Reland "Use backticks not vertical bars to denote variables in comments for /pc"
Original change's description:
> Revert "Use backticks not vertical bars to denote variables in comments for /pc"
>
> This reverts commit 37ee0f5e594dd772ec6d620b5e5ea8a751b684f0.
>
> Reason for revert: Revert in order to be able to revert https://webrtc-review.googlesource.com/c/src/+/225642
>
> Original change's description:
> > Use backticks not vertical bars to denote variables in comments for /pc
> >
> > Bug: webrtc:12338
> > Change-Id: I88cf10afa5fc810b95d2a585ab2e895dcc163b63
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226953
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Artem Titov <titovartem@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34575}
>
> TBR=hta@webrtc.org,titovartem@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I5eddd3a14e1f664bf831e5c294fbc4de5f6a88af
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12338
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227082
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34577}
Bug: webrtc:12338
Change-Id: I96bd229b73613c162d11d75fa4f5934e1b4295c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227087
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34611}
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 1fdc736..c2b453e 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -377,7 +377,7 @@
inbound_audio->total_audio_energy = voice_receiver_info.total_output_energy;
inbound_audio->total_samples_duration =
voice_receiver_info.total_output_duration;
- // |fir_count|, |pli_count| and |sli_count| are only valid for video and are
+ // `fir_count`, `pli_count` and `sli_count` are only valid for video and are
// purposefully left undefined for audio.
if (voice_receiver_info.last_packet_received_timestamp_ms) {
inbound_audio->last_packet_received_timestamp = static_cast<double>(
@@ -491,7 +491,7 @@
inbound_video->estimated_playout_timestamp = static_cast<double>(
*video_receiver_info.estimated_playout_ntp_timestamp_ms);
}
- // TODO(bugs.webrtc.org/10529): When info's |content_info| is optional
+ // TODO(bugs.webrtc.org/10529): When info's `content_info` is optional
// support the "unspecified" value.
if (video_receiver_info.content_type == VideoContentType::SCREENSHARE)
inbound_video->content_type = RTCContentType::kScreenshare;
@@ -532,7 +532,7 @@
outbound_audio->codec_id = RTCCodecStatsIDFromMidDirectionAndPayload(
mid, /*inbound=*/false, *voice_sender_info.codec_payload_type);
}
- // |fir_count|, |pli_count| and |sli_count| are only valid for video and are
+ // `fir_count`, `pli_count` and `sli_count` are only valid for video and are
// purposefully left undefined for audio.
}
@@ -585,7 +585,7 @@
video_sender_info.quality_limitation_durations_ms);
outbound_video->quality_limitation_resolution_changes =
video_sender_info.quality_limitation_resolution_changes;
- // TODO(https://crbug.com/webrtc/10529): When info's |content_info| is
+ // TODO(https://crbug.com/webrtc/10529): When info's `content_info` is
// optional, support the "unspecified" value.
if (video_sender_info.content_type == VideoContentType::SCREENSHARE)
outbound_video->content_type = RTCContentType::kScreenshare;
@@ -629,7 +629,7 @@
std::string local_id =
RTCOutboundRTPStreamStatsIDFromSSRC(media_type, report_block.source_ssrc);
- // Look up local stat from |outbound_rtps| where the pointers are non-const.
+ // Look up local stat from `outbound_rtps` where the pointers are non-const.
auto local_id_it = outbound_rtps.find(local_id);
if (local_id_it != outbound_rtps.end()) {
remote_inbound->local_id = local_id;
@@ -780,7 +780,7 @@
voice_sender_info.apm_statistics);
auto audio_processor(audio_track.GetAudioProcessor());
if (audio_processor.get()) {
- // The |has_remote_tracks| argument is obsolete; makes no difference if it's
+ // The `has_remote_tracks` argument is obsolete; makes no difference if it's
// set to true or false.
AudioProcessorInterface::AudioProcessorStatistics ap_stats =
audio_processor->GetStats(/*has_remote_tracks=*/false);
@@ -1213,7 +1213,7 @@
this, cached_report_, std::move(requests)));
} else if (!num_pending_partial_reports_) {
// Only start gathering stats if we're not already gathering stats. In the
- // case of already gathering stats, |callback_| will be invoked when there
+ // case of already gathering stats, `callback_` will be invoked when there
// are no more pending partial reports.
// "Now" using a system clock, relative to the UNIX epoch (Jan 1, 1970,
@@ -1224,13 +1224,13 @@
num_pending_partial_reports_ = 2;
partial_report_timestamp_us_ = cache_now_us;
- // Prepare |transceiver_stats_infos_| and |call_stats_| for use in
- // |ProducePartialResultsOnNetworkThread| and
- // |ProducePartialResultsOnSignalingThread|.
+ // Prepare `transceiver_stats_infos_` and `call_stats_` for use in
+ // `ProducePartialResultsOnNetworkThread` and
+ // `ProducePartialResultsOnSignalingThread`.
PrepareTransceiverStatsInfosAndCallStats_s_w_n();
- // Don't touch |network_report_| on the signaling thread until
+ // Don't touch `network_report_` on the signaling thread until
// ProducePartialResultsOnNetworkThread() has signaled the
- // |network_report_event_|.
+ // `network_report_event_`.
network_report_event_.Reset();
rtc::scoped_refptr<RTCStatsCollector> collector(this);
network_thread_->PostTask(
@@ -1251,7 +1251,7 @@
void RTCStatsCollector::WaitForPendingRequest() {
RTC_DCHECK_RUN_ON(signaling_thread_);
- // If a request is pending, blocks until the |network_report_event_| is
+ // If a request is pending, blocks until the `network_report_event_` is
// signaled and then delivers the result. Otherwise this is a NO-OP.
MergeNetworkReport_s();
}
@@ -1295,8 +1295,8 @@
RTC_DCHECK_RUN_ON(network_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
- // Touching |network_report_| on this thread is safe by this method because
- // |network_report_event_| is reset before this method is invoked.
+ // Touching `network_report_` on this thread is safe by this method because
+ // `network_report_event_` is reset before this method is invoked.
network_report_ = RTCStatsReport::Create(timestamp_us);
std::set<std::string> transport_names;
@@ -1318,7 +1318,7 @@
timestamp_us, transport_stats_by_name, transport_cert_stats,
network_report_.get());
- // Signal that it is now safe to touch |network_report_| on the signaling
+ // Signal that it is now safe to touch `network_report_` on the signaling
// thread, and post a task to merge it into the final results.
network_report_event_.Set();
rtc::scoped_refptr<RTCStatsCollector> collector(this);
@@ -1347,16 +1347,16 @@
void RTCStatsCollector::MergeNetworkReport_s() {
RTC_DCHECK_RUN_ON(signaling_thread_);
- // The |network_report_event_| must be signaled for it to be safe to touch
- // |network_report_|. This is normally not blocking, but if
+ // The `network_report_event_` must be signaled for it to be safe to touch
+ // `network_report_`. This is normally not blocking, but if
// WaitForPendingRequest() is called while a request is pending, we might have
- // to wait until the network thread is done touching |network_report_|.
+ // to wait until the network thread is done touching `network_report_`.
network_report_event_.Wait(rtc::Event::kForever);
if (!network_report_) {
// Normally, MergeNetworkReport_s() is executed because it is posted from
// the network thread. But if WaitForPendingRequest() is called while a
// request is pending, an early call to MergeNetworkReport_s() is made,
- // merging the report and setting |network_report_| to null. If so, when the
+ // merging the report and setting `network_report_` to null. If so, when the
// previously posted MergeNetworkReport_s() is later executed, the report is
// already null and nothing needs to be done here.
return;
@@ -1366,8 +1366,8 @@
partial_report_->TakeMembersFrom(network_report_);
network_report_ = nullptr;
--num_pending_partial_reports_;
- // |network_report_| is currently the only partial report collected
- // asynchronously, so |num_pending_partial_reports_| must now be 0 and we are
+ // `network_report_` is currently the only partial report collected
+ // asynchronously, so `num_pending_partial_reports_` must now be 0 and we are
// ready to deliver the result.
RTC_DCHECK_EQ(num_pending_partial_reports_, 0);
cache_timestamp_us_ = partial_report_timestamp_us_;
@@ -1380,7 +1380,7 @@
TRACE_EVENT_INSTANT1("webrtc_stats", "webrtc_stats", "report",
cached_report_->ToJson());
- // Deliver report and clear |requests_|.
+ // Deliver report and clear `requests_`.
std::vector<RequestInfo> requests;
requests.swap(requests_);
DeliverCachedReport(cached_report_, std::move(requests));
@@ -1704,7 +1704,7 @@
// stream, so look in both places.
auto audio_processor(audio_track->GetAudioProcessor());
if (audio_processor.get()) {
- // The |has_remote_tracks| argument is obsolete; makes no difference
+ // The `has_remote_tracks` argument is obsolete; makes no difference
// if it's set to true or false.
AudioProcessorInterface::AudioProcessorStatistics ap_stats =
audio_processor->GetStats(/*has_remote_tracks=*/false);
@@ -2218,7 +2218,7 @@
void RTCStatsCollector::OnDataChannelClosed(DataChannelInterface* channel) {
RTC_DCHECK_RUN_ON(signaling_thread_);
// Only channels that have been fully opened (and have increased the
- // |data_channels_opened_| counter) increase the closed counter.
+ // `data_channels_opened_` counter) increase the closed counter.
if (internal_record_.opened_data_channels.erase(
reinterpret_cast<uintptr_t>(channel))) {
++internal_record_.data_channels_closed;
