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Use backticks not vertical bars to denote variables in comments

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Bug: webrtc:12338
Change-Id: I89c8b3a328d04203177522cbdfd9e606fd4bce4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228246
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34696}
This commit is contained in:
Artem Titov
2021-08-10 01:22:31 +02:00
committed by WebRTC LUCI CQ
parent 603e6e3ffc
commit cfea2182f8
114 changed files with 211 additions and 211 deletions
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2
modules/audio_coding/acm2/acm_receiver.h
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@ -180,7 +180,7 @@ class AcmReceiver {
// of NACK list are in the range of [N - `max_nack_list_size`, N).
//
// `max_nack_list_size` should be positive (none zero) and less than or
// equal to |Nack::kNackListSizeLimit|. Otherwise, No change is applied and -1
// equal to `Nack::kNackListSizeLimit`. Otherwise, No change is applied and -1
// is returned. 0 is returned at success.
//
int EnableNack(size_t max_nack_list_size);

4
modules/audio_coding/acm2/audio_coding_module.cc
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@ -229,7 +229,7 @@ int32_t AudioCodingModuleImpl::Encode(
const InputData& input_data,
absl::optional<int64_t> absolute_capture_timestamp_ms) {
// TODO(bugs.webrtc.org/10739): add dcheck that
// |audio_frame.absolute_capture_timestamp_ms()| always has a value.
// `audio_frame.absolute_capture_timestamp_ms()` always has a value.
AudioEncoder::EncodedInfo encoded_info;
uint8_t previous_pltype;
@ -333,7 +333,7 @@ int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
MutexLock lock(&acm_mutex_);
int r = Add10MsDataInternal(audio_frame, &input_data_);
// TODO(bugs.webrtc.org/10739): add dcheck that
// |audio_frame.absolute_capture_timestamp_ms()| always has a value.
// `audio_frame.absolute_capture_timestamp_ms()` always has a value.
return r < 0
? r
: Encode(input_data_, audio_frame.absolute_capture_timestamp_ms());

2
modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
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@ -85,7 +85,7 @@ TEST(AnaBitrateControllerTest, ChangeBitrateOnTargetBitrateChanged) {
1000 /
kInitialFrameLengthMs;
// Frame length unchanged, bitrate changes in accordance with
// |metrics.target_audio_bitrate_bps| and |metrics.overhead_bytes_per_packet|.
// `metrics.target_audio_bitrate_bps` and `metrics.overhead_bytes_per_packet`.
UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
CheckDecision(&controller, kInitialFrameLengthMs, kBitrateBps);
}

2
modules/audio_coding/audio_network_adaptor/config.proto
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@ -169,7 +169,7 @@ message Controller {
// Shorter distance means higher significance. The significances of
// controllers determine their order in the processing pipeline. Controllers
// without `scoring_point` follow their default order in
// |ControllerManager::controllers|.
// `ControllerManager::controllers`.
optional ScoringPoint scoring_point = 1;
oneof controller {

2
modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
View File

@ -101,7 +101,7 @@ void UpdateNetworkMetrics(FecControllerPlrBasedTestStates* states,
}
// Checks that the FEC decision and `uplink_packet_loss_fraction` given by
// |states->controller->MakeDecision| matches `expected_enable_fec` and
// `states->controller->MakeDecision` matches `expected_enable_fec` and
// `expected_uplink_packet_loss_fraction`, respectively.
void CheckDecision(FecControllerPlrBasedTestStates* states,
bool expected_enable_fec,

2
modules/audio_coding/codecs/cng/webrtc_cng.cc
View File

@ -195,7 +195,7 @@ bool ComfortNoiseDecoder::Generate(rtc::ArrayView<int16_t> out_data,
/* `lpPoly` - Coefficients in Q12.
* `excitation` - Speech samples.
* |nst->dec_filtstate| - State preservation.
* `nst->dec_filtstate` - State preservation.
* `out_data` - Filtered speech samples. */
WebRtcSpl_FilterAR(lpPoly, WEBRTC_CNG_MAX_LPC_ORDER + 1, excitation,
num_samples, dec_filtstate_, WEBRTC_CNG_MAX_LPC_ORDER,

8
modules/audio_coding/codecs/isac/main/source/pitch_filter.c
View File

@ -140,9 +140,9 @@ static void FilterSegment(const double* in_data, PitchFilterParam* parameters,
int j;
double sum;
double sum2;
/* Index of |parameters->buffer| where the output is written to. */
/* Index of `parameters->buffer` where the output is written to. */
int pos = parameters->index + PITCH_BUFFSIZE;
/* Index of |parameters->buffer| where samples are read for fractional-lag
/* Index of `parameters->buffer` where samples are read for fractional-lag
* computation. */
int pos_lag = pos - parameters->lag_offset;
@ -174,9 +174,9 @@ static void FilterSegment(const double* in_data, PitchFilterParam* parameters,
/* Filter for fractional pitch. */
sum2 = 0.0;
for (m = PITCH_FRACORDER-1; m >= m_tmp; --m) {
/* |lag_index + m| is always larger than or equal to zero, see how
/* `lag_index + m` is always larger than or equal to zero, see how
* m_tmp is computed. This is equivalent to assume samples outside
* |out_dg[j]| are zero. */
* `out_dg[j]` are zero. */
sum2 += out_dg[j][lag_index + m] * parameters->interpol_coeff[m];
}
/* Add the contribution of differential gain change. */

2
modules/audio_coding/codecs/opus/audio_encoder_opus.h
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@ -139,7 +139,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
absl::optional<int64_t> link_capacity_allocation);
// TODO(minyue): remove "override" when we can deprecate
// |AudioEncoder::SetTargetBitrate|.
// `AudioEncoder::SetTargetBitrate`.
void SetTargetBitrate(int target_bps) override;
void ApplyAudioNetworkAdaptor();

2
modules/audio_coding/codecs/opus/opus_unittest.cc
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@ -116,7 +116,7 @@ class OpusTest
void TestCbrEffect(bool dtx, int block_length_ms);
// Prepare `speech_data_` for encoding, read from a hard-coded file.
// After preparation, |speech_data_.GetNextBlock()| returns a pointer to a
// After preparation, `speech_data_.GetNextBlock()` returns a pointer to a
// block of `block_length_ms` milliseconds. The data is looped every
// `loop_length_ms` milliseconds.
void PrepareSpeechData(int block_length_ms, int loop_length_ms);

6
modules/audio_coding/neteq/neteq_impl_unittest.cc
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@ -510,7 +510,7 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
// Verify |output.packet_infos_|.
// Verify `output.packet_infos_`.
ASSERT_THAT(output.packet_infos_, SizeIs(1));
{
const auto& packet_info = output.packet_infos_[0];
@ -602,7 +602,7 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
// Verify |output.packet_infos_|.
// Verify `output.packet_infos_`.
ASSERT_THAT(output.packet_infos_, SizeIs(1));
{
const auto& packet_info = output.packet_infos_[0];
@ -648,7 +648,7 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
// out-of-order packet should have been discarded.
EXPECT_TRUE(packet_buffer_->Empty());
// Verify |output.packet_infos_|. Expect to only see the second packet.
// Verify `output.packet_infos_`. Expect to only see the second packet.
ASSERT_THAT(output.packet_infos_, SizeIs(1));
{
const auto& packet_info = output.packet_infos_[0];