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platform-external-webrtc/call
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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
..
test
Reformat the WebRTC code base
2018-06-19 14:00:39 +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
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
audio_send_stream.cc
replace stringstream in call/
2018-04-04 16:09:15 +00:00
audio_send_stream.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +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
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bitrate_allocator_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bitrate_allocator.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bitrate_allocator.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bitrate_constraints.h
New api struct BitrateSettings.
2018-05-07 15:01:28 +00:00
bitrate_estimator_tests.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
BUILD.gn
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
call_config.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
call_config.h
Allows injection of network controller factory into peer connection factory.
2018-05-18 17:07:16 +00:00
call_perf_tests.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
call_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
call.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
call.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
callfactory.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
callfactory.h
Rename Call::Config to CallConfig, keep old name as alias.
2018-02-14 15:14:39 +00:00
degraded_call.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
degraded_call.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
DEPS
Make fec controller plug-able.
2018-01-22 11:48:16 +00:00
fake_network_pipe.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
fake_network_pipe.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
flexfec_receive_stream_impl.cc
replace stringstream in call/
2018-04-04 16:09:15 +00:00
flexfec_receive_stream_impl.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
flexfec_receive_stream_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
flexfec_receive_stream.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
flexfec_receive_stream.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
OWNERS
Added nisse@webrtc.org as owner in call.
2018-02-20 09:39:51 +00:00
rampup_tests.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rampup_tests.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
receive_time_calculator_unittest.cc
Added receive time calculator under field trial.
2018-03-21 15:40:39 +00:00
receive_time_calculator.cc
Fix of ReceiveTimeCalculator field trial parsing.
2018-03-24 12:36:17 +00:00
receive_time_calculator.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtcp_demuxer_unittest.cc
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
rtcp_demuxer.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +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
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +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
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_config.cc
replace stringstream in call/
2018-04-04 16:09:15 +00:00
rtp_config.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_demuxer_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_demuxer.cc
Don't check MIDs when demuxing RTP packets in Call
2018-03-29 20:36:08 +00:00
rtp_demuxer.h
Don't check MIDs when demuxing RTP packets in Call
2018-03-29 20:36:08 +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_rtcp_demuxer_helper_unittest.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +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
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +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
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_stream_receiver_controller.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_transport_controller_send_interface.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_transport_controller_send.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_transport_controller_send.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtx_receive_stream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtx_receive_stream.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtx_receive_stream.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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_config.h
Move VideoEncoderConfig from call/ to api/.
2018-05-18 12:58:16 +00:00
video_receive_stream.cc
replace stringstream in call/
2018-04-04 16:09:15 +00:00
video_receive_stream.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_send_stream.cc
Delete unused stats for preferred_bitrate.
2018-06-07 08:11:07 +00:00
video_send_stream.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00