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 7824a72..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_;
