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

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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}
This commit is contained in:
Artem Titov
2021-07-26 15:29:11 +02:00
committed by WebRTC LUCI CQ
parent 75ac5ab859
commit 37ee0f5e59
81 changed files with 535 additions and 535 deletions
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24
pc/rtp_sender_receiver_unittest.cc
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@ -494,7 +494,7 @@ class RtpSenderReceiverTest
}
// Check that minimum Jitter Buffer delay is propagated to the underlying
// |media_channel|.
// `media_channel`.
void VerifyRtpReceiverDelayBehaviour(cricket::Delayable* media_channel,
RtpReceiverInterface* receiver,
uint32_t ssrc) {
@ -509,13 +509,13 @@ class RtpSenderReceiverTest
rtc::Thread* const network_thread_;
rtc::Thread* const worker_thread_;
webrtc::RtcEventLogNull event_log_;
// The |rtp_dtls_transport_| and |rtp_transport_| should be destroyed after
// the |channel_manager|.
// The `rtp_dtls_transport_` and `rtp_transport_` should be destroyed after
// the `channel_manager`.
std::unique_ptr<cricket::DtlsTransportInternal> rtp_dtls_transport_;
std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport_;
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory_;
// |media_engine_| is actually owned by |channel_manager_|.
// `media_engine_` is actually owned by `channel_manager_`.
cricket::FakeMediaEngine* media_engine_;
std::unique_ptr<cricket::ChannelManager> channel_manager_;
cricket::FakeCall fake_call_;
@ -534,28 +534,28 @@ class RtpSenderReceiverTest
rtc::UniqueRandomIdGenerator ssrc_generator_;
};
// Test that |voice_channel_| is updated when an audio track is associated
// Test that `voice_channel_` is updated when an audio track is associated
// and disassociated with an AudioRtpSender.
TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpSender) {
CreateAudioRtpSender();
DestroyAudioRtpSender();
}
// Test that |video_channel_| is updated when a video track is associated and
// Test that `video_channel_` is updated when a video track is associated and
// disassociated with a VideoRtpSender.
TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpSender) {
CreateVideoRtpSender();
DestroyVideoRtpSender();
}
// Test that |voice_channel_| is updated when a remote audio track is
// Test that `voice_channel_` is updated when a remote audio track is
// associated and disassociated with an AudioRtpReceiver.
TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpReceiver) {
CreateAudioRtpReceiver();
DestroyAudioRtpReceiver();
}
// Test that |video_channel_| is updated when a remote video track is
// Test that `video_channel_` is updated when a remote video track is
// associated and disassociated with a VideoRtpReceiver.
TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpReceiver) {
CreateVideoRtpReceiver();
@ -1423,7 +1423,7 @@ TEST_F(RtpSenderReceiverTest, PropagatesVideoTrackContentHint) {
video_track_->set_enabled(true);
// |video_track_| is not screencast by default.
// `video_track_` is not screencast by default.
EXPECT_EQ(false, video_media_channel_->options().is_screencast);
// No content hint should be set by default.
EXPECT_EQ(VideoTrackInterface::ContentHint::kNone,
@ -1453,7 +1453,7 @@ TEST_F(RtpSenderReceiverTest,
video_track_->set_enabled(true);
// |video_track_| with a screencast source should be screencast by default.
// `video_track_` with a screencast source should be screencast by default.
EXPECT_EQ(true, video_media_channel_->options().is_screencast);
// No content hint should be set by default.
EXPECT_EQ(VideoTrackInterface::ContentHint::kNone,
@ -1518,8 +1518,8 @@ TEST_F(RtpSenderReceiverTest, VideoSenderDoesNotHaveDtmfSender) {
EXPECT_EQ(nullptr, video_rtp_sender_->GetDtmfSender());
}
// Test that the DTMF sender is really using |voice_channel_|, and thus returns
// true/false from CanSendDtmf based on what |voice_channel_| returns.
// Test that the DTMF sender is really using `voice_channel_`, and thus returns
// true/false from CanSendDtmf based on what `voice_channel_` returns.
TEST_F(RtpSenderReceiverTest, CanInsertDtmf) {
AddDtmfCodec();
CreateAudioRtpSender();