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665174fdbb4e0540eccb27cf7412348f1b65534c
platform-external-webrtc/modules/rtp_rtcp/source
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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
..
rtcp_packet
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
byte_io_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
byte_io.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
dtmf_queue.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
dtmf_queue.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
fec_private_tables_bursty_unittest.cc
Fix a couple of nits and update a few comments in forward_error_correction_internal.
2018-04-23 14:29:17 +00:00
fec_private_tables_bursty.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fec_private_tables_bursty.h
Move the FEC private tables into .cc files.
2018-04-04 15:16:10 +00:00
fec_private_tables_random.cc
Remove part of the FEC code table that covers FEC code for group of 13-48 media packets, instead generate interleaved FEC code at run time. FEC code masks for protection of group of 1 - 12 media packets is not changed.
2018-04-18 14:35:17 +00:00
fec_private_tables_random.h
Move the FEC private tables into .cc files.
2018-04-04 15:16:10 +00:00
fec_test_helper.cc
Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets"
2018-03-09 16:04:35 +00:00
fec_test_helper.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
flexfec_header_reader_writer_unittest.cc
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
flexfec_header_reader_writer.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
flexfec_header_reader_writer.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
flexfec_receiver_unittest.cc
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
flexfec_receiver.cc
Avoid lifetime issues with FlexfecReceiver packet buffer.
2017-12-12 10:12:47 +00:00
flexfec_sender_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
flexfec_sender.cc
Add MID sending to FlexfecSender
2018-04-10 16:08:35 +00:00
forward_error_correction_internal.cc
Fix a downstream test failure.
2018-06-15 13:30:26 +00:00
forward_error_correction_internal.h
Fix a couple of nits and update a few comments in forward_error_correction_internal.
2018-04-23 14:29:17 +00:00
forward_error_correction.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
forward_error_correction.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
nack_rtx_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
packet_loss_stats_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
packet_loss_stats.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
packet_loss_stats.h
Fix clang style errors in rtp_rtcp and dependant targets
2018-02-07 09:48:28 +00:00
playout_delay_oracle_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
playout_delay_oracle.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
playout_delay_oracle.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
receive_statistics_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
receive_statistics_impl.h
Drop the RTT as input to IsRetransmitOfOldPacket.
2018-05-23 13:14:40 +00:00
receive_statistics_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
remote_ntp_time_estimator_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
remote_ntp_time_estimator.cc
Move timestamp_extrapolator.h to rtc_base/time/
2018-03-22 14:36:44 +00:00
rtcp_nack_stats_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcp_nack_stats.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcp_nack_stats.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcp_packet_unittest.cc
Change RtcpPacket::PacketReadyCallback to rtc::FunctionView
2017-12-07 11:20:08 +00:00
rtcp_packet.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcp_packet.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
rtcp_receiver_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcp_receiver.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcp_receiver.h
Handle Receiver Reference Time Report from multiple receivers.
2018-04-16 09:54:07 +00:00
rtcp_sender_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcp_sender.cc
Remove "webrtc_rtp" traces.
2018-06-13 14:46:24 +00:00
rtcp_sender.h
Replace rtc::Optional with absl::optional in modules/rtp_rtcp
2018-06-15 09:53:35 +00:00
rtcp_transceiver_config.cc
Calculate RTT using ExtendedReports in RtcpTransceiver
2017-11-30 14:34:40 +00:00
rtcp_transceiver_config.h
Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets
2018-05-17 11:22:56 +00:00
rtcp_transceiver_impl_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcp_transceiver_impl.cc
Replace rtc::Optional with absl::optional in modules/rtp_rtcp
2018-06-15 09:53:35 +00:00
rtcp_transceiver_impl.h
Replace rtc::Optional with absl::optional in modules/rtp_rtcp
2018-06-15 09:53:35 +00:00
rtcp_transceiver_unittest.cc
Propagate media receiver rtcp observers to RtcpTransceiver
2017-12-14 17:39:13 +00:00
rtcp_transceiver.cc
Allow to turn RtcpTransciever on and off at runtime.
2018-03-06 15:19:11 +00:00
rtcp_transceiver.h
Allow to turn RtcpTransciever on and off at runtime.
2018-03-06 15:19:11 +00:00
rtp_fec_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_format_h264_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_format_h264.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_format_h264.h
Don't crash in SingleNalu packetization for h264 if no space in packet
2018-03-15 15:42:57 +00:00
rtp_format_video_generic_unittest.cc
Reland "Add stereo codec header and pass it through RTP"
2017-11-30 01:44:19 +00:00
rtp_format_video_generic.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_format_video_generic.h
Fix clang style errors in rtp_rtcp and dependant targets
2018-02-07 09:48:28 +00:00
rtp_format_vp8_test_helper.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_format_vp8_test_helper.h
Adding NOLINT for typedefs.h and common_types.h
2017-09-15 13:03:51 +00:00
rtp_format_vp8_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_format_vp8.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_format_vp8.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_format_vp9_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_format_vp9.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_format_vp9.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_format.cc
Delete enum RtpVideoCodecTypes, replaced with VideoCodecType.
2018-06-04 11:53:17 +00:00
rtp_format.h
Delete enum RtpVideoCodecTypes, replaced with VideoCodecType.
2018-06-04 11:53:17 +00:00
rtp_generic_frame_descriptor.cc
Introduce list of fields to put into codec agnostic descriptor
2018-06-13 14:55:09 +00:00
rtp_generic_frame_descriptor.h
Introduce list of fields to put into codec agnostic descriptor
2018-06-13 14:55:09 +00:00
rtp_header_extension_map_unittest.cc
Adding NOLINT for typedefs.h and common_types.h
2017-09-15 13:03:51 +00:00
rtp_header_extension_map.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
rtp_header_extensions.cc
Pass buffer with size when writing rtp header extension
2018-06-18 13:04:33 +00:00
rtp_header_extensions.h
Pass buffer with size when writing rtp header extension
2018-06-18 13:04:33 +00:00
rtp_header_parser.cc
Fix clang style errors in rtp_rtcp and dependant targets
2018-02-07 09:48:28 +00:00
rtp_packet_history_unittest.cc
Replace rtc::Optional with absl::optional in modules/rtp_rtcp
2018-06-15 09:53:35 +00:00
rtp_packet_history.cc
Replace rtc::Optional with absl::optional in modules/rtp_rtcp
2018-06-15 09:53:35 +00:00
rtp_packet_history.h
Replace rtc::Optional with absl::optional in modules/rtp_rtcp
2018-06-15 09:53:35 +00:00
rtp_packet_received.cc
Add move constructors and assignment operators to RtpPacketReceived and RtpPacketToSend. Since both are non-POD now, move would fall back to copy without these.
2018-02-26 13:25:50 +00:00
rtp_packet_received.h
Add move constructors and assignment operators to RtpPacketReceived and RtpPacketToSend. Since both are non-POD now, move would fall back to copy without these.
2018-02-26 13:25:50 +00:00
rtp_packet_to_send.cc
Add move constructors and assignment operators to RtpPacketReceived and RtpPacketToSend. Since both are non-POD now, move would fall back to copy without these.
2018-02-26 13:25:50 +00:00
rtp_packet_to_send.h
Add move constructors and assignment operators to RtpPacketReceived and RtpPacketToSend. Since both are non-POD now, move would fall back to copy without these.
2018-02-26 13:25:50 +00:00
rtp_packet_unittest.cc
Pass buffer with size when writing rtp header extension
2018-06-18 13:04:33 +00:00
rtp_packet.cc
Fix infinite loop in rtp packet parsing
2018-02-13 14:42:45 +00:00
rtp_packet.h
Pass buffer with size when writing rtp header extension
2018-06-18 13:04:33 +00:00
rtp_payload_registry_unittest.cc
Delete enum RtpVideoCodecTypes, replaced with VideoCodecType.
2018-06-04 11:53:17 +00:00
rtp_payload_registry.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_receiver_audio.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_receiver_audio.h
Delete OnIncomingCSRCChanged and related code.
2018-03-27 13:18:35 +00:00
rtp_receiver_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_receiver_impl.h
Replace rtc::Optional with absl::optional in modules/rtp_rtcp
2018-06-15 09:53:35 +00:00
rtp_receiver_strategy.cc
Delete dead code in RtpReceiverImpl and RTPPayloadRegistry.
2018-03-26 14:36:20 +00:00
rtp_receiver_strategy.h
Delete OnIncomingCSRCChanged and related code.
2018-03-27 13:18:35 +00:00
rtp_receiver_unittest.cc
Delete RtpFeedback. The ssrc for a receive stream should be known at
2018-05-28 11:05:19 +00:00
rtp_receiver_video.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_receiver_video.h
Delete OnIncomingCSRCChanged and related code.
2018-03-27 13:18:35 +00:00
rtp_rtcp_config.h
Make RTCP report interval configurable
2018-02-01 10:12:11 +00:00
rtp_rtcp_impl_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_rtcp_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_rtcp_impl.h
Replace rtc::Optional with absl::optional in modules/rtp_rtcp
2018-06-15 09:53:35 +00:00
rtp_sender_audio.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_sender_audio.h
Remove use of RTPFragmentationHeader from RTPSenderAudio
2017-10-30 09:56:19 +00:00
rtp_sender_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_sender_video.cc
Replace rtc::Optional with absl::optional in modules/rtp_rtcp
2018-06-15 09:53:35 +00:00
rtp_sender_video.h
Replace rtc::Optional with absl::optional in modules/rtp_rtcp
2018-06-15 09:53:35 +00:00
rtp_sender.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_sender.h
Replace rtc::Optional with absl::optional in modules/rtp_rtcp
2018-06-15 09:53:35 +00:00
rtp_utility_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_utility.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_utility.h
Delete RtpFeedback. The ssrc for a receive stream should be known at
2018-05-28 11:05:19 +00:00
time_util_unittest.cc
Delete assumption TimeMicrosToNtp can match RealTimeClock
2018-02-22 17:20:25 +00:00
time_util.cc
Add TimeMicrosToNtp to calculate current NtpTime without Clock
2017-11-28 10:11:58 +00:00
time_util.h
Delete assumption TimeMicrosToNtp can match RealTimeClock
2018-02-22 17:20:25 +00:00
tmmbr_help.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
tmmbr_help.h
Adding NOLINT for typedefs.h and common_types.h
2017-09-15 13:03:51 +00:00
ulpfec_generator_unittest.cc
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
ulpfec_generator.cc
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
ulpfec_generator.h
Fix clang style errors in rtp_rtcp and dependant targets
2018-02-07 09:48:28 +00:00
ulpfec_header_reader_writer_unittest.cc
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
ulpfec_header_reader_writer.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
ulpfec_header_reader_writer.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
ulpfec_receiver_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
ulpfec_receiver_impl.h
Fix clang style errors in rtp_rtcp and dependant targets
2018-02-07 09:48:28 +00:00
ulpfec_receiver_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00