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}
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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@ -320,7 +320,7 @@ void TestAllCodecs::RegisterSendCodec(char side,
// If G.722, store half the size to compensate for the timestamp bug in the
// RFC for G.722.
// If iSAC runs in adaptive mode, packet size in samples can change on the
// fly, so we exclude this test by setting |packet_size_samples_| to -1.
// fly, so we exclude this test by setting `packet_size_samples_` to -1.
int clockrate_hz = sampling_freq_hz;
size_t num_channels = 1;
if (absl::EqualsIgnoreCase(codec_name, "G722")) {

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@ -509,8 +509,8 @@ void TestStereo::Run(TestPackStereo* channel,
in_file_mono_->FastForward(100);
while (1) {
// Simulate packet loss by setting |packet_loss_| to "true" in
// |percent_loss| percent of the loops.
// Simulate packet loss by setting `packet_loss_` to "true" in
// `percent_loss` percent of the loops.
if (percent_loss > 0) {
if (counter_ == floor((100 / percent_loss) + 0.5)) {
counter_ = 0;

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@ -23,7 +23,7 @@
namespace webrtc {
// This class records the frame type, and delegates actual sending to the
// |next_| AudioPacketizationCallback.
// `next_` AudioPacketizationCallback.
class MonitoringAudioPacketizationCallback : public AudioPacketizationCallback {
public:
explicit MonitoringAudioPacketizationCallback(
@ -67,9 +67,9 @@ class TestVadDtx {
// the expectation. Saves result to a file.
// expects[x] means
// -1 : do not care,
// 0 : there have been no packets of type |x|,
// 1 : there have been packets of type |x|,
// with |x| indicates the following packet types
// 0 : there have been no packets of type `x`,
// 1 : there have been packets of type `x`,
// with `x` indicates the following packet types
// 0 - kEmptyFrame
// 1 - kAudioFrameSpeech
// 2 - kAudioFrameCN

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@ -277,8 +277,8 @@ void OpusTest::Run(TestPackStereo* channel,
ASSERT_GE(bitstream_len_byte_int, 0);
bitstream_len_byte = static_cast<size_t>(bitstream_len_byte_int);
// Simulate packet loss by setting |packet_loss_| to "true" in
// |percent_loss| percent of the loops.
// Simulate packet loss by setting `packet_loss_` to "true" in
// `percent_loss` percent of the loops.
// TODO(tlegrand): Move handling of loss simulation to TestPackStereo.
if (percent_loss > 0) {
if (counter_ == floor((100 / percent_loss) + 0.5)) {