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webrtc_m130/webrtc/modules/rtp_rtcp/source
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danilchap 54a20696c2 [rtcp] Bye::Parse updated not to use RTCPUtility 
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1849243002

Cr-Commit-Position: refs/heads/master@{#12244}
2016-04-05 15:48:18 +00:00
..
mock
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
rtcp_packet
[rtcp] Bye::Parse updated not to use RTCPUtility
2016-04-05 15:48:18 +00:00
bitrate.cc
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bitrate.h
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byte_io_unittest.cc
Fix ubsan warning in byteio_unittest
2016-02-26 13:13:51 +00:00
byte_io.h
Fix a bug in webrtc::ByteReader
2016-01-25 07:47:59 +00:00
CPPLINT.cfg
[rtp_rtcp] Lint errors cleaned from rtp_utility
2015-12-28 18:18:52 +00:00
dtmf_queue.cc
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dtmf_queue.h
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fec_private_tables_bursty.h
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fec_private_tables_random.h
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fec_receiver_impl.cc
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fec_receiver_impl.h
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
2015-12-15 10:54:50 +00:00
fec_receiver_unittest.cc
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
fec_test_helper.cc
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
fec_test_helper.h
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
forward_error_correction_internal.cc
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forward_error_correction_internal.h
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forward_error_correction.cc
Use pointer to generated FEC packet.
2016-01-07 14:33:57 +00:00
forward_error_correction.h
Replace RefCountImpl with rtc::RefCountedObject.
2016-03-21 15:44:41 +00:00
h264_bitstream_parser_unittest.cc
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h264_bitstream_parser.cc
Split ByteBuffer into writer/reader objects.
2016-03-30 13:43:44 +00:00
h264_bitstream_parser.h
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h264_sps_parser_unittest.cc
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h264_sps_parser.cc
Split ByteBuffer into writer/reader objects.
2016-03-30 13:43:44 +00:00
h264_sps_parser.h
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nack_rtx_unittest.cc
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
2015-12-15 10:54:50 +00:00
OWNERS
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packet_loss_stats_unittest.cc
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packet_loss_stats.cc
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packet_loss_stats.h
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producer_fec_unittest.cc
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
producer_fec.cc
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
producer_fec.h
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
receive_statistics_impl.cc
Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ )
2016-02-25 12:50:09 +00:00
receive_statistics_impl.h
Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ )
2016-02-25 12:50:09 +00:00
receive_statistics_unittest.cc
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remote_ntp_time_estimator_unittest.cc
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remote_ntp_time_estimator.cc
Add histogram stats for AV sync stream offset:
2016-03-15 08:00:54 +00:00
rtcp_format_remb_unittest.cc
Fixes a bug which incorrectly logs incoming RTCP as outgoing.
2016-01-21 13:42:10 +00:00
rtcp_packet_unittest.cc
[rtp_rtcp] rtcp::SenderReport moved into own file and got Parse function
2016-01-18 10:43:38 +00:00
rtcp_packet.cc
[rtp_rtcp] Append functionality moved from base RtcpPacket class to CompoundPacket
2016-03-14 15:19:31 +00:00
rtcp_packet.h
[rtp_rtcp] Append functionality moved from base RtcpPacket class to CompoundPacket
2016-03-14 15:19:31 +00:00
rtcp_receiver_help.cc
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rtcp_receiver_help.h
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rtcp_receiver_unittest.cc
[rtp_rtcp] Append functionality moved from base RtcpPacket class to CompoundPacket
2016-03-14 15:19:31 +00:00
rtcp_receiver.cc
Enabling rtcp-rsize negotiation and fixing some issues with it.
2016-03-18 22:02:13 +00:00
rtcp_receiver.h
Enabling rtcp-rsize negotiation and fixing some issues with it.
2016-03-18 22:02:13 +00:00
rtcp_sender_unittest.cc
RTT intermediate calculation use ntp time instead of milliseconds.
2016-02-22 17:59:47 +00:00
rtcp_sender.cc
Avoid rescheduling the next RTCP packet if the RTCP sender status doesn't change.
2016-04-01 21:46:54 +00:00
rtcp_sender.h
Make rtcp sender use max transfer unit.
2016-03-30 18:11:55 +00:00
rtcp_utility_unittest.cc
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rtcp_utility.cc
Fixed busy loop in case of partially malformed rtcp packet
2016-03-10 21:13:57 +00:00
rtcp_utility.h
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rtp_fec_unittest.cc
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rtp_format_h264_unittest.cc
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rtp_format_h264.cc
Add SEI to the list of packets counted as keyframe.
2016-02-03 11:43:50 +00:00
rtp_format_h264.h
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rtp_format_video_generic.cc
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rtp_format_video_generic.h
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rtp_format_vp8_test_helper.cc
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rtp_format_vp8_test_helper.h
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rtp_format_vp8_unittest.cc
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rtp_format_vp8.cc
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rtp_format_vp8.h
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rtp_format_vp9_unittest.cc
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rtp_format_vp9.cc
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rtp_format_vp9.h
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rtp_format.cc
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rtp_format.h
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rtp_header_extension_unittest.cc
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rtp_header_extension.cc
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rtp_header_extension.h
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
2015-12-15 10:54:50 +00:00
rtp_header_parser.cc
[rtp_rtcp] Lint errors cleaned from rtp_utility
2015-12-28 18:18:52 +00:00
rtp_packet_history_unittest.cc
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rtp_packet_history.cc
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rtp_packet_history.h
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
2015-12-15 10:54:50 +00:00
rtp_payload_registry_unittest.cc
Support multiple rtx codecs.
2016-02-03 12:30:10 +00:00
rtp_payload_registry.cc
Remove video-codec max bitrate from TMMBN.
2016-02-16 16:59:36 +00:00
rtp_receiver_audio.cc
Reland https://codereview.webrtc.org/1802993002/
2016-03-30 09:42:37 +00:00
rtp_receiver_audio.h
Reland https://codereview.webrtc.org/1802993002/
2016-03-30 09:42:37 +00:00
rtp_receiver_impl.cc
Remove deprecated RtpReceiver::CreateAudioReceiver() function.
2016-04-01 08:01:33 +00:00
rtp_receiver_impl.h
Reland https://codereview.webrtc.org/1802993002/
2016-03-30 09:42:37 +00:00
rtp_receiver_strategy.cc
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rtp_receiver_strategy.h
Reland https://codereview.webrtc.org/1802993002/
2016-03-30 09:42:37 +00:00
rtp_receiver_video.cc
Deprecate VideoDecoder::Reset() and remove calls.
2016-02-02 14:40:13 +00:00
rtp_receiver_video.h
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rtp_rtcp_config.h
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rtp_rtcp_impl_unittest.cc
Remove webrtc::ScopedVector
2016-03-31 17:24:31 +00:00
rtp_rtcp_impl.cc
Make rtcp sender use max transfer unit.
2016-03-30 18:11:55 +00:00
rtp_rtcp_impl.h
Experiment for the nack module.
2016-03-12 11:30:31 +00:00
rtp_sender_audio.cc
Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
2016-03-15 18:41:58 +00:00
rtp_sender_audio.h
Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
2016-03-15 18:41:58 +00:00
rtp_sender_unittest.cc
Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
2016-03-15 18:41:58 +00:00
rtp_sender_video.cc
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
rtp_sender_video.h
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
rtp_sender.cc
Make rtcp sender use max transfer unit.
2016-03-30 18:11:55 +00:00
rtp_sender.h
Make rtcp sender use max transfer unit.
2016-03-30 18:11:55 +00:00
rtp_utility.cc
Reland https://codereview.webrtc.org/1802993002/
2016-03-30 09:42:37 +00:00
rtp_utility.h
Reland https://codereview.webrtc.org/1802993002/
2016-03-30 09:42:37 +00:00
ssrc_database.cc
Refactor RtpSender and SSRCDatabase.
2016-02-02 16:34:16 +00:00
ssrc_database.h
Refactor RtpSender and SSRCDatabase.
2016-02-02 16:34:16 +00:00
time_util_unittest.cc
rtt calculation handles time go backwards
2016-03-09 14:14:45 +00:00
time_util.cc
rtt calculation handles time go backwards
2016-03-09 14:14:45 +00:00
time_util.h
rtt calculation handles time go backwards
2016-03-09 14:14:45 +00:00
tmmbr_help.cc
TMMBRSet become vector<rtcp::TmmbItem>
2016-02-29 13:26:05 +00:00
tmmbr_help.h
TMMBRSet become vector<rtcp::TmmbItem>
2016-02-29 13:26:05 +00:00
video_codec_information.h
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vp8_partition_aggregator_unittest.cc
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vp8_partition_aggregator.cc
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vp8_partition_aggregator.h
…