commit d00ce747c7588b1541ad6a0c0db7895487b5a618
author Artem Titov <titovartem@webrtc.org> Wed Jul 28 20:00:17 2021 +0200
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Mon Aug 02 10:45:40 2021 +0000
tree 7e75bffaa9c515f610afce30f7c377a0939ac56f
parent 0146a34b3f417181f2fabbcc19150c3fd95f1999

Use backticks not vertical bars to denote variables in comments for /modules/audio_coding

Bug: webrtc:12338
Change-Id: I02613d9fca45d00e2477f334b7a0416e7912e26b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227037
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34621}

modules/audio_coding/neteq/neteq_impl.h

diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index 88da6dc..e3d84b3 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -141,7 +141,7 @@
   bool RegisterPayloadType(int rtp_payload_type,
                            const SdpAudioFormat& audio_format) override;
 
-  // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
+  // Removes `rtp_payload_type` from the codec database. Returns 0 on success,
   // -1 on failure.
   int RemovePayloadType(uint8_t rtp_payload_type) override;
 
@@ -159,7 +159,7 @@
 
   int FilteredCurrentDelayMs() const override;
 
-  // Writes the current network statistics to |stats|. The statistics are reset
+  // Writes the current network statistics to `stats`. The statistics are reset
   // after the call.
   int NetworkStatistics(NetEqNetworkStatistics* stats) override;
 
@@ -215,7 +215,7 @@
                            rtc::ArrayView<const uint8_t> payload)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
 
-  // Delivers 10 ms of audio data. The data is written to |audio_frame|.
+  // Delivers 10 ms of audio data. The data is written to `audio_frame`.
   // Returns 0 on success, otherwise an error code.
   int GetAudioInternal(AudioFrame* audio_frame,
                        bool* muted,
@@ -223,9 +223,9 @@
       RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
 
   // Provides a decision to the GetAudioInternal method. The decision what to
-  // do is written to |operation|. Packets to decode are written to
-  // |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
-  // DTMF should be played, |play_dtmf| is set to true by the method.
+  // do is written to `operation`. Packets to decode are written to
+  // `packet_list`, and a DTMF event to play is written to `dtmf_event`. When
+  // DTMF should be played, `play_dtmf` is set to true by the method.
   // Returns 0 on success, otherwise an error code.
   int GetDecision(Operation* operation,
                   PacketList* packet_list,
@@ -234,11 +234,11 @@
                   absl::optional<Operation> action_override)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
 
-  // Decodes the speech packets in |packet_list|, and writes the results to
-  // |decoded_buffer|, which is allocated to hold |decoded_buffer_length|
-  // elements. The length of the decoded data is written to |decoded_length|.
+  // Decodes the speech packets in `packet_list`, and writes the results to
+  // `decoded_buffer`, which is allocated to hold `decoded_buffer_length`
+  // elements. The length of the decoded data is written to `decoded_length`.
   // The speech type -- speech or (codec-internal) comfort noise -- is written
-  // to |speech_type|. If |packet_list| contains any SID frames for RFC 3389
+  // to `speech_type`. If `packet_list` contains any SID frames for RFC 3389
   // comfort noise, those are not decoded.
   int Decode(PacketList* packet_list,
              Operation* operation,
@@ -293,7 +293,7 @@
                          bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
 
   // Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
-  // noise. |packet_list| can either contain one SID frame to update the
+  // noise. `packet_list` can either contain one SID frame to update the
   // noise parameters, or no payload at all, in which case the previously
   // received parameters are used.
   int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
@@ -308,20 +308,20 @@
   int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
 
-  // Overdub DTMF on top of |output|.
+  // Overdub DTMF on top of `output`.
   int DtmfOverdub(const DtmfEvent& dtmf_event,
                   size_t num_channels,
                   int16_t* output) const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
 
-  // Extracts packets from |packet_buffer_| to produce at least
-  // |required_samples| samples. The packets are inserted into |packet_list|.
+  // Extracts packets from `packet_buffer_` to produce at least
+  // `required_samples` samples. The packets are inserted into `packet_list`.
   // Returns the number of samples that the packets in the list will produce, or
   // -1 in case of an error.
   int ExtractPackets(size_t required_samples, PacketList* packet_list)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
 
   // Resets various variables and objects to new values based on the sample rate
-  // |fs_hz| and |channels| number audio channels.
+  // `fs_hz` and `channels` number audio channels.
   void SetSampleRateAndChannels(int fs_hz, size_t channels)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);