Use backticks not vertical bars to denote variables in comments for /api
Bug: webrtc:12338
Change-Id: Ib97b2c3d64dbd895f261ffa76a2e885bd934a87f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34554}
diff --git a/api/audio_codecs/audio_decoder.h b/api/audio_codecs/audio_decoder.h
index ce23594..51d20c4 100644
--- a/api/audio_codecs/audio_decoder.h
+++ b/api/audio_codecs/audio_decoder.h
@@ -53,8 +53,8 @@
// Returns true if this packet contains DTX.
virtual bool IsDtxPacket() const;
- // Decodes this frame of audio and writes the result in |decoded|.
- // |decoded| must be large enough to store as many samples as indicated by a
+ // Decodes this frame of audio and writes the result in `decoded`.
+ // `decoded` must be large enough to store as many samples as indicated by a
// call to Duration() . On success, returns an absl::optional containing the
// total number of samples across all channels, as well as whether the
// decoder produced comfort noise or speech. On failure, returns an empty
@@ -85,8 +85,8 @@
// Let the decoder parse this payload and prepare zero or more decodable
// frames. Each frame must be between 10 ms and 120 ms long. The caller must
// ensure that the AudioDecoder object outlives any frame objects returned by
- // this call. The decoder is free to swap or move the data from the |payload|
- // buffer. |timestamp| is the input timestamp, in samples, corresponding to
+ // this call. The decoder is free to swap or move the data from the `payload`
+ // buffer. `timestamp` is the input timestamp, in samples, corresponding to
// the start of the payload.
virtual std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
uint32_t timestamp);
@@ -95,12 +95,12 @@
// obsolete; callers should call ParsePayload instead. For now, subclasses
// must still implement DecodeInternal.
- // Decodes |encode_len| bytes from |encoded| and writes the result in
- // |decoded|. The maximum bytes allowed to be written into |decoded| is
- // |max_decoded_bytes|. Returns the total number of samples across all
- // channels. If the decoder produced comfort noise, |speech_type|
+ // Decodes `encode_len` bytes from `encoded` and writes the result in
+ // `decoded`. The maximum bytes allowed to be written into `decoded` is
+ // `max_decoded_bytes`. Returns the total number of samples across all
+ // channels. If the decoder produced comfort noise, `speech_type`
// is set to kComfortNoise, otherwise it is kSpeech. The desired output
- // sample rate is provided in |sample_rate_hz|, which must be valid for the
+ // sample rate is provided in `sample_rate_hz`, which must be valid for the
// codec at hand.
int Decode(const uint8_t* encoded,
size_t encoded_len,
@@ -123,11 +123,11 @@
// Calls the packet-loss concealment of the decoder to update the state after
// one or several lost packets. The caller has to make sure that the
- // memory allocated in |decoded| should accommodate |num_frames| frames.
+ // memory allocated in `decoded` should accommodate `num_frames` frames.
virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
// Asks the decoder to generate packet-loss concealment and append it to the
- // end of |concealment_audio|. The concealment audio should be in
+ // end of `concealment_audio`. The concealment audio should be in
// channel-interleaved format, with as many channels as the last decoded
// packet produced. The implementation must produce at least
// requested_samples_per_channel, or nothing at all. This is a signal to the
@@ -146,19 +146,19 @@
// Returns the last error code from the decoder.
virtual int ErrorCode();
- // Returns the duration in samples-per-channel of the payload in |encoded|
- // which is |encoded_len| bytes long. Returns kNotImplemented if no duration
+ // Returns the duration in samples-per-channel of the payload in `encoded`
+ // which is `encoded_len` bytes long. Returns kNotImplemented if no duration
// estimate is available, or -1 in case of an error.
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
// Returns the duration in samples-per-channel of the redandant payload in
- // |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no
+ // `encoded` which is `encoded_len` bytes long. Returns kNotImplemented if no
// duration estimate is available, or -1 in case of an error.
virtual int PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const;
// Detects whether a packet has forward error correction. The packet is
- // comprised of the samples in |encoded| which is |encoded_len| bytes long.
+ // comprised of the samples in `encoded` which is `encoded_len` bytes long.
// Returns true if the packet has FEC and false otherwise.
virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
diff --git a/api/audio_codecs/audio_decoder_factory_template.h b/api/audio_codecs/audio_decoder_factory_template.h
index 388668d..976f9c6 100644
--- a/api/audio_codecs/audio_decoder_factory_template.h
+++ b/api/audio_codecs/audio_decoder_factory_template.h
@@ -89,8 +89,8 @@
// Each decoder type is given as a template argument to the function; it should
// be a struct with the following static member functions:
//
-// // Converts |audio_format| to a ConfigType instance. Returns an empty
-// // optional if |audio_format| doesn't correctly specify a decoder of our
+// // Converts `audio_format` to a ConfigType instance. Returns an empty
+// // optional if `audio_format` doesn't correctly specify a decoder of our
// // type.
// absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
//
diff --git a/api/audio_codecs/audio_encoder.h b/api/audio_codecs/audio_encoder.h
index 92e42cf..047d23c 100644
--- a/api/audio_codecs/audio_encoder.h
+++ b/api/audio_codecs/audio_encoder.h
@@ -95,13 +95,13 @@
// This is the main struct for auxiliary encoding information. Each encoded
// packet should be accompanied by one EncodedInfo struct, containing the
- // total number of |encoded_bytes|, the |encoded_timestamp| and the
- // |payload_type|. If the packet contains redundant encodings, the |redundant|
+ // total number of `encoded_bytes`, the `encoded_timestamp` and the
+ // `payload_type`. If the packet contains redundant encodings, the `redundant`
// vector will be populated with EncodedInfoLeaf structs. Each struct in the
// vector represents one encoding; the order of structs in the vector is the
// same as the order in which the actual payloads are written to the byte
// stream. When EncoderInfoLeaf structs are present in the vector, the main
- // struct's |encoded_bytes| will be the sum of all the |encoded_bytes| in the
+ // struct's `encoded_bytes` will be the sum of all the `encoded_bytes` in the
// vector.
struct EncodedInfo : public EncodedInfoLeaf {
EncodedInfo();
@@ -143,7 +143,7 @@
// Accepts one 10 ms block of input audio (i.e., SampleRateHz() / 100 *
// NumChannels() samples). Multi-channel audio must be sample-interleaved.
- // The encoder appends zero or more bytes of output to |encoded| and returns
+ // The encoder appends zero or more bytes of output to `encoded` and returns
// additional encoding information. Encode() checks some preconditions, calls
// EncodeImpl() which does the actual work, and then checks some
// postconditions.
@@ -205,7 +205,7 @@
virtual void DisableAudioNetworkAdaptor();
// Provides uplink packet loss fraction to this encoder to allow it to adapt.
- // |uplink_packet_loss_fraction| is in the range [0.0, 1.0].
+ // `uplink_packet_loss_fraction` is in the range [0.0, 1.0].
virtual void OnReceivedUplinkPacketLossFraction(
float uplink_packet_loss_fraction);
diff --git a/api/audio_codecs/audio_encoder_factory_template.h b/api/audio_codecs/audio_encoder_factory_template.h
index cdc7def..4dc0672 100644
--- a/api/audio_codecs/audio_encoder_factory_template.h
+++ b/api/audio_codecs/audio_encoder_factory_template.h
@@ -103,8 +103,8 @@
// Each encoder type is given as a template argument to the function; it should
// be a struct with the following static member functions:
//
-// // Converts |audio_format| to a ConfigType instance. Returns an empty
-// // optional if |audio_format| doesn't correctly specify an encoder of our
+// // Converts `audio_format` to a ConfigType instance. Returns an empty
+// // optional if `audio_format` doesn't correctly specify an encoder of our
// // type.
// absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
//
diff --git a/api/audio_codecs/audio_format.h b/api/audio_codecs/audio_format.h
index 9f61729..0cf6779 100644
--- a/api/audio_codecs/audio_format.h
+++ b/api/audio_codecs/audio_format.h
@@ -39,7 +39,7 @@
Parameters&& param);
~SdpAudioFormat();
- // Returns true if this format is compatible with |o|. In SDP terminology:
+ // Returns true if this format is compatible with `o`. In SDP terminology:
// would it represent the same codec between an offer and an answer? As
// opposed to operator==, this method disregards codec parameters.
bool Matches(const SdpAudioFormat& o) const;
diff --git a/api/audio_codecs/opus/audio_encoder_opus_config.h b/api/audio_codecs/opus/audio_encoder_opus_config.h
index 3c412b7..d5d7256 100644
--- a/api/audio_codecs/opus/audio_encoder_opus_config.h
+++ b/api/audio_codecs/opus/audio_encoder_opus_config.h
@@ -49,10 +49,10 @@
bool cbr_enabled;
int max_playback_rate_hz;
- // |complexity| is used when the bitrate goes above
- // |complexity_threshold_bps| + |complexity_threshold_window_bps|;
- // |low_rate_complexity| is used when the bitrate falls below
- // |complexity_threshold_bps| - |complexity_threshold_window_bps|. In the
+ // `complexity` is used when the bitrate goes above
+ // `complexity_threshold_bps` + `complexity_threshold_window_bps`;
+ // `low_rate_complexity` is used when the bitrate falls below
+ // `complexity_threshold_bps` - `complexity_threshold_window_bps`. In the
// interval in the middle, we keep using the most recent of the two
// complexity settings.
int complexity;