.. |
rtcp_packet
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
byte_io.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
byte_io_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
dtmf_queue.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
dtmf_queue.h
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
fec_private_tables_bursty.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
fec_private_tables_bursty.h
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Move the FEC private tables into .cc files.
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2018-04-04 15:16:10 +00:00 |
fec_private_tables_bursty_unittest.cc
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Fix a couple of nits and update a few comments in forward_error_correction_internal.
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2018-04-23 14:29:17 +00:00 |
fec_private_tables_random.cc
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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.
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2018-04-18 14:35:17 +00:00 |
fec_private_tables_random.h
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Move the FEC private tables into .cc files.
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2018-04-04 15:16:10 +00:00 |
fec_test_helper.cc
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Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets"
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2018-03-09 16:04:35 +00:00 |
fec_test_helper.h
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Delete unneeded includes of basictypes.h.
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2018-05-21 19:35:08 +00:00 |
flexfec_header_reader_writer.cc
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Stop using LOG macros in favor of RTC_ prefixed macros.
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2017-11-09 11:56:32 +00:00 |
flexfec_header_reader_writer.h
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Delete unneeded includes of basictypes.h.
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2018-05-21 19:35:08 +00:00 |
flexfec_header_reader_writer_unittest.cc
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Delete unneeded includes of basictypes.h.
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2018-05-21 19:35:08 +00:00 |
flexfec_receiver.cc
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Avoid lifetime issues with FlexfecReceiver packet buffer.
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2017-12-12 10:12:47 +00:00 |
flexfec_receiver_unittest.cc
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Delete unneeded includes of basictypes.h.
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2018-05-21 19:35:08 +00:00 |
flexfec_sender.cc
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Add MID sending to FlexfecSender
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2018-04-10 16:08:35 +00:00 |
flexfec_sender_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
forward_error_correction.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
forward_error_correction.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
forward_error_correction_internal.cc
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Fix a downstream test failure.
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2018-06-15 13:30:26 +00:00 |
forward_error_correction_internal.h
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Fix a couple of nits and update a few comments in forward_error_correction_internal.
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2018-04-23 14:29:17 +00:00 |
nack_rtx_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
packet_loss_stats.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
packet_loss_stats.h
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Fix clang style errors in rtp_rtcp and dependant targets
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2018-02-07 09:48:28 +00:00 |
packet_loss_stats_unittest.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
playout_delay_oracle.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
playout_delay_oracle.h
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
playout_delay_oracle_unittest.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
receive_statistics_impl.cc
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Remove StreamStatistician::IsPacketInOrder
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2018-06-28 08:44:40 +00:00 |
receive_statistics_impl.h
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Remove StreamStatistician::IsPacketInOrder
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2018-06-28 08:44:40 +00:00 |
receive_statistics_unittest.cc
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Update packetsLost and jitter stats any time a packet is received.
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2018-06-25 23:56:39 +00:00 |
remote_ntp_time_estimator.cc
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Move timestamp_extrapolator.h to rtc_base/time/
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2018-03-22 14:36:44 +00:00 |
remote_ntp_time_estimator_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtcp_nack_stats.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtcp_nack_stats.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtcp_nack_stats_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtcp_packet.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtcp_packet.h
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Delete unneeded includes of basictypes.h.
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2018-05-21 19:35:08 +00:00 |
rtcp_packet_unittest.cc
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Change RtcpPacket::PacketReadyCallback to rtc::FunctionView
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2017-12-07 11:20:08 +00:00 |
rtcp_receiver.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtcp_receiver.h
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Handle Receiver Reference Time Report from multiple receivers.
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2018-04-16 09:54:07 +00:00 |
rtcp_receiver_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtcp_sender.cc
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Remove "webrtc_rtp" traces.
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2018-06-13 14:46:24 +00:00 |
rtcp_sender.h
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Replace rtc::Optional with absl::optional in modules/rtp_rtcp
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2018-06-15 09:53:35 +00:00 |
rtcp_sender_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtcp_transceiver.cc
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Allow to turn RtcpTransciever on and off at runtime.
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2018-03-06 15:19:11 +00:00 |
rtcp_transceiver.h
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Allow to turn RtcpTransciever on and off at runtime.
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2018-03-06 15:19:11 +00:00 |
rtcp_transceiver_config.cc
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Calculate RTT using ExtendedReports in RtcpTransceiver
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2017-11-30 14:34:40 +00:00 |
rtcp_transceiver_config.h
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Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets
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2018-05-17 11:22:56 +00:00 |
rtcp_transceiver_impl.cc
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Replace rtc::Optional with absl::optional in modules/rtp_rtcp
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2018-06-15 09:53:35 +00:00 |
rtcp_transceiver_impl.h
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Replace rtc::Optional with absl::optional in modules/rtp_rtcp
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2018-06-15 09:53:35 +00:00 |
rtcp_transceiver_impl_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtcp_transceiver_unittest.cc
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Propagate media receiver rtcp observers to RtcpTransceiver
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2017-12-14 17:39:13 +00:00 |
rtp_fec_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_format.cc
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Delete enum RtpVideoCodecTypes, replaced with VideoCodecType.
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2018-06-04 11:53:17 +00:00 |
rtp_format.h
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Delete enum RtpVideoCodecTypes, replaced with VideoCodecType.
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2018-06-04 11:53:17 +00:00 |
rtp_format_h264.cc
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Generalize SimulcastEncoderAdapter, use for H264 & VP8.
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2018-06-21 15:57:43 +00:00 |
rtp_format_h264.h
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Don't crash in SingleNalu packetization for h264 if no space in packet
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2018-03-15 15:42:57 +00:00 |
rtp_format_h264_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_format_video_generic.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_format_video_generic.h
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Fix clang style errors in rtp_rtcp and dependant targets
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2018-02-07 09:48:28 +00:00 |
rtp_format_video_generic_unittest.cc
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Reland "Add stereo codec header and pass it through RTP"
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2017-11-30 01:44:19 +00:00 |
rtp_format_vp8.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_format_vp8.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_format_vp8_test_helper.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_format_vp8_test_helper.h
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Adding NOLINT for typedefs.h and common_types.h
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2017-09-15 13:03:51 +00:00 |
rtp_format_vp8_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_format_vp9.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_format_vp9.h
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_format_vp9_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_generic_frame_descriptor.cc
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Discard frame self-dependency when parsing genric frame descriptor
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2018-07-03 10:28:05 +00:00 |
rtp_generic_frame_descriptor.h
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Add Parsing/Building generic frame descriptor extension
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2018-06-19 14:51:27 +00:00 |
rtp_generic_frame_descriptor_extension.cc
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Add Generic frame descritpor header extension
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2018-06-29 15:02:44 +00:00 |
rtp_generic_frame_descriptor_extension.h
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Add Generic frame descritpor header extension
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2018-06-29 15:02:44 +00:00 |
rtp_generic_frame_descriptor_extension_unittest.cc
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Add Parsing/Building generic frame descriptor extension
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2018-06-19 14:51:27 +00:00 |
rtp_header_extension_map.cc
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Add Generic frame descritpor header extension
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2018-06-29 15:02:44 +00:00 |
rtp_header_extension_map_unittest.cc
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Adding NOLINT for typedefs.h and common_types.h
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2017-09-15 13:03:51 +00:00 |
rtp_header_extensions.cc
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Pass buffer with size when writing rtp header extension
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2018-06-18 13:04:33 +00:00 |
rtp_header_extensions.h
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Pass buffer with size when writing rtp header extension
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2018-06-18 13:04:33 +00:00 |
rtp_header_parser.cc
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Fix clang style errors in rtp_rtcp and dependant targets
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2018-02-07 09:48:28 +00:00 |
rtp_packet.cc
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Fix infinite loop in rtp packet parsing
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2018-02-13 14:42:45 +00:00 |
rtp_packet.h
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Pass buffer with size when writing rtp header extension
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2018-06-18 13:04:33 +00:00 |
rtp_packet_history.cc
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Replace rtc::Optional with absl::optional in modules/rtp_rtcp
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2018-06-15 09:53:35 +00:00 |
rtp_packet_history.h
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Replace rtc::Optional with absl::optional in modules/rtp_rtcp
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2018-06-15 09:53:35 +00:00 |
rtp_packet_history_unittest.cc
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Replace rtc::Optional with absl::optional in modules/rtp_rtcp
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2018-06-15 09:53:35 +00:00 |
rtp_packet_received.cc
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Add move constructors and assignment operators to RtpPacketReceived and RtpPacketToSend. Since both are non-POD now, move would fall back to copy without these.
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2018-02-26 13:25:50 +00:00 |
rtp_packet_received.h
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Add move constructors and assignment operators to RtpPacketReceived and RtpPacketToSend. Since both are non-POD now, move would fall back to copy without these.
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2018-02-26 13:25:50 +00:00 |
rtp_packet_to_send.cc
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Add move constructors and assignment operators to RtpPacketReceived and RtpPacketToSend. Since both are non-POD now, move would fall back to copy without these.
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2018-02-26 13:25:50 +00:00 |
rtp_packet_to_send.h
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Add move constructors and assignment operators to RtpPacketReceived and RtpPacketToSend. Since both are non-POD now, move would fall back to copy without these.
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2018-02-26 13:25:50 +00:00 |
rtp_packet_unittest.cc
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Pass buffer with size when writing rtp header extension
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2018-06-18 13:04:33 +00:00 |
rtp_payload_registry.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_payload_registry_unittest.cc
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Delete enum RtpVideoCodecTypes, replaced with VideoCodecType.
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2018-06-04 11:53:17 +00:00 |
rtp_receiver_audio.cc
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Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.
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2018-06-19 16:44:19 +00:00 |
rtp_receiver_audio.h
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Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.
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2018-06-19 16:44:19 +00:00 |
rtp_receiver_impl.cc
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Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.
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2018-06-19 16:44:19 +00:00 |
rtp_receiver_impl.h
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Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.
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2018-06-19 16:44:19 +00:00 |
rtp_receiver_strategy.cc
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Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.
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2018-06-19 16:44:19 +00:00 |
rtp_receiver_strategy.h
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Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.
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2018-06-19 16:44:19 +00:00 |
rtp_receiver_unittest.cc
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Delete RtpFeedback. The ssrc for a receive stream should be known at
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2018-05-28 11:05:19 +00:00 |
rtp_receiver_video.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_receiver_video.h
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Delete OnIncomingCSRCChanged and related code.
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2018-03-27 13:18:35 +00:00 |
rtp_rtcp_config.h
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Make RTCP report interval configurable
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2018-02-01 10:12:11 +00:00 |
rtp_rtcp_impl.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_rtcp_impl.h
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Replace rtc::Optional with absl::optional in modules/rtp_rtcp
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2018-06-15 09:53:35 +00:00 |
rtp_rtcp_impl_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_sender.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_sender.h
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Replace rtc::Optional with absl::optional in modules/rtp_rtcp
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2018-06-15 09:53:35 +00:00 |
rtp_sender_audio.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_sender_audio.h
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Remove use of RTPFragmentationHeader from RTPSenderAudio
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2017-10-30 09:56:19 +00:00 |
rtp_sender_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
rtp_sender_video.cc
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Replace rtc::Optional with absl::optional in modules/rtp_rtcp
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2018-06-15 09:53:35 +00:00 |
rtp_sender_video.h
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Replace rtc::Optional with absl::optional in modules/rtp_rtcp
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2018-06-15 09:53:35 +00:00 |
rtp_utility.cc
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Add Generic frame descritpor header extension
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2018-06-29 15:02:44 +00:00 |
rtp_utility.h
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Delete RtpFeedback. The ssrc for a receive stream should be known at
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2018-05-28 11:05:19 +00:00 |
rtp_utility_unittest.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
time_util.cc
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Add TimeMicrosToNtp to calculate current NtpTime without Clock
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2017-11-28 10:11:58 +00:00 |
time_util.h
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Delete assumption TimeMicrosToNtp can match RealTimeClock
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2018-02-22 17:20:25 +00:00 |
time_util_unittest.cc
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Delete assumption TimeMicrosToNtp can match RealTimeClock
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2018-02-22 17:20:25 +00:00 |
tmmbr_help.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
tmmbr_help.h
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Adding NOLINT for typedefs.h and common_types.h
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2017-09-15 13:03:51 +00:00 |
ulpfec_generator.cc
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Delete unneeded includes of basictypes.h.
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2018-05-21 19:35:08 +00:00 |
ulpfec_generator.h
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Fix clang style errors in rtp_rtcp and dependant targets
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2018-02-07 09:48:28 +00:00 |
ulpfec_generator_unittest.cc
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Delete unneeded includes of basictypes.h.
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2018-05-21 19:35:08 +00:00 |
ulpfec_header_reader_writer.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
ulpfec_header_reader_writer.h
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Delete unneeded includes of basictypes.h.
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2018-05-21 19:35:08 +00:00 |
ulpfec_header_reader_writer_unittest.cc
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Delete unneeded includes of basictypes.h.
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2018-05-21 19:35:08 +00:00 |
ulpfec_receiver_impl.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
ulpfec_receiver_impl.h
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Fix clang style errors in rtp_rtcp and dependant targets
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2018-02-07 09:48:28 +00:00 |
ulpfec_receiver_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |