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

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Bug: webrtc:12338
Change-Id: Ifaad29ccb63b0f2f3aeefb77dae061ebc7f87e6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227024
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34561}
This commit is contained in:
Artem Titov
2021-07-27 12:23:39 +02:00
committed by WebRTC LUCI CQ
parent f0671921a1
commit d7ac581045
87 changed files with 235 additions and 235 deletions
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4
sdk/objc/components/video_codec/RTCVideoEncoderH264.mm
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@ -273,7 +273,7 @@ CFStringRef ExtractProfile(const webrtc::H264ProfileLevelId &profile_level_id) {
}
// The function returns the max allowed sample rate (pixels per second) that
// can be processed by given encoder with |profile_level_id|.
// can be processed by given encoder with `profile_level_id`.
// See https://www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-H.264-201610-S!!PDF-E&type=items
// for details.
NSUInteger GetMaxSampleRate(const webrtc::H264ProfileLevelId &profile_level_id) {
@ -723,7 +723,7 @@ NSUInteger GetMaxSampleRate(const webrtc::H264ProfileLevelId &profile_level_id)
if (_compressionSession) {
SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_AverageBitRate, bitrateBps);
// With zero |_maxAllowedFrameRate|, we fall back to automatic frame rate detection.
// With zero `_maxAllowedFrameRate`, we fall back to automatic frame rate detection.
if (_maxAllowedFrameRate > 0) {
SetVTSessionProperty(
_compressionSession, kVTCompressionPropertyKey_ExpectedFrameRate, frameRate);

2
sdk/objc/components/video_codec/nalu_rewriter.cc
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@ -111,7 +111,7 @@ bool H264CMSampleBufferToAnnexBBuffer(CMSampleBufferRef avcc_sample_buffer,
}
size_t bytes_remaining = block_buffer_size;
while (bytes_remaining > 0) {
// The size type here must match |nalu_header_size|, we expect 4 bytes.
// The size type here must match `nalu_header_size`, we expect 4 bytes.
// Read the length of the next packet of data. Must convert from big endian
// to host endian.
RTC_DCHECK_GE(bytes_remaining, (size_t)nalu_header_size);

6
sdk/objc/components/video_codec/nalu_rewriter.h
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@ -26,7 +26,7 @@ namespace webrtc {
// Converts a sample buffer emitted from the VideoToolbox encoder into a buffer
// suitable for RTP. The sample buffer is in avcc format whereas the rtp buffer
// needs to be in Annex B format. Data is written directly to |annexb_buffer|.
// needs to be in Annex B format. Data is written directly to `annexb_buffer`.
bool H264CMSampleBufferToAnnexBBuffer(CMSampleBufferRef avcc_sample_buffer,
bool is_keyframe,
rtc::Buffer* annexb_buffer);
@ -34,8 +34,8 @@ bool H264CMSampleBufferToAnnexBBuffer(CMSampleBufferRef avcc_sample_buffer,
// Converts a buffer received from RTP into a sample buffer suitable for the
// VideoToolbox decoder. The RTP buffer is in annex b format whereas the sample
// buffer is in avcc format.
// If |is_keyframe| is true then |video_format| is ignored since the format will
// be read from the buffer. Otherwise |video_format| must be provided.
// If `is_keyframe` is true then `video_format` is ignored since the format will
// be read from the buffer. Otherwise `video_format` must be provided.
// Caller is responsible for releasing the created sample buffer.
bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
size_t annexb_buffer_size,