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webrtc_m130/modules/rtp_rtcp/source
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philipel 1a4746a563 Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file. 
Bug: none
Change-Id: If28f57c5ae250afbb47c5d20c9854e9a11182642
Reviewed-on: https://webrtc-review.googlesource.com/87561
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23904}
2018-07-10 11:57:46 +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
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dtmf_queue.h
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fec_private_tables_bursty_unittest.cc
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fec_private_tables_bursty.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fec_private_tables_bursty.h
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fec_private_tables_random.cc
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fec_private_tables_random.h
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fec_test_helper.cc
Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file.
2018-07-10 11:57:46 +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
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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
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flexfec_sender_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
flexfec_sender.cc
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forward_error_correction_internal.cc
Fix a downstream test failure.
2018-06-15 13:30:26 +00:00
forward_error_correction_internal.h
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forward_error_correction.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
forward_error_correction.h
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2018-06-19 14:00:39 +00:00
nack_rtx_unittest.cc
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2018-06-19 14:00:39 +00:00
packet_loss_stats_unittest.cc
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packet_loss_stats.cc
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2018-06-19 14:00:39 +00:00
packet_loss_stats.h
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playout_delay_oracle_unittest.cc
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playout_delay_oracle.cc
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playout_delay_oracle.h
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receive_statistics_impl.cc
Remove StreamStatistician::IsPacketInOrder
2018-06-28 08:44:40 +00:00
receive_statistics_impl.h
Remove StreamStatistician::IsPacketInOrder
2018-06-28 08:44:40 +00:00
receive_statistics_unittest.cc
Update packetsLost and jitter stats any time a packet is received.
2018-06-25 23:56:39 +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
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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
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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
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rtcp_sender_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcp_sender.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
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rtcp_transceiver_config.h
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rtcp_transceiver_impl_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
rtcp_transceiver_impl.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
rtcp_transceiver.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
rtcp_transceiver.h
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rtp_fec_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_format_h264_unittest.cc
Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
2018-07-05 14:29:07 +00:00
rtp_format_h264.cc
Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
2018-07-05 14:29:07 +00:00
rtp_format_h264.h
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rtp_format_video_generic_unittest.cc
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rtp_format_video_generic.cc
Add ParsedPayload::video_header() accessor.
2018-07-04 09:31:21 +00:00
rtp_format_video_generic.h
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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
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rtp_format_vp8_unittest.cc
Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
2018-07-05 14:29:07 +00:00
rtp_format_vp8.cc
Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
2018-07-05 14:29:07 +00:00
rtp_format_vp8.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_format_vp9_unittest.cc
Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
2018-07-05 14:29:07 +00:00
rtp_format_vp9.cc
Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
2018-07-05 14:29:07 +00:00
rtp_format_vp9.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_format.cc
Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
2018-07-05 14:29:07 +00:00
rtp_format.h
Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
2018-07-05 14:29:07 +00:00
rtp_generic_frame_descriptor_extension_unittest.cc
Add Parsing/Building generic frame descriptor extension
2018-06-19 14:51:27 +00:00
rtp_generic_frame_descriptor_extension.cc
Add Generic frame descritpor header extension
2018-06-29 15:02:44 +00:00
rtp_generic_frame_descriptor_extension.h
Add Generic frame descritpor header extension
2018-06-29 15:02:44 +00:00
rtp_generic_frame_descriptor.cc
Discard frame self-dependency when parsing genric frame descriptor
2018-07-03 10:28:05 +00:00
rtp_generic_frame_descriptor.h
Add Parsing/Building generic frame descriptor extension
2018-06-19 14:51:27 +00:00
rtp_header_extension_map_unittest.cc
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rtp_header_extension_map.cc
Add Generic frame descritpor header extension
2018-06-29 15:02:44 +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
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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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
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rtp_packet_received.h
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rtp_packet_to_send.cc
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rtp_packet_to_send.h
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rtp_packet_unittest.cc
Pass buffer with size when writing rtp header extension
2018-06-18 13:04:33 +00:00
rtp_packet.cc
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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
Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.
2018-06-19 16:44:19 +00:00
rtp_receiver_audio.h
Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.
2018-06-19 16:44:19 +00:00
rtp_receiver_impl.cc
Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file.
2018-07-10 11:57:46 +00:00
rtp_receiver_impl.h
Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.
2018-06-19 16:44:19 +00:00
rtp_receiver_strategy.cc
Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.
2018-06-19 16:44:19 +00:00
rtp_receiver_strategy.h
Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.
2018-06-19 16:44:19 +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
Add ParsedPayload::video_header() accessor.
2018-07-04 09:31:21 +00:00
rtp_receiver_video.h
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rtp_rtcp_config.h
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rtp_rtcp_impl_unittest.cc
Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
2018-07-05 14:29:07 +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
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rtp_sender_unittest.cc
Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file.
2018-07-10 11:57:46 +00:00
rtp_sender_video.cc
Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
2018-07-05 14:29:07 +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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
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rtp_utility.cc
Add Generic frame descritpor header extension
2018-06-29 15:02:44 +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
rtp_video_header.cc
Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file.
2018-07-10 11:57:46 +00:00
rtp_video_header.h
Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file.
2018-07-10 11:57:46 +00:00
time_util_unittest.cc
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time_util.cc
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time_util.h
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tmmbr_help.cc
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tmmbr_help.h
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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
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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
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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
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ulpfec_receiver_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00