Use backticks not vertical bars to denote variables in comments for /modules/rtp_rtcp
Bug: webrtc:12338
Change-Id: I52eb3b6675c4705e22f51b70799ed6139a3b46bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227164
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34686}
diff --git a/modules/rtp_rtcp/include/receive_statistics.h b/modules/rtp_rtcp/include/receive_statistics.h
index ce87b99..f973b7c 100644
--- a/modules/rtp_rtcp/include/receive_statistics.h
+++ b/modules/rtp_rtcp/include/receive_statistics.h
@@ -29,7 +29,7 @@
public:
virtual ~ReceiveStatisticsProvider() = default;
// Collects receive statistic in a form of rtcp report blocks.
- // Returns at most |max_blocks| report blocks.
+ // Returns at most `max_blocks` report blocks.
virtual std::vector<rtcp::ReportBlock> RtcpReportBlocks(
size_t max_blocks) = 0;
};
diff --git a/modules/rtp_rtcp/include/remote_ntp_time_estimator.h b/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
index 6112e54..5734a50 100644
--- a/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
+++ b/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
@@ -25,21 +25,21 @@
// RemoteNtpTimeEstimator can be used to estimate a given RTP timestamp's NTP
// time in local timebase.
// Note that it needs to be trained with at least 2 RTCP SR (by calling
-// |UpdateRtcpTimestamp|) before it can be used.
+// `UpdateRtcpTimestamp`) before it can be used.
class RemoteNtpTimeEstimator {
public:
explicit RemoteNtpTimeEstimator(Clock* clock);
~RemoteNtpTimeEstimator();
- // Updates the estimator with round trip time |rtt|, NTP seconds |ntp_secs|,
- // NTP fraction |ntp_frac| and RTP timestamp |rtp_timestamp|.
+ // Updates the estimator with round trip time `rtt`, NTP seconds `ntp_secs`,
+ // NTP fraction `ntp_frac` and RTP timestamp `rtp_timestamp`.
bool UpdateRtcpTimestamp(int64_t rtt,
uint32_t ntp_secs,
uint32_t ntp_frac,
uint32_t rtp_timestamp);
- // Estimates the NTP timestamp in local timebase from |rtp_timestamp|.
+ // Estimates the NTP timestamp in local timebase from `rtp_timestamp`.
// Returns the NTP timestamp in ms when success. -1 if failed.
int64_t Estimate(uint32_t rtp_timestamp);
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 998a754..5a80cd0 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -212,7 +212,7 @@
virtual ~RtcpBandwidthObserver() {}
};
-// NOTE! |kNumMediaTypes| must be kept in sync with RtpPacketMediaType!
+// NOTE! `kNumMediaTypes` must be kept in sync with RtpPacketMediaType!
static constexpr size_t kNumMediaTypes = 5;
enum class RtpPacketMediaType : size_t {
kAudio, // Audio media packets.
@@ -220,7 +220,7 @@
kRetransmission, // Retransmisions, sent as response to NACK.
kForwardErrorCorrection, // FEC packets.
kPadding = kNumMediaTypes - 1, // RTX or plain padding sent to maintain BWE.
- // Again, don't forget to udate |kNumMediaTypes| if you add another value!
+ // Again, don't forget to udate `kNumMediaTypes` if you add another value!
};
struct RtpPacketSendInfo {
@@ -231,7 +231,7 @@
// TODO(bugs.webrtc.org/12713): Remove once downstream usage is gone.
uint32_t ssrc = 0;
absl::optional<uint32_t> media_ssrc;
- uint16_t rtp_sequence_number = 0; // Only valid if |media_ssrc| is set.
+ uint16_t rtp_sequence_number = 0; // Only valid if `media_ssrc` is set.
uint32_t rtp_timestamp = 0;
size_t length = 0;
absl::optional<RtpPacketMediaType> packet_type;
@@ -271,7 +271,7 @@
struct StreamPacketInfo {
bool received;
- // |rtp_sequence_number| and |is_retransmission| are only valid if |ssrc|
+ // `rtp_sequence_number` and `is_retransmission` are only valid if `ssrc`
// is populated.
absl::optional<uint32_t> ssrc;
uint16_t rtp_sequence_number;
@@ -434,7 +434,7 @@
// Information exposed through the GetStats api.
struct RtpReceiveStats {
- // |packets_lost| and |jitter| are defined by RFC 3550, and exposed in the
+ // `packets_lost` and `jitter` are defined by RFC 3550, and exposed in the
// RTCReceivedRtpStreamStats dictionary, see
// https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict*
int32_t packets_lost = 0;
diff --git a/modules/rtp_rtcp/include/ulpfec_receiver.h b/modules/rtp_rtcp/include/ulpfec_receiver.h
index d3981df..bf1c826 100644
--- a/modules/rtp_rtcp/include/ulpfec_receiver.h
+++ b/modules/rtp_rtcp/include/ulpfec_receiver.h
@@ -42,7 +42,7 @@
// Takes a RED packet, strips the RED header, and adds the resulting
// "virtual" RTP packet(s) into the internal buffer.
//
- // TODO(brandtr): Set |ulpfec_payload_type| during constructor call,
+ // TODO(brandtr): Set `ulpfec_payload_type` during constructor call,
// rather than as a parameter here.
virtual bool AddReceivedRedPacket(const RtpPacketReceived& rtp_packet,
uint8_t ulpfec_payload_type) = 0;
diff --git a/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h b/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h
index 89d7f08..a59e2b4 100644
--- a/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h
+++ b/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h
@@ -22,7 +22,7 @@
namespace webrtc {
//
-// Helper class for interpolating the |AbsoluteCaptureTime| header extension.
+// Helper class for interpolating the `AbsoluteCaptureTime` header extension.
//
// Supports the "timestamp interpolation" optimization:
// A receiver SHOULD memorize the capture system (i.e. CSRC/SSRC), capture
diff --git a/modules/rtp_rtcp/source/absolute_capture_time_sender.h b/modules/rtp_rtcp/source/absolute_capture_time_sender.h
index 348a283..3deff3d 100644
--- a/modules/rtp_rtcp/source/absolute_capture_time_sender.h
+++ b/modules/rtp_rtcp/source/absolute_capture_time_sender.h
@@ -22,7 +22,7 @@
namespace webrtc {
//
-// Helper class for sending the |AbsoluteCaptureTime| header extension.
+// Helper class for sending the `AbsoluteCaptureTime` header extension.
//
// Supports the "timestamp interpolation" optimization:
// A sender SHOULD save bandwidth by not sending abs-capture-time with every
diff --git a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h
index 742e7d5..4aeb430 100644
--- a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h
+++ b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h
@@ -69,7 +69,7 @@
void SetMediaHasBeenSent(bool media_sent) RTC_LOCKS_EXCLUDED(lock_);
void SetTimestampOffset(uint32_t timestamp) RTC_LOCKS_EXCLUDED(lock_);
- // For each sequence number in |sequence_number|, recall the last RTP packet
+ // For each sequence number in `sequence_number`, recall the last RTP packet
// which bore it - its timestamp and whether it was the first and/or last
// packet in that frame. If all of the given sequence numbers could be
// recalled, return a vector with all of them (in corresponding order).
@@ -96,7 +96,7 @@
void UpdateOnSendPacket(int packet_id,
int64_t capture_time_ms,
uint32_t ssrc);
- // Sends packet on to |transport_|, leaving the RTP module.
+ // Sends packet on to `transport_`, leaving the RTP module.
bool SendPacketToNetwork(const RtpPacketToSend& packet,
const PacketOptions& options,
const PacedPacketInfo& pacing_info);
diff --git a/modules/rtp_rtcp/source/fec_private_tables_bursty.h b/modules/rtp_rtcp/source/fec_private_tables_bursty.h
index 5d67292..217d950 100644
--- a/modules/rtp_rtcp/source/fec_private_tables_bursty.h
+++ b/modules/rtp_rtcp/source/fec_private_tables_bursty.h
@@ -20,7 +20,7 @@
// packets, all "consecutive" losses of size <= m are completely recoverable.
// By consecutive losses we mean consecutive with respect to the sequence
// number ordering of the list (media and FEC) of packets. The difference
-// between these masks (|kFecMaskBursty|) and |kFecMaskRandom| type, defined
+// between these masks (`kFecMaskBursty`) and `kFecMaskRandom` type, defined
// in fec_private_tables.h, is more significant for longer codes
// (i.e., more packets/symbols in the code, so larger (k,m), i.e., k > 4,
// m > 3).
diff --git a/modules/rtp_rtcp/source/fec_test_helper.h b/modules/rtp_rtcp/source/fec_test_helper.h
index b661fa8..7a24ecf 100644
--- a/modules/rtp_rtcp/source/fec_test_helper.h
+++ b/modules/rtp_rtcp/source/fec_test_helper.h
@@ -38,7 +38,7 @@
Random* random);
~MediaPacketGenerator();
- // Construct the media packets, up to |num_media_packets| packets.
+ // Construct the media packets, up to `num_media_packets` packets.
ForwardErrorCorrection::PacketList ConstructMediaPackets(
int num_media_packets,
uint16_t start_seq_num);
@@ -72,7 +72,7 @@
std::unique_ptr<AugmentedPacket> NextPacket(size_t offset, size_t length);
protected:
- // Given |header|, writes the appropriate RTP header fields in |data|.
+ // Given `header`, writes the appropriate RTP header fields in `data`.
static void WriteRtpHeader(const RTPHeader& header, uint8_t* data);
// Number of packets left to generate, in the current frame.
diff --git a/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc b/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
index 40426f1..59541c4 100644
--- a/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
+++ b/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
@@ -26,7 +26,7 @@
constexpr size_t kMaxMediaPackets = 48; // Since we are reusing ULPFEC masks.
// Maximum number of media packets tracked by FEC decoder.
-// Maintain a sufficiently larger tracking window than |kMaxMediaPackets|
+// Maintain a sufficiently larger tracking window than `kMaxMediaPackets`
// to account for packet reordering in pacer/ network.
constexpr size_t kMaxTrackedMediaPackets = 4 * kMaxMediaPackets;
diff --git a/modules/rtp_rtcp/source/flexfec_receiver.cc b/modules/rtp_rtcp/source/flexfec_receiver.cc
index 28c8b26..e01b920 100644
--- a/modules/rtp_rtcp/source/flexfec_receiver.cc
+++ b/modules/rtp_rtcp/source/flexfec_receiver.cc
@@ -62,7 +62,7 @@
// If this packet was recovered, it might be originating from
// ProcessReceivedPacket in this object. To avoid lifetime issues with
- // |recovered_packets_|, we therefore break the cycle here.
+ // `recovered_packets_`, we therefore break the cycle here.
// This might reduce decoding efficiency a bit, since we can't disambiguate
// recovered packets by RTX from recovered packets by FlexFEC.
if (packet.recovered())
diff --git a/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc b/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc
index 7261280..54ed11d 100644
--- a/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc
+++ b/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc
@@ -66,12 +66,12 @@
ForwardErrorCorrection::CreateFlexfec(kFlexfecSsrc, kMediaSsrc)),
packet_generator_(kMediaSsrc, kFlexfecSsrc) {}
- // Generates |num_media_packets| corresponding to a single frame.
+ // Generates `num_media_packets` corresponding to a single frame.
void PacketizeFrame(size_t num_media_packets,
size_t frame_offset,
PacketList* media_packets);
- // Generates |num_fec_packets| FEC packets, given |media_packets|.
+ // Generates `num_fec_packets` FEC packets, given `media_packets`.
std::list<Packet*> EncodeFec(const PacketList& media_packets,
size_t num_fec_packets);
@@ -470,7 +470,7 @@
FlexfecReceiver* receiver_;
} loopback_recovered_packet_receiver;
- // Feed recovered packets back into |receiver|.
+ // Feed recovered packets back into `receiver`.
FlexfecReceiver receiver(Clock::GetRealTimeClock(), kFlexfecSsrc, kMediaSsrc,
&loopback_recovered_packet_receiver);
loopback_recovered_packet_receiver.SetReceiver(&receiver);
@@ -594,7 +594,7 @@
bool deep_recursion_;
} loopback_recovered_packet_receiver;
- // Feed recovered packets back into |receiver|.
+ // Feed recovered packets back into `receiver`.
FlexfecReceiver receiver(Clock::GetRealTimeClock(), kFlexfecSsrc, kMediaSsrc,
&loopback_recovered_packet_receiver);
loopback_recovered_packet_receiver.SetReceiver(&receiver);
@@ -670,7 +670,7 @@
PacketizeFrame(kNumMediaPacketsPerFrame, i, &media_packets);
}
- // Receive first (|kFirstFrameNumMediaPackets| + 192) media packets.
+ // Receive first (`kFirstFrameNumMediaPackets` + 192) media packets.
// Simulate an old FEC packet by separating it from its encoded media
// packets by at least 192 packets.
auto media_it = media_packets.begin();
diff --git a/modules/rtp_rtcp/source/forward_error_correction.cc b/modules/rtp_rtcp/source/forward_error_correction.cc
index da8025d..989fb3d 100644
--- a/modules/rtp_rtcp/source/forward_error_correction.cc
+++ b/modules/rtp_rtcp/source/forward_error_correction.cc
@@ -176,7 +176,7 @@
}
packet_mask_size_ = internal::PacketMaskSize(num_mask_bits);
- // Write FEC packets to |generated_fec_packets_|.
+ // Write FEC packets to `generated_fec_packets_`.
GenerateFecPayloads(media_packets, num_fec_packets);
// TODO(brandtr): Generalize this when multistream protection support is
// added.
@@ -219,7 +219,7 @@
while (media_packets_it != media_packets.end()) {
Packet* const media_packet = media_packets_it->get();
const uint8_t* media_packet_data = media_packet->data.cdata();
- // Should |media_packet| be protected by |fec_packet|?
+ // Should `media_packet` be protected by `fec_packet`?
if (packet_masks_[pkt_mask_idx] & (1 << (7 - media_pkt_idx))) {
size_t media_payload_length =
media_packet->data.size() - kRtpHeaderSize;
@@ -391,12 +391,12 @@
void ForwardErrorCorrection::UpdateCoveringFecPackets(
const RecoveredPacket& packet) {
for (auto& fec_packet : received_fec_packets_) {
- // Is this FEC packet protecting the media packet |packet|?
+ // Is this FEC packet protecting the media packet `packet`?
auto protected_it = absl::c_lower_bound(
fec_packet->protected_packets, &packet, SortablePacket::LessThan());
if (protected_it != fec_packet->protected_packets.end() &&
(*protected_it)->seq_num == packet.seq_num) {
- // Found an FEC packet which is protecting |packet|.
+ // Found an FEC packet which is protecting `packet`.
(*protected_it)->pkt = packet.pkt;
}
}
@@ -481,8 +481,8 @@
ProtectedPacketList* protected_packets = &fec_packet->protected_packets;
std::vector<RecoveredPacket*> recovered_protected_packets;
- // Find intersection between the (sorted) containers |protected_packets|
- // and |recovered_packets|, i.e. all protected packets that have already
+ // Find intersection between the (sorted) containers `protected_packets`
+ // and `recovered_packets`, i.e. all protected packets that have already
// been recovered. Update the corresponding protected packets to point to
// the recovered packets.
auto it_p = protected_packets->cbegin();
@@ -506,16 +506,16 @@
const ReceivedPacket& received_packet,
RecoveredPacketList* recovered_packets) {
// Discard old FEC packets such that the sequence numbers in
- // |received_fec_packets_| span at most 1/2 of the sequence number space.
- // This is important for keeping |received_fec_packets_| sorted, and may
+ // `received_fec_packets_` span at most 1/2 of the sequence number space.
+ // This is important for keeping `received_fec_packets_` sorted, and may
// also reduce the possibility of incorrect decoding due to sequence number
// wrap-around.
if (!received_fec_packets_.empty() &&
received_packet.ssrc == received_fec_packets_.front()->ssrc) {
- // It only makes sense to detect wrap-around when |received_packet|
- // and |front_received_fec_packet| belong to the same sequence number
- // space, i.e., the same SSRC. This happens when |received_packet|
- // is a FEC packet, or if |received_packet| is a media packet and
+ // It only makes sense to detect wrap-around when `received_packet`
+ // and `front_received_fec_packet` belong to the same sequence number
+ // space, i.e., the same SSRC. This happens when `received_packet`
+ // is a FEC packet, or if `received_packet` is a media packet and
// RED+ULPFEC is used.
auto it = received_fec_packets_.begin();
while (it != received_fec_packets_.end()) {
@@ -523,7 +523,7 @@
if (seq_num_diff > kOldSequenceThreshold) {
it = received_fec_packets_.erase(it);
} else {
- // No need to keep iterating, since |received_fec_packets_| is sorted.
+ // No need to keep iterating, since `received_fec_packets_` is sorted.
break;
}
}
diff --git a/modules/rtp_rtcp/source/forward_error_correction.h b/modules/rtp_rtcp/source/forward_error_correction.h
index b97693d..d07bb8e 100644
--- a/modules/rtp_rtcp/source/forward_error_correction.h
+++ b/modules/rtp_rtcp/source/forward_error_correction.h
@@ -62,8 +62,8 @@
// TODO(holmer): Refactor into a proper class.
class SortablePacket {
public:
- // Functor which returns true if the sequence number of |first|
- // is < the sequence number of |second|. Should only ever be called for
+ // Functor which returns true if the sequence number of `first`
+ // is < the sequence number of `second`. Should only ever be called for
// packets belonging to the same SSRC.
struct LessThan {
template <typename S, typename T>
@@ -76,7 +76,7 @@
// Used for the input to DecodeFec().
//
- // TODO(nisse): Delete class, instead passing |is_fec| and |pkt| as separate
+ // TODO(nisse): Delete class, instead passing `is_fec` and `pkt` as separate
// arguments.
class ReceivedPacket : public SortablePacket {
public:
@@ -197,14 +197,14 @@
std::list<Packet*>* fec_packets);
// Decodes a list of received media and FEC packets. It will parse the
- // |received_packets|, storing FEC packets internally, and move
- // media packets to |recovered_packets|. The recovered list will be
+ // `received_packets`, storing FEC packets internally, and move
+ // media packets to `recovered_packets`. The recovered list will be
// sorted by ascending sequence number and have duplicates removed.
// The function should be called as new packets arrive, and
- // |recovered_packets| will be progressively assembled with each call.
- // When the function returns, |received_packets| will be empty.
+ // `recovered_packets` will be progressively assembled with each call.
+ // When the function returns, `received_packets` will be empty.
//
- // The caller will allocate packets submitted through |received_packets|.
+ // The caller will allocate packets submitted through `received_packets`.
// The function will handle allocation of recovered packets.
//
// Input: received_packets List of new received packets, of type
@@ -229,7 +229,7 @@
// accounted for as packet overhead.
size_t MaxPacketOverhead() const;
- // Reset internal states from last frame and clear |recovered_packets|.
+ // Reset internal states from last frame and clear `recovered_packets`.
// Frees all memory allocated by this class.
void ResetState(RecoveredPacketList* recovered_packets);
@@ -245,11 +245,11 @@
uint32_t protected_media_ssrc);
private:
- // Analyzes |media_packets| for holes in the sequence and inserts zero columns
- // into the |packet_mask| where those holes are found. Zero columns means that
+ // Analyzes `media_packets` for holes in the sequence and inserts zero columns
+ // into the `packet_mask` where those holes are found. Zero columns means that
// those packets will have no protection.
// Returns the number of bits used for one row of the new packet mask.
- // Requires that |packet_mask| has at least 6 * |num_fec_packets| bytes
+ // Requires that `packet_mask` has at least 6 * `num_fec_packets` bytes
// allocated.
int InsertZerosInPacketMasks(const PacketList& media_packets,
size_t num_fec_packets);
@@ -264,12 +264,12 @@
uint32_t media_ssrc,
uint16_t seq_num_base);
- // Inserts the |received_packet| into the internal received FEC packet list
- // or into |recovered_packets|.
+ // Inserts the `received_packet` into the internal received FEC packet list
+ // or into `recovered_packets`.
void InsertPacket(const ReceivedPacket& received_packet,
RecoveredPacketList* recovered_packets);
- // Inserts the |received_packet| into |recovered_packets|. Deletes duplicates.
+ // Inserts the `received_packet` into `recovered_packets`. Deletes duplicates.
void InsertMediaPacket(RecoveredPacketList* recovered_packets,
const ReceivedPacket& received_packet);
@@ -280,11 +280,11 @@
// packets covered by the FEC packet.
void UpdateCoveringFecPackets(const RecoveredPacket& packet);
- // Insert |received_packet| into internal FEC list. Deletes duplicates.
+ // Insert `received_packet` into internal FEC list. Deletes duplicates.
void InsertFecPacket(const RecoveredPacketList& recovered_packets,
const ReceivedPacket& received_packet);
- // Assigns pointers to already recovered packets covered by |fec_packet|.
+ // Assigns pointers to already recovered packets covered by `fec_packet`.
static void AssignRecoveredPackets(
const RecoveredPacketList& recovered_packets,
ReceivedFecPacket* fec_packet);
@@ -298,14 +298,14 @@
static bool StartPacketRecovery(const ReceivedFecPacket& fec_packet,
RecoveredPacket* recovered_packet);
- // Performs XOR between the first 8 bytes of |src| and |dst| and stores
- // the result in |dst|. The 3rd and 4th bytes are used for storing
+ // Performs XOR between the first 8 bytes of `src` and `dst` and stores
+ // the result in `dst`. The 3rd and 4th bytes are used for storing
// the length recovery field.
static void XorHeaders(const Packet& src, Packet* dst);
- // Performs XOR between the payloads of |src| and |dst| and stores the result
- // in |dst|. The parameter |dst_offset| determines at what byte the
- // XOR operation starts in |dst|. In total, |payload_length| bytes are XORed.
+ // Performs XOR between the payloads of `src` and `dst` and stores the result
+ // in `dst`. The parameter `dst_offset` determines at what byte the
+ // XOR operation starts in `dst`. In total, `payload_length` bytes are XORed.
static void XorPayloads(const Packet& src,
size_t payload_length,
size_t dst_offset,
@@ -320,13 +320,13 @@
static bool RecoverPacket(const ReceivedFecPacket& fec_packet,
RecoveredPacket* recovered_packet);
- // Get the number of missing media packets which are covered by |fec_packet|.
+ // Get the number of missing media packets which are covered by `fec_packet`.
// An FEC packet can recover at most one packet, and if zero packets are
// missing the FEC packet can be discarded. This function returns 2 when two
// or more packets are missing.
static int NumCoveredPacketsMissing(const ReceivedFecPacket& fec_packet);
- // Discards old packets in |recovered_packets|, which are no longer relevant
+ // Discards old packets in `recovered_packets`, which are no longer relevant
// for recovering lost packets.
void DiscardOldRecoveredPackets(RecoveredPacketList* recovered_packets);
@@ -347,7 +347,7 @@
// Arrays used to avoid dynamically allocating memory when generating
// the packet masks.
- // (There are never more than |kUlpfecMaxMediaPackets| FEC packets generated.)
+ // (There are never more than `kUlpfecMaxMediaPackets` FEC packets generated.)
uint8_t packet_masks_[kUlpfecMaxMediaPackets * kUlpfecMaxPacketMaskSize];
uint8_t tmp_packet_masks_[kUlpfecMaxMediaPackets * kUlpfecMaxPacketMaskSize];
size_t packet_mask_size_;
diff --git a/modules/rtp_rtcp/source/forward_error_correction_internal.cc b/modules/rtp_rtcp/source/forward_error_correction_internal.cc
index 2a056a6..400b640 100644
--- a/modules/rtp_rtcp/source/forward_error_correction_internal.cc
+++ b/modules/rtp_rtcp/source/forward_error_correction_internal.cc
@@ -212,7 +212,7 @@
static_cast<size_t>(num_fec_packets * mask_length)};
}
-// If |num_media_packets| is larger than the maximum allowed by |fec_mask_type|
+// If `num_media_packets` is larger than the maximum allowed by `fec_mask_type`
// for the bursty type, or the random table is explicitly asked for, then the
// random type is selected. Otherwise the bursty table callback is returned.
const uint8_t* PacketMaskTable::PickTable(FecMaskType fec_mask_type,
@@ -393,8 +393,8 @@
}
}
-// This algorithm is tailored to look up data in the |kPacketMaskRandomTbl| and
-// |kPacketMaskBurstyTbl| tables. These tables only cover fec code for up to 12
+// This algorithm is tailored to look up data in the `kPacketMaskRandomTbl` and
+// `kPacketMaskBurstyTbl` tables. These tables only cover fec code for up to 12
// media packets. Starting from 13 media packets, the fec code will be generated
// at runtime. The format of those arrays is that they're essentially a 3
// dimensional array with the following dimensions: * media packet
diff --git a/modules/rtp_rtcp/source/forward_error_correction_internal.h b/modules/rtp_rtcp/source/forward_error_correction_internal.h
index ed93f52..31acf73 100644
--- a/modules/rtp_rtcp/source/forward_error_correction_internal.h
+++ b/modules/rtp_rtcp/source/forward_error_correction_internal.h
@@ -71,7 +71,7 @@
// protection scenario.
// \param[in] use_unequal_protection Enables unequal protection: allocates
// more protection to the num_imp_packets.
-// \param[in] mask_table An instance of the |PacketMaskTable|
+// \param[in] mask_table An instance of the `PacketMaskTable`
// class, which contains the type of FEC
// packet mask used, and a pointer to the
// corresponding packet masks.
@@ -89,9 +89,9 @@
// that will be covered.
size_t PacketMaskSize(size_t num_sequence_numbers);
-// Inserts |num_zeros| zero columns into |new_mask| at position
-// |new_bit_index|. If the current byte of |new_mask| can't fit all zeros, the
-// byte will be filled with zeros from |new_bit_index|, but the next byte will
+// Inserts `num_zeros` zero columns into `new_mask` at position
+// `new_bit_index`. If the current byte of `new_mask` can't fit all zeros, the
+// byte will be filled with zeros from `new_bit_index`, but the next byte will
// be untouched.
void InsertZeroColumns(int num_zeros,
uint8_t* new_mask,
@@ -100,12 +100,12 @@
int new_bit_index);
// Copies the left most bit column from the byte pointed to by
-// |old_bit_index| in |old_mask| to the right most column of the byte pointed
-// to by |new_bit_index| in |new_mask|. |old_mask_bytes| and |new_mask_bytes|
-// represent the number of bytes used per row for each mask. |num_fec_packets|
+// `old_bit_index` in `old_mask` to the right most column of the byte pointed
+// to by `new_bit_index` in `new_mask`. `old_mask_bytes` and `new_mask_bytes`
+// represent the number of bytes used per row for each mask. `num_fec_packets`
// represent the number of rows of the masks.
-// The copied bit is shifted out from |old_mask| and is shifted one step to
-// the left in |new_mask|. |new_mask| will contain "xxxx xxn0" after this
+// The copied bit is shifted out from `old_mask` and is shifted one step to
+// the left in `new_mask`. `new_mask` will contain "xxxx xxn0" after this
// operation, where x are previously inserted bits and n is the new bit.
void CopyColumn(uint8_t* new_mask,
int new_mask_bytes,
diff --git a/modules/rtp_rtcp/source/packet_sequencer.cc b/modules/rtp_rtcp/source/packet_sequencer.cc
index db108d4..a0c27de 100644
--- a/modules/rtp_rtcp/source/packet_sequencer.cc
+++ b/modules/rtp_rtcp/source/packet_sequencer.cc
@@ -86,7 +86,7 @@
void PacketSequencer::UpdateLastPacketState(const RtpPacketToSend& packet) {
// Remember marker bit to determine if padding can be inserted with
- // sequence number following |packet|.
+ // sequence number following `packet`.
last_packet_marker_bit_ = packet.Marker();
// Remember media payload type to use in the padding packet if rtx is
// disabled.
diff --git a/modules/rtp_rtcp/source/receive_statistics_impl.cc b/modules/rtp_rtcp/source/receive_statistics_impl.cc
index f5c3eaf..b16f122 100644
--- a/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -64,7 +64,7 @@
bool StreamStatisticianImpl::UpdateOutOfOrder(const RtpPacketReceived& packet,
int64_t sequence_number,
int64_t now_ms) {
- // Check if |packet| is second packet of a stream restart.
+ // Check if `packet` is second packet of a stream restart.
if (received_seq_out_of_order_) {
// Count the previous packet as a received; it was postponed below.
--cumulative_loss_;
@@ -75,7 +75,7 @@
// Ignore sequence number gap caused by stream restart for packet loss
// calculation, by setting received_seq_max_ to the sequence number just
// before the out-of-order seqno. This gives a net zero change of
- // |cumulative_loss_|, for the two packets interpreted as a stream reset.
+ // `cumulative_loss_`, for the two packets interpreted as a stream reset.
//
// Fraction loss for the next report may get a bit off, since we don't
// update last_report_seq_max_ and last_report_cumulative_loss_ in a
@@ -92,10 +92,10 @@
// for a stream restart.
received_seq_out_of_order_ = packet.SequenceNumber();
// Postpone counting this as a received packet until we know how to update
- // |received_seq_max_|, otherwise we temporarily decrement
- // |cumulative_loss_|. The
+ // `received_seq_max_`, otherwise we temporarily decrement
+ // `cumulative_loss_`. The
// ReceiveStatisticsTest.StreamRestartDoesntCountAsLoss test expects
- // |cumulative_loss_| to be unchanged by the reception of the first packet
+ // `cumulative_loss_` to be unchanged by the reception of the first packet
// after stream reset.
++cumulative_loss_;
return true;
diff --git a/modules/rtp_rtcp/source/rtcp_packet/loss_notification.cc b/modules/rtp_rtcp/source/rtcp_packet/loss_notification.cc
index 08c75dd..0817846 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/loss_notification.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/loss_notification.cc
@@ -63,7 +63,7 @@
const size_t index_end = *index + BlockLength();
- // Note: |index| updated by the function below.
+ // Note: `index` updated by the function below.
CreateHeader(Psfb::kAfbMessageType, kPacketType, HeaderLength(), packet,
index);
diff --git a/modules/rtp_rtcp/source/rtcp_packet/loss_notification.h b/modules/rtp_rtcp/source/rtcp_packet/loss_notification.h
index 99f6d12..b23008c 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/loss_notification.h
+++ b/modules/rtp_rtcp/source/rtcp_packet/loss_notification.h
@@ -42,8 +42,8 @@
// Set all of the values transmitted by the loss notification message.
// If the values may not be represented by a loss notification message,
// false is returned, and no change is made to the object; this happens
- // when |last_recieved| is ahead of |last_decoded| by more than 0x7fff.
- // This is because |last_recieved| is represented on the wire as a delta,
+ // when `last_recieved` is ahead of `last_decoded` by more than 0x7fff.
+ // This is because `last_recieved` is represented on the wire as a delta,
// and only 15 bits are available for that delta.
ABSL_MUST_USE_RESULT
bool Set(uint16_t last_decoded,
diff --git a/modules/rtp_rtcp/source/rtcp_packet/loss_notification_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/loss_notification_unittest.cc
index 6d74225..c38e7f4 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/loss_notification_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/loss_notification_unittest.cc
@@ -80,7 +80,7 @@
test::ParseSinglePacket(packet, packet_length_bytes, &loss_notification));
// Show that after shaving off a word, the packet is no longer parsable.
- packet[3] -= 1; // Change the |length| field of the RTCP packet.
+ packet[3] -= 1; // Change the `length` field of the RTCP packet.
packet_length_bytes -= 4; // Effectively forget the last 32-bit word.
EXPECT_FALSE(
test::ParseSinglePacket(packet, packet_length_bytes, &loss_notification));
diff --git a/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc b/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
index 96c3cb3..c589a18 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
@@ -129,7 +129,7 @@
}
RTC_DCHECK_GE(size_, kMaxTwoBitCapacity);
uint16_t chunk = EncodeTwoBit(kMaxTwoBitCapacity);
- // Remove |kMaxTwoBitCapacity| encoded delta sizes:
+ // Remove `kMaxTwoBitCapacity` encoded delta sizes:
// Shift remaining delta sizes and recalculate all_same_ && has_large_delta_.
size_ -= kMaxTwoBitCapacity;
all_same_ = true;
@@ -153,7 +153,7 @@
return EncodeOneBit();
}
-// Appends content of the Lastchunk to |deltas|.
+// Appends content of the Lastchunk to `deltas`.
void TransportFeedback::LastChunk::AppendTo(
std::vector<DeltaSize>* deltas) const {
if (all_same_) {
@@ -441,7 +441,7 @@
last_chunk_.Decode(chunk, status_count - delta_sizes.size());
last_chunk_.AppendTo(&delta_sizes);
}
- // Last chunk is stored in the |last_chunk_|.
+ // Last chunk is stored in the `last_chunk_`.
encoded_chunks_.pop_back();
RTC_DCHECK_EQ(delta_sizes.size(), status_count);
num_seq_no_ = status_count;
diff --git a/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h b/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
index c2a4d43..e30d338 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
+++ b/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
@@ -54,7 +54,7 @@
TransportFeedback();
- // If |include_timestamps| is set to false, the created packet will not
+ // If `include_timestamps` is set to false, the created packet will not
// contain the receive delta block.
explicit TransportFeedback(bool include_timestamps,
bool include_lost = false);
@@ -80,7 +80,7 @@
int64_t GetBaseTimeUs() const;
TimeDelta GetBaseTime() const;
- // Get the unwrapped delta between current base time and |prev_timestamp_us|.
+ // Get the unwrapped delta between current base time and `prev_timestamp_us`.
int64_t GetBaseDeltaUs(int64_t prev_timestamp_us) const;
TimeDelta GetBaseDelta(TimeDelta prev_timestamp) const;
@@ -116,9 +116,9 @@
bool Empty() const;
void Clear();
// Return if delta sizes still can be encoded into single chunk with added
- // |delta_size|.
+ // `delta_size`.
bool CanAdd(DeltaSize delta_size) const;
- // Add |delta_size|, assumes |CanAdd(delta_size)|,
+ // Add `delta_size`, assumes `CanAdd(delta_size)`,
void Add(DeltaSize delta_size);
// Encode chunk as large as possible removing encoded delta sizes.
@@ -127,9 +127,9 @@
// Encode all stored delta_sizes into single chunk, pad with 0s if needed.
uint16_t EncodeLast() const;
- // Decode up to |max_size| delta sizes from |chunk|.
+ // Decode up to `max_size` delta sizes from `chunk`.
void Decode(uint16_t chunk, size_t max_size);
- // Appends content of the Lastchunk to |deltas|.
+ // Appends content of the Lastchunk to `deltas`.
void AppendTo(std::vector<DeltaSize>* deltas) const;
private:
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
index a8e1dc5..762255c 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -68,7 +68,7 @@
constexpr TimeDelta kDefaultVideoReportInterval = TimeDelta::Seconds(1);
constexpr TimeDelta kDefaultAudioReportInterval = TimeDelta::Seconds(5);
-// Returns true if the |timestamp| has exceeded the |interval *
+// Returns true if the `timestamp` has exceeded the |interval *
// kRrTimeoutIntervals| period and was reset (set to PlusInfinity()). Returns
// false if the timer was either already reset or if it has not expired.
bool ResetTimestampIfExpired(const Timestamp now,
@@ -127,7 +127,7 @@
uint32_t remote_ssrc = 0;
std::vector<uint16_t> nack_sequence_numbers;
- // TODO(hbos): Remove |report_blocks| in favor of |report_block_datas|.
+ // TODO(hbos): Remove `report_blocks` in favor of `report_block_datas`.
ReportBlockList report_blocks;
std::vector<ReportBlockData> report_block_datas;
int64_t rtt_ms = 0;
@@ -636,7 +636,7 @@
// Receiver rtp_rtcp module is not expected to calculate rtt using
// Sender Reports even if it accidentally can.
- // TODO(nisse): Use this way to determine the RTT only when |receiver_only_|
+ // TODO(nisse): Use this way to determine the RTT only when `receiver_only_`
// is false. However, that currently breaks the tests of the
// googCaptureStartNtpTimeMs stat for audio receive streams. To fix, either
// delete all dependencies on RTT measurements for audio receive streams, or
@@ -956,7 +956,7 @@
auto* entry = &tmmbr_info->tmmbr[sender_ssrc];
entry->tmmbr_item = rtcp::TmmbItem(sender_ssrc, request.bitrate_bps(),
request.packet_overhead());
- // FindOrCreateTmmbrInfo always sets |last_time_received_ms| to
+ // FindOrCreateTmmbrInfo always sets `last_time_received_ms` to
// |clock_->TimeInMilliseconds()|.
entry->last_updated_ms = tmmbr_info->last_time_received_ms;
diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc
index b6f44e5..597bb3c 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -821,7 +821,7 @@
if (!receive_statistics_)
return result;
- // TODO(danilchap): Support sending more than |RTCP_MAX_REPORT_BLOCKS| per
+ // TODO(danilchap): Support sending more than `RTCP_MAX_REPORT_BLOCKS` per
// compound rtcp packet when single rtcp module is used for multiple media
// streams.
result = receive_statistics_->RtcpReportBlocks(RTCP_MAX_REPORT_BLOCKS);
diff --git a/modules/rtp_rtcp/source/rtcp_sender.h b/modules/rtp_rtcp/source/rtcp_sender.h
index 133eb83..00b58b4 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/modules/rtp_rtcp/source/rtcp_sender.h
@@ -54,7 +54,7 @@
// a video version.
bool audio = false;
// SSRCs for media and retransmission, respectively.
- // FlexFec SSRC is fetched from |flexfec_sender|.
+ // FlexFec SSRC is fetched from `flexfec_sender`.
uint32_t local_media_ssrc = 0;
// The clock to use to read time. If nullptr then system clock will be used.
Clock* clock = nullptr;
@@ -225,7 +225,7 @@
void BuildNACK(const RtcpContext& context, PacketSender& sender)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
- // |duration| being TimeDelta::Zero() means schedule immediately.
+ // `duration` being TimeDelta::Zero() means schedule immediately.
void SetNextRtcpSendEvaluationDuration(TimeDelta duration)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
diff --git a/modules/rtp_rtcp/source/rtcp_transceiver.h b/modules/rtp_rtcp/source/rtcp_transceiver.h
index 52f4610..862d4be 100644
--- a/modules/rtp_rtcp/source/rtcp_transceiver.h
+++ b/modules/rtp_rtcp/source/rtcp_transceiver.h
@@ -43,7 +43,7 @@
// No other methods can be called.
// Note that interfaces provided in constructor or registered with AddObserver
// still might be used by the transceiver on the task queue
- // until |on_destroyed| runs.
+ // until `on_destroyed` runs.
void Stop(std::function<void()> on_destroyed);
// Registers observer to be notified about incoming rtcp packets.
@@ -51,7 +51,7 @@
void AddMediaReceiverRtcpObserver(uint32_t remote_ssrc,
MediaReceiverRtcpObserver* observer);
// Deregisters the observer. Might return before observer is deregistered.
- // Runs |on_removed| when observer is deregistered.
+ // Runs `on_removed` when observer is deregistered.
void RemoveMediaReceiverRtcpObserver(uint32_t remote_ssrc,
MediaReceiverRtcpObserver* observer,
std::function<void()> on_removed);
diff --git a/modules/rtp_rtcp/source/rtcp_transceiver_config.h b/modules/rtp_rtcp/source/rtcp_transceiver_config.h
index 0501b9a..5d55990 100644
--- a/modules/rtp_rtcp/source/rtcp_transceiver_config.h
+++ b/modules/rtp_rtcp/source/rtcp_transceiver_config.h
@@ -86,7 +86,7 @@
//
// Tuning parameters.
//
- // Initial state if |outgoing_transport| ready to accept packets.
+ // Initial state if `outgoing_transport` ready to accept packets.
bool initial_ready_to_send = true;
// Delay before 1st periodic compound packet.
int initial_report_delay_ms = 500;
diff --git a/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.h b/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.h
index abef371..6472216 100644
--- a/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.h
+++ b/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.h
@@ -34,7 +34,7 @@
bool ParseSuccessful() { return !parsing_failed_; }
private:
- // Reads bits from |buffer_|. If it fails, returns 0 and marks parsing as
+ // Reads bits from `buffer_`. If it fails, returns 0 and marks parsing as
// failed, but doesn't stop the parsing.
uint32_t ReadBits(size_t bit_count);
uint32_t ReadNonSymmetric(size_t num_values);
diff --git a/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.h b/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.h
index 99fefec..568e0a8 100644
--- a/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.h
+++ b/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.h
@@ -22,19 +22,19 @@
namespace webrtc {
class RtpDependencyDescriptorWriter {
public:
- // Assumes |structure| and |descriptor| are valid and
- // |descriptor| matches the |structure|.
+ // Assumes `structure` and `descriptor` are valid and
+ // `descriptor` matches the `structure`.
RtpDependencyDescriptorWriter(rtc::ArrayView<uint8_t> data,
const FrameDependencyStructure& structure,
std::bitset<32> active_chains,
const DependencyDescriptor& descriptor);
// Serializes DependencyDescriptor rtp header extension.
- // Returns false if |data| is too small to serialize the |descriptor|.
+ // Returns false if `data` is too small to serialize the `descriptor`.
bool Write();
// Returns minimum number of bits needed to serialize descriptor with respect
- // to the |structure|. Returns 0 if |descriptor| can't be serialized.
+ // to the `structure`. Returns 0 if `descriptor` can't be serialized.
int ValueSizeBits() const;
private:
diff --git a/modules/rtp_rtcp/source/rtp_fec_unittest.cc b/modules/rtp_rtcp/source/rtp_fec_unittest.cc
index a90e61a..2c01a0d 100644
--- a/modules/rtp_rtcp/source/rtp_fec_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_fec_unittest.cc
@@ -32,7 +32,7 @@
constexpr size_t kMaxMediaPackets = 48;
-// Deep copies |src| to |dst|, but only keeps every Nth packet.
+// Deep copies `src` to `dst`, but only keeps every Nth packet.
void DeepCopyEveryNthPacket(const ForwardErrorCorrection::PacketList& src,
int n,
ForwardErrorCorrection::PacketList* dst) {
@@ -62,7 +62,7 @@
kMediaSsrc,
&random_) {}
- // Construct |received_packets_|: a subset of the media and FEC packets.
+ // Construct `received_packets_`: a subset of the media and FEC packets.
//
// Media packet "i" is lost if media_loss_mask_[i] = 1, received if
// media_loss_mask_[i] = 0.
@@ -70,9 +70,9 @@
// fec_loss_mask_[i] = 0.
void NetworkReceivedPackets(int* media_loss_mask, int* fec_loss_mask);
- // Add packet from |packet_list| to list of received packets, using the
- // |loss_mask|.
- // The |packet_list| may be a media packet list (is_fec = false), or a
+ // Add packet from `packet_list` to list of received packets, using the
+ // `loss_mask`.
+ // The `packet_list` may be a media packet list (is_fec = false), or a
// FEC packet list (is_fec = true).
template <typename T>
void ReceivedPackets(const T& packet_list, int* loss_mask, bool is_fec);
@@ -168,7 +168,7 @@
// Define gTest typed test to loop over both ULPFEC and FlexFEC.
// Since the tests now are parameterized, we need to access
-// member variables using |this|, thereby enforcing runtime
+// member variables using `this`, thereby enforcing runtime
// resolution.
class FlexfecForwardErrorCorrection : public ForwardErrorCorrection {
@@ -244,7 +244,7 @@
this->media_packets_ =
this->media_packet_generator_.ConstructMediaPackets(kNumMediaPackets);
- // Create |kMaxMediaPackets| sequence number difference.
+ // Create `kMaxMediaPackets` sequence number difference.
ByteWriter<uint16_t>::WriteBigEndian(
this->media_packets_.front()->data.MutableData() + 2, 1);
ByteWriter<uint16_t>::WriteBigEndian(
diff --git a/modules/rtp_rtcp/source/rtp_format.h b/modules/rtp_rtcp/source/rtp_format.h
index b593f29..19abd3f 100644
--- a/modules/rtp_rtcp/source/rtp_format.h
+++ b/modules/rtp_rtcp/source/rtp_format.h
@@ -48,11 +48,11 @@
virtual size_t NumPackets() const = 0;
// Get the next payload with payload header.
- // Write payload and set marker bit of the |packet|.
+ // Write payload and set marker bit of the `packet`.
// Returns true on success, false otherwise.
virtual bool NextPacket(RtpPacketToSend* packet) = 0;
- // Split payload_len into sum of integers with respect to |limits|.
+ // Split payload_len into sum of integers with respect to `limits`.
// Returns empty vector on failure.
static std::vector<int> SplitAboutEqually(int payload_len,
const PayloadSizeLimits& limits);
diff --git a/modules/rtp_rtcp/source/rtp_format_h264.h b/modules/rtp_rtcp/source/rtp_format_h264.h
index 7c10dd5..f658594 100644
--- a/modules/rtp_rtcp/source/rtp_format_h264.h
+++ b/modules/rtp_rtcp/source/rtp_format_h264.h
@@ -40,7 +40,7 @@
size_t NumPackets() const override;
// Get the next payload with H264 payload header.
- // Write payload and set marker bit of the |packet|.
+ // Write payload and set marker bit of the `packet`.
// Returns true on success, false otherwise.
bool NextPacket(RtpPacketToSend* rtp_packet) override;
diff --git a/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
index 9f660b7..d217196 100644
--- a/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -404,7 +404,7 @@
limits.max_payload_len - limits.last_packet_reduction_len);
}
-// Splits frame with payload size |frame_payload_size| without fragmentation,
+// Splits frame with payload size `frame_payload_size` without fragmentation,
// Returns sizes of the payloads excluding fua headers.
std::vector<int> TestFua(size_t frame_payload_size,
const RtpPacketizer::PayloadSizeLimits& limits) {
diff --git a/modules/rtp_rtcp/source/rtp_format_video_generic.h b/modules/rtp_rtcp/source/rtp_format_video_generic.h
index f388ca22..5acd691 100644
--- a/modules/rtp_rtcp/source/rtp_format_video_generic.h
+++ b/modules/rtp_rtcp/source/rtp_format_video_generic.h
@@ -35,13 +35,13 @@
public:
// Initialize with payload from encoder.
// The payload_data must be exactly one encoded generic frame.
- // Packets returned by |NextPacket| will contain the generic payload header.
+ // Packets returned by `NextPacket` will contain the generic payload header.
RtpPacketizerGeneric(rtc::ArrayView<const uint8_t> payload,
PayloadSizeLimits limits,
const RTPVideoHeader& rtp_video_header);
// Initialize with payload from encoder.
// The payload_data must be exactly one encoded generic frame.
- // Packets returned by |NextPacket| will contain raw payload without the
+ // Packets returned by `NextPacket` will contain raw payload without the
// generic payload header.
RtpPacketizerGeneric(rtc::ArrayView<const uint8_t> payload,
PayloadSizeLimits limits);
@@ -51,7 +51,7 @@
size_t NumPackets() const override;
// Get the next payload.
- // Write payload and set marker bit of the |packet|.
+ // Write payload and set marker bit of the `packet`.
// Returns true on success, false otherwise.
bool NextPacket(RtpPacketToSend* packet) override;
diff --git a/modules/rtp_rtcp/source/rtp_format_vp8.h b/modules/rtp_rtcp/source/rtp_format_vp8.h
index 4250736..2100928 100644
--- a/modules/rtp_rtcp/source/rtp_format_vp8.h
+++ b/modules/rtp_rtcp/source/rtp_format_vp8.h
@@ -53,7 +53,7 @@
size_t NumPackets() const override;
// Get the next payload with VP8 payload header.
- // Write payload and set marker bit of the |packet|.
+ // Write payload and set marker bit of the `packet`.
// Returns true on success, false otherwise.
bool NextPacket(RtpPacketToSend* packet) override;
diff --git a/modules/rtp_rtcp/source/rtp_format_vp9.h b/modules/rtp_rtcp/source/rtp_format_vp9.h
index 5e2d52a..02458ae 100644
--- a/modules/rtp_rtcp/source/rtp_format_vp9.h
+++ b/modules/rtp_rtcp/source/rtp_format_vp9.h
@@ -36,7 +36,7 @@
class RtpPacketizerVp9 : public RtpPacketizer {
public:
- // The |payload| must be one encoded VP9 layer frame.
+ // The `payload` must be one encoded VP9 layer frame.
RtpPacketizerVp9(rtc::ArrayView<const uint8_t> payload,
PayloadSizeLimits limits,
const RTPVideoHeaderVP9& hdr);
@@ -46,13 +46,13 @@
size_t NumPackets() const override;
// Gets the next payload with VP9 payload header.
- // Write payload and set marker bit of the |packet|.
+ // Write payload and set marker bit of the `packet`.
// Returns true on success, false otherwise.
bool NextPacket(RtpPacketToSend* packet) override;
private:
// Writes the payload descriptor header.
- // |layer_begin| and |layer_end| indicates the postision of the packet in
+ // `layer_begin` and `layer_end` indicates the postision of the packet in
// the layer frame. Returns false on failure.
bool WriteHeader(bool layer_begin,
bool layer_end,
diff --git a/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc b/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
index 0dc6566..e18b8a8 100644
--- a/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
@@ -501,7 +501,7 @@
RtpPacketizer::PayloadSizeLimits limits;
limits.max_payload_len = 8;
// Calculated by hand. One packet can contain
- // |kPacketSize| - |kVp9MinDiscriptorSize| = 6 bytes of the frame payload,
+ // `kPacketSize` - `kVp9MinDiscriptorSize` = 6 bytes of the frame payload,
// thus to fit 10 bytes two packets are required.
const size_t kMinNumberOfPackets = 2;
const uint8_t kFrame[kFrameSize] = {7};
@@ -526,7 +526,7 @@
limits.last_packet_reduction_len = 5;
// Calculated by hand. VP9 payload descriptor is 2 bytes. Like in the test
// above, 1 packet is not enough. 2 packets can contain
- // 2*(|kPacketSize| - |kVp9MinDiscriptorSize|) - |kLastPacketReductionLen| = 7
+ // 2*(`kPacketSize` - `kVp9MinDiscriptorSize`) - `kLastPacketReductionLen` = 7
// But three packets are enough, since they have capacity of 3*(8-2)-5=13
// bytes.
const size_t kMinNumberOfPackets = 3;
diff --git a/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc b/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc
index ca46fa6..49ec4a1 100644
--- a/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc
+++ b/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc
@@ -20,7 +20,7 @@
// In version 00, the flags F and L in the first byte correspond to
// kFlagFirstSubframeV00 and kFlagLastSubframeV00. In practice, they were
-// always set to |true|.
+// always set to `true`.
constexpr uint8_t kFlagFirstSubframeV00 = 0x20;
constexpr uint8_t kFlagLastSubframeV00 = 0x10;
diff --git a/modules/rtp_rtcp/source/rtp_header_extension_map.cc b/modules/rtp_rtcp/source/rtp_header_extension_map.cc
index 0b5ba47..d1eee22 100644
--- a/modules/rtp_rtcp/source/rtp_header_extension_map.cc
+++ b/modules/rtp_rtcp/source/rtp_header_extension_map.cc
@@ -145,7 +145,7 @@
}
if (registered_type !=
- kInvalidType) { // |id| used by another extension type.
+ kInvalidType) { // `id` used by another extension type.
RTC_LOG(LS_WARNING) << "Failed to register extension uri:'" << uri
<< "', id:" << id
<< ". Id already in use by extension type "
diff --git a/modules/rtp_rtcp/source/rtp_header_extension_size.cc b/modules/rtp_rtcp/source/rtp_header_extension_size.cc
index 7719922..4acbcf4 100644
--- a/modules/rtp_rtcp/source/rtp_header_extension_size.cc
+++ b/modules/rtp_rtcp/source/rtp_header_extension_size.cc
@@ -26,7 +26,7 @@
int id = registered_extensions.GetId(extension.type);
if (id == RtpHeaderExtensionMap::kInvalidId)
continue;
- // All extensions should use same size header. Check if the |extension|
+ // All extensions should use same size header. Check if the `extension`
// forces to switch to two byte header that allows larger id and value size.
if (id > RtpExtension::kOneByteHeaderExtensionMaxId ||
extension.value_size >
diff --git a/modules/rtp_rtcp/source/rtp_header_extension_size.h b/modules/rtp_rtcp/source/rtp_header_extension_size.h
index 8047fcc..1fb2eb2 100644
--- a/modules/rtp_rtcp/source/rtp_header_extension_size.h
+++ b/modules/rtp_rtcp/source/rtp_header_extension_size.h
@@ -22,8 +22,8 @@
};
// Calculates rtp header extension size in bytes assuming packet contain
-// all |extensions| with provided |value_size|.
-// Counts only extensions present among |registered_extensions|.
+// all `extensions` with provided `value_size`.
+// Counts only extensions present among `registered_extensions`.
int RtpHeaderExtensionSize(rtc::ArrayView<const RtpExtensionSize> extensions,
const RtpHeaderExtensionMap& registered_extensions);
diff --git a/modules/rtp_rtcp/source/rtp_header_extensions.cc b/modules/rtp_rtcp/source/rtp_header_extensions.cc
index 1dd4f54..12359a4 100644
--- a/modules/rtp_rtcp/source/rtp_header_extensions.cc
+++ b/modules/rtp_rtcp/source/rtp_header_extensions.cc
@@ -315,9 +315,9 @@
// |seq count cont.|
// +-+-+-+-+-+-+-+-+
//
-// The bit |T| determines whether the feedback should include timing information
-// or not and |seq_count| determines how many packets the feedback packet should
-// cover including the current packet. If |seq_count| is zero no feedback is
+// The bit `T` determines whether the feedback should include timing information
+// or not and `seq_count` determines how many packets the feedback packet should
+// cover including the current packet. If `seq_count` is zero no feedback is
// requested.
constexpr RTPExtensionType TransportSequenceNumberV2::kId;
constexpr uint8_t TransportSequenceNumberV2::kValueSizeBytes;
@@ -344,7 +344,7 @@
(feedback_request_raw & kIncludeTimestampsBit) != 0;
uint16_t sequence_count = feedback_request_raw & ~kIncludeTimestampsBit;
- // If |sequence_count| is zero no feedback is requested.
+ // If `sequence_count` is zero no feedback is requested.
if (sequence_count != 0) {
*feedback_request = {include_timestamps, sequence_count};
}
@@ -487,7 +487,7 @@
// Video Timing.
// 6 timestamps in milliseconds counted from capture time stored in rtp header:
// encode start/finish, packetization complete, pacer exit and reserved for
-// modification by the network modification. |flags| is a bitmask and has the
+// modification by the network modification. `flags` is a bitmask and has the
// following allowed values:
// 0 = Valid data, but no flags available (backwards compatibility)
// 1 = Frame marked as timing frame due to cyclic timer.
@@ -804,7 +804,7 @@
if (data.empty() || data[0] == 0) // Valid string extension can't be empty.
return false;
const char* cstr = reinterpret_cast<const char*>(data.data());
- // If there is a \0 character in the middle of the |data|, treat it as end
+ // If there is a \0 character in the middle of the `data`, treat it as end
// of the string. Well-formed string extensions shouldn't contain it.
str->assign(cstr, strnlen(cstr, data.size()));
RTC_DCHECK(!str->empty());
diff --git a/modules/rtp_rtcp/source/rtp_packet.h b/modules/rtp_rtcp/source/rtp_packet.h
index e2e291c..b87d213 100644
--- a/modules/rtp_rtcp/source/rtp_packet.h
+++ b/modules/rtp_rtcp/source/rtp_packet.h
@@ -26,9 +26,9 @@
using ExtensionType = RTPExtensionType;
using ExtensionManager = RtpHeaderExtensionMap;
- // |extensions| required for SetExtension/ReserveExtension functions during
+ // `extensions` required for SetExtension/ReserveExtension functions during
// packet creating and used if available in Parse function.
- // Adding and getting extensions will fail until |extensions| is
+ // Adding and getting extensions will fail until `extensions` is
// provided via constructor or IdentifyExtensions function.
// |*extensions| is only accessed during construction; the pointer is not
// stored.
@@ -99,7 +99,7 @@
// which are modified after FEC protection is generated.
void ZeroMutableExtensions();
- // Removes extension of given |type|, returns false is extension was not
+ // Removes extension of given `type`, returns false is extension was not
// registered in packet's extension map or not present in the packet. Only
// extension that should be removed must be registered, other extensions may
// not be registered and will be preserved as is.
@@ -136,11 +136,11 @@
template <typename Extension>
bool ReserveExtension();
- // Find or allocate an extension |type|. Returns view of size |length|
+ // Find or allocate an extension `type`. Returns view of size `length`
// to write raw extension to or an empty view on failure.
rtc::ArrayView<uint8_t> AllocateExtension(ExtensionType type, size_t length);
- // Find an extension |type|.
+ // Find an extension `type`.
// Returns view of the raw extension or empty view on failure.
rtc::ArrayView<const uint8_t> FindExtension(ExtensionType type) const;
diff --git a/modules/rtp_rtcp/source/rtp_packet_history.cc b/modules/rtp_rtcp/source/rtp_packet_history.cc
index 317b808..fe5ccc7 100644
--- a/modules/rtp_rtcp/source/rtp_packet_history.cc
+++ b/modules/rtp_rtcp/source/rtp_packet_history.cc
@@ -54,7 +54,7 @@
void RtpPacketHistory::StoredPacket::IncrementTimesRetransmitted(
PacketPrioritySet* priority_set) {
// Check if this StoredPacket is in the priority set. If so, we need to remove
- // it before updating |times_retransmitted_| since that is used in sorting,
+ // it before updating `times_retransmitted_` since that is used in sorting,
// and then add it back.
const bool in_priority_set = priority_set && priority_set->erase(this) > 0;
++times_retransmitted_;
diff --git a/modules/rtp_rtcp/source/rtp_packet_history.h b/modules/rtp_rtcp/source/rtp_packet_history.h
index 44adc8c..f87ad4d 100644
--- a/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -31,7 +31,7 @@
public:
enum class StorageMode {
kDisabled, // Don't store any packets.
- kStoreAndCull // Store up to |number_to_store| packets, but try to remove
+ kStoreAndCull // Store up to `number_to_store` packets, but try to remove
// packets as they time out or as signaled as received.
};
@@ -78,7 +78,7 @@
// a packet in the history before we are reasonably sure it has been received.
void SetRtt(int64_t rtt_ms);
- // If |send_time| is set, packet was sent without using pacer, so state will
+ // If `send_time` is set, packet was sent without using pacer, so state will
// be set accordingly.
void PutRtpPacket(std::unique_ptr<RtpPacketToSend> packet,
absl::optional<int64_t> send_time_ms);
@@ -206,13 +206,13 @@
// the front and new packets being added to the back. Note that there may be
// wrap-arounds so the back may have a lower sequence number.
// Packets may also be removed out-of-order, in which case there will be
- // instances of StoredPacket with |packet_| set to nullptr. The first and last
+ // instances of StoredPacket with `packet_` set to nullptr. The first and last
// entry in the queue will however always be populated.
std::deque<StoredPacket> packet_history_ RTC_GUARDED_BY(lock_);
// Total number of packets with inserted.
uint64_t packets_inserted_ RTC_GUARDED_BY(lock_);
- // Objects from |packet_history_| ordered by "most likely to be useful", used
+ // Objects from `packet_history_` ordered by "most likely to be useful", used
// in GetPayloadPaddingPacket().
PacketPrioritySet padding_priority_ RTC_GUARDED_BY(lock_);
};
diff --git a/modules/rtp_rtcp/source/rtp_packetizer_av1.cc b/modules/rtp_rtcp/source/rtp_packetizer_av1.cc
index 4408bee..9cca983 100644
--- a/modules/rtp_rtcp/source/rtp_packetizer_av1.cc
+++ b/modules/rtp_rtcp/source/rtp_packetizer_av1.cc
@@ -70,7 +70,7 @@
return size;
}
-// Given |remaining_bytes| free bytes left in a packet, returns max size of an
+// Given `remaining_bytes` free bytes left in a packet, returns max size of an
// OBU fragment that can fit into the packet.
// i.e. MaxFragmentSize + Leb128Size(MaxFragmentSize) <= remaining_bytes.
int MaxFragmentSize(int remaining_bytes) {
@@ -191,7 +191,7 @@
const bool is_last_obu = obu_index == obus.size() - 1;
const Obu& obu = obus[obu_index];
- // Putting |obu| into the last packet would make last obu element stored in
+ // Putting `obu` into the last packet would make last obu element stored in
// that packet not last. All not last OBU elements must be prepend with the
// element length. AdditionalBytesForPreviousObuElement calculates how many
// bytes are needed to store that length.
@@ -242,12 +242,12 @@
: packet_remaining_bytes;
// Because available_bytes might be different than
// packet_remaining_bytes it might happen that max_first_fragment_size >=
- // obu.size. Also, since checks above verified |obu| should not be put
- // completely into the |packet|, leave at least 1 byte for later packet.
+ // obu.size. Also, since checks above verified `obu` should not be put
+ // completely into the `packet`, leave at least 1 byte for later packet.
int first_fragment_size = std::min(obu.size - 1, max_first_fragment_size);
if (first_fragment_size == 0) {
// Rather than writing 0-size element at the tail of the packet,
- // 'uninsert' the |obu| from the |packet|.
+ // 'uninsert' the `obu` from the `packet`.
packet.num_obu_elements--;
packet.packet_size -= previous_obu_extra_size;
} else {
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index f3e7e30..e3fd8ab 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -128,9 +128,9 @@
bool process_rtt = now >= last_rtt_process_time_ + kRtpRtcpRttProcessTimeMs;
if (rtcp_sender_.Sending()) {
// Process RTT if we have received a report block and we haven't
- // processed RTT for at least |kRtpRtcpRttProcessTimeMs| milliseconds.
+ // processed RTT for at least `kRtpRtcpRttProcessTimeMs` milliseconds.
// Note that LastReceivedReportBlockMs() grabs a lock, so check
- // |process_rtt| first.
+ // `process_rtt` first.
if (process_rtt && rtt_stats_ != nullptr &&
rtcp_receiver_.LastReceivedReportBlockMs() > last_rtt_process_time_) {
int64_t max_rtt_ms = 0;
@@ -530,7 +530,7 @@
if (expected_retransmission_time_ms > 0) {
return expected_retransmission_time_ms;
}
- // No rtt available (|kRtpRtcpRttProcessTimeMs| not yet passed?), so try to
+ // No rtt available (`kRtpRtcpRttProcessTimeMs` not yet passed?), so try to
// poll avg_rtt_ms directly from rtcp receiver.
if (rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), nullptr,
&expected_retransmission_time_ms, nullptr,
@@ -666,7 +666,7 @@
wait_time = kStartUpRttMs;
}
- // Send a full NACK list once within every |wait_time|.
+ // Send a full NACK list once within every `wait_time`.
return now - nack_last_time_sent_full_ms_ > wait_time;
}
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 45cfdb4..c5d0b3a 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -279,7 +279,7 @@
// Handles final time timestamping/stats/etc and handover to Transport.
DEPRECATED_RtpSenderEgress packet_sender;
// If no paced sender configured, this class will be used to pass packets
- // from |packet_generator_| to |packet_sender_|.
+ // from `packet_generator_` to `packet_sender_`.
DEPRECATED_RtpSenderEgress::NonPacedPacketSender non_paced_sender;
// Handles creation of RTP packets to be sent.
RTPSender packet_generator;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index f20fe87..136c11c 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -53,7 +53,7 @@
int DelayMillisForDuration(TimeDelta duration) {
// TimeDelta::ms() rounds downwards sometimes which leads to too little time
- // slept. Account for this, unless |duration| is exactly representable in
+ // slept. Account for this, unless `duration` is exactly representable in
// millisecs.
return (duration.us() + rtc::kNumMillisecsPerSec - 1) /
rtc::kNumMicrosecsPerMillisec;
@@ -528,9 +528,9 @@
: -1;
}
-// TODO(tommi): Check if |avg_rtt_ms|, |min_rtt_ms|, |max_rtt_ms| params are
+// TODO(tommi): Check if `avg_rtt_ms`, `min_rtt_ms`, `max_rtt_ms` params are
// actually used in practice (some callers ask for it but don't use it). It
-// could be that only |rtt| is needed and if so, then the fast path could be to
+// could be that only `rtt` is needed and if so, then the fast path could be to
// just call rtt_ms() and rely on the calculation being done periodically.
int32_t ModuleRtpRtcpImpl2::RTT(const uint32_t remote_ssrc,
int64_t* rtt,
@@ -550,7 +550,7 @@
if (expected_retransmission_time_ms > 0) {
return expected_retransmission_time_ms;
}
- // No rtt available (|kRttUpdateInterval| not yet passed?), so try to
+ // No rtt available (`kRttUpdateInterval` not yet passed?), so try to
// poll avg_rtt_ms directly from rtcp receiver.
if (rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), nullptr,
&expected_retransmission_time_ms, nullptr,
@@ -686,7 +686,7 @@
wait_time = kStartUpRttMs;
}
- // Send a full NACK list once within every |wait_time|.
+ // Send a full NACK list once within every `wait_time`.
return now - nack_last_time_sent_full_ms_ > wait_time;
}
@@ -865,7 +865,7 @@
TimeDelta duration) {
// We end up here under various sequences including the worker queue, and
// the RTCPSender lock is held.
- // See note in ScheduleRtcpSendEvaluation about why |worker_queue_| can be
+ // See note in ScheduleRtcpSendEvaluation about why `worker_queue_` can be
// accessed.
worker_queue_->PostDelayedTask(
ToQueuedTask(task_safety_,
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
index 2d8de02..14d1409 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
@@ -274,7 +274,7 @@
// Handles final time timestamping/stats/etc and handover to Transport.
RtpSenderEgress packet_sender;
// If no paced sender configured, this class will be used to pass packets
- // from |packet_generator_| to |packet_sender_|.
+ // from `packet_generator_` to `packet_sender_`.
RtpSenderEgress::NonPacedPacketSender non_paced_sender;
// Handles creation of RTP packets to be sent.
RTPSender packet_generator;
@@ -295,7 +295,7 @@
// Used from RtcpSenderMediator to maybe send rtcp.
void MaybeSendRtcp() RTC_RUN_ON(worker_queue_);
- // Called when |rtcp_sender_| informs of the next RTCP instant. The method may
+ // Called when `rtcp_sender_` informs of the next RTCP instant. The method may
// be called on various sequences, and is called under a RTCPSenderLock.
void ScheduleRtcpSendEvaluation(TimeDelta duration);
@@ -305,7 +305,7 @@
void MaybeSendRtcpAtOrAfterTimestamp(Timestamp execution_time)
RTC_RUN_ON(worker_queue_);
- // Schedules a call to MaybeSendRtcpAtOrAfterTimestamp delayed by |duration|.
+ // Schedules a call to MaybeSendRtcpAtOrAfterTimestamp delayed by `duration`.
void ScheduleMaybeSendRtcpAtOrAfterTimestamp(Timestamp execution_time,
TimeDelta duration);
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index 7e644be..e90d866 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -124,12 +124,12 @@
// done by RTCP RR acking.
bool always_send_mid_and_rid = false;
- // If set, field trials are read from |field_trials|, otherwise
+ // If set, field trials are read from `field_trials`, otherwise
// defaults to webrtc::FieldTrialBasedConfig.
const WebRtcKeyValueConfig* field_trials = nullptr;
// SSRCs for media and retransmission, respectively.
- // FlexFec SSRC is fetched from |flexfec_sender|.
+ // FlexFec SSRC is fetched from `flexfec_sender`.
uint32_t local_media_ssrc = 0;
absl::optional<uint32_t> rtx_send_ssrc;
@@ -203,7 +203,7 @@
int payload_frequency) = 0;
// Unregisters a send payload.
- // |payload_type| - payload type of codec
+ // `payload_type` - payload type of codec
// Returns -1 on failure else 0.
virtual int32_t DeRegisterSendPayload(int8_t payload_type) = 0;
@@ -259,7 +259,7 @@
virtual void SetMid(const std::string& mid) = 0;
// Sets CSRC.
- // |csrcs| - vector of CSRCs
+ // `csrcs` - vector of CSRCs
virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) = 0;
// Turns on/off sending RTX (RFC 4588). The modes can be set as a combination
@@ -355,7 +355,7 @@
virtual RtcpMode RTCP() const = 0;
// Sets RTCP status i.e on(compound or non-compound)/off.
- // |method| - RTCP method to use.
+ // `method` - RTCP method to use.
virtual void SetRTCPStatus(RtcpMode method) = 0;
// Sets RTCP CName (i.e unique identifier).
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index a47d774..ccc72a6 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -393,7 +393,7 @@
packet_history_->GetPayloadPaddingPacket(
[&](const RtpPacketToSend& packet)
-> std::unique_ptr<RtpPacketToSend> {
- // Limit overshoot, generate <= |max_padding_size_factor_| *
+ // Limit overshoot, generate <= `max_padding_size_factor_` *
// target_size_bytes.
const size_t max_overshoot_bytes = static_cast<size_t>(
((max_padding_size_factor_ - 1.0) * target_size_bytes) +
@@ -555,7 +555,7 @@
// sender can reduce overhead by omitting these header extensions once it
// knows that the receiver has "bound" the SSRC.
// This optimization can be configured by setting
- // |always_send_mid_and_rid_| appropriately.
+ // `always_send_mid_and_rid_` appropriately.
//
// The algorithm here is fairly simple: Always attach a MID and/or RID (if
// configured) to the outgoing packets until an RTCP receiver report comes
diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index 0eb6558..4919b40 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -165,7 +165,7 @@
return flexfec_ssrc_;
}
- // Sends packet to |transport_| or to the pacer, depending on configuration.
+ // Sends packet to `transport_` or to the pacer, depending on configuration.
// TODO(bugs.webrtc.org/XXX): Remove in favor of EnqueuePackets().
bool SendToNetwork(std::unique_ptr<RtpPacketToSend> packet)
RTC_LOCKS_EXCLUDED(send_mutex_);
@@ -201,7 +201,7 @@
const absl::optional<uint32_t> rtx_ssrc_;
const absl::optional<uint32_t> flexfec_ssrc_;
// Limits GeneratePadding() outcome to <=
- // |max_padding_size_factor_| * |target_size_bytes|
+ // `max_padding_size_factor_` * `target_size_bytes`
const double max_padding_size_factor_;
RtpPacketHistory* const packet_history_;
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.cc b/modules/rtp_rtcp/source/rtp_sender_egress.cc
index 53e4adb..99697c4 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.cc
@@ -295,7 +295,7 @@
}
if (send_success) {
- // |media_has_been_sent_| is used by RTPSender to figure out if it can send
+ // `media_has_been_sent_` is used by RTPSender to figure out if it can send
// padding in the absence of transport-cc or abs-send-time.
// In those cases media must be sent first to set a reference timestamp.
media_has_been_sent_ = true;
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.h b/modules/rtp_rtcp/source/rtp_sender_egress.h
index 9c1baea..747471c 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.h
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.h
@@ -80,7 +80,7 @@
void SetMediaHasBeenSent(bool media_sent) RTC_LOCKS_EXCLUDED(lock_);
void SetTimestampOffset(uint32_t timestamp) RTC_LOCKS_EXCLUDED(lock_);
- // For each sequence number in |sequence_number|, recall the last RTP packet
+ // For each sequence number in `sequence_number`, recall the last RTP packet
// which bore it - its timestamp and whether it was the first and/or last
// packet in that frame. If all of the given sequence numbers could be
// recalled, return a vector with all of them (in corresponding order).
@@ -112,7 +112,7 @@
void UpdateOnSendPacket(int packet_id,
int64_t capture_time_ms,
uint32_t ssrc);
- // Sends packet on to |transport_|, leaving the RTP module.
+ // Sends packet on to `transport_`, leaving the RTP module.
bool SendPacketToNetwork(const RtpPacketToSend& packet,
const PacketOptions& options,
const PacedPacketInfo& pacing_info);
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index d265cc6..e7ac1e4 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -588,7 +588,7 @@
first_packet->HasExtension<RtpDependencyDescriptorExtension>();
// Minimization of the vp8 descriptor may erase temporal_id, so use
- // |temporal_id| rather than reference |video_header| beyond this point.
+ // `temporal_id` rather than reference `video_header` beyond this point.
if (has_generic_descriptor) {
MinimizeDescriptor(&video_header);
}
@@ -687,7 +687,7 @@
red_packet->SetPayloadType(*red_payload_type_);
red_packet->set_is_red(true);
- // Append |red_packet| instead of |packet| to output.
+ // Append `red_packet` instead of `packet` to output.
red_packet->set_packet_type(RtpPacketMediaType::kVideo);
red_packet->set_allow_retransmission(packet->allow_retransmission());
rtp_packets.emplace_back(std::move(red_packet));
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.h b/modules/rtp_rtcp/source/rtp_sender_video.h
index ba8d7e8..226c406 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -67,7 +67,7 @@
Config(const Config&) = delete;
Config(Config&&) = default;
- // All members of this struct, with the exception of |field_trials|, are
+ // All members of this struct, with the exception of `field_trials`, are
// expected to outlive the RTPSenderVideo object they are passed to.
Clock* clock = nullptr;
RTPSender* rtp_sender = nullptr;
@@ -91,7 +91,7 @@
// expected_retransmission_time_ms.has_value() -> retransmission allowed.
// `capture_time_ms` and `clock::CurrentTime` should be using the same epoch.
// Calls to this method is assumed to be externally serialized.
- // |estimated_capture_clock_offset_ms| is an estimated clock offset between
+ // `estimated_capture_clock_offset_ms` is an estimated clock offset between
// this sender and the original capturer, for this video packet. See
// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time for more
// details. If the sender and the capture has the same clock, it is supposed
@@ -208,12 +208,12 @@
RTC_GUARDED_BY(send_checker_);
absl::optional<VideoLayersAllocation> allocation_
RTC_GUARDED_BY(send_checker_);
- // Flag indicating if we should send |allocation_|.
+ // Flag indicating if we should send `allocation_`.
SendVideoLayersAllocation send_allocation_ RTC_GUARDED_BY(send_checker_);
// Current target playout delay.
VideoPlayoutDelay current_playout_delay_ RTC_GUARDED_BY(send_checker_);
- // Flag indicating if we need to send |current_playout_delay_| in order
+ // Flag indicating if we need to send `current_playout_delay_` in order
// to guarantee it gets delivered.
bool playout_delay_pending_;
// Set by the field trial WebRTC-ForceSendPlayoutDelay to override the playout
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h
index 8573869..10d0241 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h
+++ b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h
@@ -45,26 +45,26 @@
absl::optional<int64_t> expected_retransmission_time_ms);
// Implements TransformedFrameCallback. Can be called on any thread. Posts
- // the transformed frame to be sent on the |encoder_queue_|.
+ // the transformed frame to be sent on the `encoder_queue_`.
void OnTransformedFrame(
std::unique_ptr<TransformableFrameInterface> frame) override;
- // Delegates the call to RTPSendVideo::SendVideo on the |encoder_queue_|.
+ // Delegates the call to RTPSendVideo::SendVideo on the `encoder_queue_`.
void SendVideo(std::unique_ptr<TransformableFrameInterface> frame) const;
// Delegates the call to RTPSendVideo::SetVideoStructureAfterTransformation
- // under |sender_lock_|.
+ // under `sender_lock_`.
void SetVideoStructureUnderLock(
const FrameDependencyStructure* video_structure);
// Delegates the call to
// RTPSendVideo::SetVideoLayersAllocationAfterTransformation under
- // |sender_lock_|.
+ // `sender_lock_`.
void SetVideoLayersAllocationUnderLock(VideoLayersAllocation allocation);
- // Unregisters and releases the |frame_transformer_| reference, and resets
- // |sender_| under lock. Called from RTPSenderVideo destructor to prevent the
- // |sender_| to dangle.
+ // Unregisters and releases the `frame_transformer_` reference, and resets
+ // `sender_` under lock. Called from RTPSenderVideo destructor to prevent the
+ // `sender_` to dangle.
void Reset();
protected:
diff --git a/modules/rtp_rtcp/source/rtp_sequence_number_map.cc b/modules/rtp_rtcp/source/rtp_sequence_number_map.cc
index 28ae9c8..441429d 100644
--- a/modules/rtp_rtcp/source/rtp_sequence_number_map.cc
+++ b/modules/rtp_rtcp/source/rtp_sequence_number_map.cc
@@ -23,7 +23,7 @@
RtpSequenceNumberMap::RtpSequenceNumberMap(size_t max_entries)
: max_entries_(max_entries) {
- RTC_DCHECK_GT(max_entries_, 4); // See code paring down to |max_entries_|.
+ RTC_DCHECK_GT(max_entries_, 4); // See code paring down to `max_entries_`.
RTC_DCHECK_LE(max_entries_, 1 << 15);
}
@@ -42,7 +42,7 @@
if (AheadOrAt(sequence_number, associations_.front().sequence_number) &&
AheadOrAt(associations_.back().sequence_number, sequence_number)) {
// The sequence number has wrapped around and is within the range
- // currently held by |associations_| - we should invalidate all entries.
+ // currently held by `associations_` - we should invalidate all entries.
RTC_LOG(LS_WARNING) << "Sequence number wrapped-around unexpectedly.";
associations_.clear();
associations_.emplace_back(sequence_number, info);
@@ -59,7 +59,7 @@
erase_to = std::next(erase_to, max_entries_ - new_size);
}
- // It is guaranteed that |associations_| can be split into two partitions,
+ // It is guaranteed that `associations_` can be split into two partitions,
// either partition possibly empty, such that:
// * In the first partition, all elements are AheadOf the new element.
// This is the partition of the obsolete elements.
diff --git a/modules/rtp_rtcp/source/rtp_sequence_number_map.h b/modules/rtp_rtcp/source/rtp_sequence_number_map.h
index 56979a3..8a036c2 100644
--- a/modules/rtp_rtcp/source/rtp_sequence_number_map.h
+++ b/modules/rtp_rtcp/source/rtp_sequence_number_map.h
@@ -22,7 +22,7 @@
// Records the association of RTP sequence numbers to timestamps and to whether
// the packet was first and/or last in the frame.
//
-// 1. Limits number of entries. Whenever |max_entries| is about to be exceeded,
+// 1. Limits number of entries. Whenever `max_entries` is about to be exceeded,
// the size is reduced by approximately 25%.
// 2. RTP sequence numbers wrap around relatively infrequently.
// This class therefore only remembers at most the last 2^15 RTP packets,
diff --git a/modules/rtp_rtcp/source/rtp_sequence_number_map_unittest.cc b/modules/rtp_rtcp/source/rtp_sequence_number_map_unittest.cc
index 324350c..78c9e4a 100644
--- a/modules/rtp_rtcp/source/rtp_sequence_number_map_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sequence_number_map_unittest.cc
@@ -438,7 +438,7 @@
uut.InsertPacket(new_association.sequence_number, new_association.info);
associations.push_back(new_association);
- // The +1 is for |new_association|.
+ // The +1 is for `new_association`.
const size_t kExpectedAssociationCount = 3 * kMaxEntries / 4 + 1;
const auto expected_begin =
std::prev(associations.end(), kExpectedAssociationCount);
@@ -466,7 +466,7 @@
uut.InsertPacket(new_association.sequence_number, new_association.info);
associations.push_back(new_association);
- // The +1 is for |new_association|.
+ // The +1 is for `new_association`.
const size_t kExpectedAssociationCount =
std::min(3 * max_entries / 4, max_entries - obsoleted_count) + 1;
const auto expected_begin =
diff --git a/modules/rtp_rtcp/source/source_tracker.cc b/modules/rtp_rtcp/source/source_tracker.cc
index d6c7445..f9aa003 100644
--- a/modules/rtp_rtcp/source/source_tracker.cc
+++ b/modules/rtp_rtcp/source/source_tracker.cc
@@ -72,7 +72,7 @@
SourceTracker::SourceEntry& SourceTracker::UpdateEntry(const SourceKey& key) {
// We intentionally do |find() + emplace()|, instead of checking the return
- // value of |emplace()|, for performance reasons. It's much more likely for
+ // value of `emplace()`, for performance reasons. It's much more likely for
// the key to already exist than for it not to.
auto map_it = map_.find(key);
if (map_it == map_.end()) {
diff --git a/modules/rtp_rtcp/source/source_tracker.h b/modules/rtp_rtcp/source/source_tracker.h
index 0c7627c..3f3ef8c 100644
--- a/modules/rtp_rtcp/source/source_tracker.h
+++ b/modules/rtp_rtcp/source/source_tracker.h
@@ -48,8 +48,8 @@
// RTCRtpReceiver's MediaStreamTrack.
void OnFrameDelivered(const RtpPacketInfos& packet_infos);
- // Returns an |RtpSource| for each unique SSRC and CSRC identifier updated in
- // the last |kTimeoutMs| milliseconds. Entries appear in reverse chronological
+ // Returns an `RtpSource` for each unique SSRC and CSRC identifier updated in
+ // the last `kTimeoutMs` milliseconds. Entries appear in reverse chronological
// order (i.e. with the most recently updated entries appearing first).
std::vector<RtpSource> GetSources() const;
@@ -58,7 +58,7 @@
SourceKey(RtpSourceType source_type, uint32_t source)
: source_type(source_type), source(source) {}
- // Type of |source|.
+ // Type of `source`.
RtpSourceType source_type;
// CSRC or SSRC identifier of the contributing or synchronization source.
@@ -81,12 +81,12 @@
struct SourceEntry {
// Timestamp indicating the most recent time a frame from an RTP packet,
// originating from this source, was delivered to the RTCRtpReceiver's
- // MediaStreamTrack. Its reference clock is the outer class's |clock_|.
+ // MediaStreamTrack. Its reference clock is the outer class's `clock_`.
int64_t timestamp_ms;
// Audio level from an RFC 6464 or RFC 6465 header extension received with
// the most recent packet used to assemble the frame associated with
- // |timestamp_ms|. May be absent. Only relevant for audio receivers. See the
+ // `timestamp_ms`. May be absent. Only relevant for audio receivers. See the
// specs for `RTCRtpContributingSource` for more info.
absl::optional<uint8_t> audio_level;
@@ -96,7 +96,7 @@
absl::optional<AbsoluteCaptureTime> absolute_capture_time;
// RTP timestamp of the most recent packet used to assemble the frame
- // associated with |timestamp_ms|.
+ // associated with `timestamp_ms`.
uint32_t rtp_timestamp;
};
diff --git a/modules/rtp_rtcp/source/source_tracker_unittest.cc b/modules/rtp_rtcp/source/source_tracker_unittest.cc
index 8514e84..b64f03c 100644
--- a/modules/rtp_rtcp/source/source_tracker_unittest.cc
+++ b/modules/rtp_rtcp/source/source_tracker_unittest.cc
@@ -37,7 +37,7 @@
constexpr size_t kPacketInfosCountMax = 5;
-// Simple "guaranteed to be correct" re-implementation of |SourceTracker| for
+// Simple "guaranteed to be correct" re-implementation of `SourceTracker` for
// dual-implementation testing purposes.
class ExpectedSourceTracker {
public:
diff --git a/modules/rtp_rtcp/source/ulpfec_generator.cc b/modules/rtp_rtcp/source/ulpfec_generator.cc
index a5ce8c9..2d585d7 100644
--- a/modules/rtp_rtcp/source/ulpfec_generator.cc
+++ b/modules/rtp_rtcp/source/ulpfec_generator.cc
@@ -40,17 +40,17 @@
constexpr size_t kMinMediaPackets = 4;
// Threshold on the received FEC protection level, above which we enforce at
-// least |kMinMediaPackets| packets for the FEC code. Below this
-// threshold |kMinMediaPackets| is set to default value of 1.
+// least `kMinMediaPackets` packets for the FEC code. Below this
+// threshold `kMinMediaPackets` is set to default value of 1.
//
// The range is between 0 and 255, where 255 corresponds to 100% overhead
// (relative to the number of protected media packets).
constexpr uint8_t kHighProtectionThreshold = 80;
-// This threshold is used to adapt the |kMinMediaPackets| threshold, based
+// This threshold is used to adapt the `kMinMediaPackets` threshold, based
// on the average number of packets per frame seen so far. When there are few
// packets per frame (as given by this threshold), at least
-// |kMinMediaPackets| + 1 packets are sent to the FEC code.
+// `kMinMediaPackets` + 1 packets are sent to the FEC code.
constexpr float kMinMediaPacketsAdaptationThreshold = 2.0f;
// At construction time, we don't know the SSRC that is used for the generated
@@ -129,7 +129,7 @@
}
const bool complete_frame = packet.Marker();
if (media_packets_.size() < kUlpfecMaxMediaPackets) {
- // Our packet masks can only protect up to |kUlpfecMaxMediaPackets| packets.
+ // Our packet masks can only protect up to `kUlpfecMaxMediaPackets` packets.
auto fec_packet = std::make_unique<ForwardErrorCorrection::Packet>();
fec_packet->data = packet.Buffer();
media_packets_.push_back(std::move(fec_packet));
@@ -148,8 +148,8 @@
// Produce FEC over at most |params_.max_fec_frames| frames, or as soon as:
// (1) the excess overhead (actual overhead - requested/target overhead) is
- // less than |kMaxExcessOverhead|, and
- // (2) at least |min_num_media_packets_| media packets is reached.
+ // less than `kMaxExcessOverhead`, and
+ // (2) at least `min_num_media_packets_` media packets is reached.
if (complete_frame &&
(num_protected_frames_ >= params.max_fec_frames ||
(ExcessOverheadBelowMax() && MinimumMediaPacketsReached()))) {
@@ -203,7 +203,7 @@
}
// Wrap FEC packet (including FEC headers) in a RED packet. Since the
- // FEC packets in |generated_fec_packets_| don't have RTP headers, we
+ // FEC packets in `generated_fec_packets_` don't have RTP headers, we
// reuse the header from the last media packet.
RTC_CHECK(last_media_packet_.has_value());
last_media_packet_->SetPayloadSize(0);
diff --git a/modules/rtp_rtcp/source/ulpfec_generator.h b/modules/rtp_rtcp/source/ulpfec_generator.h
index 934a1d5..c9924581 100644
--- a/modules/rtp_rtcp/source/ulpfec_generator.h
+++ b/modules/rtp_rtcp/source/ulpfec_generator.h
@@ -81,14 +81,14 @@
int Overhead() const;
// Returns true if the excess overhead (actual - target) for the FEC is below
- // the amount |kMaxExcessOverhead|. This effects the lower protection level
+ // the amount `kMaxExcessOverhead`. This effects the lower protection level
// cases and low number of media packets/frame. The target overhead is given
// by |params_.fec_rate|, and is only achievable in the limit of large number
// of media packets.
bool ExcessOverheadBelowMax() const;
// Returns true if the number of added media packets is at least
- // |min_num_media_packets_|. This condition tries to capture the effect
+ // `min_num_media_packets_`. This condition tries to capture the effect
// that, for the same amount of protection/overhead, longer codes
// (e.g. (2k,2m) vs (k,m)) are generally more effective at recovering losses.
bool MinimumMediaPacketsReached() const;
diff --git a/modules/rtp_rtcp/source/ulpfec_generator_unittest.cc b/modules/rtp_rtcp/source/ulpfec_generator_unittest.cc
index c07e81d..18f5685 100644
--- a/modules/rtp_rtcp/source/ulpfec_generator_unittest.cc
+++ b/modules/rtp_rtcp/source/ulpfec_generator_unittest.cc
@@ -103,12 +103,12 @@
}
TEST_F(UlpfecGeneratorTest, OneFrameFec) {
- // The number of media packets (|kNumPackets|), number of frames (one for
+ // The number of media packets (`kNumPackets`), number of frames (one for
// this test), and the protection factor (|params->fec_rate|) are set to make
// sure the conditions for generating FEC are satisfied. This means:
// (1) protection factor is high enough so that actual overhead over 1 frame
- // of packets is within |kMaxExcessOverhead|, and (2) the total number of
- // media packets for 1 frame is at least |minimum_media_packets_fec_|.
+ // of packets is within `kMaxExcessOverhead`, and (2) the total number of
+ // media packets for 1 frame is at least `minimum_media_packets_fec_`.
constexpr size_t kNumPackets = 4;
FecProtectionParams params = {15, 3, kFecMaskRandom};
packet_generator_.NewFrame(kNumPackets);
@@ -137,13 +137,13 @@
}
TEST_F(UlpfecGeneratorTest, TwoFrameFec) {
- // The number of media packets/frame (|kNumPackets|), the number of frames
- // (|kNumFrames|), and the protection factor (|params->fec_rate|) are set to
+ // The number of media packets/frame (`kNumPackets`), the number of frames
+ // (`kNumFrames`), and the protection factor (|params->fec_rate|) are set to
// make sure the conditions for generating FEC are satisfied. This means:
// (1) protection factor is high enough so that actual overhead over
- // |kNumFrames| is within |kMaxExcessOverhead|, and (2) the total number of
- // media packets for |kNumFrames| frames is at least
- // |minimum_media_packets_fec_|.
+ // `kNumFrames` is within `kMaxExcessOverhead`, and (2) the total number of
+ // media packets for `kNumFrames` frames is at least
+ // `minimum_media_packets_fec_`.
constexpr size_t kNumPackets = 2;
constexpr size_t kNumFrames = 2;
diff --git a/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc b/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc
index 49f483d..8378a8f 100644
--- a/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc
+++ b/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc
@@ -25,7 +25,7 @@
constexpr size_t kMaxMediaPackets = 48;
// Maximum number of media packets tracked by FEC decoder.
-// Maintain a sufficiently larger tracking window than |kMaxMediaPackets|
+// Maintain a sufficiently larger tracking window than `kMaxMediaPackets`
// to account for packet reordering in pacer/ network.
constexpr size_t kMaxTrackedMediaPackets = 4 * kMaxMediaPackets;
diff --git a/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc b/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc
index fdfa475..c993923 100644
--- a/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc
+++ b/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc
@@ -157,10 +157,10 @@
int32_t UlpfecReceiverImpl::ProcessReceivedFec() {
RTC_DCHECK_RUN_ON(&sequence_checker_);
- // If we iterate over |received_packets_| and it contains a packet that cause
+ // If we iterate over `received_packets_` and it contains a packet that cause
// us to recurse back to this function (for example a RED packet encapsulating
// a RED packet), then we will recurse forever. To avoid this we swap
- // |received_packets_| with an empty vector so that the next recursive call
+ // `received_packets_` with an empty vector so that the next recursive call
// wont iterate over the same packet again. This also solves the problem of
// not modifying the vector we are currently iterating over (packets are added
// in AddReceivedRedPacket).
diff --git a/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc b/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc
index 53d363d..b16ef3d 100644
--- a/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc
+++ b/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc
@@ -54,27 +54,27 @@
{})),
packet_generator_(kMediaSsrc) {}
- // Generates |num_fec_packets| FEC packets, given |media_packets|.
+ // Generates `num_fec_packets` FEC packets, given `media_packets`.
void EncodeFec(const ForwardErrorCorrection::PacketList& media_packets,
size_t num_fec_packets,
std::list<ForwardErrorCorrection::Packet*>* fec_packets);
- // Generates |num_media_packets| corresponding to a single frame.
+ // Generates `num_media_packets` corresponding to a single frame.
void PacketizeFrame(size_t num_media_packets,
size_t frame_offset,
std::list<AugmentedPacket*>* augmented_packets,
ForwardErrorCorrection::PacketList* packets);
- // Build a media packet using |packet_generator_| and add it
+ // Build a media packet using `packet_generator_` and add it
// to the receiver.
void BuildAndAddRedMediaPacket(AugmentedPacket* packet,
bool is_recovered = false);
- // Build a FEC packet using |packet_generator_| and add it
+ // Build a FEC packet using `packet_generator_` and add it
// to the receiver.
void BuildAndAddRedFecPacket(Packet* packet);
- // Ensure that |recovered_packet_receiver_| will be called correctly
+ // Ensure that `recovered_packet_receiver_` will be called correctly
// and that the recovered packet will be identical to the lost packet.
void VerifyReconstructedMediaPacket(const AugmentedPacket& packet,
size_t times);
@@ -139,7 +139,7 @@
const AugmentedPacket& packet,
size_t times) {
// Verify that the content of the reconstructed packet is equal to the
- // content of |packet|, and that the same content is received |times| number
+ // content of `packet`, and that the same content is received `times` number
// of times in a row.
EXPECT_CALL(recovered_packet_receiver_,
OnRecoveredPacket(_, packet.data.size()))
diff --git a/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc b/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
index dffdf2e..8941f1c 100644
--- a/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
+++ b/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
@@ -155,7 +155,7 @@
uint8_t fec_packet_masks_[kMaxNumberMediaPackets][kMaxNumberMediaPackets];
FILE* fp_mask_;
- // Measure of the gap of the loss for configuration given by |state|.
+ // Measure of the gap of the loss for configuration given by `state`.
// This is to measure degree of consecutiveness for the loss configuration.
// Useful if the packets are sent out in order of sequence numbers and there
// is little/no re-ordering during transmission.
@@ -183,8 +183,8 @@
}
// Returns the number of recovered media packets for the XOR code, given the
- // packet mask |fec_packet_masks_|, for the loss state/configuration given by
- // |state|.
+ // packet mask `fec_packet_masks_`, for the loss state/configuration given by
+ // `state`.
int RecoveredMediaPackets(int num_media_packets,
int num_fec_packets,
uint8_t* state) {
@@ -241,7 +241,7 @@
}
// Compute the probability of occurence of the loss state/configuration,
- // given by |state|, for all the loss models considered in this test.
+ // given by `state`, for all the loss models considered in this test.
void ComputeProbabilityWeight(double* prob_weight,
uint8_t* state,
int tot_num_packets) {
@@ -317,8 +317,8 @@
}
// Compute the residual loss per gap, by summing the
- // |residual_loss_per_loss_gap| over all loss configurations up to loss number
- // = |num_fec_packets|.
+ // `residual_loss_per_loss_gap` over all loss configurations up to loss number
+ // = `num_fec_packets`.
double ComputeResidualLossPerGap(MetricsFecCode metrics,
int gap_number,
int num_fec_packets,
@@ -339,7 +339,7 @@
}
// Compute the recovery rate per loss number, by summing the
- // |residual_loss_per_loss_gap| over all gap configurations.
+ // `residual_loss_per_loss_gap` over all gap configurations.
void ComputeRecoveryRatePerLoss(MetricsFecCode* metrics,
int num_media_packets,
int num_fec_packets,
@@ -358,7 +358,7 @@
if (tot_num_configs > 0) {
arl = arl / static_cast<double>(tot_num_configs);
}
- // Recovery rate for a given loss |loss| is 1 minus the scaled |arl|,
+ // Recovery rate for a given loss `loss` is 1 minus the scaled `arl`,
// where the scale factor is relative to code size/parameters.
double scaled_loss =
static_cast<double>(loss * num_media_packets) /
@@ -376,8 +376,8 @@
sizeof(double) * 2 * kMaxMediaPacketsTest + 1);
}
- // Compute the metrics for an FEC code, given by the code type |code_type|
- // (XOR-random/ bursty or RS), and by the code index |code_index|
+ // Compute the metrics for an FEC code, given by the code type `code_type`
+ // (XOR-random/ bursty or RS), and by the code index `code_index`
// (which containes the code size parameters/protection length).
void ComputeMetricsForCode(CodeType code_type, int code_index) {
std::unique_ptr<double[]> prob_weight(new double[kNumLossModels]);
@@ -393,7 +393,7 @@
int num_loss_configurations = 1 << tot_num_packets;
// Loop over all loss configurations for the symbol sequence of length
- // |tot_num_packets|. In this version we process up to (k=12, m=12) codes,
+ // `tot_num_packets`. In this version we process up to (k=12, m=12) codes,
// and get exact expressions for the residual loss.
// TODO(marpan): For larger codes, loop over some random sample of loss
// configurations, sampling driven by the underlying statistical loss model
@@ -470,16 +470,16 @@
metrics_code.residual_loss_per_loss_gap[index] += residual_loss;
if (code_type == xor_random_code) {
// The configuration density is only a function of the code length and
- // only needs to computed for the first |code_type| passed here.
+ // only needs to computed for the first `code_type` passed here.
code_params_[code_index].configuration_density[index]++;
}
} // Done with loop over configurations.
// Normalize the average residual loss and compute/normalize the variance.
for (int k = 0; k < kNumLossModels; k++) {
// Normalize the average residual loss by the total number of packets
- // |tot_num_packets| (i.e., the code length). For a code with no (zero)
+ // `tot_num_packets` (i.e., the code length). For a code with no (zero)
// recovery, the average residual loss for that code would be reduced like
- // ~|average_loss_rate| * |num_media_packets| / |tot_num_packets|. This is
+ // ~`average_loss_rate` * `num_media_packets` / `tot_num_packets`. This is
// the expected reduction in the average residual loss just from adding
// FEC packets to the symbol sequence.
metrics_code.average_residual_loss[k] =
@@ -516,7 +516,7 @@
void WriteOutMetricsAllFecCodes() {
std::string filename = test::OutputPath() + "data_metrics_all_codes";
FILE* fp = fopen(filename.c_str(), "wb");
- // Loop through codes up to |kMaxMediaPacketsTest|.
+ // Loop through codes up to `kMaxMediaPacketsTest`.
int code_index = 0;
for (int num_media_packets = 1; num_media_packets <= kMaxMediaPacketsTest;
num_media_packets++) {
@@ -714,7 +714,7 @@
const int packet_mask_max = kMaxMediaPackets[fec_mask_type];
std::unique_ptr<uint8_t[]> packet_mask(
new uint8_t[packet_mask_max * kUlpfecMaxPacketMaskSize]);
- // Loop through codes up to |kMaxMediaPacketsTest|.
+ // Loop through codes up to `kMaxMediaPacketsTest`.
for (int num_media_packets = 1; num_media_packets <= kMaxMediaPacketsTest;
++num_media_packets) {
const int mask_bytes_fec_packet =
@@ -955,7 +955,7 @@
for (int code_index = 0; code_index < max_num_codes_; code_index++) {
int num_fec_packets = code_params_[code_index].num_fec_packets;
for (int loss = 1; loss <= num_fec_packets; loss++) {
- int index = loss; // |gap| is zero.
+ int index = loss; // `gap` is zero.
EXPECT_EQ(kMetricsXorBursty[code_index].residual_loss_per_loss_gap[index],
0.0);
}
@@ -1010,8 +1010,8 @@
for (int code_index = 0; code_index < max_num_codes_; code_index++) {
int num_media_packets = code_params_[code_index].num_media_packets;
int num_fec_packets = code_params_[code_index].num_fec_packets;
- // Perfect recovery (|recovery_rate_per_loss| == 1) is expected for
- // |loss_number| = 1, for all codes.
+ // Perfect recovery (`recovery_rate_per_loss` == 1) is expected for
+ // `loss_number` = 1, for all codes.
int loss_number = 1;
EXPECT_EQ(
kMetricsReedSolomon[code_index].recovery_rate_per_loss[loss_number],
@@ -1020,7 +1020,7 @@
1.0);
EXPECT_EQ(kMetricsXorBursty[code_index].recovery_rate_per_loss[loss_number],
1.0);
- // For |loss_number| = |num_fec_packets| / 2, we expect the following:
+ // For `loss_number` = `num_fec_packets` / 2, we expect the following:
// Perfect recovery for RS, and recovery for XOR above the threshold.
loss_number = num_fec_packets / 2 > 0 ? num_fec_packets / 2 : 1;
EXPECT_EQ(
@@ -1030,7 +1030,7 @@
kRecoveryRateXorRandom[0]);
EXPECT_GE(kMetricsXorBursty[code_index].recovery_rate_per_loss[loss_number],
kRecoveryRateXorBursty[0]);
- // For |loss_number| = |num_fec_packets|, we expect the following:
+ // For `loss_number` = `num_fec_packets`, we expect the following:
// Perfect recovery for RS, and recovery for XOR above the threshold.
loss_number = num_fec_packets;
EXPECT_EQ(
@@ -1040,7 +1040,7 @@
kRecoveryRateXorRandom[1]);
EXPECT_GE(kMetricsXorBursty[code_index].recovery_rate_per_loss[loss_number],
kRecoveryRateXorBursty[1]);
- // For |loss_number| = |num_fec_packets| + 1, we expect the following:
+ // For `loss_number` = `num_fec_packets` + 1, we expect the following:
// Zero recovery for RS, but non-zero recovery for XOR.
if (num_fec_packets > 1 && num_media_packets > 2) {
loss_number = num_fec_packets + 1;
