Use backticks not vertical bars to denote variables in comments
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}
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index 9963603..18b662a 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -180,7 +180,7 @@
// 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);
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index d629139..8ba1b9f 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -229,7 +229,7 @@
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 @@
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());
diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
index 76f52ad..3155f19 100644
--- a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
@@ -85,7 +85,7 @@
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);
}
diff --git a/modules/audio_coding/audio_network_adaptor/config.proto b/modules/audio_coding/audio_network_adaptor/config.proto
index 4f8b2c7..63b220d 100644
--- a/modules/audio_coding/audio_network_adaptor/config.proto
+++ b/modules/audio_coding/audio_network_adaptor/config.proto
@@ -169,7 +169,7 @@
// 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 {
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
index 355431a..743b087 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
@@ -101,7 +101,7 @@
}
// 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,
diff --git a/modules/audio_coding/codecs/cng/webrtc_cng.cc b/modules/audio_coding/codecs/cng/webrtc_cng.cc
index bfe77c7..48f1b8c 100644
--- a/modules/audio_coding/codecs/cng/webrtc_cng.cc
+++ b/modules/audio_coding/codecs/cng/webrtc_cng.cc
@@ -195,7 +195,7 @@
/* `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,
diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_filter.c b/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
index 899d842..bf03dff 100644
--- a/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
+++ b/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
@@ -140,9 +140,9 @@
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 @@
/* 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. */
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index ab954fe..c7ee4f4 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -139,7 +139,7 @@
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();
diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc
index b507a32..b40d738 100644
--- a/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -116,7 +116,7 @@
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);
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 875e62c..b0fee47 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -510,7 +510,7 @@
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 @@
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 @@
// 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];