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webrtc_m130/modules/rtp_rtcp/source
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Per Kjellander f81170b48f Add error logs to RtpPacketHistory::GetBestFittingPacket when no packet is found. 
Make sure nullptr is returned if the packet is not in history.

Bug: webrtc:9863
Change-Id: I9658b1b271071a4bd38f062ed68c60cc04c63123
Reviewed-on: https://webrtc-review.googlesource.com/c/105300
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25114}
2018-10-11 12:33:07 +00:00
..
rtcp_packet
Delete support for VoIP metrics (RFC 3611 4.7)
2018-10-05 10:07:57 +00:00
byte_io_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
byte_io.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
contributing_sources_unittest.cc
Refactor RtpReceiverImpl, extracting CSRC book-keeping to its own class
2018-08-13 11:59:08 +00:00
contributing_sources.cc
Refactor voe::Channel to not use RtpReceiver.
2018-08-16 10:18:20 +00:00
contributing_sources.h
Reland "[cleanup] Remove useless includes."
2018-10-08 07:44:19 +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
Delete root header file typedef.h.
2018-07-25 14:59:26 +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
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
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
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
Move RtpHeaderExtensionMap::GetTotalLengthInBytes into own file
2018-10-03 17:25:31 +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
Revert "Fix buffer overflow in ulpfec recovery"
2018-07-16 12:31:57 +00:00
forward_error_correction.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
nack_rtx_unittest.cc
Add extended header containing frame ID to the generic packetizer.
2018-08-09 14:05:39 +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
Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1
2018-09-28 12:00:28 +00:00
receive_statistics_impl.h
Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1
2018-09-28 12:00:28 +00:00
receive_statistics_unittest.cc
Use zero octets for rtp packet padding
2018-10-11 10:22:36 +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
Delete support for VoIP metrics (RFC 3611 4.7)
2018-10-05 10:07:57 +00:00
rtcp_receiver.cc
Exposing video bitrate allocator into API
2018-07-23 21:23:21 +00:00
rtcp_receiver.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
rtcp_sender_unittest.cc
Turning off a stream should results in target bitrate 0 signal
2018-10-05 12:00:26 +00:00
rtcp_sender.cc
Turning off a stream should results in target bitrate 0 signal
2018-10-05 12:00:26 +00:00
rtcp_sender.h
Turning off a stream should results in target bitrate 0 signal
2018-10-05 12:00:26 +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
Replace WeakPtr with CancelablePeriodicTask in RtcpTransceiverImpl
2018-09-05 11:06:16 +00:00
rtcp_transceiver_impl.cc
Replace WeakPtr with CancelablePeriodicTask in RtcpTransceiverImpl
2018-09-05 11:06:16 +00:00
rtcp_transceiver_impl.h
Move RtcpTransceiver deletion of copy and assign methods to public section
2018-09-10 12:23:19 +00:00
rtcp_transceiver_unittest.cc
Make RtcpTransceiver destructor non-blocking
2018-09-07 15:34:08 +00:00
rtcp_transceiver.cc
Make RtcpTransceiver destructor non-blocking
2018-09-07 15:34:08 +00:00
rtcp_transceiver.h
Move RtcpTransceiver deletion of copy and assign methods to public section
2018-09-10 12:23:19 +00:00
rtp_fec_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_format_h264_unittest.cc
Support first packet reduction in H264 packetizer
2018-09-25 11:43:29 +00:00
rtp_format_h264.cc
Support first packet reduction in H264 packetizer
2018-09-25 11:43:29 +00:00
rtp_format_h264.h
Support first packet reduction in H264 packetizer
2018-09-25 11:43:29 +00:00
rtp_format_unittest.cc
Use signed integers for limiting packet size in video packetizers
2018-09-07 09:24:18 +00:00
rtp_format_video_generic_unittest.cc
Use signed integers for limiting packet size in video packetizers
2018-09-07 09:24:18 +00:00
rtp_format_video_generic.cc
Make RtpGenericFrameDescriptor available for E2EE.
2018-10-02 13:35:29 +00:00
rtp_format_video_generic.h
Use signed integers for limiting packet size in video packetizers
2018-09-07 09:24:18 +00:00
rtp_format_vp8_test_helper.cc
Cleanup RtpPacketizerVP8 tests
2018-08-31 08:13:45 +00:00
rtp_format_vp8_test_helper.h
Cleanup RtpPacketizerVP8 tests
2018-08-31 08:13:45 +00:00
rtp_format_vp8_unittest.cc
Reland "Remove RTPVideoHeader::vp8() accessors."
2018-09-07 13:04:07 +00:00
rtp_format_vp8.cc
Reland "Remove RTPVideoHeader::vp8() accessors."
2018-09-07 13:04:07 +00:00
rtp_format_vp8.h
Use signed integers for limiting packet size in video packetizers
2018-09-07 09:24:18 +00:00
rtp_format_vp9_unittest.cc
Cleanup RtpPacketizerVp9
2018-09-07 22:01:46 +00:00
rtp_format_vp9.cc
Cleanup RtpPacketizerVp9
2018-09-07 22:01:46 +00:00
rtp_format_vp9.h
Cleanup RtpPacketizerVp9
2018-09-07 22:01:46 +00:00
rtp_format.cc
Cleanup RtpPacketizerH264 constructor
2018-09-12 11:11:18 +00:00
rtp_format.h
Use signed integers for limiting packet size in video packetizers
2018-09-07 09:24:18 +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
Reland "Added field trial WebRTC-GenericDescriptor for the new generic descriptor."
2018-09-26 10:26:43 +00:00
rtp_generic_frame_descriptor.cc
Make RtpGenericFrameDescriptor available for E2EE.
2018-10-02 13:35:29 +00:00
rtp_generic_frame_descriptor.h
Make RtpGenericFrameDescriptor available for E2EE.
2018-10-02 13:35:29 +00:00
rtp_header_extension_map_unittest.cc
Move RtpHeaderExtensionMap::GetTotalLengthInBytes into own file
2018-10-03 17:25:31 +00:00
rtp_header_extension_map.cc
Add support for sending RTP two-byte header extensions.
2018-10-05 08:45:52 +00:00
rtp_header_extension_size_unittest.cc
Move RtpHeaderExtensionMap::GetTotalLengthInBytes into own file
2018-10-03 17:25:31 +00:00
rtp_header_extension_size.cc
Add support for sending RTP two-byte header extensions.
2018-10-05 08:45:52 +00:00
rtp_header_extension_size.h
Move RtpHeaderExtensionMap::GetTotalLengthInBytes into own file
2018-10-03 17:25:31 +00:00
rtp_header_extensions.cc
Add support for sending RTP two-byte header extensions.
2018-10-05 08:45:52 +00:00
rtp_header_extensions.h
Frame marking RTP header extension (PART 1: implement extension)
2018-09-11 22:35:30 +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
Reland "Decrease complexity of RtpPacketHistory::GetBestFittingPacket.""
2018-09-11 13:54:30 +00:00
rtp_packet_history.cc
Add error logs to RtpPacketHistory::GetBestFittingPacket when no packet is found.
2018-10-11 12:33:07 +00:00
rtp_packet_history.h
Reland "Decrease complexity of RtpPacketHistory::GetBestFittingPacket.""
2018-09-11 13:54:30 +00:00
rtp_packet_received.cc
Frame marking RTP header extension (PART 1: implement extension)
2018-09-11 22:35:30 +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
Use zero octets for rtp packet padding
2018-10-11 10:22:36 +00:00
rtp_packet.cc
Use zero octets for rtp packet padding
2018-10-11 10:22:36 +00:00
rtp_packet.h
Use zero octets for rtp packet padding
2018-10-11 10:22:36 +00:00
rtp_rtcp_config.h
Delete unused constants from rtp_rtcp_config.h
2018-08-09 08:38:51 +00:00
rtp_rtcp_impl_unittest.cc
Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1
2018-09-28 12:00:28 +00:00
rtp_rtcp_impl.cc
Makes RtpSender indicate allocation and feedback status on packets.
2018-10-10 08:28:34 +00:00
rtp_rtcp_impl.h
Makes RtpSender indicate allocation and feedback status on packets.
2018-10-10 08:28:34 +00:00
rtp_sender_audio.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_sender_audio.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
rtp_sender_unittest.cc
Makes RtpSender indicate allocation and feedback status on packets.
2018-10-10 08:28:34 +00:00
rtp_sender_video.cc
Use minimalistic vp8 descriptor when generic descriptor is in use
2018-09-25 18:05:52 +00:00
rtp_sender_video.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
rtp_sender.cc
Use zero octets for rtp packet padding
2018-10-11 10:22:36 +00:00
rtp_sender.h
Remove unlimited retransmission for screenshare experiment code
2018-10-11 07:53:47 +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
Avoid wrong parsing of padding length and its use in NetEq simulation.
2018-09-12 11:23:03 +00:00
rtp_utility.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
rtp_video_header.cc
Convert VP8 descriptor to generic descriptor.
2018-08-27 13:03:20 +00:00
rtp_video_header.h
Added is_last_packet_in_frame to match is_first_packet_in_frame.
2018-09-13 11:07:10 +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
Delete root header file typedef.h.
2018-07-25 14:59:26 +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
Delete RtpReceiver and related code.
2018-10-04 08:46:16 +00:00
ulpfec_receiver_impl.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
ulpfec_receiver_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00