cricket::VideoRecvParameters parameters; parameters.codecs.push_back(GetEngineCodec("VP8")); @@ -1128,7 +1128,7 @@ } TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, EmptyFactories) { - // |engine| take ownership of the factories. + // `engine` take ownership of the factories. webrtc::MockVideoEncoderFactory* encoder_factory = new webrtc::MockVideoEncoderFactory(); webrtc::MockVideoDecoderFactory* decoder_factory = @@ -1151,7 +1151,7 @@ // from the engine and that we will create a Vp8 encoder and decoder using the // new factories. TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) { - // |engine| take ownership of the factories. + // `engine` take ownership of the factories. webrtc::MockVideoEncoderFactory* encoder_factory = new webrtc::MockVideoEncoderFactory(); webrtc::MockVideoDecoderFactory* decoder_factory = @@ -1207,7 +1207,7 @@ VerifyCodecHasDefaultFeedbackParams(engine_codecs.at(0), /*lntf_expected=*/false); - // Mock encoder creation. |engine| take ownership of the encoder. + // Mock encoder creation. `engine` take ownership of the encoder. webrtc::VideoEncoderFactory::CodecInfo codec_info; codec_info.has_internal_source = false; const webrtc::SdpVideoFormat format("VP8"); @@ -1219,7 +1219,7 @@ return std::make_unique<FakeWebRtcVideoEncoder>(nullptr); }); - // Mock decoder creation. |engine| take ownership of the decoder. + // Mock decoder creation. `engine` take ownership of the decoder. EXPECT_CALL(*decoder_factory, CreateVideoDecoder(format)).WillOnce([] { return std::make_unique<FakeWebRtcVideoDecoder>(nullptr); }); @@ -1276,7 +1276,7 @@ // Test behavior when decoder factory fails to create a decoder (returns null). TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, NullDecoder) { - // |engine| take ownership of the factories. + // `engine` take ownership of the factories. webrtc::MockVideoEncoderFactory* encoder_factory = new webrtc::MockVideoEncoderFactory(); webrtc::MockVideoDecoderFactory* decoder_factory = @@ -1373,7 +1373,7 @@ options.video_noise_reduction.emplace(false); EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder)); // Change back to regular video content, update encoder. Also change - // a non |is_screencast| option just to verify it doesn't affect recreation. + // a non `is_screencast` option just to verify it doesn't affect recreation. frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(3)); EXPECT_EQ(webrtc::VideoCodecMode::kRealtimeVideo, @@ -3573,7 +3573,7 @@ EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); EXPECT_EQ(1, send_stream->num_encoder_reconfigurations()); - // Change |options| and expect 2 reconfigurations. + // Change `options` and expect 2 reconfigurations. options.video_noise_reduction = true; EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); EXPECT_EQ(2, send_stream->num_encoder_reconfigurations()); @@ -4367,7 +4367,7 @@ EXPECT_EQ(video_stream_config.rtp.rtcp_mode, flexfec_stream_config.rtcp_mode); EXPECT_EQ(video_stream_config.rtcp_send_transport, flexfec_stream_config.rtcp_send_transport); - // TODO(brandtr): Update this EXPECT when we set |transport_cc| in a + // TODO(brandtr): Update this EXPECT when we set `transport_cc` in a // spec-compliant way. EXPECT_EQ(video_stream_config.rtp.transport_cc, flexfec_stream_config.rtp.transport_cc); @@ -7476,7 +7476,7 @@ &frame_forwarder)); channel_->SetSend(true); - // Set |scale_resolution_down_by|'s. + // Set `scale_resolution_down_by`'s. auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_); ASSERT_EQ(rtp_parameters.encodings.size(), 3u); rtp_parameters.encodings[0].scale_resolution_down_by = 1.0; @@ -7632,7 +7632,7 @@ &frame_forwarder)); channel_->SetSend(true); - // Set |scale_resolution_down_by|'s. + // Set `scale_resolution_down_by`'s. auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_); ASSERT_EQ(rtp_parameters.encodings.size(), 3u); rtp_parameters.encodings[0].scale_resolution_down_by = 1.0; @@ -7868,7 +7868,7 @@ // FakeVideoSendStream calls CreateEncoderStreams, test that the vector of // VideoStreams are created appropriately for the simulcast case. - // The maximum |max_framerate| is used, kDefaultVideoMaxFramerate: 60. + // The maximum `max_framerate` is used, kDefaultVideoMaxFramerate: 60. EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size()); EXPECT_EQ(15, stream->GetVideoStreams()[0].max_framerate); EXPECT_EQ(kDefaultVideoMaxFramerate, @@ -8640,7 +8640,7 @@ rtp_packet.SetSsrc(kIncomingUnsignalledSsrc); ReceivePacketAndAdvanceTime(rtp_packet.Buffer(), /* packet_time_us */ -1); - // The |ssrc| member should still be unset. + // The `ssrc` member should still be unset. rtp_parameters = channel_->GetDefaultRtpReceiveParameters(); ASSERT_EQ(1u, rtp_parameters.encodings.size()); EXPECT_FALSE(rtp_parameters.encodings[0].ssrc);
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc index a2741f7..e9ffb21 100644 --- a/media/engine/webrtc_voice_engine.cc +++ b/media/engine/webrtc_voice_engine.cc
@@ -171,8 +171,8 @@ return std::min(a, b); } -// |max_send_bitrate_bps| is the bitrate from "b=" in SDP. -// |rtp_max_bitrate_bps| is the bitrate from RtpSender::SetParameters. +// `max_send_bitrate_bps` is the bitrate from "b=" in SDP. +// `rtp_max_bitrate_bps` is the bitrate from RtpSender::SetParameters. absl::optional<int> ComputeSendBitrate(int max_send_bitrate_bps, absl::optional<int> rtp_max_bitrate_bps, const webrtc::AudioCodecSpec& spec) { @@ -186,8 +186,8 @@ } if (bps < spec.info.min_bitrate_bps) { - // If codec is not multi-rate and |bps| is less than the fixed bitrate then - // fail. If codec is not multi-rate and |bps| exceeds or equal the fixed + // If codec is not multi-rate and `bps` is less than the fixed bitrate then + // fail. If codec is not multi-rate and `bps` exceeds or equal the fixed // bitrate then ignore. RTC_LOG(LS_ERROR) << "Failed to set codec " << spec.format.name << " to bitrate " << bps @@ -1003,7 +1003,7 @@ number_of_frames, sample_rate, audio_frame->speech_type_, audio_frame->vad_activity_, number_of_channels); // TODO(bugs.webrtc.org/10739): add dcheck that - // |absolute_capture_timestamp_ms| always receives a value. + // `absolute_capture_timestamp_ms` always receives a value. if (absolute_capture_timestamp_ms) { audio_frame->set_absolute_capture_timestamp_ms( *absolute_capture_timestamp_ms); @@ -1011,11 +1011,11 @@ stream_->SendAudioData(std::move(audio_frame)); } - // Callback from the |source_| when it is going away. In case Start() has + // Callback from the `source_` when it is going away. In case Start() has // never been called, this callback won't be triggered. void OnClose() override { RTC_DCHECK_RUN_ON(&worker_thread_checker_); - // Set |source_| to nullptr to make sure no more callback will get into + // Set `source_` to nullptr to make sure no more callback will get into // the source. source_ = nullptr; UpdateSendState(); @@ -1498,8 +1498,8 @@ // |WebRtcAudioSendStream::SetRtpParameters()| which is called at the end, // though there are two difference: // 1. |WebRtcVoiceMediaChannel::SetChannelSendParameters()| only calls - // |SetSendCodec| while |WebRtcAudioSendStream::SetRtpParameters()| calls - // |SetSendCodecs|. The outcome should be the same. + // `SetSendCodec` while |WebRtcAudioSendStream::SetRtpParameters()| calls + // `SetSendCodecs`. The outcome should be the same. // 2. AudioSendStream can be recreated. // Codecs are handled at the WebRtcVoiceMediaChannel level. @@ -1998,7 +1998,7 @@ RTC_DCHECK_RUN_ON(worker_thread_); RTC_LOG(LS_INFO) << "ResetUnsignaledRecvStream."; unsignaled_stream_params_ = StreamParams(); - // Create a copy since RemoveRecvStream will modify |unsignaled_recv_ssrcs_|. + // Create a copy since RemoveRecvStream will modify `unsignaled_recv_ssrcs_`. std::vector<uint32_t> to_remove = unsignaled_recv_ssrcs_; for (uint32_t ssrc : to_remove) { RemoveRecvStream(ssrc);
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc index c570b1a..4b2742c 100644 --- a/media/engine/webrtc_voice_engine_unittest.cc +++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -395,10 +395,10 @@ } // Test that send bandwidth is set correctly. - // |codec| is the codec under test. - // |max_bitrate| is a parameter to set to SetMaxSendBandwidth(). - // |expected_result| is the expected result from SetMaxSendBandwidth(). - // |expected_bitrate| is the expected audio bitrate afterward. + // `codec` is the codec under test. + // `max_bitrate` is a parameter to set to SetMaxSendBandwidth(). + // `expected_result` is the expected result from SetMaxSendBandwidth(). + // `expected_bitrate` is the expected audio bitrate afterward. void TestMaxSendBandwidth(const cricket::AudioCodec& codec, int max_bitrate, bool expected_result, @@ -1470,7 +1470,7 @@ // Receive PCMU packet (SSRC=1). DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); - // The |ssrc| member should still be unset. + // The `ssrc` member should still be unset. rtp_parameters = channel_->GetDefaultRtpReceiveParameters(); ASSERT_EQ(1u, rtp_parameters.encodings.size()); EXPECT_FALSE(rtp_parameters.encodings[0].ssrc); @@ -3611,11 +3611,11 @@ // Tests when GetSources is called with non-existing ssrc, it will return an // empty list of RtpSource without crashing. TEST_P(WebRtcVoiceEngineTestFake, GetSourcesWithNonExistingSsrc) { - // Setup an recv stream with |kSsrcX|. + // Setup an recv stream with `kSsrcX`. SetupRecvStream(); cricket::WebRtcVoiceMediaChannel* media_channel = static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_); - // Call GetSources with |kSsrcY| which doesn't exist. + // Call GetSources with `kSsrcY` which doesn't exist. std::vector<webrtc::RtpSource> sources = media_channel->GetSources(kSsrcY); EXPECT_EQ(0u, sources.size()); }
diff --git a/media/sctp/sctp_transport_internal.h b/media/sctp/sctp_transport_internal.h index b132716..e44efb5 100644 --- a/media/sctp/sctp_transport_internal.h +++ b/media/sctp/sctp_transport_internal.h
@@ -86,11 +86,11 @@ // completes. This method can be called multiple times, though not if either // of the ports are changed. // - // |local_sctp_port| and |remote_sctp_port| are passed along the wire and the + // `local_sctp_port` and `remote_sctp_port` are passed along the wire and the // listener and connector must be using the same port. They are not related // to the ports at the IP level. If set to -1, we default to // kSctpDefaultPort. - // |max_message_size_| sets the max message size on the connection. + // `max_message_size_` sets the max message size on the connection. // It must be smaller than or equal to kSctpSendBufferSize. // It can be changed by a secons Start() call. // @@ -104,10 +104,10 @@ // NOTE: Initially there was a "Stop" method here, but it was never used, so // it was removed. - // Informs SctpTransport that |sid| will start being used. Returns false if - // it is impossible to use |sid|, or if it's already in use. - // Until calling this, can't send data using |sid|. - // TODO(deadbeef): Actually implement the "returns false if |sid| can't be + // Informs SctpTransport that `sid` will start being used. Returns false if + // it is impossible to use `sid`, or if it's already in use. + // Until calling this, can't send data using `sid`. + // TODO(deadbeef): Actually implement the "returns false if `sid` can't be // used" part. See: // https://bugs.chromium.org/p/chromium/issues/detail?id=619849 virtual bool OpenStream(int sid) = 0;
diff --git a/media/sctp/usrsctp_transport.cc b/media/sctp/usrsctp_transport.cc index 7824a729..ce868a1 100644 --- a/media/sctp/usrsctp_transport.cc +++ b/media/sctp/usrsctp_transport.cc
@@ -304,7 +304,7 @@ return map_.erase(id) > 0; } - // Posts |action| to the network thread of the transport identified by |id| + // Posts `action` to the network thread of the transport identified by `id` // and returns true if found, all while holding a lock to protect against the // transport being simultaneously deleted/deregistered, or returns false if // not found.
diff --git a/media/sctp/usrsctp_transport.h b/media/sctp/usrsctp_transport.h index 5dcf57b..06988fd 100644 --- a/media/sctp/usrsctp_transport.h +++ b/media/sctp/usrsctp_transport.h
@@ -68,10 +68,10 @@ class UsrsctpTransport : public SctpTransportInternal, public sigslot::has_slots<> { public: - // |network_thread| is where packets will be processed and callbacks from + // `network_thread` is where packets will be processed and callbacks from // this transport will be posted, and is the only thread on which public // methods can be called. - // |transport| is not required (can be null). + // `transport` is not required (can be null). UsrsctpTransport(rtc::Thread* network_thread, rtc::PacketTransportInternal* transport); ~UsrsctpTransport() override; @@ -163,7 +163,7 @@ // buffered message was accepted by the sctp lib. bool SendBufferedMessage(); - // Tries to send the |payload| on the usrsctp lib. The message will be + // Tries to send the `payload` on the usrsctp lib. The message will be // advanced by the amount that was sent. SendDataResult SendMessageInternal(OutgoingMessage* message); @@ -180,7 +180,7 @@ void OnSendThresholdCallback(); sockaddr_conn GetSctpSockAddr(int port); - // Called using |invoker_| to send packet on the network. + // Called using `invoker_` to send packet on the network. void OnPacketFromSctpToNetwork(const rtc::CopyOnWriteBuffer& buffer); // Called on the network thread. @@ -189,10 +189,10 @@ size_t length, struct sctp_rcvinfo rcv, int flags); - // Called using |invoker_| to decide what to do with the data. + // Called using `invoker_` to decide what to do with the data. void OnDataFromSctpToTransport(const ReceiveDataParams& params, const rtc::CopyOnWriteBuffer& buffer); - // Called using |invoker_| to decide what to do with the notification. + // Called using `invoker_` to decide what to do with the notification. void OnNotificationFromSctp(const rtc::CopyOnWriteBuffer& buffer); void OnNotificationAssocChange(const sctp_assoc_change& change); @@ -226,7 +226,7 @@ // Has Start been called? Don't create SCTP socket until it has. bool started_ = false; // Are we ready to queue data (SCTP socket created, and not blocked due to - // congestion control)? Different than |transport_|'s "ready to send". + // congestion control)? Different than `transport_`'s "ready to send". bool ready_to_send_data_ = false; // Used to keep track of the status of each stream (or rather, each pair of @@ -268,7 +268,7 @@ } }; - // Entries should only be removed from this map if |reset_complete| is + // Entries should only be removed from this map if `reset_complete` is // true. std::map<uint32_t, StreamStatus> stream_status_by_sid_;