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webrtc/call
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Jonas Olsson a4d873786f Format almost everything. 
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00
..
test Format almost everything. 2019-07-08 13:45:15 +00:00
audio_receive_stream.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_receive_stream.h Delete RTCInboundRTPStreamStats::fraction_lost 2019-06-26 11:43:23 +00:00
audio_send_stream.cc Introduce MediaTransportConfig 2019-05-21 18:58:33 +00:00
audio_send_stream.h [getStats] Implement "media-source" audio levels, fixing Chrome bug. 2019-07-04 08:13:45 +00:00
audio_state.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_state.h [getStats] Implement "media-source" audio levels, fixing Chrome bug. 2019-07-04 08:13:45 +00:00
bitrate_allocator_unittest.cc Format almost everything. 2019-07-08 13:45:15 +00:00
bitrate_allocator.cc Removes SendSideCongestionController. 2019-06-11 11:09:24 +00:00
bitrate_allocator.h Enable experiments with audio bitrate priority. 2019-05-08 14:21:01 +00:00
bitrate_estimator_tests.cc Use explicit TaskQueueFactory for FrameGeneratorCapturer in BitrateEstimatorTest. 2019-04-18 14:17:12 +00:00
BUILD.gn Make TaskQueueFactory required construction parameter for Call 2019-07-03 14:02:45 +00:00
call_config.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
call_config.h Make TaskQueueFactory required construction parameter for Call 2019-07-03 14:02:45 +00:00
call_factory.cc Format almost everything. 2019-07-08 13:45:15 +00:00
call_factory.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
call_perf_tests.cc Reland "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" 2019-06-12 14:44:01 +00:00
call_unittest.cc Format almost everything. 2019-07-08 13:45:15 +00:00
call.cc Format almost everything. 2019-07-08 13:45:15 +00:00
call.h Move TaskQueueFactory from Call::Create parameter to CallConfig 2019-04-01 09:27:44 +00:00
degraded_call.cc Format almost everything. 2019-07-08 13:45:15 +00:00
degraded_call.h Format almost everything. 2019-07-08 13:45:15 +00:00
DEPS Make fec controller plug-able. 2018-01-22 11:48:16 +00:00
fake_network_pipe_unittest.cc ClangTidy fixes for call/ 2019-03-14 09:38:01 +00:00
fake_network_pipe.cc Format almost everything. 2019-07-08 13:45:15 +00:00
fake_network_pipe.h Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
flexfec_receive_stream_impl.cc Format almost everything. 2019-07-08 13:45:15 +00:00
flexfec_receive_stream_impl.h Injecting Clock in video receive. 2019-03-04 21:53:57 +00:00
flexfec_receive_stream_unittest.cc Format almost everything. 2019-07-08 13:45:15 +00:00
flexfec_receive_stream.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
flexfec_receive_stream.h Format almost everything. 2019-07-08 13:45:15 +00:00
OWNERS Remove myself from OWNERS in a few places. 2019-06-10 07:57:46 +00:00
packet_receiver.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
rampup_tests.cc Delete RtcEventLogFactory factory as now unused 2019-06-27 10:03:22 +00:00
rampup_tests.h Refactor bitrate configuration in CallTest 2018-11-13 16:03:00 +00:00
receive_time_calculator_unittest.cc Format almost everything. 2019-07-08 13:45:15 +00:00
receive_time_calculator.cc [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +00:00
receive_time_calculator.h Format almost everything. 2019-07-08 13:45:15 +00:00
rtcp_demuxer_unittest.cc Format almost everything. 2019-07-08 13:45:15 +00:00
rtcp_demuxer.cc Format almost everything. 2019-07-08 13:45:15 +00:00
rtcp_demuxer.h Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
rtcp_packet_sink_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_bitrate_configurator_unittest.cc Format almost everything. 2019-07-08 13:45:15 +00:00
rtp_bitrate_configurator.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_bitrate_configurator.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
rtp_config.cc Negotiate use of RTCP loss notification feedback (LNTF) 2019-05-24 12:44:14 +00:00
rtp_config.h Format almost everything. 2019-07-08 13:45:15 +00:00
rtp_demuxer_unittest.cc Fully qualify googletest symbols. 2019-04-09 17:18:20 +00:00
rtp_demuxer.cc Log details when RtpDemuxer fails to deliver a packet 2019-04-16 00:47:53 +00:00
rtp_demuxer.h Log details when RtpDemuxer fails to deliver a packet 2019-04-16 00:47:53 +00:00
rtp_packet_sink_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_payload_params_unittest.cc Format almost everything. 2019-07-08 13:45:15 +00:00
rtp_payload_params.cc Format almost everything. 2019-07-08 13:45:15 +00:00
rtp_payload_params.h Reland "Populate the GFD-00 for H264 and generic codecs." 2019-06-14 14:47:06 +00:00
rtp_rtcp_demuxer_helper_unittest.cc Format almost everything. 2019-07-08 13:45:15 +00:00
rtp_rtcp_demuxer_helper.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_rtcp_demuxer_helper.h [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +00:00
rtp_stream_receiver_controller_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_stream_receiver_controller.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
rtp_stream_receiver_controller.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
rtp_transport_controller_send_interface.h Switch from RtpPacketSender to RtpPacketPacer interface usage. 2019-06-24 10:46:06 +00:00
rtp_transport_controller_send.cc Format almost everything. 2019-07-08 13:45:15 +00:00
rtp_transport_controller_send.h Switch from RtpPacketSender to RtpPacketPacer interface usage. 2019-06-24 10:46:06 +00:00
rtp_video_sender_interface.h Define FecControllerOverride and plumb it down to VideoEncoder 2019-06-28 15:57:22 +00:00
rtp_video_sender_unittest.cc Format almost everything. 2019-07-08 13:45:15 +00:00
rtp_video_sender.cc Implement RtpVideoSender::SetFecAllowed() 2019-06-28 18:24:16 +00:00
rtp_video_sender.h Implement RtpVideoSender::SetFecAllowed() 2019-06-28 18:24:16 +00:00
rtx_receive_stream_unittest.cc Format almost everything. 2019-07-08 13:45:15 +00:00
rtx_receive_stream.cc Format almost everything. 2019-07-08 13:45:15 +00:00
rtx_receive_stream.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
simulated_network_unittest.cc Format almost everything. 2019-07-08 13:45:15 +00:00
simulated_network.cc Adds CoDel implementation to network simulation. 2019-02-25 09:54:03 +00:00
simulated_network.h Format almost everything. 2019-07-08 13:45:15 +00:00
simulated_packet_receiver.h Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
ssrc_binding_observer.h Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
syncable.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
syncable.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
video_receive_stream.cc Format almost everything. 2019-07-08 13:45:15 +00:00
video_receive_stream.h Add totalDecodeTime to RTCInboundRTPStreamStats 2019-07-02 08:28:06 +00:00
video_send_stream.cc Reland "Inform VideoEncoder of negotiated capabilities" 2019-06-11 14:49:37 +00:00
video_send_stream.h Format almost everything. 2019-07-08 13:45:15 +00:00