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Reformat the WebRTC code base

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Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This commit is contained in:
Yves Gerey
2018-06-19 15:03:05 +02:00
parent b602123a5a
commit 665174fdbb
1569 changed files with 30495 additions and 30309 deletions
Download Patch File Download Diff File Expand all files Collapse all files

21
ortc/ortcfactory_integrationtest.cc
View File

@ -227,8 +227,7 @@ class OrtcFactoryIntegrationTest : public testing::Test {
new rtc::RefCountedObject<FakePeriodicVideoTrackSource>(
false /* remote */));
return rtc::scoped_refptr<VideoTrackInterface>(
ortc_factory->CreateVideoTrack(
id, fake_video_track_sources_.back()));
ortc_factory->CreateVideoTrack(id, fake_video_track_sources_.back()));
}
// Helper function used to test two way RTP senders and receivers with basic
@ -325,15 +324,15 @@ class OrtcFactoryIntegrationTest : public testing::Test {
fake_audio_capture_module2_->frames_received() >
kDefaultNumFrames &&
fake_video_renderer2.num_rendered_frames() > kDefaultNumFrames,
kDefaultTimeout) << "Audio capture module 1 received "
<< fake_audio_capture_module1_->frames_received()
<< " frames, Video renderer 1 rendered "
<< fake_video_renderer1.num_rendered_frames()
<< " frames, Audio capture module 2 received "
<< fake_audio_capture_module2_->frames_received()
<< " frames, Video renderer 2 rendered "
<< fake_video_renderer2.num_rendered_frames()
<< " frames.";
kDefaultTimeout)
<< "Audio capture module 1 received "
<< fake_audio_capture_module1_->frames_received()
<< " frames, Video renderer 1 rendered "
<< fake_video_renderer1.num_rendered_frames()
<< " frames, Audio capture module 2 received "
<< fake_audio_capture_module2_->frames_received()
<< " frames, Video renderer 2 rendered "
<< fake_video_renderer2.num_rendered_frames() << " frames.";
} else {
WAIT(false, kReceivingDuration);
rendered_video_frames1_ = fake_video_renderer1.num_rendered_frames();

6
ortc/rtptransportcontrolleradapter.cc
View File

@ -127,8 +127,7 @@ RtpTransportControllerAdapter::~RtpTransportControllerAdapter() {
}
// Call must be destroyed on the worker thread.
worker_thread_->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&RtpTransportControllerAdapter::Close_w, this));
RTC_FROM_HERE, rtc::Bind(&RtpTransportControllerAdapter::Close_w, this));
}
RTCErrorOr<std::unique_ptr<RtpTransportInterface>>
@ -636,8 +635,7 @@ RtpTransportControllerAdapter::RtpTransportControllerAdapter(
remote_video_description_.AddCodec(dummy_video);
worker_thread_->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&RtpTransportControllerAdapter::Init_w, this));
RTC_FROM_HERE, rtc::Bind(&RtpTransportControllerAdapter::Init_w, this));
}
// TODO(nisse): Duplicates corresponding method in PeerConnection (used