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Yves Gerey 665174fdbb Reformat the WebRTC code base 
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}
2018-06-19 14:00:39 +00:00
..
end_to_end_tests
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
test
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
BUILD.gn
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
call_stats_unittest.cc
Reduce locking for CallStats (preparation for TaskQueue).
2018-03-28 13:24:07 +00:00
call_stats.cc
Reduce locking for CallStats (preparation for TaskQueue).
2018-03-28 13:24:07 +00:00
call_stats.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
DEPS
Move remaining traces of VoiceEngine
2018-01-17 13:27:47 +00:00
encoder_rtcp_feedback_unittest.cc
Adds unit tests for VideoSendStreamImpl.
2018-04-20 13:12:13 +00:00
encoder_rtcp_feedback.cc
Adds unit tests for VideoSendStreamImpl.
2018-04-20 13:12:13 +00:00
encoder_rtcp_feedback.h
Adds unit tests for VideoSendStreamImpl.
2018-04-20 13:12:13 +00:00
full_stack_tests_plot.py
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
full_stack_tests.cc
Add full stack test with link capacity of 150 kbit/s.
2018-06-08 11:33:30 +00:00
overuse_frame_detector_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
overuse_frame_detector.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
overuse_frame_detector.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
OWNERS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
payload_router_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
payload_router.cc
Moved timing related logic into its own function in webrtc::PayloadRouter.
2018-06-13 11:28:46 +00:00
payload_router.h
Move BitrateAllocation to api/ and rename it VideoBitrateAllocation
2018-04-23 15:31:27 +00:00
picture_id_tests.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
quality_scaling_tests.cc
Add tests for quality scaling.
2018-05-04 07:51:04 +00:00
quality_threshold_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
quality_threshold.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
quality_threshold.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
receive_statistics_proxy_unittest.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
receive_statistics_proxy.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
receive_statistics_proxy.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
replay.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
report_block_stats_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
report_block_stats.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
report_block_stats.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_streams_synchronizer.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_streams_synchronizer.h
Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY.
2018-02-07 10:07:28 +00:00
rtp_video_stream_receiver_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_video_stream_receiver.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_video_stream_receiver.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
screenshare_loopback.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
send_delay_stats_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
send_delay_stats.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
send_delay_stats.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
send_statistics_proxy_unittest.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
send_statistics_proxy.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
send_statistics_proxy.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
stats_counter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
stats_counter.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
stats_counter.h
Adding NOLINT for typedefs.h and common_types.h
2017-09-15 13:03:51 +00:00
stream_synchronization_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
stream_synchronization.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
stream_synchronization.h
Adding NOLINT for typedefs.h and common_types.h
2017-09-15 13:03:51 +00:00
sv_loopback.cc
Add full stack test with link capacity of 150 kbit/s.
2018-06-08 11:33:30 +00:00
transport_adapter.cc
Replace Atomic32 with std::atomic in video/
2017-10-24 23:40:29 +00:00
transport_adapter.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_loopback.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_quality_observer.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_quality_observer.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_quality_test.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_quality_test.h
Add full stack test with link capacity of 150 kbit/s.
2018-06-08 11:33:30 +00:00
video_receive_stream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_receive_stream.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_receive_stream.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_send_stream_impl_unittest.cc
Adds unit tests for VideoSendStreamImpl.
2018-04-20 13:12:13 +00:00
video_send_stream_impl.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_send_stream_impl.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_send_stream_tests.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_send_stream.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_send_stream.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_stream_decoder_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_stream_decoder_impl.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_stream_decoder.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_stream_decoder.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_stream_encoder_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_stream_encoder.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_stream_encoder.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00