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webrtc_m130/webrtc/modules/rtp_rtcp/source
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sprang b32aaf97bd Reland of Verify sender ssrc when receiving rtcp target bitrate (patchset #1 id:1 of https://codereview.webrtc.org/3005633002/ ) 
Reason for revert:
Landed fix in upstream project.

Original issue's description:
> Revert of Verify sender ssrc when receiving rtcp target bitrate (patchset #3 id:40001 of https://codereview.webrtc.org/3000373002/ )
>
> Reason for revert:
> Might be the root cause of an internal test failure.
>
> Original issue's description:
> > Verify sender ssrc when receiving rtcp target bitrate
> >
> > BUG=webrtc:8137
> >
> > Review-Url: https://codereview.webrtc.org/3000373002
> > Cr-Commit-Position: refs/heads/master@{#19524}
> > Committed: a7a4beb419
>
> TBR=danilchap@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:8137
>
> Review-Url: https://codereview.webrtc.org/3005633002
> Cr-Commit-Position: refs/heads/master@{#19529}
> Committed: 95a64ca8aa

TBR=danilchap@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8137

Review-Url: https://codereview.webrtc.org/3006683002
Cr-Commit-Position: refs/heads/master@{#19557}
2017-08-28 12:49:12 +00:00
..
rtcp_packet
When using clang, switch on -Wc++11-narrowing
2017-08-16 11:52:35 +00:00
byte_io_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
byte_io.h
Fix a bug in webrtc::ByteReader
2016-01-25 07:47:59 +00:00
dtmf_queue.cc
Support multiple timestamp rates for sending DTMF.
2016-11-17 13:25:45 +00:00
dtmf_queue.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
fec_private_tables_bursty.h
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
fec_private_tables_random.h
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
fec_test_helper.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
fec_test_helper.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
flexfec_header_reader_writer_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
flexfec_header_reader_writer.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
flexfec_header_reader_writer.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
flexfec_receiver_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
flexfec_receiver.cc
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
flexfec_sender_unittest.cc
Persist RTP state for FlexFEC.
2017-05-30 09:32:12 +00:00
flexfec_sender.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
forward_error_correction_internal.cc
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
forward_error_correction_internal.h
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
forward_error_correction.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
forward_error_correction.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
nack_rtx_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
packet_loss_stats_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
packet_loss_stats.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
packet_loss_stats.h
Add statistics gathering for packet loss.
2015-07-11 01:10:08 +00:00
playout_delay_oracle_unittest.cc
Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock
2017-08-09 13:13:45 +00:00
playout_delay_oracle.cc
Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock
2017-08-09 13:13:45 +00:00
playout_delay_oracle.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
receive_statistics_impl.cc
Reduce locking when collecting receive statistic
2017-08-14 12:51:02 +00:00
receive_statistics_impl.h
Reduce locking when collecting receive statistic
2017-08-14 12:51:02 +00:00
receive_statistics_unittest.cc
Renamed fields in common_types.h/RtcpStatistics.
2017-08-04 12:03:53 +00:00
remote_ntp_time_estimator_unittest.cc
Replace Clock::CurrentNtp with Clock::CurrentNtpTime
2017-03-07 10:51:09 +00:00
remote_ntp_time_estimator.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcp_nack_stats_unittest.cc
Remove rtcp_utility as mostly unused.
2017-02-09 13:21:42 +00:00
rtcp_nack_stats.cc
Remove rtcp_utility as mostly unused.
2017-02-09 13:21:42 +00:00
rtcp_nack_stats.h
Remove rtcp_utility as mostly unused.
2017-02-09 13:21:42 +00:00
rtcp_packet_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
rtcp_packet.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcp_packet.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcp_receiver_unittest.cc
Reland of Verify sender ssrc when receiving rtcp target bitrate (patchset #1 id:1 of https://codereview.webrtc.org/3005633002/ )
2017-08-28 12:49:12 +00:00
rtcp_receiver.cc
Reland of Verify sender ssrc when receiving rtcp target bitrate (patchset #1 id:1 of https://codereview.webrtc.org/3005633002/ )
2017-08-28 12:49:12 +00:00
rtcp_receiver.h
Reland of Verify sender ssrc when receiving rtcp target bitrate (patchset #1 id:1 of https://codereview.webrtc.org/3005633002/ )
2017-08-28 12:49:12 +00:00
rtcp_sender_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcp_sender.cc
Take smaller interface for RtpRtcp::Configuration::receive_statistics
2017-07-27 11:44:18 +00:00
rtcp_sender.h
Take smaller interface for RtpRtcp::Configuration::receive_statistics
2017-07-27 11:44:18 +00:00
rtp_fec_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_format_h264_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_format_h264.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_format_h264.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_format_video_generic_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_format_video_generic.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_format_video_generic.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_format_vp8_test_helper.cc
Modify VP8 RTP to always use 2 bytes for picture Id
2017-08-09 11:17:48 +00:00
rtp_format_vp8_test_helper.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_format_vp8_unittest.cc
Modify VP8 RTP to always use 2 bytes for picture Id
2017-08-09 11:17:48 +00:00
rtp_format_vp8.cc
Modify VP8 RTP to always use 2 bytes for picture Id
2017-08-09 11:17:48 +00:00
rtp_format_vp8.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_format_vp9_unittest.cc
Fix packetization logic to leave space for extensions in the last packet
2017-05-23 16:34:21 +00:00
rtp_format_vp9.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_format_vp9.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_format.cc
Fix packetization logic to leave space for extensions in the last packet
2017-05-23 16:34:21 +00:00
rtp_format.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_header_extension_map_unittest.cc
Rename rtp_header_extension.h to rtp_header_extension_map.h
2017-06-02 09:11:27 +00:00
rtp_header_extension_map.cc
Add parsing/serializing for MID RTP header extension.
2017-07-21 17:33:25 +00:00
rtp_header_extensions.cc
Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )
2017-08-18 09:51:12 +00:00
rtp_header_extensions.h
Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )
2017-08-18 09:51:12 +00:00
rtp_header_parser.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_packet_history_unittest.cc
Fix cpplint errors in locations that are already being checked
2017-03-24 10:22:49 +00:00
rtp_packet_history.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_packet_history.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_packet_received.h
Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket.
2017-05-11 15:00:58 +00:00
rtp_packet_to_send.h
Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )
2017-08-18 09:51:12 +00:00
rtp_packet_unittest.cc
Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )
2017-08-18 09:51:12 +00:00
rtp_packet.cc
Add parsing/serializing for MID RTP header extension.
2017-07-21 17:33:25 +00:00
rtp_packet.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_payload_registry_unittest.cc
RTPPayloadRegistry: Stop using the rate to keep track of receive codecs
2016-12-06 11:52:26 +00:00
rtp_payload_registry.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_receiver_audio.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_receiver_audio.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_receiver_impl.cc
Add audio_level member to RtpSource and set it from RtpReceiverImpl::IncomingRtpPacket.
2017-08-24 21:52:17 +00:00
rtp_receiver_impl.h
Add audio_level member to RtpSource and set it from RtpReceiverImpl::IncomingRtpPacket.
2017-08-24 21:52:17 +00:00
rtp_receiver_strategy.cc
Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
2016-04-14 10:05:37 +00:00
rtp_receiver_strategy.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_receiver_unittest.cc
Add audio_level member to RtpSource and set it from RtpReceiverImpl::IncomingRtpPacket.
2017-08-24 21:52:17 +00:00
rtp_receiver_video.cc
Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )
2017-08-18 09:51:12 +00:00
rtp_receiver_video.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_rtcp_config.h
Delete support for receiving RTCP RPSI and SLI message.
2017-03-22 14:15:09 +00:00
rtp_rtcp_impl_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_rtcp_impl.cc
Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock
2017-08-09 13:13:45 +00:00
rtp_rtcp_impl.h
Remove deprecated RtpRtcp::SetAudioPacketSize
2017-07-25 14:46:54 +00:00
rtp_sender_audio.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_sender_audio.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_sender_unittest.cc
Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )
2017-08-18 09:51:12 +00:00
rtp_sender_video.cc
Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )
2017-08-18 09:51:12 +00:00
rtp_sender_video.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_sender.cc
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
rtp_sender.h
Remove deprecated RtpRtcp::SetAudioPacketSize
2017-07-25 14:46:54 +00:00
rtp_utility_unittest.cc
Rename rtp_header_extension.h to rtp_header_extension_map.h
2017-06-02 09:11:27 +00:00
rtp_utility.cc
Remove NullObjectReceiveStatistics() in rtp_rtcp module
2017-07-25 11:20:12 +00:00
rtp_utility.h
Remove NullObjectReceiveStatistics() in rtp_rtcp module
2017-07-25 11:20:12 +00:00
time_util_unittest.cc
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
time_util.cc
rtt calculation handles time go backwards
2016-03-09 14:14:45 +00:00
time_util.h
replace NtpTime->Clock with Clock->NtpTime dependency
2017-02-09 19:15:25 +00:00
tmmbr_help.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
tmmbr_help.h
Remove TMMBRSet class
2016-08-18 13:17:48 +00:00
ulpfec_generator_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
ulpfec_generator.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
ulpfec_generator.h
Fix forward declaration typo in UlpfecGenerator.
2016-11-08 08:29:06 +00:00
ulpfec_header_reader_writer_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
ulpfec_header_reader_writer.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
ulpfec_header_reader_writer.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
ulpfec_receiver_impl.cc
When Ulpfec recovers a packet, set |returned| flag earlier.
2017-08-25 16:08:44 +00:00
ulpfec_receiver_impl.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
ulpfec_receiver_unittest.cc
Reland of Only compare sequence numbers from the same SSRC in ForwardErrorCorrection. (patchset #1 id:1 of https://codereview.webrtc.org/2919313005/ )
2017-06-29 09:45:35 +00:00
video_codec_information.h
lint whitespace warning removed from most rtp_rtcp/source/ files
2015-12-10 10:39:45 +00:00