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

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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}
This commit is contained in:
Artem Titov
2021-07-28 20:00:17 +02:00
committed by WebRTC LUCI CQ
parent 0146a34b3f
commit d00ce747c7
143 changed files with 809 additions and 809 deletions
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22
modules/audio_coding/codecs/opus/opus_interface.cc
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@ -574,8 +574,8 @@ void WebRtcOpus_DecoderInit(OpusDecInst* inst) {
/* For decoder to determine if it is to output speech or comfort noise. */
static int16_t DetermineAudioType(OpusDecInst* inst, size_t encoded_bytes) {
// Audio type becomes comfort noise if |encoded_byte| is 1 and keeps
// to be so if the following |encoded_byte| are 0 or 1.
// Audio type becomes comfort noise if `encoded_byte` is 1 and keeps
// to be so if the following `encoded_byte` are 0 or 1.
if (encoded_bytes == 0 && inst->in_dtx_mode) {
return 2; // Comfort noise.
} else if (encoded_bytes == 1 || encoded_bytes == 2) {
@ -595,7 +595,7 @@ static int16_t DetermineAudioType(OpusDecInst* inst, size_t encoded_bytes) {
}
}
/* |frame_size| is set to maximum Opus frame size in the normal case, and
/* `frame_size` is set to maximum Opus frame size in the normal case, and
* is set to the number of samples needed for PLC in case of losses.
* It is up to the caller to make sure the value is correct. */
static int DecodeNative(OpusDecInst* inst,
@ -632,9 +632,9 @@ static int DecodePlc(OpusDecInst* inst, int16_t* decoded) {
FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz);
if (inst->plc_use_prev_decoded_samples) {
/* The number of samples we ask for is |number_of_lost_frames| times
* |prev_decoded_samples_|. Limit the number of samples to maximum
* |MaxFrameSizePerChannel()|. */
/* The number of samples we ask for is `number_of_lost_frames` times
* `prev_decoded_samples_`. Limit the number of samples to maximum
* `MaxFrameSizePerChannel()`. */
plc_samples = inst->prev_decoded_samples;
const int max_samples_per_channel =
MaxFrameSizePerChannel(inst->sample_rate_hz);
@ -729,9 +729,9 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst,
int WebRtcOpus_PlcDuration(OpusDecInst* inst) {
if (inst->plc_use_prev_decoded_samples) {
/* The number of samples we ask for is |number_of_lost_frames| times
* |prev_decoded_samples_|. Limit the number of samples to maximum
* |MaxFrameSizePerChannel()|. */
/* The number of samples we ask for is `number_of_lost_frames` times
* `prev_decoded_samples_`. Limit the number of samples to maximum
* `MaxFrameSizePerChannel()`. */
const int plc_samples = inst->prev_decoded_samples;
const int max_samples_per_channel =
MaxFrameSizePerChannel(inst->sample_rate_hz);
@ -826,8 +826,8 @@ int WebRtcOpus_PacketHasFec(const uint8_t* payload,
// as binary values with uniform probability, they can be extracted directly
// from the most significant bits of the first byte of compressed data.
for (int n = 0; n < channels; n++) {
// The LBRR bit for channel 1 is on the (|silk_frames| + 1)-th bit, and
// that of channel 2 is on the |(|silk_frames| + 1) * 2 + 1|-th bit.
// The LBRR bit for channel 1 is on the (`silk_frames` + 1)-th bit, and
// that of channel 2 is on the |(`silk_frames` + 1) * 2 + 1|-th bit.
if (frame_data[0][0] & (0x80 >> ((n + 1) * (silk_frames + 1) - 1)))
return 1;
}