| 2622ea73e33bf4269dcccff89a7ba224a80975b9 |
24-Feb-2017 |
Chih-Hung Hsieh <chh@google.com> |
Leave only an empty top level OWNERS file.
We should not copy OWNERS files from upstream,
or the owners should be registered in Gerrit Code Review.
Bug: 33166666
Test: default build targets
Change-Id: Ibfd47e643f03678bb65880653383adb84809169d
WNERS
ource/OWNERS
est/OWNERS
est/testFec/OWNERS
|
| 2f7dea164dc49ae8a0322e3c9edb1dd23266c664 |
13-Jan-2016 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] rtcp::Empty moved into own file and renamed to CompoundPacket on the way
Class renamed to indicated use of the rtcp::Empty class: it can only be used as container for other rtcp packets.
All tests that use Append function moved from rtcp_packet_unittest, even if they did not use Empty class.
This is because there is plan to make RtcpPacket class lighter by moving Append functionality to CompoundPacket class.
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1582503002
Cr-Commit-Position: refs/heads/master@{#11234}
UILD.gn
tp_rtcp.gypi
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/compound_packet.cc
ource/rtcp_packet/compound_packet.h
ource/rtcp_packet/compound_packet_unittest.cc
ource/rtcp_packet_unittest.cc
ource/rtcp_sender.cc
|
| 6955870806624479723addfae6dcf5d13968796c |
13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
nclude/rtp_payload_registry.h
nclude/rtp_receiver.h
nclude/rtp_rtcp_defines.h
ource/mock/mock_rtp_payload_strategy.h
ource/rtp_payload_registry.cc
ource/rtp_payload_registry_unittest.cc
ource/rtp_receiver_impl.cc
ource/rtp_receiver_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_audio.cc
ource/rtp_sender_audio.h
est/testAPI/test_api_audio.cc
|
| 92e677a1f8d24dfa0031d307c4a7d8e530cd4eb4 |
12-Jan-2016 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] rtcp::Sli packet moved into own file and got Parse function
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1551893002
Cr-Commit-Position: refs/heads/master@{#11228}
UILD.gn
tp_rtcp.gypi
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/sli.cc
ource/rtcp_packet/sli.h
ource/rtcp_packet/sli_unittest.cc
ource/rtcp_packet_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
|
| 3842c5c7f73639527e653f41c65334245d2317a1 |
12-Jan-2016 |
Stefan Holmer <stefan@webrtc.org> |
Wire-up BWE feedback for audio receive streams.
Also wires up receiving transport sequence numbers.
BUG=webrtc:5263
R=mflodman@webrtc.org, pbos@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1535963002 .
Cr-Commit-Position: refs/heads/master@{#11220}
ource/rtp_utility.cc
|
| 7e8145f05d5f6921ffca3d62e9c4d1301c1d8bcb |
11-Jan-2016 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] rtcp::Tmmbr moved into own file
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1575023002
Cr-Commit-Position: refs/heads/master@{#11206}
UILD.gn
tp_rtcp.gypi
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/tmmbr.cc
ource/rtcp_packet/tmmbr.h
ource/rtcp_packet/tmmbr_unittest.cc
ource/rtcp_packet_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
|
| ef3d805f6e50bc488f8e4e9e353068b78c73d17f |
11-Jan-2016 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] rtcp::Tmmbn moved into own file
explicetly unchanged.
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1578713002
Cr-Commit-Position: refs/heads/master@{#11201}
UILD.gn
tp_rtcp.gypi
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/tmmbn.cc
ource/rtcp_packet/tmmbn.h
ource/rtcp_packet/tmmbn_unittest.cc
ource/rtcp_packet_unittest.cc
ource/rtcp_sender.cc
|
| 3886fc8f57770ade76fb4bca29474659fedc7609 |
07-Jan-2016 |
Peter Boström <pbos@webrtc.org> |
Use pointer to generated FEC packet.
Removes multiple index lookups to generated_fec_packets_ speeding up
FecTest.FecTest with >2x in both Debug and Release, improving
performance but also readability.
On Debug this means that the slowest test in modules_tests now takes
~15-20 seconds instead of 50+ seconds, reducing the overall bottleneck.
BUG=webrtc:4712
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1552563003 .
Cr-Commit-Position: refs/heads/master@{#11166}
ource/forward_error_correction.cc
|
| 2df2ba7ae1cb89ee0a8e46b17b1d43d0f283aa4b |
29-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] Fix CL#1539423003
public function RtpHeaderParser::Parse with old signature restored as deprecated.
BUG=webrtc:5277
TBR=åsapersson
NOTRY=True
Review URL: https://codereview.webrtc.org/1550283002
Cr-Commit-Position: refs/heads/master@{#11135}
ource/rtp_utility.h
|
| f6975f46131981f83e0c88d276dee6b6c5753180 |
28-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] Lint errors cleaned from rtp_utility
R=åsapersson
BUG=webrtc:5277
Review URL: https://codereview.webrtc.org/1539423003
Cr-Commit-Position: refs/heads/master@{#11131}
ource/CPPLINT.cfg
ource/rtp_header_parser.cc
ource/rtp_sender.cc
ource/rtp_sender_audio.cc
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
ource/rtp_utility.cc
ource/rtp_utility.h
|
| a72e7349d52366655076e609e9e32d456da7f5a2 |
22-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] cleanup in RTCPSender class internals.
PrepareReportBlock and AddReportBlock private functions merged:
PrepareReportBlock moved report block from statistic to temporary structure
AddReportBlock copied that temporary structure into temporary map right after.
Thanks to rtcp packet classes that temporary structure is now unneccesary.
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1538833002
Cr-Commit-Position: refs/heads/master@{#11112}
ource/receive_statistics_impl.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
|
| a8890a57a5d03f942924ff61d3c62244f2bdab10 |
22-Dec-2015 |
danilchap <danilchap@webrtc.org> |
rtcp::Nack packet moved into own file and got Parse function
Review URL: https://codereview.webrtc.org/1461623003
Cr-Commit-Position: refs/heads/master@{#11111}
UILD.gn
tp_rtcp.gypi
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/nack.cc
ource/rtcp_packet/nack.h
ource/rtcp_packet/nack_unittest.cc
ource/rtcp_packet/rtpfb.cc
ource/rtcp_packet/rtpfb.h
ource/rtcp_packet_unittest.cc
ource/rtcp_sender.cc
ource/rtp_rtcp_impl_unittest.cc
|
| 1227e8b3451b1a2f2a765bf6101cb0862f625568 |
21-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] time helper functions
RTP timestams helper functions moved from rtp_utility
added functions to deal with CompactNtp timestamps
R=åsapersson
BUG=webrtc:5260
Review URL: https://codereview.webrtc.org/1535113002
Cr-Commit-Position: refs/heads/master@{#11106}
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/rtp_receiver_impl.cc
ource/rtp_sender.cc
ource/rtp_utility.cc
ource/rtp_utility.h
ource/time_util.h
ource/time_util_unittest.cc
|
| 0eb15ed7b806125774bd13fb214aeb403e2c6857 |
17-Dec-2015 |
kwiberg <kwiberg@webrtc.org> |
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
ource/rtcp_packet/app_unittest.cc
ource/rtcp_packet/bye_unittest.cc
ource/rtcp_packet/extended_jitter_report_unittest.cc
ource/rtcp_packet/receiver_report_unittest.cc
|
| 361888c324fe78f2427a2517d69777764d5f7564 |
16-Dec-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
OWNERS: Add * to .gyp{i,} everywhere.
Also convert DOS->Unix line endings in two of the OWNERS files.
NOTRY=True
NOPRESUBMIT=True
R=niklas.enbom@webrtc.org
Review URL: https://codereview.webrtc.org/1530003003 .
Cr-Commit-Position: refs/heads/master@{#11056}
WNERS
|
| 54999d411b97e3df54121e5f7bfb28846f3c8086 |
16-Dec-2015 |
danilchap <danilchap@webrtc.org> |
rtcp::Dlrr block moved into own file and got Parse function
BUG=webrtc:5260
Review URL: https://codereview.webrtc.org/1453973005
Cr-Commit-Position: refs/heads/master@{#11044}
UILD.gn
tp_rtcp.gypi
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/dlrr.cc
ource/rtcp_packet/dlrr.h
ource/rtcp_packet/dlrr_unittest.cc
|
| 91941ae493cb37a4e1250c7d3aad1c7394b5850e |
15-Dec-2015 |
danilchap <danilchap@webrtc.org> |
rtcp::VoipMetric block moved into own file and got Parse function
Review URL: https://codereview.webrtc.org/1452733002
Cr-Commit-Position: refs/heads/master@{#11030}
UILD.gn
tp_rtcp.gypi
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/voip_metric.cc
ource/rtcp_packet/voip_metric.h
ource/rtcp_packet/voip_metric_unittest.cc
ource/rtcp_packet_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
|
| 6db6cdc604f9a866991ecf8454eb7f7aa69918ea |
15-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1513303003
Cr-Commit-Position: refs/heads/master@{#11025}
ocks/mock_rtp_rtcp.h
ource/CPPLINT.cfg
ource/fec_receiver_impl.h
ource/nack_rtx_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_sender.cc
ource/rtp_header_extension.h
ource/rtp_packet_history.h
ource/rtp_receiver_impl.cc
ource/rtp_receiver_impl.h
ource/rtp_receiver_strategy.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender_audio.cc
ource/rtp_sender_audio.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
|
| e005cf2c93555125e7446a790779f8984cf9fa67 |
15-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] SSRCDatabase class cleaned (including all lint errors)
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1507313005
Cr-Commit-Position: refs/heads/master@{#11023}
ource/ssrc_database.cc
ource/ssrc_database.h
|
| 47a740bc5e36bcaf19385f9d4c0afb0cad070a05 |
15-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] lint errors about rand() usage fixed.
rand() usage replaced with new Random class, which also makes it clearer what interval random number is in.
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1519503002
Cr-Commit-Position: refs/heads/master@{#11019}
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_fec_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
est/testFec/test_fec.cc
|
| 40f349fddafd97c3f4cd0e37407bd1968496cb09 |
14-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] Lint errors cleared from rtp_rtcp/test
except rand() function that is subject of CL#1519503002
and namespace that is fixed in CL#1506823002
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1511413005
Cr-Commit-Position: refs/heads/master@{#11012}
est/testAPI/test_api_audio.cc
est/testAPI/test_api_rtcp.cc
est/testFec/test_fec.cc
est/testFec/test_packet_masks_metrics.cc
|
| b2f80e3a28d37c7c06b7765196b8de925898e0f2 |
14-Dec-2015 |
danilchap <danilchap@webrtc.org> |
rtp_rtcp/test/BWEStandAlone deleted as obsolete
BUG=webrtc:5277
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1525573002
Cr-Commit-Position: refs/heads/master@{#11008}
est/BWEStandAlone/BWEStandAlone.cc
est/BWEStandAlone/MatlabPlot.cc
est/BWEStandAlone/MatlabPlot.h
est/BWEStandAlone/TestLoadGenerator.cc
est/BWEStandAlone/TestLoadGenerator.h
est/BWEStandAlone/TestSenderReceiver.cc
est/BWEStandAlone/TestSenderReceiver.h
est/bwe_standalone.gypi
|
| 4c1093b86f4d0a1c8ade68a4b6a411b2674deac8 |
11-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
Add FEC producer fuzzing and a unittest for one of the issues found.
BUG=webrtc:4800
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1522463002 .
Cr-Commit-Position: refs/heads/master@{#10990}
ource/forward_error_correction.cc
ource/producer_fec_unittest.cc
|
| 6a6f0893dd1e653410ba4b22e7f33947d15aeb65 |
10-Dec-2015 |
danilchap <danilchap@webrtc.org> |
in rtp_rtcp module:
fixed build/namespaces lint errors
fixed readability/namespace lint errors
BUG=webrtc:5277
R=mflodman,stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1506823002
Cr-Commit-Position: refs/heads/master@{#10978}
ource/byte_io.h
ource/fec_private_tables_bursty.h
ource/fec_private_tables_random.h
ource/fec_test_helper.h
ource/forward_error_correction_internal.cc
ource/nack_rtx_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver_help.cc
ource/rtcp_utility.cc
ource/rtcp_utility.h
ource/rtp_header_extension.h
ource/vp8_partition_aggregator.cc
ource/vp8_partition_aggregator.h
ource/vp8_partition_aggregator_unittest.cc
est/BWEStandAlone/MatlabPlot.cc
est/BWEStandAlone/TestSenderReceiver.h
est/testAPI/test_api.cc
est/testAPI/test_api_audio.cc
est/testAPI/test_api_rtcp.cc
est/testFec/average_residual_loss_xor_codes.h
est/testFec/test_fec.cc
est/testFec/test_packet_masks_metrics.cc
|
| 5c1def8892390a336d1eecd9b61adacece858898 |
10-Dec-2015 |
danilchap <danilchap@webrtc.org> |
modules/rtp_rtcp/include folder cleared of lint warnings
Functions that do not follow lint are marked deprecated, including function in the interface.
BUG=webrtc:5308
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1493403003
Cr-Commit-Position: refs/heads/master@{#10975}
nclude/rtp_payload_registry.h
nclude/rtp_rtcp.h
nclude/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/rtp_payload_registry.cc
ource/rtp_payload_registry_unittest.cc
ource/rtp_receiver_impl.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
est/testAPI/test_api_audio.cc
|
| b8b6fbb7a5d2f5a14f7f6f81c253747aa28e4c7f |
10-Dec-2015 |
danilchap <danilchap@webrtc.org> |
lint build/include errors fixed in rtp_rtcp module
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1505993003
Cr-Commit-Position: refs/heads/master@{#10971}
nclude/rtp_payload_registry.h
nclude/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/h264_bitstream_parser.h
ource/receive_statistics_impl.h
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/extended_jitter_report.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_help.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_utility.cc
ource/rtp_packet_history.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_utility.cc
ource/rtp_utility.h
ource/tmmbr_help.cc
est/BWEStandAlone/MatlabPlot.cc
est/testAPI/test_api.cc
|
| 162abd3562d7b08ab36569800d757b52739b9249 |
10-Dec-2015 |
danilchap <danilchap@webrtc.org> |
lint whitespace warning removed from most rtp_rtcp/source/ files
rtcp_utility, rtp_utility, tmmbr_help, rtcp_receiver, rtcp_receiver_help are explicetly excluded from the cleanup becaues there are short plans (or cls) to do a deeper cleaning there.
BUG=webrtc:5277
R=pbos@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1512493002
Cr-Commit-Position: refs/heads/master@{#10966}
ource/dtmf_queue.cc
ource/forward_error_correction.cc
ource/forward_error_correction_internal.cc
ource/nack_rtx_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_packet.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_header_extension.h
ource/rtp_payload_registry_unittest.cc
ource/rtp_receiver_audio.h
ource/rtp_rtcp_config.h
ource/rtp_sender.h
ource/rtp_sender_audio.cc
ource/rtp_sender_audio.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.h
ource/ssrc_database.cc
ource/ssrc_database.h
ource/video_codec_information.h
ource/vp8_partition_aggregator.cc
|
| 84e78f9102dfbe9fc17aecd8d9d816042425a294 |
10-Dec-2015 |
terelius <terelius@webrtc.org> |
Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/.
Created a simple unit test for the new random number generator. (It mostly tests
that the generated numbers are consistent with the intended distribution, e.g. uniform.
It is not a comprehensive test of the quality of the random numbers.)
Several assertions in OveruseDetectorTest seem to depend on the exact sequence of random numbers. I updated those numbers to work with the new PRNG.
Compute the standard deviation of the expected result in TestReorderFilter instead of passing an uncertainty parameter.
BUG=webrtc:5177
Review URL: https://codereview.webrtc.org/1457023002
Cr-Commit-Position: refs/heads/master@{#10965}
ource/rtcp_packet/report_block_unittest.cc
|
| 0b3d7eec07100a9df006e679408a8e015af643d6 |
10-Dec-2015 |
mflodman <mflodman@webrtc.org> |
Prevent RTCP SR to be sent with bogus timestamp.
This CL makes sure no RTCP SR is sent before there is a valid timestamp
to set in the SR, based on the first sent media packet.
BUG=webrtc:1600
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1506103006 .
Cr-Commit-Position: refs/heads/master@{#10964}
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl_unittest.cc
|
| 5eb4988c0ac0665701e9bccba0fad3dcadfcfcd0 |
09-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] Lint build/header_guard errors fixed
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1506043003
Cr-Commit-Position: refs/heads/master@{#10949}
ource/fec_private_tables_random.h
ource/mock/mock_rtp_payload_strategy.h
ource/rtcp_packet.h
ource/rtp_header_extension.h
ource/rtp_packet_history.h
est/testAPI/test_api.h
est/testFec/average_residual_loss_xor_codes.h
|
| 4654d204e42d00dea43ce1e5b2200063e8272c8b |
08-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
Add test which verifies that the RTP header extensions are set correctly for FEC packets.
Also taking the opportunity to do a little bit of clean up.
BUG=webrtc:705
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1506863002 .
Cr-Commit-Position: refs/heads/master@{#10927}
ource/forward_error_correction.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
|
| fc47ed6c0524d7ee0bc7947f0ad65fcefda34a29 |
07-Dec-2015 |
Danil Chapovalov <danilchap@webrtc.org> |
rtcp::Rrtr block moved into own file and got Parse function
BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson
Review URL: https://codereview.webrtc.org/1496883002 .
Cr-Commit-Position: refs/heads/master@{#10912}
UILD.gn
tp_rtcp.gypi
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/rrtr.cc
ource/rtcp_packet/rrtr.h
ource/rtcp_packet/rrtr_unittest.cc
ource/rtcp_packet_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
|
| b86d4e4a8dec1eb1b801244a2a97cda66f561d8e |
07-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
Prepare the AudioSendStream to be hooked up to send-side BWE.
This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.
BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1479023002 .
Cr-Commit-Position: refs/heads/master@{#10909}
ource/rtp_sender.cc
ource/rtp_sender_audio.cc
ource/rtp_sender_video.cc
|
| 97f7e13c23ddb26543f33bce944d501e58d1dd9b |
04-Dec-2015 |
Danil Chapovalov <danilchap@webrtc.org> |
rtcp::ReceiverReport moved into own file and got Parse function
BUG=webrtc:5260
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1453083002 .
Cr-Commit-Position: refs/heads/master@{#10897}
UILD.gn
tp_rtcp.gypi
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/receiver_report.cc
ource/rtcp_packet/receiver_report.h
ource/rtcp_packet/receiver_report_unittest.cc
ource/rtcp_packet_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
|
| f7c5776d4254e31e51107388a05c66d14108a8af |
04-Dec-2015 |
Erik Språng <sprang@webrtc.org> |
Refactorings to send RTCP packets directly via the RtcpPacket callback, with some simplifications enabled by this. NACK now also sent via RtcpPacket.
BUG=webrtc:2450
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1309833002 .
Cr-Commit-Position: refs/heads/master@{#10888}
ource/rtcp_packet.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
|
| f8385aded0943c7889d6e9b92f3c0978f3657bb2 |
27-Nov-2015 |
danilchap <danilchap@webrtc.org> |
rtcp::Pli moved into own file and got a Parse function
Created rtcp::Psfb abstract class between rtcp::Pli and rtcp::RtcpPacket to hold common data for Feedback Message.
BUG=webrtc:5260
Review URL: https://codereview.webrtc.org/1446513002
Cr-Commit-Position: refs/heads/master@{#10823}
UILD.gn
tp_rtcp.gypi
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/pli.cc
ource/rtcp_packet/pli.h
ource/rtcp_packet/pli_unittest.cc
ource/rtcp_packet/psfb.cc
ource/rtcp_packet/psfb.h
ource/rtcp_packet_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
|
| 64c0a0a1110a69d722b6f7610e4096fe3288fe67 |
27-Nov-2015 |
stefan <stefan@webrtc.org> |
Revert of Make overuse estimator one dimensional. (patchset #5 id:80001 of https://codereview.webrtc.org/1376423002/ )
Reason for revert:
Broke webrtc_perf_tests on bots.
Original issue's description:
> Make overuse estimator one dimensional.
>
> This drops the payload size difference dimension of the Kalman filter,
> which doesn't improve the quality of the estimation when pacing packets
> on the send-side.
>
> R=gaetano.carlucci@gmail.com, mflodman@webrtc.org, terelius@webrtc.org
>
> Committed: https://crrev.com/06e05a85b9e4def75ed5e6b582c4df842616f25f
> Cr-Commit-Position: refs/heads/master@{#10809}
TBR=terelius@webrtc.org,mflodman@webrtc.org,gaetano.carlucci@gmail.com
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1481003002
Cr-Commit-Position: refs/heads/master@{#10816}
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver_unittest.cc
|
| 8c38e8b9b96d72317d6ce94c1442113b4e385dcb |
26-Nov-2015 |
Peter Boström <pbos@webrtc.org> |
Clean up PlatformThread.
* Move PlatformThread to rtc::.
* Remove ::CreateThread factory method.
* Make non-scoped_ptr from a lot of invocations.
* Make Start/Stop void.
* Remove rtc::Thread priorities, which were unused and would collide.
* Add ::IsRunning() to PlatformThread.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1476453002 .
Cr-Commit-Position: refs/heads/master@{#10812}
est/BWEStandAlone/MatlabPlot.cc
est/BWEStandAlone/MatlabPlot.h
est/BWEStandAlone/TestLoadGenerator.cc
est/BWEStandAlone/TestLoadGenerator.h
est/BWEStandAlone/TestSenderReceiver.cc
est/BWEStandAlone/TestSenderReceiver.h
|
| 06e05a85b9e4def75ed5e6b582c4df842616f25f |
26-Nov-2015 |
Stefan Holmer <stefan@webrtc.org> |
Make overuse estimator one dimensional.
This drops the payload size difference dimension of the Kalman filter,
which doesn't improve the quality of the estimation when pacing packets
on the send-side.
R=gaetano.carlucci@gmail.com, mflodman@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/1376423002 .
Cr-Commit-Position: refs/heads/master@{#10809}
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver_unittest.cc
|
| 12411ef40e08c5e28ccde54ab3418c96676ffcbc |
23-Nov-2015 |
pbos <pbos@webrtc.org> |
Move ThreadWrapper to ProcessThread in base.
Also removes all virtual methods. Permits using a thread from
rtc_base_approved (namely event tracing).
BUG=webrtc:5158
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1469013002
Cr-Commit-Position: refs/heads/master@{#10760}
est/BWEStandAlone/MatlabPlot.cc
est/BWEStandAlone/MatlabPlot.h
est/BWEStandAlone/TestLoadGenerator.cc
est/BWEStandAlone/TestLoadGenerator.h
est/BWEStandAlone/TestSenderReceiver.cc
est/BWEStandAlone/TestSenderReceiver.h
|
| 50c5136cb2ad11eb9ba3df1a1d54d527c8a0dc77 |
22-Nov-2015 |
danilchap <danilchap@webrtc.org> |
RTCP Bye packet moved to own file
Bye class got support for Parsing
Reason field implemented
Review URL: https://codereview.webrtc.org/1430013003
Cr-Commit-Position: refs/heads/master@{#10741}
UILD.gn
tp_rtcp.gypi
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/bye.cc
ource/rtcp_packet/bye.h
ource/rtcp_packet/bye_unittest.cc
ource/rtcp_packet_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
|
| 0219c9b4bfcbb778137756210eb95f40d936cc66 |
18-Nov-2015 |
danilchap <danilchap@webrtc.org> |
rtcp::App moved into own file and got Parse function
Review URL: https://codereview.webrtc.org/1437353003
Cr-Commit-Position: refs/heads/master@{#10688}
UILD.gn
tp_rtcp.gypi
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/app.cc
ource/rtcp_packet/app.h
ource/rtcp_packet/app_unittest.cc
ource/rtcp_packet_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
|
| 740c4f11e0d3b409c5444b328859754d2a717e33 |
17-Nov-2015 |
pbos <pbos@webrtc.org> |
Remove packet initializer in RtpRtcpRtxNackTest.
Fixes RtpRtcpRtxNackTest to not use uninitialized data when not sending
RTX.
BUG=webrtc:3183
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1427653007
Cr-Commit-Position: refs/heads/master@{#10665}
ource/nack_rtx_unittest.cc
|
| f8506cbdd88ce538d9e6c28ee39111345189778f |
13-Nov-2015 |
danilchap <danilchap@webrtc.org> |
rtcp::Ij renamed to rtcp::ExtendedJitterReport
to match name given in the RFC5450
private member renamed to inter_arrival_jitters_ for the same reason.
rtcp::ExtendedJitterReport moved into own file
accessors and Parse function added
to make class usable for parsing packet
Review URL: https://codereview.webrtc.org/1434213004
Cr-Commit-Position: refs/heads/master@{#10636}
UILD.gn
tp_rtcp.gypi
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/extended_jitter_report.cc
ource/rtcp_packet/extended_jitter_report.h
ource/rtcp_packet/extended_jitter_report_unittest.cc
ource/rtcp_packet_unittest.cc
ource/rtcp_receiver_unittest.cc
|
| df948f03b34dc652c2b3a944535fc01ec22395ce |
13-Nov-2015 |
danilchap <danilchap@webrtc.org> |
rtcp::ReportBlock refactored to contain parsing
Review URL: https://codereview.webrtc.org/1420283022
Cr-Commit-Position: refs/heads/master@{#10633}
UILD.gn
tp_rtcp.gypi
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/report_block.cc
ource/rtcp_packet/report_block.h
ource/rtcp_packet/report_block_unittest.cc
ource/rtcp_sender.cc
|
| 5237aaf243d29732f59557361b7a993c0a18cf0e |
11-Nov-2015 |
tfarina <tfarina@chromium.org> |
Convert usage of ARRAY_SIZE to arraysize.
ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.
BUG=None
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1405023016
Cr-Commit-Position: refs/heads/master@{#10594}
ource/h264_sps_parser_unittest.cc
|
| cfc319be1d6afec77bd41eeb70d3e7886dd524db |
10-Nov-2015 |
philipel <philipel@webrtc.org> |
Reland of Work on flexible mode and screen sharing. (patchset #1 id:1 of https://codereview.webrtc.org/1438543002/ )
Reason for revert:
Failed test not related to this CL (test fails on
master at an earlier date), re-landing original CL..
(This time from my @webrtc account.)
Original issue's description:
> Revert of Work on flexible mode and screen sharing. (patchset #28 id:520001 of https://codereview.webrtc.org/1328113004/ )
>
> Reason for revert:
> Seems to break VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Linux Memcheck buildbot.
>
> Original issue's description:
> > Work on flexible mode and screen sharing.
> >
> > Implement VP8 style screensharing but with spatial layers.
> > Implement flexible mode.
> >
> > Files from other patches:
> > generic_encoder.cc
> > layer_filtering_transport.cc
> >
> > BUG=webrtc:4914
> >
> > Committed: https://crrev.com/77ccfb4d16c148e61a316746bb5d9705e8b39f4a
> > Cr-Commit-Position: refs/heads/master@{#10572}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,philipel@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4914
>
> Committed: https://crrev.com/0be8f1d347bdb171462df89c2a4c69b3f3eb7519
> Cr-Commit-Position: refs/heads/master@{#10578}
TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,terelius@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4914
Review URL: https://codereview.webrtc.org/1431283002
Cr-Commit-Position: refs/heads/master@{#10581}
ource/rtp_format_vp9.cc
|
| 0be8f1d347bdb171462df89c2a4c69b3f3eb7519 |
10-Nov-2015 |
terelius <terelius@webrtc.org> |
Revert of Work on flexible mode and screen sharing. (patchset #28 id:520001 of https://codereview.webrtc.org/1328113004/ )
Reason for revert:
Seems to break VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Linux Memcheck buildbot.
Original issue's description:
> Work on flexible mode and screen sharing.
>
> Implement VP8 style screensharing but with spatial layers.
> Implement flexible mode.
>
> Files from other patches:
> generic_encoder.cc
> layer_filtering_transport.cc
>
> BUG=webrtc:4914
>
> Committed: https://crrev.com/77ccfb4d16c148e61a316746bb5d9705e8b39f4a
> Cr-Commit-Position: refs/heads/master@{#10572}
TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,philipel@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4914
Review URL: https://codereview.webrtc.org/1438543002
Cr-Commit-Position: refs/heads/master@{#10578}
ource/rtp_format_vp9.cc
|
| 77ccfb4d16c148e61a316746bb5d9705e8b39f4a |
10-Nov-2015 |
philipel <philipel@webrtc.org> |
Work on flexible mode and screen sharing.
Implement VP8 style screensharing but with spatial layers.
Implement flexible mode.
Files from other patches:
generic_encoder.cc
layer_filtering_transport.cc
BUG=webrtc:4914
Review URL: https://codereview.webrtc.org/1328113004
Cr-Commit-Position: refs/heads/master@{#10572}
ource/rtp_format_vp9.cc
|
| 19299fb28b2578d721649fff65419d4eb9ea1af3 |
07-Nov-2015 |
kjellander <kjellander@webrtc.org> |
Remove interface directories kept to avoid breaking downstream.
This is a follow-up CL for https://codereview.webrtc.org/1417683006
now that downstream code has been updated to use the 'include' directories
for header files instead.
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel -m tryserver.webrtc --bot=ios_rel
Review URL: https://codereview.webrtc.org/1414793020
Cr-Commit-Position: refs/heads/master@{#10547}
nterface/fec_receiver.h
nterface/receive_statistics.h
nterface/remote_ntp_time_estimator.h
nterface/rtp_cvo.h
nterface/rtp_header_parser.h
nterface/rtp_payload_registry.h
nterface/rtp_receiver.h
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
|
| c4a1c370aa7e4ec467ff16162ca0eef0cfaf53b0 |
06-Nov-2015 |
mflodman <mflodman@webrtc.org> |
Removed vie_defines.h
The defines still in use was only used in single files, so they were
moved to these specific cc-files.
Review URL: https://codereview.webrtc.org/1411573007
Cr-Commit-Position: refs/heads/master@{#10539}
ource/rtp_packet_history_unittest.cc
|
| c253a1c00eefd966aa59e00885fae4714806094f |
06-Nov-2015 |
asapersson <asapersson@webrtc.org> |
Reland of "Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile."
BUG=webrtc:5144
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1409753007
Cr-Commit-Position: refs/heads/master@{#10533}
ource/rtp_format_vp9.cc
ource/rtp_format_vp9_unittest.cc
|
| 394c537b21e6e2d6a93f2982f1e01d57497a98dc |
05-Nov-2015 |
asapersson <asapersson@webrtc.org> |
Update layer indices for non-flexible mode according to updates in the RTP payload profile.
https://tools.ietf.org/id/draft-ietf-payload-vp9-01.txt
BUG=chromium:500602
TBR=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1426813002
Cr-Commit-Position: refs/heads/master@{#10522}
ource/rtp_format_vp9.cc
ource/rtp_format_vp9_unittest.cc
|
| 5d4e944391043dddc36fc3d5570a34be1b286a5a |
04-Nov-2015 |
asapersson <asapersson@webrtc.org> |
Revert of Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile. (patchset #3 id:40001 of https://codereview.webrtc.org/1427253002/ )
Reason for revert:
Breaks bot.
Original issue's description:
> Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile. Max p_diff is 8 bits.
>
> Change type of number of reference pictures (size_t -> uint8_t). Max is 2 bits.
>
> Size of WebRtcRTPHeader: 4352 -> 1784 bytes.
>
> BUG=webrtc:5144, chromium:500602
>
> Committed: https://crrev.com/81c5c7f8157f767747bd97419eb0a589207354cf
> Cr-Commit-Position: refs/heads/master@{#10504}
TBR=stefan@webrtc.org,mflodman@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5144, chromium:500602
Review URL: https://codereview.webrtc.org/1423493005
Cr-Commit-Position: refs/heads/master@{#10508}
ource/rtp_format_vp9.cc
ource/rtp_format_vp9_unittest.cc
|
| 81c5c7f8157f767747bd97419eb0a589207354cf |
04-Nov-2015 |
asapersson <asapersson@webrtc.org> |
Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile. Max p_diff is 8 bits.
Change type of number of reference pictures (size_t -> uint8_t). Max is 2 bits.
Size of WebRtcRTPHeader: 4352 -> 1784 bytes.
BUG=webrtc:5144, chromium:500602
Review URL: https://codereview.webrtc.org/1427253002
Cr-Commit-Position: refs/heads/master@{#10504}
ource/rtp_format_vp9.cc
ource/rtp_format_vp9_unittest.cc
|
| ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
UILD.gn
nclude/fec_receiver.h
nclude/receive_statistics.h
nclude/remote_ntp_time_estimator.h
nclude/rtp_cvo.h
nclude/rtp_header_parser.h
nclude/rtp_payload_registry.h
nclude/rtp_receiver.h
nclude/rtp_rtcp.h
nclude/rtp_rtcp_defines.h
nterface/fec_receiver.h
nterface/receive_statistics.h
nterface/remote_ntp_time_estimator.h
nterface/rtp_cvo.h
nterface/rtp_header_parser.h
nterface/rtp_payload_registry.h
nterface/rtp_receiver.h
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
tp_rtcp.gypi
ource/fec_receiver_impl.h
ource/fec_receiver_unittest.cc
ource/fec_test_helper.h
ource/forward_error_correction.cc
ource/forward_error_correction.h
ource/mock/mock_rtp_payload_strategy.h
ource/nack_rtx_unittest.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/remote_ntp_time_estimator.cc
ource/remote_ntp_time_estimator_unittest.cc
ource/rtcp_packet.h
ource/rtcp_packet/transport_feedback.h
ource/rtcp_receiver.h
ource/rtcp_receiver_help.h
ource/rtcp_sender.h
ource/rtcp_utility.h
ource/rtp_format.h
ource/rtp_format_h264.cc
ource/rtp_format_h264_unittest.cc
ource/rtp_format_video_generic.cc
ource/rtp_format_vp8.h
ource/rtp_format_vp8_test_helper.h
ource/rtp_format_vp9.h
ource/rtp_header_extension.h
ource/rtp_header_extension_unittest.cc
ource/rtp_header_parser.cc
ource/rtp_packet_history.h
ource/rtp_packet_history_unittest.cc
ource/rtp_payload_registry.cc
ource/rtp_payload_registry_unittest.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_impl.cc
ource/rtp_receiver_impl.h
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_audio.cc
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
ource/rtp_utility.h
ource/vp8_partition_aggregator.h
est/BWEStandAlone/BWEStandAlone.cc
est/BWEStandAlone/TestLoadGenerator.h
est/BWEStandAlone/TestSenderReceiver.cc
est/BWEStandAlone/TestSenderReceiver.h
est/testAPI/test_api.h
est/testAPI/test_api_audio.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
|
| a41ab9326c8f0f7eb738e5d51a239a2b9e276361 |
31-Oct-2015 |
tfarina <tfarina@chromium.org> |
Switch usage of _DEBUG macro to NDEBUG.
http://stackoverflow.com/a/29253284/5237416
BUG=None
R=tommi@webrtc.org
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1429513004
Cr-Commit-Position: refs/heads/master@{#10468}
ource/rtp_utility.cc
|
| 6449990387f57e04f9912e8002c620a0f247eed5 |
29-Oct-2015 |
asapersson <asapersson@webrtc.org> |
Update scalability structure data according to updates in the RTP payload profile.
https://tools.ietf.org/id/draft-ietf-payload-vp9-01.txt
BUG=chromium:500602
TBR=mflodman
Review URL: https://codereview.webrtc.org/1411923004
Cr-Commit-Position: refs/heads/master@{#10445}
ource/rtp_format_vp9.cc
ource/rtp_format_vp9_unittest.cc
|
| 32df5efc6db394ff4c535c2049c079b8c0e31183 |
29-Oct-2015 |
asapersson <asapersson@webrtc.org> |
Update reference indices according to updates in the RTP payload profile.
https://tools.ietf.org/id/draft-ietf-payload-vp9-01.txt
BUG=chromium:500602
Review URL: https://codereview.webrtc.org/1406283008
Cr-Commit-Position: refs/heads/master@{#10442}
ource/rtp_format_vp9.cc
ource/rtp_format_vp9_unittest.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
nterface/remote_ntp_time_estimator.h
nterface/rtp_rtcp_defines.h
ource/bitrate.cc
ource/dtmf_queue.h
ource/fec_receiver_impl.cc
ource/forward_error_correction.h
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/remote_ntp_time_estimator.cc
ource/remote_ntp_time_estimator_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_sender.cc
ource/rtp_header_parser.cc
ource/rtp_packet_history.cc
ource/rtp_packet_history_unittest.cc
ource/rtp_receiver_audio.cc
ource/rtp_receiver_impl.h
ource/rtp_receiver_strategy.cc
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
ource/rtp_utility.cc
ource/ssrc_database.cc
ource/ssrc_database.h
ource/tmmbr_help.h
est/BWEStandAlone/BWEStandAlone.cc
est/BWEStandAlone/MatlabPlot.cc
est/BWEStandAlone/MatlabPlot.h
est/BWEStandAlone/TestLoadGenerator.cc
est/BWEStandAlone/TestLoadGenerator.h
est/BWEStandAlone/TestSenderReceiver.cc
est/BWEStandAlone/TestSenderReceiver.h
|
| ebc0b4e99365443111857a0c7cfcc8944d8f1b6e |
28-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use webrtc/base/logging.h for rtp_rtcp.
BUG=webrtc:5118
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1422023002 .
Cr-Commit-Position: refs/heads/master@{#10437}
ource/fec_receiver_impl.cc
ource/forward_error_correction.cc
ource/h264_sps_parser.cc
ource/remote_ntp_time_estimator.cc
ource/rtcp_packet.cc
ource/rtcp_receiver.cc
ource/rtcp_sender.cc
ource/rtp_format_vp8.cc
ource/rtp_format_vp9.cc
ource/rtp_packet_history.cc
ource/rtp_payload_registry.cc
ource/rtp_receiver_audio.cc
ource/rtp_receiver_impl.cc
ource/rtp_receiver_video.cc
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
ource/rtp_sender_video.cc
ource/rtp_utility.cc
|
| f116bd0d7a3cdad20bb638d5a87427bd920c8904 |
27-Oct-2015 |
stefan <stefan@webrtc.org> |
Call OnSentPacket for all packets sent in the test framework.
Required a bit of refactoring to make it possible to pass a Call to DirectTransport on construction. This also lead to me having to remove the shared lock between PacketTransport and RtpRtcpObserver. Now RtpRtcpObserver has a SetTransports method instead of a SetReceivers method.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1419193002
Cr-Commit-Position: refs/heads/master@{#10430}
ource/rtp_sender.cc
|
| a99069db6397ca9377ed473cdbfc6c4a53e22d98 |
23-Oct-2015 |
pbos <pbos@webrtc.org> |
Fix win32 header include order in rtp_utility.h.
Matches the include order in webrtc/base/criticalsection.h and makes use
of winsock2.h instead of winsock.h for consistency.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1407053008
Cr-Commit-Position: refs/heads/master@{#10389}
ource/rtp_utility.cc
|
| bbe876f0d30ec806c7c4a12629eb1f19ab45fb86 |
23-Oct-2015 |
stefan <stefan@webrtc.org> |
Set send times in send time history via OnSentPacket.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1419503004
Cr-Commit-Position: refs/heads/master@{#10384}
nterface/rtp_rtcp_defines.h
ource/rtp_sender.cc
|
| affa39cb39c77408109fef691533021533d969e1 |
21-Oct-2015 |
sprang <sprang@webrtc.org> |
Remove time constraint on first retransmit of a packet.
We don't allow more than one retransmission within one RTT, but the RTT
estimate might be off. Reasonably, the remote end will not send a NACK
until the packet after has been received - so always resend on first
request.
Review URL: https://codereview.webrtc.org/1414563003
Cr-Commit-Position: refs/heads/master@{#10362}
ource/rtp_packet_history.cc
ource/rtp_packet_history.h
ource/rtp_packet_history_unittest.cc
|
| 0a87ffcaad6a5e8cd4ead9c4d4957bd8a77fd7f2 |
21-Oct-2015 |
Stefan Holmer <stefan@webrtc.org> |
Fix bug in how send timestamps are converted to 24 bits.
BUG=webrtc:4173
R=sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1412683004 .
Cr-Commit-Position: refs/heads/master@{#10356}
ource/rtp_sender.cc
ource/rtp_sender_unittest.cc
|
| e4f96501fc5b3e6de0d1ccd262372afcda1f5b4f |
21-Oct-2015 |
tommi <tommi@webrtc.org> |
Remove system_wrappers/interface/trace_event.h
BUG=
Review URL: https://codereview.webrtc.org/1417773002
Cr-Commit-Position: refs/heads/master@{#10346}
ource/rtcp_receiver.cc
ource/rtcp_sender.cc
ource/rtp_receiver_audio.cc
ource/rtp_receiver_video.cc
ource/rtp_sender.cc
ource/rtp_sender_audio.cc
ource/rtp_sender_video.cc
|
| 22993e1a0c114122fc1b9de0fc74d4096ec868bd |
19-Oct-2015 |
pbos <pbos@webrtc.org> |
Unify FrameType and VideoFrameType.
Prevents some heap allocation and frame-type conversion since interfaces
mismatch. Also it's less confusing to have one type for this.
BUG=webrtc:5042
R=magjed@webrtc.org, mflodman@webrtc.org, henrik.lundin@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1371043003
Cr-Commit-Position: refs/heads/master@{#10320}
ource/rtp_format_h264_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender_audio.cc
ource/rtp_sender_unittest.cc
|
| 3eab10d629a5f549ddd62ec3053088205155d5b8 |
17-Oct-2015 |
noahric <noahric@chromium.org> |
Add back an override of RestoreOriginalPacket.
External consumers may have a dependency on the old name, so this will give them the opportunity to switch over.
BUG=
Review URL: https://codereview.webrtc.org/1414543002
Cr-Commit-Position: refs/heads/master@{#10310}
nterface/rtp_payload_registry.h
ource/rtp_payload_registry.cc
|
| 861c55e58311383b7f4f61af463ddea53eb3f30f |
16-Oct-2015 |
sprang <sprang@webrtc.org> |
Transport sequence number should be set also for retransmissions.
This is a reland of https://codereview.webrtc.org/1385563005 which was
reverted since the test was flaky. The reason was a race condition (in
the test) and that NACK wasn't properly set up.
BUG=
Review URL: https://codereview.webrtc.org/1406193002
Cr-Commit-Position: refs/heads/master@{#10303}
ource/rtp_sender.cc
|
| c1aeaf0dc37d96f31c92f893f4e30e7a5f7cc2b7 |
15-Oct-2015 |
stefan <stefan@webrtc.org> |
Wire up packet_id / send time callbacks to webrtc via libjingle.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1363573002
Cr-Commit-Position: refs/heads/master@{#10289}
nterface/rtp_rtcp_defines.h
ource/rtp_sender.cc
|
| 65220a70a38ffe252b587775c5e9104606ab7c2c |
14-Oct-2015 |
noahric <noahric@chromium.org> |
Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
Also updated the RTPPayloadRegistry::RestoreOriginalPacket signature to not take the first arg as a **, since it isn't modified.
Review URL: https://codereview.webrtc.org/1394573004
Cr-Commit-Position: refs/heads/master@{#10276}
nterface/rtp_payload_registry.h
ource/nack_rtx_unittest.cc
ource/rtp_payload_registry.cc
ource/rtp_payload_registry_unittest.cc
|
| 7dc39f331a161a24a4d4c6aac7cfb6850f43fb56 |
13-Oct-2015 |
sprang <sprang@webrtc.org> |
Avoid data race in RtcpReceiver.
See eg https://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan%20v2/builds/3930/steps/video_engine_tests/logs/stdio
Also some cleanup, lock annotations.
BUG=
Review URL: https://codereview.webrtc.org/1401463003
Cr-Commit-Position: refs/heads/master@{#10266}
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
|
| e23e737177cf5d131a6d4a4d229aa513c5270a59 |
08-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Disable pacer disabling.
Since the pacer is always enabled, removing enable/disable which makes
all packet queueing succeed. Also renaming one of the ::SendPackets
::InsertPacket to avoid confusion.
BUG=webrtc:1695, webrtc:2629
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1392513002 .
Cr-Commit-Position: refs/heads/master@{#10211}
nterface/rtp_rtcp_defines.h
ource/rtp_packet_history.cc
ource/rtp_packet_history.h
ource/rtp_packet_history_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
|
| 0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
07-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.
BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1362503003 .
Cr-Commit-Position: refs/heads/master@{#10196}
ource/h264_bitstream_parser.cc
ource/h264_sps_parser.cc
ource/h264_sps_parser.h
ource/h264_sps_parser_unittest.cc
|
| 10950692d67af5cfdcb19d93b40f25193d1db8c6 |
06-Oct-2015 |
Alejandro Luebs <aluebs@webrtc.org> |
Revert "Transport sequence number should be set also for retransmissions."
After this CL, video_engine_test started failing flakily in different bots for different CLs.
TBR=sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1393553003 .
Cr-Commit-Position: refs/heads/master@{#10188}
ource/rtp_sender.cc
|
| af4ced986bc62c263fbdb6eab68aef5c0d4e7c78 |
06-Oct-2015 |
sprang <sprang@webrtc.org> |
Transport sequence number should be set also for retransmissions.
When fetching a packet from the rtp packet history, cuased by a
retransmission, the transport seq extension header is enabled but the
sequence number is set to 0. A new transport seq should be assigned in
this case.
BUG=
Review URL: https://codereview.webrtc.org/1385563005
Cr-Commit-Position: refs/heads/master@{#10183}
ource/rtp_sender.cc
|
| 1d8a506405734d0cef9653704b036ca4f1388960 |
02-Oct-2015 |
stefan <stefan@webrtc.org> |
Add a PacketOptions struct to webrtc::Transport.
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.
BUG=4173
Review URL: https://codereview.webrtc.org/1376673004
Cr-Commit-Position: refs/heads/master@{#10144}
ource/nack_rtx_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender_unittest.cc
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
est/testAPI/test_api.cc
est/testAPI/test_api.h
|
| da903eaabbb6c6830efcafc3c2ade1d36f511e43 |
02-Oct-2015 |
pbos <pbos@webrtc.org> |
Unify newapi::RtcpMode and RTCPMethod.
BUG=webrtc:1695
R=solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1373903003
Cr-Commit-Position: refs/heads/master@{#10143}
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/nack_rtx_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
est/BWEStandAlone/TestSenderReceiver.cc
est/testAPI/test_api.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
|
| 49f9cdba02248d216dfc875dc0ab3c5ae187bc42 |
01-Oct-2015 |
sprang <sprang@webrtc.org> |
Fix bug where rtcp::TransportFeedback may generate incorrect messages.
In particular, if 14 short deltas were inserted (2 * capacity of status
vector chunk with 2bit items) followed by a large delta, that status
item would be dropped.
BUG=
Review URL: https://codereview.webrtc.org/1367193002
Cr-Commit-Position: refs/heads/master@{#10132}
ource/rtcp_packet/transport_feedback.cc
ource/rtcp_packet/transport_feedback_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_utility.cc
|
| 38778b046f058565bd4bae266f79c46cde806aa1 |
29-Sep-2015 |
sprang <sprang@webrtc.org> |
Add unit test for nack bandwidth constraint.
BUG=
Review URL: https://codereview.webrtc.org/1341743002
Cr-Commit-Position: refs/heads/master@{#10111}
ource/rtp_sender_unittest.cc
|
| 86fd9ed6f9e2a38aa343db8c62764659633231fa |
29-Sep-2015 |
sprang <sprang@webrtc.org> |
Set RtcpSender transport at construction.
BUG=
Review URL: https://codereview.webrtc.org/1365043002
Cr-Commit-Position: refs/heads/master@{#10106}
ource/rtcp_format_remb_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_rtcp_impl.cc
est/testAPI/test_api.cc
|
| 2d566686a23fe93ada58f1c38a0d4b9a0d68556e |
28-Sep-2015 |
pbos <pbos@webrtc.org> |
Unify Transport and newapi::Transport interfaces.
BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1369263002
Cr-Commit-Position: refs/heads/master@{#10096}
ource/nack_rtx_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
est/testAPI/test_api.cc
est/testAPI/test_api.h
|
| 4fbd145dcefd23169a9b1612d5ca92dace8196d6 |
28-Sep-2015 |
stefan <stefan@webrtc.org> |
Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-side.
In addition to this the ramp-up tests are refactored to use a receive call instead of only a remote bitrate estimator, and to make use of BaseTest.
BUG=webrtc:4836
Review URL: https://codereview.webrtc.org/1368943002
Cr-Commit-Position: refs/heads/master@{#10087}
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver_unittest.cc
|
| 6b8d3551681f40b880507cecc88f478a12383cc7 |
24-Sep-2015 |
Erik Språng <sprang@webrtc.org> |
Reland "Wire up send-side bandwidth estimation."
Revert was patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/
The culprit was RTC_DCHECK(poller_thread_->Start()); in rampup_test.cc
BUG=webrtc:4173
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1362303002 .
Cr-Commit-Position: refs/heads/master@{#10052}
nterface/rtp_rtcp_defines.h
ource/rtcp_format_remb_unittest.cc
ource/rtcp_packet.h
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_help.cc
ource/rtcp_receiver_help.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.h
ource/rtcp_utility.cc
ource/rtcp_utility.h
ource/rtp_rtcp_impl.cc
|
| 8c266e6baff043a1fa5c9134f46042908a376d5b |
24-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
H264 bitstream parser.
Parsing the encoded bitstream is required for doing downscaling
decisions based on average encoded QP to improve perceived quality.
BUG=webrtc:4968
R=noahric@chromium.org, stefan@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1314473008 .
Cr-Commit-Position: refs/heads/master@{#10051}
UILD.gn
tp_rtcp.gypi
ource/h264_bitstream_parser.cc
ource/h264_bitstream_parser.h
ource/h264_bitstream_parser_unittest.cc
|
| c9bbeb03542cffc14b7d306e5f88b6c0e593864d |
23-Sep-2015 |
Erik Språng <sprang@webrtc.org> |
Revert of Wire up send-side bandwidth estimation. (patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ )
Reason for revert:
Breaking some Android bots.
https://chromegw.corp.google.com/i/client.webrtc/builders/Android32%20Tests%20%28L%20Nexus5%29
Original issue's description:
> Wire up send-side bandwidth estimation.
>
> BUG=webrtc:4173
>
> Committed: https://crrev.com/ef165eefc79cf28bb67779afe303cc2365885547
> Cr-Commit-Position: refs/heads/master@{#10012}
TBR=stefan@webrtc.org, kjellander@webrtc.org
NOPRESUBMIT=false
NOTREECHECKS=false
NOTRY=false
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1362923002 .
Cr-Commit-Position: refs/heads/master@{#10029}
nterface/rtp_rtcp_defines.h
ource/rtcp_format_remb_unittest.cc
ource/rtcp_packet.h
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_help.cc
ource/rtcp_receiver_help.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.h
ource/rtcp_utility.cc
ource/rtcp_utility.h
ource/rtp_rtcp_impl.cc
|
| ef165eefc79cf28bb67779afe303cc2365885547 |
22-Sep-2015 |
sprang <sprang@webrtc.org> |
Wire up send-side bandwidth estimation.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1338203003
Cr-Commit-Position: refs/heads/master@{#10012}
nterface/rtp_rtcp_defines.h
ource/rtcp_format_remb_unittest.cc
ource/rtcp_packet.h
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_help.cc
ource/rtcp_receiver_help.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.h
ource/rtcp_utility.cc
ource/rtcp_utility.h
ource/rtp_rtcp_impl.cc
|
| ebbf8a805b45613b4cb118e4eb0cebe7eeee69ac |
22-Sep-2015 |
sprang <sprang@webrtc.org> |
Make sure rtp_rtcp module doesn't directly reference anything in the pacer module, and remove build dependencies on it.
BUG=
Review URL: https://codereview.webrtc.org/1350163005
Cr-Commit-Position: refs/heads/master@{#10005}
UILD.gn
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
tp_rtcp.gypi
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_audio.cc
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
|
| 586b19bdb615dde34cdcf107272d8857fe2f5631 |
18-Sep-2015 |
Stefan Holmer <stefan@webrtc.org> |
Enable probing with repeated payload packets by default.
To make this possible padding only packets will have the same timestamp
as the previously sent media packet, as long as RTX is not enabled. This
has the side effect that if we send only padding for a long time without
sending media, a receive-side jitter buffer could potentially overflow.
In practice this shouldn't be an issue, partly because RTX is recommended and
used by default, but also because padding typically is terminated before being
received by a client. It is also not an issue for bandwidth estimation as long
as abs-send-time is used instead of toffset.
BUG=chromium:425925
R=mflodman@webrtc.org, sprang@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1327933003 .
Cr-Commit-Position: refs/heads/master@{#9984}
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
|
| ac547a653862744d0aae560713f8418ad2852085 |
17-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Remove channel ids from various interfaces.
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.
IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately
BUG=webrtc:1695
R=henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1335353005 .
Cr-Commit-Position: refs/heads/master@{#9978}
nterface/rtp_receiver.h
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ource/nack_rtx_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_impl.cc
ource/rtp_receiver_impl.h
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_audio.cc
ource/rtp_sender_audio.h
ource/rtp_sender_unittest.cc
est/testAPI/test_api.cc
est/testAPI/test_api.h
est/testAPI/test_api_audio.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
|
| e64fbce0d92949b2928a1a7427b24f37ba90f526 |
17-Sep-2015 |
terelius <terelius@webrtc.org> |
Changed loopback transport in RtxNackTest to not store sequence numbers for retransmitted packets.
The unit test currently works as follows:
RtxLoopBackTransport logs the sequence numbers for all sent packets in expected_sequence_numbers_. Since the transport is configured to drop some of the packets there will be requests for retransmissions and the RTX sequence numbers will also be stored in the same list.
The (non-rtx) packets are received by VerifyingRtxReceiver which also stores the sequence numbers in a list sequence_numbers_. Both lists are then sorted and sequence_numbers_ is compared to whatever is in the start of expected_sequence_numbers_.
This works assuming that the RTX sequence numbers are greater than the regular RTP sequence numbers. In the RTP sender, both RTP and RTX are set to start at "random" 15-bit sequence numbers. The RTP sequence number is then changed to 2345 in the unit test, which would imply that the RTX sequence number is lower than the ones for RTP with probability ~1%. The reason why the test works anyway is that the test sets up a fake clock, which is used to initialize the random number generator in RTPSender, and the fixed starting point for the clock happens to result in RTX sequence numbers greater than 2345. However, any change to the initialization of the sequence numbers, the seeding of the PRNG or the fake clock causes a test failure with probability ~1%.
The new code omits the RTX sequence numbers from expected_sequence_numbers_, thus avoiding the problem with low RTX sequence numbers. The initialization of the sequence numbers in RTPSender is also bad, but I'll fix that in another CL.
Review URL: https://codereview.webrtc.org/1263383002
Cr-Commit-Position: refs/heads/master@{#9967}
ource/nack_rtx_unittest.cc
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
ource/packet_loss_stats.cc
ource/rtcp_packet.cc
ource/rtcp_packet/transport_feedback.cc
ource/rtcp_receiver.cc
ource/rtcp_sender.cc
ource/rtcp_utility.cc
ource/rtp_format_vp9.cc
ource/rtp_receiver_video.cc
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
ource/rtp_sender_video.cc
|
| 384194369b4be41912353631a68689129a49e58c |
16-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Consolidate constructormagic macros with Chromium version and remove Chromium override.
Part of work removing dependency on Chromium's base.
Only adds "= delete". From https://codereview.chromium.org/1151443003 :
"This will guarantee the error to be at compile time, and not rely on the call visibility (private)."
In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.
Depends on https://codereview.webrtc.org/1345433002/
BUG=chromium:468375
(in particular comment #37)
NOTRY=true
Review URL: https://codereview.webrtc.org/1342543004
Cr-Commit-Position: refs/heads/master@{#9954}
ource/rtp_format_h264.cc
ource/rtp_format_h264.h
|
| 3c089d751ede283e21e186885eaf705c3257ccd2 |
16-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to contructormagic macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
nterface/remote_ntp_time_estimator.h
ource/rtcp_packet.h
ource/rtcp_packet/transport_feedback.h
ource/rtcp_receiver_help.h
ource/rtp_format_h264.h
ource/rtp_format_video_generic.h
ource/rtp_format_vp8.h
ource/rtp_format_vp8_test_helper.h
ource/rtp_format_vp9.h
ource/vp8_partition_aggregator.h
|
| 73a93e82579d6eeb3a1c4a63ef4b64c3c4d9bb18 |
14-Sep-2015 |
sprang <sprang@webrtc.org> |
Add a ParseHeader method to RtcpPacket, for parsing common RTCP header.
Also refactor TransportFeedback to use this.
BUG=
Review URL: https://codereview.webrtc.org/1307663004
Cr-Commit-Position: refs/heads/master@{#9935}
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/transport_feedback.cc
ource/rtcp_utility.cc
ource/rtcp_utility.h
ource/rtcp_utility_unittest.cc
|
| 5e023eb337eed9746ecea7fc6fbb0fca386f1961 |
14-Sep-2015 |
sprang <sprang@webrtc.org> |
Add TransportFeedback adapter, adapting remote feedback to bwe estiamtor
When using send-side bandwidth estimation, the inter-packet delay is
reported back to the sender using RTCP TransportFeedback messages.
Theis data needs to be translated into a format which the bandwidth
estimator (now instantiated on the send side) can use, including looking
up the local absolute send time from the send time history.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1329083005
Cr-Commit-Position: refs/heads/master@{#9929}
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
ource/rtp_sender.h
|
| c32d2db69bc94480ecb312268c6e6769d4a1cac6 |
11-Sep-2015 |
pbos <pbos@webrtc.org> |
Refactor RTPPacketHistory to use a packet struct.
Collects packet information within a struct instead of spreading it out
over different vectors. Adds a fixed-size buffer to the stored packet
instead of using vectors.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1340573002
Cr-Commit-Position: refs/heads/master@{#9926}
ource/rtp_packet_history.cc
ource/rtp_packet_history.h
ource/rtp_packet_history_unittest.cc
ource/rtp_sender.cc
|
| 9a78d22822880884f9fa495e4cbe33f5224296c4 |
10-Sep-2015 |
tommi <tommi@webrtc.org> |
Revert of Consolidate constructormagic macros with Chromium version and remove Chromium override. (patchset #4 id:60001 of https://codereview.webrtc.org/1316363005/ )
Reason for revert:
Had to revert since FYI bots stopped compiling. Example failure:
[94/9470] CXX obj\third_party\webrtc\modules\video_processing\main\source\video_processing_sse2.content_analysis_sse2.obj
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\third_party\webrtc\modules\video_coding\codecs\h264\webrtc_h264.h264.obj.rsp /c ..\..\third_party\webrtc\modules\video_coding\codecs\h264\h264.cc /Foobj\third_party\webrtc\modules\video_coding\codecs\h264\webrtc_h264.h264.obj /Fdobj\third_party\webrtc\modules\webrtc_h264.cc.pdb
e:\b\build\slave\win\build\src\base\macros.h(28) : error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win\build\src\base\macros.h(28) : warning C4005: 'DISALLOW_COPY_AND_ASSIGN' : macro redefinition
e:\b\build\slave\win\build\src\third_party\webrtc\base\constructormagic.h(27) : see previous definition of 'DISALLOW_COPY_AND_ASSIGN'
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\third_party\webrtc\base\rtc_base_approved.bitbuffer.obj.rsp /c ..\..\third_party\webrtc\base\bitbuffer.cc /Foobj\third_party\webrtc\base\rtc_base_approved.bitbuffer.obj /Fdobj\third_party\webrtc\base\rtc_base_approved.cc.pdb
e:\b\build\slave\win\build\src\base\macros.h(28) : error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win\build\src\base\macros.h(28) : warning C4005: 'DISALLOW_COPY_AND_ASSIGN' : macro redefinition
e:\b\build\slave\win\build\src\third_party\webrtc\base\constructormagic.h(27) : see previous definition of 'DISALLOW_COPY_AND_ASSIGN'
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\third_party\webrtc\modules\audio_processing\logging\audio_processing.aec_logging_file_handling.obj.rsp /c ..\..\third_party\webrtc\modules\audio_processing\logging\aec_logging_file_handling.cc /Foobj\third_party\webrtc\modules\audio_processing\logging\audio_processing.aec_logging_file_handling.obj /Fdobj\third_party\webrtc\modules\audio_processing.cc.pdb
e:\b\build\slave\win\build\src\base\macros.h(28) : error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win\build\src\base\macros.h(28) : warning C4005: 'DISALLOW_COPY_AND_ASSIGN' : macro redefinition
e:\b\build\slave\win\build\src\third_party\webrtc\base\constructormagic.h(27) : see previous definition of 'DISALLOW_COPY_AND_ASSIGN'
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\third_party\webrtc\modules\audio_processing\beamformer\audio_processing.nonlinear_beamformer.obj.rsp /c ..\..\third_party\webrtc\modules\audio_processing\beamformer\nonlinear_beamformer.cc /Foobj\third_party\webrtc\modules\audio_processing\beamformer\audio_processing.nonlinear_beamformer.obj /Fdobj\third_party\webrtc\modules\audio_processing.cc.pdb
e:\b\build\slave\win\build\src\base\macros.h(28) : error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win\build\src\base\macros.h(28) : warning C4005: 'DISALLOW_COPY_AND_ASSIGN' : macro redefinition
e:\b\build\slave\win\build\src\third_party\webrtc\base\constructormagic.h(27) : see previous definition of 'DISALLOW_COPY_AND_ASSIGN'
Original issue's description:
> Consolidate constructormagic macros with Chromium version and remove Chromium override.
>
> Part of work removing dependency on Chromium's base.
>
> Only adds "= delete". From https://codereview.chromium.org/1151443003 :
> "This will guarantee the error to be at compile time, and not rely on the call visibility (private)."
>
> In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.
>
> BUG=chromium:468375 (in particular comment #37)
> NOTRY=true
>
> Committed: https://crrev.com/0de8ff488d92e0bc6b7b65662898ff5e955cda93
> Cr-Commit-Position: refs/heads/master@{#9913}
TBR=andrew@webrtc.org,henrikg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:468375 (in particular comment #37)
Review URL: https://codereview.webrtc.org/1330283002
Cr-Commit-Position: refs/heads/master@{#9914}
ource/rtp_format_h264.cc
ource/rtp_format_h264.h
|
| 0de8ff488d92e0bc6b7b65662898ff5e955cda93 |
10-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Consolidate constructormagic macros with Chromium version and remove Chromium override.
Part of work removing dependency on Chromium's base.
Only adds "= delete". From https://codereview.chromium.org/1151443003 :
"This will guarantee the error to be at compile time, and not rely on the call visibility (private)."
In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.
BUG=chromium:468375 (in particular comment #37)
NOTRY=true
Review URL: https://codereview.webrtc.org/1316363005
Cr-Commit-Position: refs/heads/master@{#9913}
ource/rtp_format_h264.cc
ource/rtp_format_h264.h
|
| 233bd87d45bbeeec50d7687e7d98c1cfc7f65562 |
08-Sep-2015 |
sprang <sprang@webrtc.org> |
Add RemoteEstimatorProxy for capturing receive times
For use when send-side bandwidth estimation is enabled.
Receive times need to be captured, buffered and then sent using
TransportFeedback RTCP messaged back to the send side.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1290813008
Cr-Commit-Position: refs/heads/master@{#9898}
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_packet/transport_feedback.cc
ource/rtcp_packet/transport_feedback.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| 7f6a6fc0b23795cd4f0aacbf707618c1f3d0a878 |
08-Sep-2015 |
ivica <ivica@webrtc.org> |
Enabling spatial layers in VP9Impl. Filter layers in the loopback test.
Handling the case when encoder drops only the higher layer.
Added options to screenshare loopback test to discard high temporal or spatial layers (to view the lower layers).
Review URL: https://codereview.webrtc.org/1287643002
Cr-Commit-Position: refs/heads/master@{#9883}
ource/rtp_format_vp9.cc
ource/rtp_format_vp9_unittest.cc
|
| c8a1cccd0a80b35a5b6846f6efe6082f96c29083 |
04-Sep-2015 |
sprang <sprang@webrtc.org> |
Fixed base time in TransportFeedback message writing.
Value was incorrectly truncated to 16 bits when serializing the message.
Fixed, with added regression tests.
BUG=
Review URL: https://codereview.webrtc.org/1294393002
Cr-Commit-Position: refs/heads/master@{#9858}
ource/rtcp_packet/transport_feedback.cc
ource/rtcp_packet/transport_feedback.h
ource/rtcp_packet/transport_feedback_unittest.cc
|
| be9b7b6881e5b0e0b54e7d2fb79c5af5f68c015b |
04-Sep-2015 |
sprang <sprang@webrtc.org> |
Make sure ByteReader and ByteWriter classes (and their specializations) don't perform operations that have implementation-specific or undefined behavior.
Pitfalls:
* Left shift of signed integer has undefined behavior
* Right-shift of signed integer has platform-specific behavior is value is negative
* Cast from unsigned to signed has undefined behavior if value is negative
BUG=webrtc:4824
Review URL: https://codereview.webrtc.org/1226993003
Cr-Commit-Position: refs/heads/master@{#9854}
ource/byte_io.h
|
| 521875a9a48256ffa865f7c4a635f260358fa4a7 |
01-Sep-2015 |
Erik Språng <sprang@webrtc.org> |
Use RtcpPacket to send APP in RtcpSender
BUG=webrtc:2450
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1311453002 .
Cr-Commit-Position: refs/heads/master@{#9827}
ource/rtcp_sender.cc
ource/rtcp_sender_unittest.cc
|
| ca28fdcf9f061193671ef71c492ad3cd8c193a59 |
31-Aug-2015 |
Erik Språng <sprang@webrtc.org> |
Use RtcpPacket to send XR (RTRR, DLRR, VOIP) in RtcpSender
BUG=webrtc:2450
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1304123003 .
Cr-Commit-Position: refs/heads/master@{#9820}
ource/rtcp_sender.cc
|
| d83df50e95a73859bee1568ec7375ff832e1d628 |
27-Aug-2015 |
sprang <sprang@webrtc.org> |
Use RtcpPacket to send TMMBN in RtcpSender
BUG=webrtc:2450
Review URL: https://codereview.webrtc.org/1302403002
Cr-Commit-Position: refs/heads/master@{#9793}
ource/rtcp_sender.cc
|
| d8ee4f99154691752dd8d2f2d70750554dca7ca7 |
24-Aug-2015 |
sprang <sprang@webrtc.org> |
Use RtcpPacket to send BYE in RtcpSender
BUG=webrtc:2450
Review URL: https://codereview.webrtc.org/1306893003
Cr-Commit-Position: refs/heads/master@{#9763}
ource/rtcp_sender.cc
|
| 81a3e60c639b5b05486acd1fb84e376271e50012 |
21-Aug-2015 |
sprang <sprang@webrtc.org> |
Use RtcpPacket to send TMMBR in RtcpSender
BUG=webrtc:2450
Review URL: https://codereview.webrtc.org/1296163004
Cr-Commit-Position: refs/heads/master@{#9755}
ource/rtcp_sender.cc
|
| dd4edc5813a0331049f53a93ac2404a8899e6ae8 |
21-Aug-2015 |
sprang <sprang@webrtc.org> |
Reland of Use RtcpPacket to send REMB in RtcpSender (patchset #1 id:1 of https://codereview.webrtc.org/1300863002/ )
Reason for revert:
This wasn't the cause of the breakage. Re-reverting.
https://code.google.com/p/webrtc/issues/detail?id=4923
Original issue's description:
> Revert of Use RtcpPacket to send REMB in RtcpSender (patchset #1 id:1 of https://codereview.webrtc.org/1290573004/ )
>
> Reason for revert:
> A few bots started failing rtc_unittests after this was commited. Ex https://build.chromium.org/p/client.webrtc/builders/Linux64%20Debug/builds/5048
>
> Original issue's description:
> > Use RtcpPacket to send REMB in RtcpSender
> >
> > BUG=webrtc:2450
> > R=asapersson@webrtc.org
> >
> > Committed: https://chromium.googlesource.com/external/webrtc/+/35ab4baa20a730de71b390008900a16563cbbe8e
>
> TBR=asapersson@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:2450
>
> Committed: https://crrev.com/141c5951f4beda868797c2746002a4b1b267ab2a
> Cr-Commit-Position: refs/heads/master@{#9723}
TBR=asapersson@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:2450
Review URL: https://codereview.webrtc.org/1309723002
Cr-Commit-Position: refs/heads/master@{#9754}
ource/rtcp_sender.cc
|
| 22ff75a1635597d96644084707645b11bb3e6f95 |
21-Aug-2015 |
asapersson <asapersson@webrtc.org> |
Add unit tests for more packet types in rtcp_sender_unittest.
BUG=webrtc:2450
Review URL: https://codereview.webrtc.org/1291113004
Cr-Commit-Position: refs/heads/master@{#9751}
ource/rtcp_sender_unittest.cc
|
| 141c5951f4beda868797c2746002a4b1b267ab2a |
18-Aug-2015 |
sprang <sprang@webrtc.org> |
Revert of Use RtcpPacket to send REMB in RtcpSender (patchset #1 id:1 of https://codereview.webrtc.org/1290573004/ )
Reason for revert:
A few bots started failing rtc_unittests after this was commited. Ex https://build.chromium.org/p/client.webrtc/builders/Linux64%20Debug/builds/5048
Original issue's description:
> Use RtcpPacket to send REMB in RtcpSender
>
> BUG=webrtc:2450
> R=asapersson@webrtc.org
>
> Committed: https://chromium.googlesource.com/external/webrtc/+/35ab4baa20a730de71b390008900a16563cbbe8e
TBR=asapersson@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:2450
Review URL: https://codereview.webrtc.org/1300863002
Cr-Commit-Position: refs/heads/master@{#9723}
ource/rtcp_sender.cc
|
| 35ab4baa20a730de71b390008900a16563cbbe8e |
18-Aug-2015 |
Erik Språng <sprang@webrtc.org> |
Use RtcpPacket to send REMB in RtcpSender
BUG=webrtc:2450
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1290573004 .
Cr-Commit-Position: refs/heads/master@{#9722}
ource/rtcp_sender.cc
|
| cf7f54d6f40db4bb751d8ec0d5df2f81b4eda690 |
13-Aug-2015 |
sprang <sprang@webrtc.org> |
Use RtcpPacket to send RPSI in RtcpSender
BUG=webrtc:2450
Review URL: https://codereview.webrtc.org/1291013002
Cr-Commit-Position: refs/heads/master@{#9704}
ource/rtcp_sender.cc
|
| 0365a27f56aa2d2376d2f356bf70d161c3450244 |
11-Aug-2015 |
sprang <sprang@webrtc.org> |
Use RtcpPacket to send SLI in RtcpSender
BUG=webrtc:2450
Review URL: https://codereview.webrtc.org/1268383002
Cr-Commit-Position: refs/heads/master@{#9695}
ource/rtcp_sender.cc
est/testAPI/test_api_rtcp.cc
|
| 4cee419e0777dcbfbd0837e26bed202e35e696a9 |
10-Aug-2015 |
Minyue <minyue@webrtc.org> |
Separating voice activity flag from audio level in RtpHeaderExtension.
VAD flag was embedded in RtpHeaderExtension.audioLevel, which is not easy to interpret. This CL tries to separate the flag with the actual audio level.
BUG=
R=andrew@webrtc.org, henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1272343003 .
Cr-Commit-Position: refs/heads/master@{#9691}
ource/rtp_sender_unittest.cc
ource/rtp_utility.cc
|
| 62dae190985454188de112e35a16e35fc6e912a4 |
05-Aug-2015 |
sprang <sprang@webrtc.org> |
Use RtcpPacket to send FIR in RtcpSender
BUG=webrtc:2450
Review URL: https://codereview.webrtc.org/1261323003
Cr-Commit-Position: refs/heads/master@{#9677}
ource/rtcp_sender.cc
|
| 867fb5224e1ba6a1c2cd523c005499a93ed61a08 |
03-Aug-2015 |
sprang <sprang@webrtc.org> |
Add support for transport wide sequence numbers
Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.
BUG=webrtc:4311
Review URL: https://codereview.webrtc.org/1247293002
Cr-Commit-Position: refs/heads/master@{#9670}
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| 72aa9a6c6e8d05c744496ad8c53273ec49556d28 |
31-Jul-2015 |
Erik Språng <sprang@webrtc.org> |
Use RtcpPacket to send PLI in RtcpSender
BUG=webrtc:2450
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1262153003 .
Cr-Commit-Position: refs/heads/master@{#9666}
ource/rtcp_sender.cc
|
| a9455ab235e1169572f9eae3175cd9310d6729e2 |
31-Jul-2015 |
asapersson <asapersson@webrtc.org> |
Integration of VP9 packetization.
Supports running 1 spatial and 1-3 temporal layers in non-flexible mode.
BUG=webrtc:4148, webrtc:4168, chromium:500602
TBR=mflodman
Review URL: https://codereview.webrtc.org/1211353002
Cr-Commit-Position: refs/heads/master@{#9665}
ource/rtp_format_vp9.cc
ource/rtp_format_vp9_unittest.cc
ource/rtp_payload_registry.cc
ource/rtp_sender_video.cc
|
| 5f5f11cc8bf86de2e6ccf32eef1fe9f7c8e6c924 |
30-Jul-2015 |
pbos <pbos@webrtc.org> |
FEC protect H264 delta frames as well.
BUG=webrtc:4800
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1266593003
Cr-Commit-Position: refs/heads/master@{#9662}
ource/rtp_format_h264.cc
ource/rtp_format_vp8.cc
|
| a3b8769860bdb0a45dbff6d1e0092486fa59aaa4 |
29-Jul-2015 |
Erik Språng <sprang@webrtc.org> |
Add packetization and coding/decoding of feedback message format.
BUG=webrtc:4312
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1175263002 .
Cr-Commit-Position: refs/heads/master@{#9651}
UILD.gn
tp_rtcp.gypi
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet/transport_feedback.cc
ource/rtcp_packet/transport_feedback.h
ource/rtcp_packet/transport_feedback_unittest.cc
ource/rtcp_packet_unittest.cc
|
| f1828e8ed96ae1aa3ea9dc1eb96e2e703d2e78cf |
28-Jul-2015 |
pbos <pbos@webrtc.org> |
Prevent OOB reads for truncated H264 STAP-A packets.
BUG=webrtc:4771, webrtc:4834
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1238033003
Cr-Commit-Position: refs/heads/master@{#9650}
ource/rtp_format_h264.cc
|
| f38ea3caa39887c63e7d4862dcf420d4a35c1073 |
28-Jul-2015 |
asapersson <asapersson@webrtc.org> |
Add support for VP9 packetization/depacketization.
RTP payload format for VP9:
https://www.ietf.org/id/draft-uberti-payload-vp9-01.txt
BUG=webrtc:4148, webrtc:4168, chromium:500602
TBR=mflodman
Review URL: https://codereview.webrtc.org/1232023006
Cr-Commit-Position: refs/heads/master@{#9649}
UILD.gn
tp_rtcp.gypi
ource/rtp_format.cc
ource/rtp_format_vp9.cc
ource/rtp_format_vp9.h
ource/rtp_format_vp9_unittest.cc
|
| a38233a586dd865c0cd728ce523b3a82ca52ea8b |
24-Jul-2015 |
Erik Språng <sprang@webrtc.org> |
Removed extended jitter report from RtcpSender.
This was never used (value always 0, when sent)
BUG=2450
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1208843003 .
Cr-Commit-Position: refs/heads/master@{#9631}
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| 43e7d3bc150788045b549f4ab94a91095980d059 |
14-Jul-2015 |
noahric <noahric@chromium.org> |
Avoid overflow in checking for emulation bytes in rbsp.
Also fixed an arithmetic issue where a 0 0 3 at the end of the rbsp would include the 3 (that's not a legal bitstream anyway, so it probably wasn't a real bug, but it was incorrect).
This maintains the underflow fix from an earlier CL (https://codereview.webrtc.org/1219493004/). The overflow fix is virtually impossible to hit (hence no unit tests), but is there for strict correctness.
BUG=
Review URL: https://codereview.webrtc.org/1226203002
Cr-Commit-Position: refs/heads/master@{#9581}
ource/h264_sps_parser.cc
|
| ba8c15b857c0f341d9c1e02d41b6ccd56f9f1030 |
14-Jul-2015 |
pbos <pbos@webrtc.org> |
Merge methods for configuring NACK/FEC/hybrid.
BUG=webrtc:1695
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1226143013
Cr-Commit-Position: refs/heads/master@{#9580}
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
|
| d6f1a38165455d743fbe61f6980f22be6a3c4de9 |
14-Jul-2015 |
Peter Boström <pbos@webrtc.org> |
Remove ViEChannel simulcast lock.
Since the number of streams is now known on construction we can
initialize all RTP modules on construction. They are internally locked
so we don't nede a simulcast lock anymore.
BUG=1695
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52639004 .
Cr-Commit-Position: refs/heads/master@{#9577}
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
est/testAPI/test_api.cc
|
| 30409b4dca3d9cfdb0e714a5932b135becb0f822 |
11-Jul-2015 |
bcornell <bcornell@google.com> |
Add statistics gathering for packet loss.
Adds a class used to classify whether packet loss events are a single packet or multiple packets as well as how many packets have been lost. Also exposes a new function in the RtpRtcp interface to retrieve these statistics.
BUG=
Review URL: https://codereview.webrtc.org/1198853004
Cr-Commit-Position: refs/heads/master@{#9568}
UILD.gn
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
tp_rtcp.gypi
ource/packet_loss_stats.cc
ource/packet_loss_stats.h
ource/packet_loss_stats_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| d436298332c7a7ecb51241f3a66588539c2ece83 |
07-Jul-2015 |
pbos <pbos@webrtc.org> |
Remove ResetStatistics from RTP feedback.
BUG=
R=asapersson@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1213603002
Cr-Commit-Position: refs/heads/master@{#9548}
nterface/receive_statistics.h
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_impl.cc
ource/rtp_receiver_impl.h
ource/rtp_receiver_strategy.cc
ource/rtp_receiver_strategy.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 2bad88d164754f1f0694e9fea1051e71b3cb5347 |
06-Jul-2015 |
pbos <pbos@webrtc.org> |
Prevent heap overflows for incorrect FEC packet lengths.
Bugs found by manual inspection of code, not by fuzzing or packet
replays. At least one of them confirmed by local fuzzing.
BUG=chromium:496094, webrtc:4771
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1182793002
Cr-Commit-Position: refs/heads/master@{#9542}
ource/fec_receiver_unittest.cc
ource/forward_error_correction.cc
ource/forward_error_correction.h
|
| 468e62a97426a8d001e9187f3ca1d1e43f80b970 |
06-Jul-2015 |
Erik Språng <sprang@webrtc.org> |
Remove MimdRateControl and factories for RemoteBitrateEstimor.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1208083002.
Cr-Commit-Position: refs/heads/master@{#9541}
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender_unittest.cc
|
| 545727ecce444320328b825d65b287e844dca7cb |
01-Jul-2015 |
pbos <pbos@webrtc.org> |
Move early-return in TimeToSendPadding.
Prevents taking send_critsect_ for checking sending status when not
actually intending to send padding.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1218093002
Cr-Commit-Position: refs/heads/master@{#9526}
ource/rtp_sender.cc
|
| bd2522abf75891f34da6f83c247c47ca95641cee |
01-Jul-2015 |
pbos <pbos@webrtc.org> |
Fail RTP parsing on excessive padding length.
BUG=webrtc:4771
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1220863002
Cr-Commit-Position: refs/heads/master@{#9525}
ource/fec_receiver_unittest.cc
ource/rtp_payload_registry.cc
ource/rtp_receiver_video.cc
ource/rtp_sender_unittest.cc
ource/rtp_utility.cc
|
| 4daa90eed7591f37d7d157f9ec5000d83272a604 |
01-Jul-2015 |
pbos <pbos@webrtc.org> |
Prevent size_t underflow in H264 SPS parsing.
BUG=webrtc:4771
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1219493004
Cr-Commit-Position: refs/heads/master@{#9523}
ource/h264_sps_parser.cc
ource/rtp_format_h264_unittest.cc
|
| 2f1509395b56fe3175b27dc2ac76e8f749c809f7 |
30-Jun-2015 |
pbos <pbos@webrtc.org> |
Prevent OOB read on truncated H264 headers.
Prevents OOB reads on truncated FU-A NAL units, StapA headers and past
truncation just after StapA headers.
BUG=webrtc:4771
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1218023003
Cr-Commit-Position: refs/heads/master@{#9522}
ource/rtp_format_h264.cc
ource/rtp_format_h264_unittest.cc
|
| 7ada923a94de3cd95142b3052996f9b38e134f39 |
30-Jun-2015 |
pbos <pbos@webrtc.org> |
Prevent OOB reads for zero-length H264 payloads.
Also fixes zero-length OOB reads for generic packetization.
BUG=webrtc:4771
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1218013002
Cr-Commit-Position: refs/heads/master@{#9521}
ource/rtp_format_h264.cc
ource/rtp_format_h264_unittest.cc
ource/rtp_format_video_generic.cc
|
| 48c3839e703f4186570590a9c7d966af6407d3ab |
30-Jun-2015 |
pbos <pbos@webrtc.org> |
Prevent depacketizer OOB reads on zero-length VP8 payload.
BUG=webrtc:4771
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1221643009
Cr-Commit-Position: refs/heads/master@{#9520}
ource/rtp_format_vp8.cc
ource/rtp_format_vp8_unittest.cc
|
| 2e43b26c78f465d71dfd180d55d04be1b8d4f1fb |
30-Jun-2015 |
pbos <pbos@webrtc.org> |
Prevent OOB reads in FEC packets without complete RED headers.
BUG=webrtc:4771
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1220753003
Cr-Commit-Position: refs/heads/master@{#9518}
ource/fec_receiver_impl.cc
ource/fec_receiver_unittest.cc
|
| 70d5c475ddef7ed9f848df02446d222729ed04ec |
29-Jun-2015 |
pbos <pbos@webrtc.org> |
Prevent out-of-bounds reads for short FEC packets.
BUG=webrtc:4771
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1219703002
Cr-Commit-Position: refs/heads/master@{#9514}
ource/fec_receiver_impl.cc
ource/fec_receiver_unittest.cc
|
| 0ea42d319e2a18785f5de5fe8d52e0a7a5fd1448 |
25-Jun-2015 |
Erik Språng <sprang@webrtc.org> |
Send Sdes using RtcpPacket
BUG=2450
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1196863003.
Cr-Commit-Position: refs/heads/master@{#9504}
nterface/rtp_rtcp.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| bdc0b0d869e9a14bbfafcbb84e294a13383e6fa6 |
22-Jun-2015 |
Erik Språng <sprang@webrtc.org> |
Use RtcpPacket classes for SenderReport/ReceiveReport in RTCPSender
BUG=2450
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1170723002.
Cr-Commit-Position: refs/heads/master@{#9483}
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
|
| c1b9d4e686c184e4b1779d442d447128477d3b8b |
08-Jun-2015 |
Erik Språng <sprang@webrtc.org> |
Add support for fragmentation in RtcpPacket.
If the buffer becomes full an OnPacketReady callback will be used to
send the packets created so far. On success the buffer can be reused.
The same callback will be called when the last packet has beed created.
Also made some changes to RawPacket. Buffer will now be heap-allocated
rather than (potentially) stack-allocated, but on the plus side it can
now be allocted with variable size and also avoids one memcpy.
BUG=
patch from issue 56429004 at patchset 160001 (http://crrev.com/56429004#ps160001)
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1165113002
Cr-Commit-Position: refs/heads/master@{#9390}
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtp_rtcp_impl_unittest.cc
|
| 26b08605e2b99136fcc1cab0800234f469d6e236 |
04-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
Use one scoped_refptr.
Uses webrtc/base/scoped_ref_ptr.h and removes the copy in
system_wrappers.
BUG=
R=kwiberg@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1152733005
Cr-Commit-Position: refs/heads/master@{#9370}
ource/forward_error_correction.cc
ource/forward_error_correction.h
|
| 9ba52f89acd1b9bc88115880dfe2716147bf3b5d |
01-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
Remove intermediate RTCP CNAME buffers.
Sets CNAME using a pointer to only perform a copy inside the RTCP
sender.
BUG=
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50169005
Cr-Commit-Position: refs/heads/master@{#9346}
nterface/rtp_rtcp.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| 71861a0e2039e1729ad34758474d5e569012fd2f |
28-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove GetSendSideDelay from RtpRtcp.
These stats are reported using a callback either way, removing a getter
+ an old related deadlock suppression.
BUG=1695, 2999
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50119004
Cr-Commit-Position: refs/heads/master@{#9314}
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 11beccd712dd52ae73c078332122070de3cb5c3d |
28-May-2015 |
Erik Språng <sprang@webrtc.org> |
Remove external report blocks from RtcpSender and rtp_rtcp interface.
Feature does not seem to be used and complicates other refactoring of
the rtcp module.
BUG=
R=asapersson@webrtc.org, henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54569004
Cr-Commit-Position: refs/heads/master@{#9304}
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
est/testAPI/test_api_rtcp.cc
|
| 57e5fd2e604ff7e60425c3f7654b40da03fc763c |
25-May-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
PRESUBMIT: Improve PyLint check and add GN format check.
Add pylintrc file based on
https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc
bit tightened up quite a bit (the one in depot_tools is far
more relaxed).
Remove a few excluded directories from pylint check and fixed/
suppressed all warnings generated.
Add GN format check + formatted all GN files using 'gn format'.
Cleanup redundant rules in tools/PRESUBMIT.py
TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations.
Ran it again with a modification in webrtc/build/webrtc.gni, formatted
all the GN files and ran it again.
R=henrika@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50069004
Cr-Commit-Position: refs/heads/master@{#9274}
UILD.gn
|
| 242e22b055940be70b1df3031e2363b0d02397b2 |
11-May-2015 |
Erik Språng <sprang@webrtc.org> |
Refactor RTCP sender
The main purpose of this CL is to clean up RTCPSender::PrepareRTCP, but
it has quite a few ramifications. Notable changes:
* Removed the rtcpPacketTypeFlags bit vector and don't assume
RTCPPacketType values have a single unique bit set. This will allow
making this an enum class once rtcp_receiver has been overhauled.
* Flags are now stored in a map that is a member of the class. This
meant we could remove some bool flags (eg send_remb_) which was
previously masked into rtcpPacketTypeFlags and then masked out again
when testing if a remb packet should be sent.
* Make all build methods, eg. BuildREMB(), have the same signature.
An RtcpContext struct was introduced for this purpose. This allowed
the use of a map from RTCPPacketType to method pointer. Instead of
18 consecutive if-statements, there is now a single loop.
The context class also allowed some simplifications in the build
methods themselves.
* A few minor simplifications and cleanups.
The next step is to gradually replace the builder methods with the
builders from the new RtcpPacket classes.
BUG=2450
R=asapersson@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48329004
Cr-Commit-Position: refs/heads/master@{#9166}
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/rtcp_receiver.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_utility.cc
ource/rtcp_utility.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
|
| c56ac1ec298630ba95e44a9da9efeb9d1a6d43d4 |
04-May-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
rtc::Buffer: Remove backwards compatibility band-aids
This CL makes two changes to rtc::Buffer that have had to wait for
Chromium's use of it to be modernized:
1. Change default return type of rtc::Buffer::data() from char* to
uint8_t*. uint8_t is a more natural type for bytes, and won't
accidentally convert to a string. (Chromium previously expected
the default return type to be char, which is why
rtc::Buffer::data() initially got char as default return type in
9478437f, but that's been fixed now.)
2. Stop accepting void* inputs in constructors and methods. While
this is convenient, it's also dangerous since any pointer type
will implicitly convert to void*.
(This was previously committed (9e1a6d7c) but had to be reverted
(cbf09274) because Chromium on Android wasn't quite ready for it).
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47109004
Cr-Commit-Position: refs/heads/master@{#9132}
ource/rtp_sender_unittest.cc
|
| cbf0927473c10a0a25bbf55707f1ca2b2fd57708 |
30-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
Revert "rtc::Buffer: Remove backwards compatibility band-aids"
This reverts commit 9e1a6d7c236c9a8a322bef54d4ec2a087e5baa07, because
Chromium for Android still isn't happy with it.
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49869004
Cr-Commit-Position: refs/heads/master@{#9122}
ource/rtp_sender_unittest.cc
|
| 9e1a6d7c236c9a8a322bef54d4ec2a087e5baa07 |
30-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
rtc::Buffer: Remove backwards compatibility band-aids
This CL makes two changes to rtc::Buffer that have had to wait for
Chromium's use of it to be modernized:
1. Change default return type of rtc::Buffer::data() from char* to
uint8_t*. uint8_t is a more natural type for bytes, and won't
accidentally convert to a string. (Chromium previously expected
the default return type to be char, which is why
rtc::Buffer::data() initially got char as default return type in
9478437f, but that's been fixed now.)
2. Stop accepting void* inputs in constructors and methods. While
this is convenient, it's also dangerous since any pointer type
will implicitly convert to void*.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44269004
Cr-Commit-Position: refs/heads/master@{#9121}
ource/rtp_sender_unittest.cc
|
| 97f13c5f7fcada0e419347e55449e08856d512b9 |
29-Apr-2015 |
Noah Richards <noahric@chromium.org> |
Fixed incorrect RBSP parsing. The original version would eat 0x3 as an emulation byte in places where it shouldn't, whereas the real parsing is only supposed to eat 0x3 preceded by 0x0 0x0.
Also, now that BitBuffer is getting a writer (https://webrtc-codereview.appspot.com/45259005/), I wrote a function that creates a fake SPS of a given resolution. The created SPS has an emulation 0x3 and a real 0x3, so it ensures the parser has the correct behavior.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44349004
Cr-Commit-Position: refs/heads/master@{#9108}
ource/h264_sps_parser.cc
ource/h264_sps_parser_unittest.cc
|
| 61be2a401635eed1d13c169dc104b9ff4a2f477b |
27-Apr-2015 |
Erik Språng <sprang@google.com> |
Clean up RTCPSender.
Reformat to current code style, remove non-const references, use
scoped_ptr, remove empty comments and dead code, etc..
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49019004
Cr-Commit-Position: refs/heads/master@{#9086}
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| f955b5d3f52db0f7456bd6c6bd4068d3599967da |
24-Apr-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Add h.264 AVC SPS parsing for resolution (re-land)
Re-land of noharic@'s CL at https://webrtc-codereview.appspot.com/48129004
which was reverted due to a Mac compile error which most
likely was a Goma flake (it passed on all trybots).
TBR=stefan@webrtc.org, noharic@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44329005
Cr-Commit-Position: refs/heads/master@{#9079}
UILD.gn
tp_rtcp.gypi
ource/h264_sps_parser.cc
ource/h264_sps_parser.h
ource/h264_sps_parser_unittest.cc
ource/rtp_format_h264.cc
ource/rtp_format_h264_unittest.cc
|
| e3827f27c393d31c919142928f50ce04b09636c6 |
24-Apr-2015 |
Noah Richards <noahric@chromium.org> |
Revert "Add h.264 AVC SPS parsing for resolution."
The Mac64 Debug builder is broken for an unknown failure (trybot is
green, no failure obvious in the commit break). Reverting this CL to see
if it goes green again, and then relanding to see if it is just some
weird flaky build issue.
This reverts commit 5ea8eff55ec21a1d81aaf7d29c0106fe13256150.
BUG=
TBR=rollback
Review URL: https://webrtc-codereview.appspot.com/47019004
Cr-Commit-Position: refs/heads/master@{#9074}
UILD.gn
tp_rtcp.gypi
ource/h264_sps_parser.cc
ource/h264_sps_parser.h
ource/h264_sps_parser_unittest.cc
ource/rtp_format_h264.cc
ource/rtp_format_h264_unittest.cc
|
| 5ea8eff55ec21a1d81aaf7d29c0106fe13256150 |
24-Apr-2015 |
Noah Richards <noahric@chromium.org> |
Add h.264 AVC SPS parsing for resolution.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48129004
Cr-Commit-Position: refs/heads/master@{#9073}
UILD.gn
tp_rtcp.gypi
ource/h264_sps_parser.cc
ource/h264_sps_parser.h
ource/h264_sps_parser_unittest.cc
ource/rtp_format_h264.cc
ource/rtp_format_h264_unittest.cc
|
| 9728241e6a57b0ac6c994cded1b3e87bafd241f1 |
23-Apr-2015 |
Noah Richards <noahric@chromium.org> |
Record H264 NALU type in the h264 header.
BUG=
R=niklas.enbom@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48999004
Cr-Commit-Position: refs/heads/master@{#9072}
ource/rtp_format_h264.cc
ource/rtp_format_h264_unittest.cc
|
| fe7a80c38c2cc023e5cfd96e879c98ffac68888b |
23-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Prevent sender RTCP signals for receive-only channels.
Since RTCP packets are delivered to both senders and receivers that
correspond the receivers currently log that NACKed packets are missing,
since they have no direct connection to the sending side or the RTP
packet history. Also preventing triggering on SR requests and PLI/FIR.
BUG=
R=asapersson@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45249004
Cr-Commit-Position: refs/heads/master@{#9071}
nterface/rtp_rtcp.h
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender_unittest.cc
ource/rtp_rtcp_impl.cc
|
| e62202fedf57b74cc263246c0586ee353978caf8 |
21-Apr-2015 |
Shao Changbin <changbin.shao@webrtc.org> |
Support handling multiple RTX but only generate SDP with RTX associated with VP8.
This implementation registers RTX-APT map inside RTP sender and receiver.
While it only generates SDP with RTX associated with VP8 to make it
compatible with previous Chrome versions.
Should add following changes after reaches stable,
* Use RTX-APT map for building and restoring RTP packets.
* Add RTX support for RED or VP9 in Video engine.
* Set RTX payload type for RED inside FecConfig in EndToEndTest.
BUG=4024
R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36889004
Cr-Commit-Position: refs/heads/master@{#9040}
nterface/rtp_payload_registry.h
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/nack_rtx_unittest.cc
ource/rtp_payload_registry.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
est/testAPI/test_api.cc
|
| 9478437fdea4eb31b92ffe0c10368fe5bc9b9e16 |
20-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
rtc::Buffer improvements
1. Constructors, SetData(), and AppendData() now accept uint8_t*,
int8_t*, and char*. Previously, they accepted void*, meaning that
any kind of pointer was accepted. I think requiring an explicit
cast in cases where the input array isn't already of a byte-sized
type is a better compromise between convenience and safety.
2. data() can now return a uint8_t* instead of a char*, which seems
more appropriate for a byte array, and is harder to mix up with
zero-terminated C strings. data<int8_t>() is also available so
that callers that want that type instead won't have to cast, as
is data<char>() (which remains the default until all existing
callers have been fixed).
3. Constructors, SetData(), and AppendData() now accept arrays
natively, not just decayed to pointers. The advantage of this is
that callers don't have to pass the size separately.
4. There are new constructors that allow setting size and capacity
without initializing the array. Previously, this had to be done
separately after construction.
5. Instead of TransferTo(), Buffer now supports swap(), and move
construction and assignment, and has a Pass() method that works
just like std::move(). (The Pass method is modeled after
scoped_ptr::Pass().)
R=jmarusic@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42989004
Cr-Commit-Position: refs/heads/master@{#9033}
ource/rtp_sender_unittest.cc
|
| 61c2a6f241ac9db626aeab755e49897030b289e1 |
16-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
Remove rtc::Buffer::length(), since no one uses it anymore
Chromium now uses size() instead, just like WebRTC.
This CL also fixes a new length() call that had crept in.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44119004
Cr-Commit-Position: refs/heads/master@{#9024}
ource/rtp_sender_unittest.cc
|
| 352b2d7a19d6313273608c26edf8900e592a3831 |
15-Apr-2015 |
Åsa Persson <asapersson@webrtc.org> |
Fix for sent/received RTCP packet counters returned by GetRtcpPacketTypeCounters. The returned counters are incorrect: sent_packets returns stats from a sent stream (and received_packets returns stats from a receive stream).
Add separate functions for returning stats from send/receive stream and updated how functions are used.
Add test implementation for histogram methods in system_wrappers/interface/metrics.h.
BUG=4519
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49639004
Cr-Commit-Position: refs/heads/master@{#9009}
ocks/mock_rtp_rtcp.h
|
| fcf54bdabbdf495cef7aa587b5cabdcb899ba24f |
14-Apr-2015 |
mflodman <mflodman@webrtc.org> |
Reland "Avoid critsect for protection- and qm setting callbacks in
VideoSender."
The original Cl is uploaded as patch set 1, the fix in ps#2 and I'll rebase in ps#3.
BUG=4534
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46769004
Cr-Commit-Position: refs/heads/master@{#9000}
ource/producer_fec.cc
ource/producer_fec.h
ource/producer_fec_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
|
| 6ae2572fa6dc29349b946e3cfd926289e54d9371 |
13-Apr-2015 |
Åsa Persson <asapersson@webrtc.org> |
Add missing configuration of rtx payload type for rtp/rtcp module.
BUG=4528
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51639004
Cr-Commit-Position: refs/heads/master@{#8989}
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 64c1e8cda5cb4db85c5c296bf2f6a8181af7de9d |
02-Apr-2015 |
Guo-wei Shieh <guoweis@chromium.org> |
Enable CVO by default through webrtc pipeline.
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.
Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome
BUG=4145
R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Committed: https://crrev.com/1b1c15cad16de57053bb6aa8a916079e0534bdae
Cr-Commit-Position: refs/heads/master@{#8905}
Review URL: https://webrtc-codereview.appspot.com/47399004
Cr-Commit-Position: refs/heads/master@{#8917}
ource/rtp_header_extension.cc
ource/rtp_header_extension.h
ource/rtp_header_extension_unittest.cc
ource/rtp_receiver_video.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
|
| 31331cfd2d3d17958942b67190c8b943c05b084f |
01-Apr-2015 |
Minyue <minyue@webrtc.org> |
Revert "Enable CVO by default through webrtc pipeline."
This reverts commit 1b1c15cad16de57053bb6aa8a916079e0534bdae.
Due to failure on
http://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/4092
and following builds (the test hangs and never finishes).
R=kjellander@webrtc.org
TBR=guoweis@chromium.org
TESTED=Local revert + execution of libjingle_peerconnection_java_unittest show that this is the culprit.
Review URL: https://webrtc-codereview.appspot.com/47909004
Cr-Commit-Position: refs/heads/master@{#8911}
ource/rtp_header_extension.cc
ource/rtp_header_extension.h
ource/rtp_header_extension_unittest.cc
ource/rtp_receiver_video.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
|
| 1b1c15cad16de57053bb6aa8a916079e0534bdae |
01-Apr-2015 |
Guo-wei Shieh <guoweis@chromium.org> |
Enable CVO by default through webrtc pipeline.
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.
Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome
BUG=4145
R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47399004
Cr-Commit-Position: refs/heads/master@{#8905}
ource/rtp_header_extension.cc
ource/rtp_header_extension.h
ource/rtp_header_extension_unittest.cc
ource/rtp_receiver_video.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
|
| 0828a0c09440cb7edbfacc94d362bf08414c7655 |
31-Mar-2015 |
mflodman <mflodman@webrtc.org> |
Revert "Avoid critsect for protection- and qm setting callbacks in VideoSender."
This reverts commit 903c0f2e7649a2b98659286dc228447facd49bb7,
aka #8899.
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46759004
Cr-Commit-Position: refs/heads/master@{#8901}
ource/producer_fec.cc
ource/producer_fec.h
ource/producer_fec_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
|
| 903c0f2e7649a2b98659286dc228447facd49bb7 |
31-Mar-2015 |
mflodman <mflodman@webrtc.org> |
Avoid critsect for protection- and qm setting callbacks in VideoSender.
This CL avoids changing the mentioned callbacks during a call, to avoid
a potential deadlock when acquiring _sendCritSect and calling
_mediaOpt.SetTargetRates.
Moving the critsect revealed a race for the FEC parameters in RtpVideoSender, so the CL grew a bit to avoid this. I also cleaned up some code here at the same time, but tried to keep it at a minimum since this CL had already increased a lot in size.
BUG=769
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42939004
Cr-Commit-Position: refs/heads/master@{#8899}
ource/producer_fec.cc
ource/producer_fec.h
ource/producer_fec_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
|
| 190c3ca7a9494ad0f98c0152c13d72616122a2e9 |
25-Mar-2015 |
Minyue Li <minyue@webrtc.org> |
Register sample rate of Audio RED in RTPPayloadRegistry.
Sample rate of RED payload type was not registered. And therefore VoE can fail when it receives RED packets. This is a fix to this problem.
BUG=3619
R=henrik.lundin@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43919004
Cr-Commit-Position: refs/heads/master@{#8859}
ource/rtp_payload_registry.cc
ource/rtp_payload_registry_unittest.cc
|
| eebcab5ce99d3e8641dd92a569916b0d24e29fca |
24-Mar-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
rtc::Buffer: Rename length to size, for conformance with the STL
And add a constructor for creating an uninitialized Buffer of a
specified size.
(I intend to follow up with more Buffer changes, but since it's rather
widely used, the rename is quite noisy and works better as a separate
CL.)
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48579004
Cr-Commit-Position: refs/heads/master@{#8841}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8841 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender_unittest.cc
|
| 38492c5b6fbb615159fa32b9cc24cd887295573b |
22-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Re-land 8810 "- Add a SetPriority method to ThreadWr..."
> Revert 8810 "- Add a SetPriority method to ThreadWrapper"
> Seeing if this is causing roll issues.
>
> > - Add a SetPriority method to ThreadWrapper
> > - Remove 'priority' from CreateThread and related member variables from implementations
> > - Make supplying a name for threads, non-optional
> >
> > BUG=
> > R=magjed@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/44729004
>
> TBR=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/48609004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50459005
Cr-Commit-Position: refs/heads/master@{#8819}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8819 4adac7df-926f-26a2-2b94-8c16560cd09d
est/BWEStandAlone/TestLoadGenerator.cc
est/BWEStandAlone/TestSenderReceiver.cc
|
| 90a1cb463092c5189b1a69837731a3395d79f61c |
22-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Revert 8810 "- Add a SetPriority method to ThreadWrapper"
Seeing if this is causing roll issues.
> - Add a SetPriority method to ThreadWrapper
> - Remove 'priority' from CreateThread and related member variables from implementations
> - Make supplying a name for threads, non-optional
>
> BUG=
> R=magjed@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/44729004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48609004
Cr-Commit-Position: refs/heads/master@{#8818}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8818 4adac7df-926f-26a2-2b94-8c16560cd09d
est/BWEStandAlone/TestLoadGenerator.cc
est/BWEStandAlone/TestSenderReceiver.cc
|
| b6817d793fa647ec77aaaaf74df82a94e46632bb |
20-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
- Add a SetPriority method to ThreadWrapper
- Remove 'priority' from CreateThread and related member variables from implementations
- Make supplying a name for threads, non-optional
BUG=
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44729004
Cr-Commit-Position: refs/heads/master@{#8810}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8810 4adac7df-926f-26a2-2b94-8c16560cd09d
est/BWEStandAlone/TestLoadGenerator.cc
est/BWEStandAlone/TestSenderReceiver.cc
|
| 361981faa86668cd9b20a2837d0b166fc024cd9b |
19-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Use scoped_ptr for ThreadWrapper::CreateThread.
BUG=
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45799004
Cr-Commit-Position: refs/heads/master@{#8794}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8794 4adac7df-926f-26a2-2b94-8c16560cd09d
est/BWEStandAlone/MatlabPlot.cc
est/BWEStandAlone/MatlabPlot.h
est/BWEStandAlone/TestLoadGenerator.cc
est/BWEStandAlone/TestLoadGenerator.h
est/BWEStandAlone/TestSenderReceiver.cc
est/BWEStandAlone/TestSenderReceiver.h
|
| 779c3d16b9623f38a72439bc013102aeb0077a62 |
17-Mar-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Use ByteReader/ByteWriter instead of rtputility and manual shift/add.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41289004
Cr-Commit-Position: refs/heads/master@{#8761}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8761 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/fec_receiver_impl.cc
ource/fec_test_helper.cc
ource/forward_error_correction.cc
ource/producer_fec.cc
ource/rtcp_packet.cc
ource/rtcp_sender.cc
ource/rtp_fec_unittest.cc
ource/rtp_format_h264.cc
ource/rtp_payload_registry.cc
ource/rtp_sender.cc
ource/rtp_sender_audio.cc
ource/rtp_sender_video.cc
ource/rtp_utility.cc
ource/rtp_utility.h
est/testAPI/test_api_video.cc
est/testFec/test_fec.cc
|
| 30933904797ab220a7a1532a535904f9d8ee3149 |
17-Mar-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Parsing of transport wide sequence number rtp extension header.
Plus some refactoring to correctly handle padding.
BUG=4311
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45429004
Cr-Commit-Position: refs/heads/master@{#8757}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8757 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
ource/rtp_header_extension.cc
ource/rtp_header_extension.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_utility.cc
ource/rtp_utility.h
|
| 86639737b83d8877abc4810100e30a8af863189d |
13-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove thread id from ThreadWrapper::Start().
Removes ThreadPosix::InitParams and a corresponding wait for an event.
This unblocks ThreadPosix::Start which had to wait for thread scheduling
for an event to trigger on the spawned thread, giving faster Start()
calls.
BUG=4413
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43699004
Cr-Commit-Position: refs/heads/master@{#8709}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8709 4adac7df-926f-26a2-2b94-8c16560cd09d
est/BWEStandAlone/MatlabPlot.cc
est/BWEStandAlone/TestLoadGenerator.cc
est/BWEStandAlone/TestSenderReceiver.cc
|
| fdd10579496123c9a7fdc0bf185e2a26a12ed340 |
12-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add CVO support to Vie layer.
1. standard plumbing CVO through vie layer.
2. added a rtp_cvo.h which has both conversion functions from rtp header byte to/from VideoRotation.
WebRTCVideoEngine will later pass the rotation info in SendFrame() through VieVideoFrameI420.
BUG=4145
R=mflodman@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46429007
Cr-Commit-Position: refs/heads/master@{#8703}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8703 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_cvo.h
ource/rtp_receiver_video.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| ece4b2869c356ffd74efa5ceeae982cb0ce334f4 |
10-Mar-2015 |
marpan@webrtc.org <marpan@webrtc.org> |
FecTest: Reduce loop over numMediaPackets in test_fec.
Speed up the test by factor of ~2.
TBR=pbos@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/40289004
Cr-Commit-Position: refs/heads/master@{#8676}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8676 4adac7df-926f-26a2-2b94-8c16560cd09d
est/testFec/test_fec.cc
|
| 1b32bbe0a78adfe5f2d38561ba6d90b754239cd4 |
09-Mar-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Removing private and unused method in RTPReceiver.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42269004
Cr-Commit-Position: refs/heads/master@{#8650}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8650 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
|
| 4536289353cdcc315cc5e6218893e4843cf528e6 |
04-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add CVO support to RTP sender side.
According to http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126114v120700p.pdf,
CVO byte should only be added in the last packet of each key frame or when the rotation changes. Currently, we're adding this byte in each frame to start with.
BUG=4145
R=mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42439004
Cr-Commit-Position: refs/heads/master@{#8606}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8606 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
ource/rtp_header_extension.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
ource/rtp_utility.cc
|
| 14665ff7d4024d07e58622f498b23fd980001871 |
04-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: https://chromium.googlesource.com/chromium/src/+/e144d30..6fdb142/tools/clang/scripts/update.sh
Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h
The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"`
which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override
Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h
Remaining uses of OVERRIDE was fixed by search+replace.
Manual edits were done to fix virtual destructors that were
overriding inherited ones.
Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc
This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.
BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41069004
Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
nterface/rtp_rtcp_defines.h
ource/fec_receiver_impl.h
ource/nack_rtx_unittest.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_packet.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender_unittest.cc
ource/rtp_format_h264.h
ource/rtp_format_video_generic.h
ource/rtp_format_vp8.h
ource/rtp_header_parser.cc
ource/rtp_payload_registry.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_impl.h
ource/rtp_receiver_video.h
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
est/BWEStandAlone/BWEStandAlone.cc
est/BWEStandAlone/TestSenderReceiver.h
est/testAPI/test_api.cc
est/testAPI/test_api.h
est/testAPI/test_api_audio.cc
|
| 00b8f6b3643332cce1ee711715f7fbb824d793ca |
26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36229004
Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/remote_ntp_time_estimator.h
nterface/rtp_payload_registry.h
ource/bitrate.h
ource/fec_receiver_impl.cc
ource/fec_receiver_impl.h
ource/fec_receiver_unittest.cc
ource/nack_rtx_unittest.cc
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver_help.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_format_h264_unittest.cc
ource/rtp_format_vp8_unittest.cc
ource/rtp_header_parser.cc
ource/rtp_packet_history.h
ource/rtp_payload_registry_unittest.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_impl.h
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.h
ource/rtp_sender_audio.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
est/testAPI/test_api.cc
est/testAPI/test_api.h
est/testAPI/test_api_audio.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
est/testFec/test_packet_masks_metrics.cc
|
| ac2d27d9ae74eb8d28ec0d5f12f70fa64461ab90 |
26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Fix style violations in common_types.h and config.h
Mostly, it's about moving constructors and descructors to the .cc
files, so that they won't be inlined everywhere.
The reason this CL is so big is that a lot of code was using
common_types.h without declaring a dependency on webrtc_common, which
broke the build once common_types.h started to depend on
common_types.cc.
BUG=163
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26089004
Cr-Commit-Position: refs/heads/master@{#8516}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
UILD.gn
|
| 9dd0ebc379d449d91c61884997e451d2c52a350f |
26-Feb-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Remove the default RTP module.
This CL removes the default module owned by ViEEncoder, functionality in
the module to register default modules and the final changes in
rtp_rtcp_impl using default/child modules.
BUG=769
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42509004
Cr-Commit-Position: refs/heads/master@{#8514}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8514 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| 96abda0316312183307a0c95e9417f10eab7e05b |
25-Feb-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Removing FEC functionality from the default RTP module.
This CL removes the last default module methods used from ViEEncoder and
the default module itself will be removed in a separate CL.
BUG=769
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35309004
Cr-Commit-Position: refs/heads/master@{#8505}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8505 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender_unittest.cc
ource/rtp_rtcp_impl.cc
|
| 749c60217d00a18544a72c7e9e4fe3b944dce9f2 |
25-Feb-2015 |
andresp@webrtc.org <andresp@webrtc.org> |
Moved gypi to avoid presubmit warning about '..' when touching the files.
R=kjellander@webrtc.org,mflodman@webrtc.org
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39299004
Cr-Commit-Position: refs/heads/master@{#8503}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8503 4adac7df-926f-26a2-2b94-8c16560cd09d
tp_rtcp.gypi
ource/rtp_rtcp.gypi
|
| 49096de442f6131e90925daff6dc9888d085de00 |
24-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
DCHECK send DataCountersUpdated for valid SSRCs.
Also updates RTPSender to not update RTX stats when RTX is disabled.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42399004
Cr-Commit-Position: refs/heads/master@{#8489}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8489 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
|
| d324546ced76d4e792338af4f7d02a5cd8819f92 |
23-Feb-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :
* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes
BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36179004
Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
ocks/mock_rtp_rtcp.h
ource/fec_test_helper.h
ource/rtcp_sender_unittest.cc
ource/rtp_payload_registry_unittest.cc
ource/rtp_receiver_audio.cc
ource/rtp_sender.cc
ource/rtp_utility.cc
est/testAPI/test_api_video.cc
|
| 50e28166afcdf4b2fcc6e331b70e77a284c3a560 |
23-Feb-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Move SetTargetSendBitrates logic from default module to payload router.
This cl just moves the logic form the default module
SetTargetSendBitrates to PayloadRouter. There might be glitch / mismatch
in size between trate the vector and rtp modules. This was the same in
the default module and is quite hard to protect from before we have the
new video API.
I also removed some test form rtp_rtcp_impl_unittest that were affected
by this change. The test tests code that isn't implemented, hence the
DISABLED_, and this will never be implemented in the RTP module, rather
the payload router in the future.
BUG=769
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42419004
Cr-Commit-Position: refs/heads/master@{#8453}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8453 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
|
| 1d0fa5d352fe12092201fade249905c7e1ff974b |
19-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Add RtcpPacketTypeCounter stats to new API.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788
Review URL: https://webrtc-codereview.appspot.com/37489004
Cr-Commit-Position: refs/heads/master@{#8429}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8429 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
|
| 47d657b68e753d7afb9656c1fa2f421674ed742d |
19-Feb-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Remove Set/Get sending status from the default RTP module.
This is now taken care of by the payload router and the calls to set_active.
BUG=769
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42379004
Cr-Commit-Position: refs/heads/master@{#8427}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8427 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| 0abc6011b968dab31635841cec64195441732992 |
17-Feb-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Remove SetCaptureDelay from the RTP module.
This is a small step in getting rid of the default module, but also to
eventually delete FrameProviderBase completely.
BUG=769
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34229004
Cr-Commit-Position: refs/heads/master@{#8396}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8396 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| a28a91d2f00480c998112ceb47fa2ddca1a642c4 |
17-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Fix data race for RTCPReceiver stats callback.
Annotates the callback which identifies the bug, then fixes it.
R=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/40009004
Cr-Commit-Position: refs/heads/master@{#8390}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8390 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
|
| 290cb56dcaf29dc15cc3fffc10565d2c3fb7095d |
17-Feb-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Remove TimeToSendPacket and TimeToSendPadding from the default module.
Thie CL moves the default RTP module logic for TimeToSendPacket and
TimeToSendPadding to PayloadRouter class and asserts on usage of the
default module.
BUG=769
TEST=New unittest.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33319004
Cr-Commit-Position: refs/heads/master@{#8383}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8383 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| 0200f70792982c4b5987cf4364dcd53f8aa94779 |
16-Feb-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Set webrtc_rtp category to be disabled by default.
Should not affect webrtc standalone. For chromium, disabling helps
mitigate viewing performance problems.
BUG=chromium:441440
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41909004
Cr-Commit-Position: refs/heads/master@{#8375}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8375 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver.cc
ource/rtcp_sender.cc
ource/rtp_receiver_audio.cc
ource/rtp_receiver_video.cc
ource/rtp_sender.cc
ource/rtp_sender_audio.cc
ource/rtp_sender_video.cc
|
| 2bd299a1720e93913d3e1cd5f3da81100c010d82 |
13-Feb-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Remove call to RtpRtcp::RegisterSendPayload for the default RTP module.
The send payload type is only checked in RTPSender::CheckPayloadType,
which in turn is only called from SendOutgoingData and never from the
default module anylonger.
BUG=769
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39949004
Cr-Commit-Position: refs/heads/master@{#8357}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8357 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| 7c4d20fd6c95f76cf909669b94effdbef05ecb54 |
12-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove potential deadlock in RTPSenderAudio.
Removes lock-order inversion formed by RTPSenderAudio->RTPSender calls
by doing a lot shorter locking which fetches a current state of
RTPSenderAudio variables before sending.
Thread annotates locked variables and removes one lock in
RTPSenderAudio, bonus fixes data races reported in voe_auto_test
--automated under TSan (DTMF data race).
Also includes some bonus cleanup of RTPSenderVideo which removes the
send critsect completely as all methods using it was always called
from RTPSender under its send_critsect.
R=henrik.lundin@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
BUG=3001, chromium:454654
Review URL: https://webrtc-codereview.appspot.com/41869004
Cr-Commit-Position: refs/heads/master@{#8348}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8348 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/dtmf_queue.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_audio.cc
ource/rtp_sender_audio.h
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
est/testAPI/test_api_audio.cc
|
| a4ef2ce29de0c68b869f8d66276bc5acba54cc79 |
12-Feb-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Remove getting max payload length from default module.
Moving functionality to get max payload length from default RTP module
to the payload router.
I'll make a follow up CL changing asserts to DCHECK in rtp_rtcp_impl.cc.
BUG=769
TEST=New unittest and existing sender mtu test
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36119004
Cr-Commit-Position: refs/heads/master@{#8345}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8345 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| a98e796615ddb42b07ae5513d5ed4be30ec5c556 |
11-Feb-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Remove default RTP module functionality for setting CSRC.
ViECapturer is always calling DeliverFrame with an empty CSRC vector, so
this is basically a dead path already. I added a DCHECK in ViEEncoder to
verify this is true.
BUG=769
TEST=Manually verified in Chromium.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39059004
Cr-Commit-Position: refs/heads/master@{#8335}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8335 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| 02270cd718fd2047bbbf99fbe344e3d988480b57 |
06-Feb-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Implementing a packet router class, used to route RTP packets to the
sending RTP module for the specified simulcast layer a frame belongs to.
This CL also removes the corresponding functionality from the RTP RTCP
module and fixes lint warnings in the files touched.
BUG=769
TEST=New unittest and manual tests
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39629004
Cr-Commit-Position: refs/heads/master@{#8267}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8267 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| 875c97ed9dc25a8eac8075a42742863aa1b45d3e |
04-Feb-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Remove SetNotAlive method from the thread class.
Also cleaning up methods with the same name in other classes that are derived from the above method.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41759004
Cr-Commit-Position: refs/heads/master@{#8242}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8242 4adac7df-926f-26a2-2b94-8c16560cd09d
est/BWEStandAlone/MatlabPlot.cc
est/BWEStandAlone/TestLoadGenerator.cc
est/BWEStandAlone/TestSenderReceiver.cc
|
| 4414939954fd908b6490ce1bb88271e161219aa3 |
04-Feb-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Add method for incrementing RtpPacketCounter. Removes duplicate code.
Correction to check if rtx is enabled on send-side (and not receive) when updating rtx send bitrate stat.
Remove unneeded guarded by annotations.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41729004
Cr-Commit-Position: refs/heads/master@{#8239}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8239 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/receive_statistics_impl.cc
ource/receive_statistics_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender_unittest.cc
|
| 0e81fdf5d2c2665bc3d23e07cfd9ea7f7d36aed9 |
03-Feb-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting.
BUG=chromium:82439
TEST=none
R=henrik.lundin@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40569004
Cr-Commit-Position: refs/heads/master@{#8229}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8229 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/producer_fec.cc
ource/producer_fec_unittest.cc
|
| c957ffc6dc36879e5ad72d7f0af2a014707d70fa |
02-Feb-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Fixed potential crash if rtp packet history is completely full.
Also performance enhanecement in rtp_sender (don't lookup if kDontStore)
BUG=4171
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39759004
Cr-Commit-Position: refs/heads/master@{#8226}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8226 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_packet_history.cc
ource/rtp_packet_history.h
ource/rtp_packet_history_unittest.cc
ource/rtp_sender.cc
|
| 026b892e724c3f47bde92d773d84099768e57ec8 |
30-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Using << on an int8_t or uint8_t will output a character rather than a number.
Places that do this need to cast to int to get the desired behavior.
BUG=none
TEST=none
R=henrik.lundin@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40579004
Cr-Commit-Position: refs/heads/master@{#8223}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8223 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_payload_registry.cc
ource/rtp_receiver_audio.cc
ource/rtp_receiver_impl.cc
ource/rtp_receiver_video.cc
|
| 4161715e3f7e744bc9ef3d3ae437da1e8e4de38d |
29-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Remove ChangeUniqueID.
This fixes a two year old TODO of deleting dead code :)
In cases where the _id or id_ member variable is being used for tracing,
I changed the member to at least be const.
It doesn't look like id's are that useful anymore so maybe the next step is to get rid of them.
BUG=
R=henrika@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37849004
Cr-Commit-Position: refs/heads/master@{#8201}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8201 4adac7df-926f-26a2-2b94-8c16560cd09d
ocks/mock_rtp_rtcp.h
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
|
| 43c883954f5edc84bd8e0e901ef770fead218ed5 |
29-Jan-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Allow rtp packet history to dynamically expand in size.
When using the paced sender, packets will be put into the rtp packet
history and then retreived from there again when it is time to send.
In some cases (low send bitrate and very large frames created) this
may overflow, causing packets to be overwritten in the packet history
before they have been sent.
Check this condition and expand history size if needed.
This is primarily triggered during screenshare, when
switching to a large picture with lots of high frequency
details in it.
BUG=4171
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34879004
Cr-Commit-Position: refs/heads/master@{#8195}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8195 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/nack_rtx_unittest.cc
ource/rtp_packet_history.cc
ource/rtp_packet_history.h
ource/rtp_packet_history_unittest.cc
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
|
| 37c0559c1edd108b345abcce1939f9b8d78d02a3 |
28-Jan-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Notify jitter buffer about received FEC packets (to avoid sending NACK request for these packets).
Don't copy codec specific header for empty packets in the jitter buffer.
BUG=3135
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37659004
Cr-Commit-Position: refs/heads/master@{#8184}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8184 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_packet_unittest.cc
|
| 2a6558c2a51a0aa610e85455ea7a35cfaf39bec8 |
28-Jan-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Make sure ByteReader<T>::Read* is properly constified.
Also, start using it in real code...
BUG=
R=holmer@google.com, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37809004
Cr-Commit-Position: refs/heads/master@{#8181}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8181 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/byte_io.h
ource/byte_io_unittest.cc
|
| 273fbbb921e61273c3d83eb494d0a68db7834d3d |
27-Jan-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Update StreamDataCounter with FEC bytes.
Add histograms stats for send/receive FEC bitrate:
- "WebRTC.Video.FecBitrateReceivedInKbps"
- "WebRTC.Video.FecBitrateSentInKbps"
Correct media payload bytes in StreamDataCounter to not include FEC bytes.
Fix stats for rtcp packets sent/received per minute (regression from r7910).
BUG=crbug/419657
R=holmer@google.com, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8164 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/rtp_sender.cc
|
| cfd82dfc1156f6610388bec0ebbdeacaf47e9719 |
22-Jan-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Split packets/bytes in StreamDataCounter into RtpPacketCounter struct.
Prepares for adding FEC bytes to the StreamDataCounter.
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8122 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/receive_statistics_impl.cc
ource/receive_statistics_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender_unittest.cc
|
| df7b65ba014da72fea2bbe0b6074aceaa0a51318 |
21-Jan-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Change CreateOrGetReportBlockInformation to have one return path.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8113 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
|
| 9ffd8fe96b6f7126420200ac78317756e855f1f1 |
21-Jan-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Indentation changes.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8107 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_sender.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| 9691b369951d8406b32bda2fb40667d55a3da96a |
20-Jan-2015 |
changbin.shao@intel.com <changbin.shao@intel.com> |
Cleanup for Rtp Rtcp API test.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8098 4adac7df-926f-26a2-2b94-8c16560cd09d
est/testAPI/test_api.cc
est/testAPI/test_api.h
|
| 0800db74b991dec8ef750c428eb611360a1286f4 |
15-Jan-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Add percentage of fec packets and recovered media packets to histogram stats:
- "WebRTC.Video.ReceivedFecPacketsInPercent"
- "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec"
BUG=crbug/419657
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8072 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/fec_receiver.h
ource/fec_receiver_impl.cc
ource/fec_receiver_impl.h
ource/fec_receiver_unittest.cc
|
| 2ebfac5649a5e48fbbc501b42a4336ff979c03e6 |
14-Jan-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Remove COMPILE_ASSERT and use static_assert everywhere
COMPILE_ASSERT is no longer needed now that we have C++11's
static_assert.
R=aluebs@webrtc.org, andrew@webrtc.org, hellner@chromium.org, henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8058 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_format_vp8_unittest.cc
|
| 86e1e487e73ec33177d8c03989042a31cc157575 |
14-Jan-2015 |
andresp@webrtc.org <andresp@webrtc.org> |
Move system_wrappers.gyp files to the proper directory.
Build targets should not refer to non-subpaths as was happening before when
source/system_wrappers.gyp refers to ../interface/ files.
R=kjellander@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp.gypi
est/bwe_standalone.gypi
|
| 0b0c24177bac6eaa27cd520595ba799e48e84a0c |
13-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Only return Rtx mode in RTXSendStatus().
There is no need to return 'ssrc' and 'payloadtype' inside this function
since they are never used.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38569004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8049 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/nack_rtx_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
est/testAPI/test_api.cc
|
| 3df38b442f6ba29722049b4c4d7121053003a1f8 |
13-Jan-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Unify the two copies of compile_assert.h
This patch basically deletes webrtc/base/compile_assert.h (which is
the more outdated copy) and moves
webrtc/system_wrappers/source/compile_assert.h to take its place.
R=aluebs@webrtc.org, andrew@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8048 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_format_vp8_unittest.cc
|
| 16825b1a828bb4ff40f7682040e43a239b7b8ca3 |
12-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t more consistently for times, in particular for RTT values.
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t. Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
nterface/remote_ntp_time_estimator.h
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/remote_ntp_time_estimator.cc
ource/remote_ntp_time_estimator_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_help.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_packet_history.cc
ource/rtp_packet_history.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
est/testAPI/test_api_rtcp.cc
|
| 8f27fcce79584378da97f0d84574564799e138d6 |
09-Jan-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
Revert 8028 "Support associated payload type when registering Rt..."
Reasons for revert:
1. glaznev discovered potentially related problems using the Android AppRTCDemo.
2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky.
> Support associated payload type when registering Rtx payload type.
>
> Major changes include,
> - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
> - Receiver: Restore RTP packets by the new RTX-APT map.
> - Sender: Send RTP packets by checking RTX-APT map.
> - Add RTX payload type for RED in the default codec list.
>
> BUG=4024
> R=pbos@webrtc.org, stefan@webrtc.org
> TBR=mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/26259004
>
> Patch from Changbin Shao <changbin.shao@intel.com>.
TBR=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_payload_registry.h
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/nack_rtx_unittest.cc
ource/rtp_payload_registry.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
est/testAPI/test_api.cc
|
| 2a169640a3225a559f926fe74f1fe1af239e191f |
09-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Support associated payload type when registering Rtx payload type.
Major changes include,
- Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
- Receiver: Restore RTP packets by the new RTX-APT map.
- Sender: Send RTP packets by checking RTX-APT map.
- Add RTX payload type for RED in the default codec list.
BUG=4024
R=pbos@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26259004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_payload_registry.h
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/nack_rtx_unittest.cc
ource/rtp_payload_registry.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
est/testAPI/test_api.cc
|
| 8649fed1b83882d2f25d3c58a3464a0a59a22225 |
08-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
GN: Fix Windows build.
This required a tiny include fix in
src/third_party/winsdk_samples/src
which was committed in
https://code.google.com/p/webrtc/source/detail?r=7951
This incorporates contribution from vchigrin@yandex-team.ru
in https://webrtc-codereview.appspot.com/29299004/
BUG=261,1348,4105
R=pbos@webrtc.org
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8027 4adac7df-926f-26a2-2b94-8c16560cd09d
UILD.gn
|
| d16e839c6d29831e79312180085b6a19149df43f |
19-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Rtp-Rtcp sender cleanup.
Some setter functions from Rtp and Rtcp Sender never return negative values. Remove return results from those functions.
Also removed const on non-pointer/reference types for related files.
BUG=
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34469004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7962 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/nack_rtx_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
est/testAPI/test_api.cc
est/testAPI/test_api_audio.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
|
| 11d8176cb3383a2f96e118ff054e92e97a8d9db4 |
19-Dec-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Move updating nack bitrate inside UpdateNACKBitRate.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7960 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
|
| cb79141eabdf1d2de736fd4285dc59bb44de4682 |
18-Dec-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Store the received report blocks map (mapped per remote ssrc) in a map per source ssrc.
When using rtx, receiver reports with two report blocks are received. The report blocks have the same remote ssrc and therefore the first report block was overwritten by the second report block when stored in the ReportBlockInfoMap.
Removed unused function ResetRTT.
BUG=4114
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33659005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7952 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| ce4e9a356200170abcdd44ff2af95f87a6781b8e |
18-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Refactor some receive-side stats.
Removes polling of CName as well as receive codec statistics in favor of
internal callbacks keeping a statistics struct up to date.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/28259005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| 0198933b3d49941a567f3957984a06750865d0b1 |
16-Dec-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Cleanup: Remove 'const' qualifier from OnReceivedEstimatedBitrate().
This should fix the following error I'm seeing in Win8 GN trybot:
e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\bitrate_controller\bitrate_controller_impl.cc(78)
: error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\bitrate_controller\bitrate_controller_impl.cc(30)
: warning C4373:
'webrtc::BitrateControllerImpl::RtcpBandwidthObserverImpl::OnReceivedEstimatedBitrate':
virtual function overrides 'webrtc::RtcpBandwidthObserver::OnReceivedEstimatedBitrate',
previous versions of the compiler did not override when parameters only differed by const/volatile qualifiers
e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\rtp_rtcp\interface\rtp_rtcp_defines.h(286)
: see declaration of 'webrtc::RtcpBandwidthObserver::OnReceivedEstimatedBitrate'
http://build.chromium.org/p/tryserver.chromium.win/builders/win8_chromium_gn_dbg/builds/23/steps/compile/logs/stdio
The above was triggered in CL https://codereview.chromium.org/802113002/
BUG=None
R=kjellander@google.com, kjellander@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37409004
Patch from Thiago Farina <tfarina@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7911 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
|
| d08d389ce836238030ec31e45c5f9a5535e0855f |
16-Dec-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Add field to counters for when first rtp/rtcp packet is sent/received.
Use this time for histogram statistics (send/receive bitrates, sent/received rtcp fir/nack packets/min).
BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7910 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/receive_statistics_impl.cc
ource/receive_statistics_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_sender.cc
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender_unittest.cc
|
| 0b1534c52eab79372557a6d81aaf4dd9407f55d3 |
15-Dec-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.
This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".
BUG=chromium:81439
TEST=none
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
ocks/mock_rtp_rtcp.h
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/rtp_rtcp_config.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| 97d0489058ae7a983f7306f32cfd49a2356c6488 |
09-Dec-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Add video send bitrates to histogram stats:
- total bitrate ("WebRTC.Video.BitrateSentInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateSentInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateSentInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateSentInKbps")
- retransmitted bitrate ("WebRTC.Video.RetransmittedBitrateInKbps")
Add retransmitted bytes to StreamDataCounters.
Change in UpdateRtpStats to also update counters for retransmitted packet.
BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7838 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| ba8138ba384f72674dad9b91b9a095a0fd1b27dd |
08-Dec-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Change type of nack_last_time_sent_full_ from uint32_t to int64_t.
Could cause nack requests to be sent too frequently.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7825 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
|
| 7f722492f141c1fe9a855d6ef45e9cf2dc756ab8 |
01-Dec-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Set simulcastIdx field to zero even if it has no meaning.
Helps to be able to memcmp between 2 parses of the same packet.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7773 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_format_vp8.cc
|
| d952c40c7e31c1603988c1f09ebfba9f17c6a866 |
27-Nov-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Add receive bitrates to histogram stats:
- total bitrate ("WebRTC.Video.BitrateReceivedInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateReceivedInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateReceivedInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateReceivedInKbps")
BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27189005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7756 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
nterface/rtp_payload_registry.h
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/rtp_payload_registry.cc
|
| aff1751c961c3efdae250309c6231de8925d77b0 |
24-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Add new test for VP8 packetizer to test tight partitions
It was discovered that if remaining_bytes is an exact multiple of
max_payload_len in RtpPacketizerVp8::CalcNextSize, then the packetizer
will produce too many packets (i.e., split the payload into more
packets than needed).
This CL adds a test to trigger the problem.
BUG=4019
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7739 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_format_vp8_unittest.cc
|
| 9334ac2d78f760b37f512ef6c12bff220d1654c1 |
24-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Use vector of CSRCs for DeliverFrame & SetCSRCs.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28029004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7734 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
est/testAPI/test_api.cc
est/testAPI/test_api_rtcp.cc
|
| 4591fbd09f9cb6e83433c49a12dd8524c2806502 |
20-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/fec_receiver.h
nterface/receive_statistics.h
nterface/rtp_payload_registry.h
nterface/rtp_receiver.h
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/bitrate.cc
ource/bitrate.h
ource/fec_receiver_impl.cc
ource/fec_receiver_impl.h
ource/fec_test_helper.cc
ource/fec_test_helper.h
ource/forward_error_correction.cc
ource/forward_error_correction.h
ource/nack_rtx_unittest.cc
ource/producer_fec.cc
ource/producer_fec.h
ource/producer_fec_unittest.cc
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_fec_unittest.cc
ource/rtp_format_video_generic.cc
ource/rtp_format_video_generic.h
ource/rtp_format_vp8.cc
ource/rtp_format_vp8.h
ource/rtp_format_vp8_test_helper.cc
ource/rtp_format_vp8_test_helper.h
ource/rtp_format_vp8_unittest.cc
ource/rtp_header_extension.cc
ource/rtp_header_extension.h
ource/rtp_header_extension_unittest.cc
ource/rtp_packet_history.cc
ource/rtp_packet_history.h
ource/rtp_packet_history_unittest.cc
ource/rtp_payload_registry.cc
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_impl.cc
ource/rtp_receiver_impl.h
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_audio.cc
ource/rtp_sender_audio.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
ource/rtp_utility.cc
ource/vp8_partition_aggregator.cc
ource/vp8_partition_aggregator.h
ource/vp8_partition_aggregator_unittest.cc
est/BWEStandAlone/BWEStandAlone.cc
est/BWEStandAlone/TestLoadGenerator.cc
est/BWEStandAlone/TestLoadGenerator.h
est/BWEStandAlone/TestSenderReceiver.cc
est/BWEStandAlone/TestSenderReceiver.h
est/testAPI/test_api.h
est/testAPI/test_api_audio.cc
est/testAPI/test_api_video.cc
est/testFec/test_fec.cc
|
| ece3890d3a40fe911ae895e28c329491e795b14d |
14-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Report total bitrate for all streams in GetStats.
This regression wasn't caught because I accidentally disabled multiple
streams for EndToEndTest.GetStats in a refactoring.
R=stefan@webrtc.org, xians@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/27179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtp_rtcp_impl.cc
est/testAPI/test_api_rtcp.cc
|
| 49ff40e32e408bc77e8c9bec6090f6aa2e445173 |
13-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Make SetREMBData accept vector of SSRCs.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7697 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_format_remb_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| d42a3adf429ad27779bea1789f53b76e52388583 |
07-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove partially defined WebRtcRTPHeader from Parse().
It' bit ugly that RtpDepacketizer::ParsedPayload partially defines WebRtcRTPHeader, and then sent to Parse() function for internal change.
To make it clearer, the CL gets rid of using partially-defined WebRtcRTPHeader.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28919004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7660 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_format.h
ource/rtp_format_h264.cc
ource/rtp_format_h264_unittest.cc
ource/rtp_format_video_generic.cc
ource/rtp_format_vp8.cc
ource/rtp_format_vp8_unittest.cc
ource/rtp_receiver_video.cc
|
| 0bae1fab4adb9bb8164e53142bf419049eafec38 |
05-Nov-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Wire up bandwidth stats to the new API and webrtcvideoengine2.
Adds stats to verify bandwidth and pacer stats.
BUG=1788
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| dcebf2daa76aebd021dbb778f3908375b819e59a |
04-Nov-2014 |
sprang@webrtc.org <sprang@webrtc.org> |
Reworked paced sender queue
Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage.
Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these.
Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 2dd3134e50f884f6a9e16fb643b2a8f2f6920c1d |
29-Oct-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Add stats for duplicate sent and received NACK requests.
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7559 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_utility.cc
ource/rtcp_utility.h
ource/rtcp_utility_unittest.cc
ource/rtp_rtcp_impl_unittest.cc
|
| 76960d5f742194ca2de6c900603dc72124bdcf4d |
22-Oct-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
For FIR packet, payload length is zero, so SendToNetwork function is failing.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7490 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
|
| eb24b04f1631aeee670230aa6600d28ae23890d0 |
14-Oct-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Add periodic logging of received RTP headers and estimated clock offsets for e2e delay.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7444 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/remote_ntp_time_estimator.h
ource/remote_ntp_time_estimator.cc
|
| 3cefbc99f4cc2db744cb130ca629768401a59eb4 |
10-Oct-2014 |
xians@webrtc.org <xians@webrtc.org> |
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
This also marks all virtual overrides of other classes in the same files.
This will make a subsequent change I intend to do safer, where I'll change the
argument types of the base Transport functions, by breaking the compile if I
miss any overrides.
This also highlighted a number of unused functions. I've removed some of these.
TBR=mflodman@webrtc.org, pkasting@chromium.org
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/28709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/nack_rtx_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender_unittest.cc
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender_unittest.cc
est/BWEStandAlone/TestSenderReceiver.cc
est/BWEStandAlone/TestSenderReceiver.h
est/testAPI/test_api.h
est/testAPI/test_api_audio.cc
|
| 2c0cdbce226137a8f755ae0fb51c28a335b2ea5d |
09-Oct-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Estimating NTP time with a given RTT.
RemoteNtpTimeEstimator needed user to give a remote SSRC and it intended to call RtpRtcp module to obtain RTT, to be able to calculate Ntp time.
When RTT cannot be directly obtained from the RtpRtcp module with the specified SSRC, RemoteNtpTimeEstimator would fail.
This change allows RemoteNtpTimeEstimator to calculate NTP with an external RTT estimate.
An immediate benefit is that capture_start_ntp_time_ms_ can be obtained in a Google hangout call.
BUG=
TEST=chromium + hangout call
R=stefan@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7407 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/remote_ntp_time_estimator.h
ource/remote_ntp_time_estimator.cc
ource/remote_ntp_time_estimator_unittest.cc
|
| 730d2707713c4240070af17e56edd10d039bafd2 |
29-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove callback from RtpDepacketizer::Parse().
BUG=
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30489004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7318 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_format.cc
ource/rtp_format.h
ource/rtp_format_h264.cc
ource/rtp_format_h264.h
ource/rtp_format_h264_unittest.cc
ource/rtp_format_video_generic.cc
ource/rtp_format_video_generic.h
ource/rtp_format_vp8.cc
ource/rtp_format_vp8.h
ource/rtp_format_vp8_unittest.cc
ource/rtp_receiver_video.cc
|
| f21ea918ad9e4dcbe7f372fd32d130c082641e36 |
28-Sep-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
GN: Add common configs to all targets.
This is needed to ensure we have the same build with GN
as with GYP, since GYP includes the common.gypi on a global level.
Several fixes has been needed in the past because some code have
been built without the right defines.
BUG=3441
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/28589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7317 4adac7df-926f-26a2-2b94-8c16560cd09d
UILD.gn
|
| 315669939afc8461b40612c905eaec95c2ee645d |
25-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Fix typo from RtpPacketizerH264.
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27609004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7295 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_format_h264.h
|
| 38344ed2806c8fed60d67d280ca44c32e36707c0 |
24-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Move thread_annotations.h to webrtc/base/.
R=andresp@webrtc.org, mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_sender.h
ource/rtp_packet_history.h
ource/rtp_sender.h
|
| 5a098c51ea75ea08921bfef634c59336eaae4edf |
17-Sep-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Refactor VP8 de-packetizer.
It's duplicated to parse VP8 RTP packet at the moment. We firstly call
RTPPayloadParser functions to save parsed information in RTPPayload
structure, then copy them to RTP header.
This CL removes RTPPayloadParser class and directly saves parsed data in
RTP header.
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7211 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_format_vp8.cc
ource/rtp_format_vp8_unittest.cc
ource/rtp_receiver_impl.cc
ource/rtp_utility.cc
ource/rtp_utility.h
ource/rtp_utility_unittest.cc
|
| dae612ebf8044be2eccda45053805cc7289f8106 |
16-Sep-2014 |
henrikg@webrtc.org <henrikg@webrtc.org> |
Mark all virtual overrides in the hierarchies of UdpTransportData and
UdpSocketWrapper as such.
This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.
This also removes an unused function.
BUG=none
TEST=none
R=henrik.lundin@webrtc.org, henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7195 4adac7df-926f-26a2-2b94-8c16560cd09d
est/BWEStandAlone/BWEStandAlone.cc
est/BWEStandAlone/TestSenderReceiver.h
|
| 1fb5d1204b4378f45d13e200a1900b4a7e8b385a |
12-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Initialize restored_packet in nack_rtx_unittest.cc.
This is to get the DrMemory Full bots to go green, this was previously
suppressed. This fix is likely hiding a real bug that should be
investigated, but it's not a regression from before. The issue should
not be closed before we figure out why this is the case and revert this
"fix".
TBR=stefan@webrtc.org
BUG=3183
Review URL: https://webrtc-codereview.appspot.com/30369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7169 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/nack_rtx_unittest.cc
|
| b5e6bfc76a32a588da2400636688d34a71a2f47d |
12-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Make RTPSender/RTPReceiver generic.
Changes include,
1) Introduce class RtpPacketizerGeneric & RtpDePacketizerGeneric.
2) Introduce class RtpDepacketizerVp8.
3) Make RTPSenderVideo::SendH264 generic and used by all packetizers.
4) Move codec specific functions from RTPSenderVideo/RTPReceiverVideo to
RtpPacketizer/RtpDePacketizer sub-classes.
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26399004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7163 4adac7df-926f-26a2-2b94-8c16560cd09d
UILD.gn
nterface/rtp_rtcp_defines.h
ource/rtp_format.cc
ource/rtp_format.h
ource/rtp_format_h264.cc
ource/rtp_format_h264.h
ource/rtp_format_h264_unittest.cc
ource/rtp_format_video_generic.cc
ource/rtp_format_video_generic.h
ource/rtp_format_vp8.cc
ource/rtp_format_vp8.h
ource/rtp_format_vp8_unittest.cc
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp.gypi
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
|
| 6071b0636df9072206790c650bf6d07e709aca15 |
12-Sep-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such.
This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.
This also highlighted a number of unused functions which I've removed.
-- This is was reviewed in https://webrtc-codereview.appspot.com/19309004/, but
-- a new cl was needed to resolve a small conflict before committing.
BUG=none
TEST=none
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7162 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
ource/rtp_receiver_impl.cc
ource/rtp_receiver_impl.h
|
| 1972ff8a6e45f7ad3fb7e4ed51dc0135c72f6c9d |
11-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.
This will make a subsequent change I intend to do safer, where I'll change the
return type of one of the base Module functions, by breaking the compile if I
miss any overrides.
This also highlighted a number of unused functions (in many cases apparently
virtual "overrides" of no-longer-existent base functions). I've removed some of
these.
This also highlighted several cases where "virtual" was used unnecessarily to
mark a function that was only defined in one class. Removed "virtual" in those
cases.
BUG=none
TEST=none
R=andrew@webrtc.org, henrik.lundin@webrtc.org, mallinath@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7146 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_receiver_impl.cc
ource/rtp_receiver_impl.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| c30e9e230065ddde4cc439d9ba430273413e70d7 |
08-Sep-2014 |
sprang@webrtc.org <sprang@webrtc.org> |
Ignore FEC packet in stats, if it is first packet on ssrc.
BUG=chrome:410456
R=mflodman@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7096 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
|
| 0a214ffa8ad5c2d52d0f2d20bf5f1d686994f552 |
03-Sep-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Setting marker bit on DTMF correctly
BUG=1157
R=braveyao@webrtc.org, pbos@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7037 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender_audio.cc
ource/rtp_sender_unittest.cc
|
| f8723d666a74506d66ef91ba916c93437125e3a9 |
28-Aug-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Add unit tests to rtcp_receiver_test.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6994 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver_unittest.cc
|
| c3c29113d1733a4f97ca4b8e212f22f718a876b7 |
27-Aug-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Expose setPayloadType on the rtp_sender. Thus allowing other users of this module
to set the payload type to be used without having to call SendOutgoingData.
BUG=3694
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6988 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 047a46f8b49e7100d7727377c89f109542125b9c |
26-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove Android.mk build files.
These files are generally not maintained and break, some contain files
that don't exist anymore and do not build anymore. If we need to add
some of these back we should really set up a bot for them.
R=andrew@webrtc.org, glaznev@webrtc.org, henrike@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/15249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/Android.mk
|
| b96ea2aab5a5b2f170f374427f22159048bd1c1e |
26-Aug-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Remove former team members from OWNERS and WATCHLISTS
Remove the following (CCed) former team members from all
OWNERS files and the WATCHLISTS file:
* fischman@
* leozwang@
* mikhal@
* pwestin@
* wu@
BUG=
R=henrike@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6973 4adac7df-926f-26a2-2b94-8c16560cd09d
WNERS
|
| 42ee5b54b59d766066bac540c3e8ddf7d49b649f |
25-Aug-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
GN: Disable Chromium clang plugins for standalone build.
Now that WebRTC has rolled the chromium_revision past
http://crrev.com/284372 in r6784, clang has become the
default compiler. Since WebRTC standalone code doesn't
yet compile the Chromium Clang plugins enabled, this CL
disables them for the parts of the code that doesn't yet pass
compilation with them enabled.
The buildbots are using Goma which is not yet switched
over to Clang by default. That's why they're not red yet.
BUG=163
TEST=Passing compile locally on Linux using:
gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja
-C out/Debug
gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja
-C out/Release
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/16279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
UILD.gn
|
| a84b0a6dabdf5c0c6f120bd72ad15653a0d3ddcf |
14-Aug-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Small refactor on ViE to remove redudant conditions and long ifdefs.
BUG=3694
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6905 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| 58e2d262fc6a67d069f6c5b8c5043748570521f9 |
14-Aug-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics().
Fixes issues where statistics only was reported for the first stream if
configured with simulcast, and in case of RTX the reported statistics was
depending on the order of the report blocks.
Also fixes issues with multiple report blocks in the SendStatisticsProxy and the
RtcpStatisticsCallback. SendStatisticsProxy is now aware of RTX ssrcs, and the
RTCPReceiver is calling the RtcpStatisticsCallback with the correct SSRCs, and
not only the primary stream SSRC.
R=mflodman@webrtc.org, sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6903 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver.cc
|
| 817a034cf25ea2232c54ac2f3afcffe85bd50c47 |
14-Aug-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix TimeToSendPadding return to be 0 if no padding bytes are sent.
BUG=3694
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15149005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6900 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
|
| c27543d297c2aff191605f892cad573ea5c25305 |
13-Aug-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix STAP-A bug where we might overflow the packet buffer due to not accounting for the length of the length field.
BUG=3679
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6881 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_format_h264.cc
ource/rtp_format_h264_unittest.cc
|
| c891fee7aba4e6bcc33f6e03ec9e7f3a2940e03c |
13-Aug-2014 |
fbarchard@google.com <fbarchard@google.com@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make a int64 constant use ULL suffix so it wont get truncated.
BUG=3690
TESTED=try bots
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6878 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_packet.cc
|
| 59a2f9f5848057db42ba8c782cc9b4854762a16b |
07-Aug-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the old H264 code now that a new H.264 packetizer has been implemented.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6847 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/H264/bitstream_builder.cc
ource/H264/bitstream_builder.h
ource/H264/bitstream_parser.cc
ource/H264/bitstream_parser.h
ource/H264/h264_information.cc
ource/H264/h264_information.h
ource/H264/rtp_sender_h264.cc
ource/H264/rtp_sender_h264.h
|
| 9d74f7ce8c02deb7bfea8194a7e211384cf0f2d3 |
07-Aug-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix single nalu packetization bug.
Nalus which had the same size as the max payload size would cause the payload size accounting to wrap.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6846 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_format_h264.cc
ource/rtp_format_h264_unittest.cc
|
| 8b033adb19f8be63603f8b9b79082ac952d01a2e |
06-Aug-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Change the way we reference enumerators in H.264 packetization code to be standard C++ compliant.
R=kjellander@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6833 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_format_h264.cc
ource/rtp_format_h264_unittest.cc
|
| 84b9e1e9d9407c48ff85a28f0825fe3a23a1f614 |
04-Aug-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix for retransmission. Base layer packets were not retransmitted.
Issue introduced in r6669.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6816 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender_video.cc
|
| e1c9caf6eee2d97824d2ecae75dbd5aae2f0a3b4 |
31-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804.
TEST=buildtools/linux64/gn gen out/Default
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6805 4adac7df-926f-26a2-2b94-8c16560cd09d
UILD.gn
|
| 2ec560606be6519dc4e32a1e6855b0f362ca498d |
31-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add H.264 packetization.
This also includes:
- Creating new packetizer and depacketizer interfaces.
- Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition.
- Created a Create() factory method for packetizers and depacketizers.
R=niklas.enbom@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
UILD.gn
ocks/mock_rtp_rtcp.h
ource/fec_receiver_unittest.cc
ource/rtp_format.cc
ource/rtp_format.h
ource/rtp_format_h264.cc
ource/rtp_format_h264.h
ource/rtp_format_h264_unittest.cc
ource/rtp_format_vp8.cc
ource/rtp_format_vp8.h
ource/rtp_format_vp8_test_helper.cc
ource/rtp_format_vp8_test_helper.h
ource/rtp_format_vp8_unittest.cc
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp.gypi
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
ource/rtp_utility_unittest.cc
|
| e75d78d32d7283adc53cd91a85094245a7428d84 |
29-Jul-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Integrate rtcp packet class to rtcp receiver tests.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6795 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver_unittest.cc
|
| f9460688a61ccac0067feef07192e05a44e5d7e3 |
24-Jul-2014 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make sure padding is sent on the first sending RTP module.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6774 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl_unittest.cc
|
| 5ab7616983d8db80a52aa347114642c94c71a19e |
22-Jul-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove remains of WEBRTC_NO_STL.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6752 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/ssrc_database.cc
ource/ssrc_database.h
|
| 8b94e3da0f35638529d6640e4dfcd7f04057d3f4 |
17-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled.
This broke bandwidth estimation for calls without abs-send-time is enabled, but where RTX was.
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6719 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
ource/rtp_header_extension.cc
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 63c60ed22457d45444d29b33a622ea2bedd12ea5 |
16-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove old padding path in RTPSender.
Removing RTPSender::SendPaddingAccordingToBitrate() as well as a couple
of arguments from SendPadData().
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6703 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 376b4ea93f439d85754c081650710ce1265d9cd4 |
15-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix breakage introduced by r6691.
ModuleRtpRtcpImpl returned incorrectly on RemoteNTP as the
RTCPReceiver::NTP changed return type.
BUG=
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6693 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| 2f4b14e3f31b34a50310357c6c7be86c3bca1537 |
15-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make RTCP sender report send media bytes.
r6654 changed RtpSender::Bytes() to return the number of bytes sent
instead of number of media bytes. This is used by VideoEngine for stats.
This change broke RTCP which sends this same count as the number of
payload bytes sent (excluding headers and padding).
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6691 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| 9e1acc872859ffd6dc2827af81a9446b50a9a53f |
11-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 .
A few places were relying on temporalIdx being signed. Fix to explicitly check
for kNoTemporalIdx.
TBR=pbos,stefan
Review URL: https://webrtc-codereview.appspot.com/13939005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6669 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender_video.cc
|
| dd6780d85d2491a4cabc81e737d503d7d879a2b9 |
11-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove always-true expression.
TBR=pbos
Review URL: https://webrtc-codereview.appspot.com/16059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6668 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_format_vp8.cc
|
| eec6ecdb1e249871dd25d04b62fc9ddc03dc8a34 |
11-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.
---
Fixes for re-enabling more MSVC level 4 warnings: webrtc/ edition
This contains fixes for the following sorts of issues:
* Possibly-uninitialized local variable
* Signedness mismatch
* Assignment inside conditional
This also contains a small number of other cleanups to nearby code. In
particular several warning-disables for MSVC are removed because they don't seem
to be necessary (either that warning is not enabled or the code does not trigger
it).
BUG=crbug.com/81439
TEST=none
R=henrika@webrtc.org, pkasting@chromium.org
Review URL: https://webrtc-codereview.appspot.com/18769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6667 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
ource/tmmbr_help.cc
|
| 180e516bef1f2929ef22bc7324861cfe18227ed2 |
11-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Thread annotate RTCPSender.
Also fixes data races in RTCPSender::SetCSRCStatus() and
RTCPSender::SetStartTimestamp().
BUG=
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6666 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_format_remb_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_rtcp_impl.cc
|
| 168f23faa5b8a49d4dd709c6649e77d5fecf36bf |
11-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21869005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| 4ef438e2defd6c46404f6b367287364cde66b7fb |
11-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the send-side cname getter APIs from voice and video engine.
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
est/testAPI/test_api.cc
|
| 72491b9a90bfd4e2339f42e353560c9c33875151 |
10-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Count total bytes sent in RTPSender::Bytes().
Previously only media bytes were included, this adds header bytes and
padding bytes to the calculation.
BUG=
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6654 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| 8f1512140ed57ce57635a1cd561b631dfdc5e05f |
10-Jul-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6649 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| 5ac876bae08611e3f4f75d12eb9d33b825a76453 |
10-Jul-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Remove remains of WEBRTC_NO_STL." (rev 6641).
Reason breaks linux_memcheck.
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6645 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/ssrc_database.cc
ource/ssrc_database.h
|
| 47d1c98a4ee728b83dbb105522a3720ae70dfa24 |
09-Jul-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove remains of WEBRTC_NO_STL.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6641 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/ssrc_database.cc
ource/ssrc_database.h
|
| d11bec40b25e5990bf05b410676587f6f38b9b8c |
08-Jul-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6626 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| 62bafae6618fe3aefbd18657062abc98a40c3375 |
08-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Some refactoring inside rtp_rtcp/.
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.
BUG=
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_header_parser.h
nterface/rtp_payload_registry.h
ource/H264/rtp_sender_h264.cc
ource/fec_receiver_impl.cc
ource/fec_test_helper.cc
ource/forward_error_correction.cc
ource/mock/mock_rtp_payload_strategy.h
ource/producer_fec.cc
ource/receive_statistics_impl.cc
ource/rtcp_packet.cc
ource/rtcp_sender.cc
ource/rtp_fec_unittest.cc
ource/rtp_header_parser.cc
ource/rtp_payload_registry.cc
ource/rtp_payload_registry_unittest.cc
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_impl.cc
ource/rtp_receiver_video.cc
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_audio.cc
ource/rtp_sender_audio.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
ource/rtp_utility.cc
ource/rtp_utility.h
ource/rtp_utility_unittest.cc
est/testAPI/test_api.h
est/testAPI/test_api_video.cc
est/testFec/test_fec.cc
|
| 2bb1bdab8d11f5445693c028335fb3ace631f636 |
07-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Preserve RTP states for restarted VideoSendStreams.
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.
Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/rtcp_receiver.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
|
| b9f5453e2997253addb87706a43b4484e1139972 |
04-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add boilerplate code for H.264.
R=mflodman@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17849005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_payload_registry.cc
ource/rtp_receiver_video.cc
ource/rtp_sender_video.cc
|
| 88e0dda475e1f6a5fa5855eec0be111bddbf00ac |
04-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Introduces PacedVideoSender to test framework and moves the Pacer to use Clock.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6600 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
|
| 720964faac9250618a12c3fbf6af74bf92d534ba |
03-Jul-2014 |
marpan@webrtc.org <marpan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix memcheck error in r6594.
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6596 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_fec_unittest.cc
|
| c8364539d30d68fb3b12bfcddfa65aa0d91a19e3 |
03-Jul-2014 |
marpan@webrtc.org <marpan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix for FEC decoding with sequence number wrap-around.
BUG=3507
R=stefan@webrtc.org
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6594 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/forward_error_correction.cc
ource/rtp_fec_unittest.cc
est/testFec/test_fec.cc
|
| aa0e56e8e8384dea0a2dea2945d019777371577f |
26-Jun-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes a bug causing NACKs to be dropped excessively at the send-side.
This was introduced in r6472 where the target bitrate was changed to be stored in bits/s instead of kbits/s, but the storage type was accidentally left as uint16_t. This caused the bitrate to be truncated, which at times causes NACKs to be dropped due to insufficient bitrate available.
BUG=3518
TEST=Tested in Chrome, trybots and verified that it fixes the bug in vie_auto_test loopback test.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6544 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.h
|
| fe526ff10fea5cc9f456f9a9313499f19bd7c8d0 |
25-Jun-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
fix after r6472 in rtp_sender, comparison between signed and unsigned integer expressions.
BUG=N/A
R=pwestin@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6539 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
|
| 3b84b3a58cf4093204749fa7ba782f49b8934246 |
25-Jun-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add RTCP packet types to packet builder:
REMB, TMMBR, TMMBN and
extended reports: RRTR, DLRR, VoIP metric.
BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9299005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6537 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet_unittest.cc
|
| 1227ab89a7c08e4e5af051a63daba889ea0d2da7 |
23-Jun-2014 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
GN: Add BUILD.gn files + kjellander to OWNERS
This should work as a foundation for all the work that is
left to do to make the parts of WebRTC that Chromium uses
to build with GN.
I implemented some the smaller modules myself in this CL.
The remaining work (TODO's in the .gn files) will be distributed
to various team members.
I'm adding myself to OWNERS files for BUILD.gn files in all the
directories where I'm adding a BUILD.gn file.
BUG=3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default
I built successfully from a Chromium checkout (with
https://codereview.chromium.org/321313006/ applied) using:
gn gen out/Default && ninja -C out/Default webrtc
R=brettw@chromium.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
UILD.gn
WNERS
|
| a15fbfdcdee391bd87bb1c2721f0fbb824f5fbfb |
17-Jun-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add round-robin selection of send stream to pad on.
BUG=1812
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6472 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| 4b12d400089f324293b8c313ba8257d9247e9cc6 |
16-Jun-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class.
BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6449 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet_unittest.cc
|
| ef92755780253c6a7940c89598a206e58e05b810 |
05-Jun-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC.
This makes it easier to disable RTX by filtering out the RTX codec during call setup/signaling, and won't require that also the SSRCs are filtered out.
BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15629005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6335 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_payload_registry.h
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/nack_rtx_unittest.cc
ource/rtp_payload_registry.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
est/testAPI/test_api.cc
est/testAPI/test_api_audio.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
|
| 4436b4436adfb608bb4e62e67906b9f5e72b7c7f |
04-Jun-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Moved verbose logging in rtcp_receiver.cc to LS_VERBOSE.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6324 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver.cc
|
| 420b2567f38241099907d30d8bece1c4db50262d |
30-May-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix bug where RTP headers in the packet history were replaced with the RTX wrapped headers.
This caused only the first retransmission to be successful.
Introduced with https://code.google.com/p/webrtc/source/detail?r=5728.
BUG=1811
R=asapersson@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12609005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6284 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_packet_history.cc
ource/rtp_packet_history.h
ource/rtp_packet_history_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 88fbb2d86b33a3886bba1af4d098efa2c19eb1e7 |
21-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
Same as https://webrtc-codereview.appspot.com/19519004. The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing...
(tested locally).
BUG=3380
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver_help.h
ource/rtp_format_vp8.h
ource/rtp_format_vp8_test_helper.h
ource/vp8_partition_aggregator.h
|
| 2fa7f79094bfa283e0ff2b086be511e65c24c69e |
21-May-2014 |
mcasas@webrtc.org <mcasas@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6202 "Switch to using base/constructormagic.h and remove ..."
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
>
> BUG=N/A
> R=andrew@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/19519004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14579007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver_help.h
ource/rtp_format_vp8.h
ource/rtp_format_vp8_test_helper.h
ource/vp8_partition_aggregator.h
|
| 125ffd709dee39214e54d80fb277da66adc9ebd3 |
20-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver_help.h
ource/rtp_format_vp8.h
ource/rtp_format_vp8_test_helper.h
ource/vp8_partition_aggregator.h
|
| a826006132b3606b7325befcbffd608df6714f6c |
20-May-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add NACK and RPSI packet types to RTCP packet builder.
Fixes bug found when parsing received RPSI packet.
BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6194 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet_unittest.cc
ource/rtcp_utility.cc
|
| 88abf11cadc0eb8986561a942ecc13ad9a324f16 |
14-May-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe.
BUG=3111
TEST=try bots
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6152 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/remote_ntp_time_estimator.h
ource/remote_ntp_time_estimator.cc
ource/remote_ntp_time_estimator_unittest.cc
ource/rtp_rtcp.gypi
|
| 8f69330310bf786cff373c225967e7459fb0b560 |
26-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Replace scoped_array<T> with scoped_ptr<T[]>.
scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar...
except for the few not-built-on-Linux files which were updated manually.
TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
est/testFec/test_packet_masks_metrics.cc
|
| cd70119a106e42ab7eac58050d067bc050610739 |
25-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase.
BUG=3111
TEST=new performance tests
R=niklas.enbom@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5976 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| 93fd25c20c688961569d3631b875c8ee0dfc2a80 |
24-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.
* Cast rtp header extension to int in log in rtp_utility.cc.
BUG=3237
TEST=try bots
R=stefan@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5975 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_utility.cc
|
| 2c3f1abb69376e66cf15e5fb6fe5bcd88f185aca |
15-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Replace flooding logs in rtp_sender.cc with a comment.
Started occurring after:
https://webrtc-codereview.appspot.com/11129004
BUG=3153
R=andresp@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5916 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
|
| 2f8d5f330279f42ac79174dbbc2e4722f5cf535e |
15-Apr-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Check if a header extension is registered before updating it and fail silently if it's not.
BUG=
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5909 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_header_extension.h
ource/rtp_sender.cc
|
| 2c89b5cb27536eac2ca298c4a36f3a5ccb903141 |
14-Apr-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done
(and then removed the talk/ impact)
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/OWNERS
est/OWNERS
est/testFec/OWNERS
|
| dc80bae2a62a1bdbe0d342b3260a7e5b2cb958df |
08-Apr-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
Clean some logs and add asserts in the way.
BUG=3153
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5861 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/fec_receiver.h
nterface/rtp_payload_registry.h
ource/fec_receiver_impl.cc
ource/fec_receiver_impl.h
ource/fec_receiver_unittest.cc
ource/forward_error_correction.cc
ource/forward_error_correction.h
ource/nack_rtx_unittest.cc
ource/producer_fec_unittest.cc
ource/rtcp_packet.cc
ource/rtcp_receiver.cc
ource/rtcp_sender.cc
ource/rtcp_sender_unittest.cc
ource/rtp_fec_unittest.cc
ource/rtp_header_parser.cc
ource/rtp_packet_history.cc
ource/rtp_packet_history.h
ource/rtp_packet_history_unittest.cc
ource/rtp_payload_registry.cc
ource/rtp_payload_registry_unittest.cc
ource/rtp_receiver_audio.cc
ource/rtp_receiver_impl.cc
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
ource/rtp_utility.cc
ource/rtp_utility.h
ource/rtp_utility_unittest.cc
ource/ssrc_database.cc
est/testAPI/test_api.cc
est/testAPI/test_api_audio.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
est/testFec/test_fec.cc
|
| d09d0748270f40c35330837069523245839b7258 |
26-Mar-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Protect write of send_target_bitrate.
This issue was catch by tsan bot.
BUG=3065
R=stefan@webrtc.org, andrew
Review URL: https://webrtc-codereview.appspot.com/10619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5790 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 440fa235539cfbf1819f2366c488f587be80caae |
25-Mar-2014 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out.
BUG=2954
R=mflodman@webrtc.org, stefan@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5786 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_utility.cc
|
| efcad39f778276296ef45e2f14427154841e911f |
25-Mar-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix race condition in RTPSEnder.
In RTPSender::SendPayloadType(), payload_type_ should not be read
without owning send_critsect_.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5778 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 9d4762e8b65b6694d06220c2a34b8b953c53c3c5 |
24-Mar-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Have changes to REMB trigger RTCP to be sent immediately.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5763 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_sender.cc
|
| b1f50100757036cf475072c26f5f374eee9588ca |
24-Mar-2014 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
VoE changes to allow forwarding of packets from VoE to ViE BWE.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5757 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl_unittest.cc
|
| af839b28b073be3c58a76433d7a4d96013e869f3 |
24-Mar-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add AIMD option to BWE API.
TEST=trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10319005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5755 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender_unittest.cc
|
| 16395228f5a6ae6f5d4f85441873d8408f5c11d6 |
19-Mar-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Properly account for retransmitted packets when not using the pacer.
This regression was introduced in r5728.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5729 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
|
| 7c6ff2da261699628e7253d9d10068bc531fe0f8 |
19-Mar-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes RTX related bugs.
- An RTX packet with no payload should be dropped prior to parsing RTX header since it doesn't have an RTX header. This can for example happen when sending padding-only packets over the RTX stream.
- The retransmit code path when the pacer is disabled doesn't properly update the abs-send-time and ts-offset header extensions.
TEST=trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5728 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 5a320fb06fadb8378b76112556473af7b9e0c82a |
13-Mar-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Race condition in RTPSender
RTPSender::sending_media_ should be guarded by send_critsect_. Fix this
in GetSendSideDelay, SendPadData and TimeToSendPadding.
Also add appropriate thread annotations.
BUG=3029
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5697 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
|
| ebdb0e3ad0a787bee066d12cdcd115a38b0a10d1 |
07-Mar-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005.
- Add ability to VoE to send Absolute Sender Time header extension.
- Refactor handling of RTP header extensions in VoE to work the same as in ViE.
- Add API to enable receiving Absolute Sender Time in VoE.
This is part of the work to include audio packets in bandwidth estimation, for
better accuracy in estimates.
BUG=
TBR=solenberg@webrtc.org,henrikg@webrtc.org,stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ource/rtp_header_extension.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_audio.cc
ource/rtp_sender_audio.h
ource/rtp_sender_unittest.cc
ource/rtp_utility.cc
|
| 0f2809a5ac5477a6134ebafb4680597252f8a5c5 |
21-Feb-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add RTCP packet class.
Adds packet types: sr, rr, bye, fir.
BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5592 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_packet.cc
ource/rtcp_packet.h
ource/rtcp_packet_unittest.cc
ource/rtp_rtcp.gypi
|
| 8098e0747879b191335e8de1e16b87cf6adbdf54 |
19-Feb-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR).
Add counter to RTCP sender and RTCP receiver.
Add video api GetRtcpPacketTypes().
BUG=2638
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5575 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
|
| c320027d6a86abb620f25a2248484b4bcc23a193 |
18-Feb-2014 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Don't print a warning if RTPPacketHistory::SetStorePacketStatus is called
twice with the same settings.
Without this change, setting up a call with the new video API will
print a trace warning.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5566 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_packet_history.cc
|
| 346094cb01ef2ffbf0398f465d61c9a4f77b465c |
18-Feb-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Incorrect overhead calculation when using FEC + RTP extension headers.
When frames are fragmented inte multiple RTP packets in order to not
exceed a maximum packet size, the header overhead calculation must
take into account that FEC redundancy packets may use more than the
12 bytes of the basic RTP header. For example, a csrc list or extension
headers may be present.
BUG=2899
R=phoglund@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8769005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5562 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
ource/fec_test_helper.h
ource/forward_error_correction.cc
ource/rtp_sender.cc
ource/rtp_sender_video.cc
|
| 0e5a2b5de6feffde32ced00c923c25c3bca3a278 |
04-Feb-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Handle the invalid case of setting multiple stream_bitrates if there is only a single send stream registered.
This can happen when switching between multiple streams and a single while getting feedback from the receiver.
BUG=2881
TEST=trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5486 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| a45cac0fb79782fd4bfe9c6ef1e1a74074a33aee |
27-Jan-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Avoid potential dead lock in StreamStatisticianImpl
Extract callbacks for rtp/rtcp data, from StreamStatisticianImpl to
ReceiveStatisticsImpl, into separate methods with guards agains having
incorrect lock order.
BUG=2856
R=andresp@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5441 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
|
| 5314e859262c03e8c212fee53245e91851c1e5cc |
27-Jan-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Race condition in RTPSender::UpdateRtpStats
The ssrc should not be access directly from the ssrc_ field, without
holding the send_critsect_ lock. A better way is to just use the SSRC()
getter method.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7539006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5439 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
|
| c00adbed7388c7c3a2e6214e6ab06242997e1825 |
27-Jan-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Race in StreamStatisticianImpl::GetStatistics vs. ::IncomingPacket
StreamStatisticianImpl.ssrc_ is protected by stream_lock_, should
be cached while holding lock to avoid race condition.
Also, rtp_callback_ do not need to be called in GetStatistics() at all
BUG=2853
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5435 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
|
| 871d949299a4ec27efe9805ad5c2289e7e2f68b3 |
24-Jan-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtcp modules.
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5426 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl_unittest.cc
|
| 0e93257cee79c0d19ddaef1f14ba750bf469a084 |
23-Jan-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add callbacks for receive channel RTP statistics
This allows a listener to receive new statistics (byte/packet counts, etc) as it
is generated - avoiding the need to poll. This also makes handling stats from
multiple RTP streams more tractable. The change is primarily targeted at the new
video engine API.
TEST=Unit test in ReceiveStatisticsTest.
Integration tests to follow as call tests when fully wired up.
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5416 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
|
| 7dba27c740048c92692d4e1cf6fee1fee7827901 |
21-Jan-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Potential dead lock in receive statistics
A dead lock could occur if the following to code paths are called
concurrently:
ReceiveStatisticsImpl::IncomingPacket() ->
StreamStatisticianImpl::IncomingPacket()
StreamStatisticianImpl::GetStatistics() ->
ReceiveStatisticsImpl::StatisticsUpdated()
Solution is to release ReceiveStatisticsImpl lock after lookup/lazy-init of StreamStatisticianImpl. Don't need to hold it when doing the call to StreamStatisticianImpl::IncomingPacket().
BUG=2818
R=asapersson@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5406 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
|
| efaeda0c76fbf9a58c44931d525348ab59dd52b0 |
20-Jan-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add configuration and test for extended RTCP reference time reports to new video api.
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5401 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_sender.cc
|
| 2fb72cfeecbfe7660767556a5128d21dba94c922 |
24-Dec-2013 |
braveyao@webrtc.org <braveyao@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add include guards to forward_error_correction_internal.h
R=henrika@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5789005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5335 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/forward_error_correction_internal.h
|
| 7fb75ecbd4226ca3fccdb7e64ce19850059c8c13 |
20-Dec-2013 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add thread_annotations for clang targets.
TESTED: As expected clang bots catched a few issues which are fixed with this CL, other bots ignore the annotations and compile fine.
R=niklas.enbom@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5328 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/fec_receiver_impl.cc
ource/rtcp_receiver.cc
|
| 54ae4ffb9e235a9742e2b11298327e02d870571c |
19-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add callbacks for receive channel RTCP statistics.
This allows a listener to receive new statistics as it is generated - avoiding the need to poll. This also makes handling stats from multiple RTP streams more tractable.
The change is primarily targeted at the new video engine API.
TEST=Unit test in ReceiveStatisticsTest. Integration tests to follow as call tests when fully wired up.
BUG=2235
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5323 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/rtcp_sender.cc
est/testAPI/test_api_rtcp.cc
|
| e6b871bb29dcdd9da08509ab3a39d90424f73781 |
17-Dec-2013 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Added method for getting default module state and protect agains a
read/write race for child_modules_.
BUG=2731
TEST=tsan
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5919005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5306 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| e7b1e112833517c334a12aac16be17a27d798944 |
16-Dec-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..."
> Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."
>
> > Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
> >
> > R=holmer@google.com
> >
> > Review URL: https://webrtc-codereview.appspot.com/5049004
>
> TBR=asapersson@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5799004
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5299 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
|
| 5ab756703ea32f2c2ff9878d6eae628c7380bc14 |
16-Dec-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r5294 to re-roll r5293.
To fix races in test each stream now owns its own encoder/decoder.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/5919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5297 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_utility.cc
|
| 87ad57bc753f6745e0d4c77b493485ce7ea1846a |
16-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver
The iterator is incremented both in loop header and loop body. Should
only be incremented in header.
BUG=2727
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5295 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| 41e2615e020311172b937f527c13d9e090437eca |
15-Dec-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."
> Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
>
> BUG=
> R=mflodman@webrtc.org, stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5409004
TBR=solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5294 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_utility.cc
|
| 341e91441aaa9c2c5a638082c3ee4530aa21612c |
14-Dec-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_utility.cc
|
| 86bb56a7f5b08ac285656bd95ddac34a7922c43a |
13-Dec-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."
> Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
>
> R=holmer@google.com
>
> Review URL: https://webrtc-codereview.appspot.com/5049004
TBR=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5285 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
|
| 6811b6e308d16f160ba4c32650f195d5d3d9a2b1 |
13-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Callback for send bitrate estimates - new roll
Issue https://webrtc-codereview.appspot.com/4459004/ was commited as
r5259, after which flakiness was detected and a rollback was performed
at r5261.
Patch Set 1 of this issue is the code submitted in r5259. Subsequent
patch sets fixes a race condition which caused the seen problems.
The root cause was a dead lock between a thread sending rtp packets and
and a timed module processing thread:
webrtc::RTPSender::BitrateUpdated() // Get RTPSender stats lock
webrtc::Bitrate::Process() // Get Bitrate lock
webrtc::RTPSender::ProcessBitrate()
webrtc::ModuleRtpRtcpImpl::Process()
...
webrtc::Bitrate::Update() // Get Bitrate lock
webrtc::RTPSender::UpdateRtpStats() // Get RTPSender stats lock
webrtc::RTPSender::SendToNetwork()
...
This is fixed in Bitrate::Process() by releasing the lock before
calling the callback.
BUG=2235
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5281 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/bitrate.cc
ource/bitrate.h
ource/receive_statistics_impl.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
|
| e9abd591d73218e11a8bd3e7c72d4d7af9a3cea8 |
13-Dec-2013 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Making RemoteRateControl::min_configured_bit_rate_ configurable
The minimum bitrate can now be configured from WrappingBitrateEstimator.
BUG=2698
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5279 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender_unittest.cc
|
| 096e8d9f944abeee5fb75d165d91f7a68258f073 |
11-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5259 "Callback for send bitrate estimates"
CL is causing flakiness in RampUpTest.WithoutPacing.
> Callback for send bitrate estimates
>
> BUG=2235
> R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/4459004
R=mflodman@webrtc.org, pbos@webrtc.org
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/5579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5261 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/bitrate.cc
ource/bitrate.h
ource/receive_statistics_impl.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
|
| 2656cf9f4c37fe1360e2392a5b0101df38660403 |
11-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Callback for send bitrate estimates
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5259 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/bitrate.cc
ource/bitrate.h
ource/receive_statistics_impl.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
|
| eb7def234e2fc6fd16cc627eaef813d2316c6ed6 |
09-Dec-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix compilation errors on Fedora 20.
peerconnection_jni.cc: syscall() comes from <unistd.h>
RTPtimeshift.cc: char* being compared to 0 instead of the atoi() of it
rtp_payload_registry_unittest.cc: avoid narrowing int to uint32.
BUG=2700
R=andrew@webrtc.org, fischman@webrtc.org, henrik.lundin@webrtc.org, henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5019004
Patch from Victor Costan <costan@gmail.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5248 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_payload_registry_unittest.cc
|
| 88615f0659948ff0cb87e6e467ea650b304b030d |
06-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5236 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender_unittest.cc
|
| 96a9b2dcdcaee150f7c19f229f4b7297df76e13b |
05-Dec-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
R=holmer@google.com
Review URL: https://webrtc-codereview.appspot.com/5049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5228 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
|
| ebad765ee00b90c48507bff1997ea8c1070a9316 |
05-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add callbacks for send channel rtp statistics
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5227 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| 0a3c1471b873ea7f81bff2faa7cf0d9e563c7d53 |
05-Dec-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add API to query video engine for the send-side delay.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4559005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5225 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
|
| a6ad6e5b589465f6a51ce46ee87d50e00bfd85b2 |
05-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add callbacks for send channel rtcp statistics
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5220 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| c4726d06fa0553b4d673ecbbd632effc37e0f2e3 |
05-Dec-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make RTPSender::SendPadData public.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5219 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.h
|
| 71f055fb41336316324942f828e022e2f7d93ec7 |
04-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add send frame rate statistics callback
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5213 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| 79b63206b99912d9a5f97a35b546409886a8fed2 |
04-Dec-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes a crash in fullstack tests introduced with r5209.
TBR=mflodman@webrtc.org
BUG=1812
Review URL: https://webrtc-codereview.appspot.com/4689005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5211 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_packet_history.cc
ource/rtp_packet_history_unittest.cc
ource/rtp_sender.cc
|
| 7e9315b42ebe8f7df860030af93618de81326503 |
04-Dec-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds support for sending redundant payloads over RTX.
TEST=trybots
BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5209 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/nack_rtx_unittest.cc
ource/rtp_packet_history.cc
ource/rtp_packet_history.h
ource/rtp_packet_history_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
est/testAPI/test_api.cc
|
| 499631c1e4ad2672a333898e652d905c372793a1 |
03-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Utility class for reading/writing network-byte-ordered integers.
BUG=
R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2151008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5203 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/byte_io.h
ource/byte_io_unittest.cc
ource/rtp_rtcp.gypi
|
| 47fadba7502f852629cb635426047efb797c1e31 |
25-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add include stdlib.h to files using abs.
abs function is declared in stdlib.h
Committing for alextaran@chromium.org.
Reviewed here: http://review.webrtc.org/4239004/
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5170 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/forward_error_correction.cc
|
| ffe1b17b57a6f617f33a85ff43905a145b4fed92 |
21-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Lock access to ModuleRtpRtcpImpl::simulcast_.
Fixes race between RegisterSendPayload and SendOutgoingData.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4099006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5152 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| 8d02f5dc7146ebc35c30fc3f7e1cbfa6802486a2 |
21-Nov-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Added API for enabling/disabling RTCP Receiver Reference Time extension.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3419005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5147 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
|
| 1ae1d0c47145f1036c3844a5cd1b536c22565325 |
20-Nov-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module).
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2383004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5139 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
|
| 6e95d7afab12dcc6cd3893210baf56d49df74ea0 |
15-Nov-2013 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Increment RTP timestamps for padding packets
This CL makes the padding packets get their own RTP timestamps,
rather than having the same timestamp as the last sent video
packet. The purpose is to solve Issue 2611, where the overuse-
detector does not react to padding packets.
A test was implemented to verify that the padding packets do
get their own timestamps.
BUG=2611
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5125 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| 9b82f5a6ed2ceb04f72b66c1d3cca67a2bbcec3a |
13-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix for RTX in combination with pacing.
Retransmissions didn't get sent over RTX when pacing was enabled since
the pacer didn't keep track of whether a packet was a retransmit or not.
BUG=1811
TEST=trybots
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5117 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| 38599510dfdcd1ee2cd8ce147b5b46ff8df15720 |
12-Nov-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Parse next RTCP XR report block after an unsupported block type.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5114 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver_unittest.cc
ource/rtcp_utility.cc
ource/rtcp_utility.h
|
| 57eb8586986a2c77b99124c270bc6caa11165f7f |
11-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove ".." from include_dirs in build/common.
BUG=1662
TEST=compile on trybots
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2332004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5107 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/H264/h264_information.h
ource/H264/rtp_sender_h264.h
|
| 48df38114d9502f4b4ad700c011190c608a702d5 |
08-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix for making sure that the packet in order checks are done prior to updating the last received packet state.
Without this fix all packets are considered out-of-order by the rtp receiver, causing the last received state
in the rtp receiver to never get valid.
Also makes sure that only valid timestamps and receive times are used for audio/video sync.
BUG=2608
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5102 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_receiver.h
ource/rtp_receiver_impl.cc
ource/rtp_receiver_impl.h
est/testAPI/test_api_audio.cc
|
| 766154aa1d9cdb7a8f9ac16611a1e4a13060b85b |
04-Nov-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removed unused code.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5073 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| 7d6bd2201938e4b6b7e5219c0fc971b0e1ba05b1 |
31-Oct-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Propagate estimated RTT from receivers to rtt observer.
BUG=1613
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5063 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_impl_unittest.cc
|
| 042e91c2b24b3bf2acea6e59e0303ff50ff36970 |
23-Oct-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix for incorrect RTT estimation. A too low RTT value could be estimated.
R=andrew@webrtc.org, holmer@google.com, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5023 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_sender.cc
ource/rtp_rtcp_impl_unittest.cc
|
| 31628aae7e0d5a00e816f1f5db4b65101319a307 |
22-Oct-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Upgrade scoped_ptr to Chromium's latest version.
Analogous to the recent libjingle change: http://cl/54929753-p10.
This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather
than scoped_array and scoped_ptr_malloc respectively.
- Add Chromium's template-based COMPILE_ASSERT. We didn't have this
previously in order to support the macro in C. Instead, move the
existing macro to compile_assert_c.h.
- Additionally copy the move.h and template_util.h depedencies and add
the WARN_UNUSED_RESULT macro.
- Leave scoped_array and scoped_ptr_malloc for now, but mark as
deprecated.
- Remove scoped_ptr foo(NULL) use. The default constructor handles it.
- Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc.
- Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove
some repeated code.
TESTED=trybots
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2449005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_format_vp8_unittest.cc
|
| 621df678c8690f36875b0b34d45393df58662172 |
22-Oct-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN.
Mostly to remove a long-standing TODO...
TESTED=trybots
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5013 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_utility.cc
|
| 3555303cb0d37a3b4a86883ab3c62234d91998c3 |
15-Oct-2013 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Roll chromium_revision 226126:228675 and fix clang warnings
By request from thakis@chromium.org, I disabled the
-Wno-unused-const-variable setting that is set in Chromium's
common.gypi so we can prepare our code for it's removal.
This required some cleanup in order to get the code to compile
with Clang having the -Wunused-const-variable warning enabled.
TEST=all trybots passing
BUG=none
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2400004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4966 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/receive_statistics_unittest.cc
|
| e5021fe5905e3cad792738e6aaadc6ddc742d42b |
15-Oct-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make RtpData and RtpFeedback destructors public.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4965 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
|
| 8469f7b328ec980f80fa79931b4e07872d0feb23 |
02-Oct-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Added support for sending and receiving RTCP XR packets:
- Receiver reference time report block
- DLRR report block (RFC3611).
BUG=1613
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2196010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4898 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_help.cc
ource/rtcp_receiver_help.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtcp_utility.cc
ource/rtcp_utility.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| 663da0a8fccef6ecfde780e42fda21ad46010038 |
26-Sep-2013 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
With ACM2 and NetEq4, VoE fuzz test very often fails.
As far as I observe, there are several reasons for this:
1. webrtc/modules/audio_coding/neteq4/neteq_impl.cc : 870
assert(new_codec_);
This is related to webrtc/modules/audio_coding/neteq4/decision_logic_normal.cc : 81
kUndefined can happen without new_codec_ being set
2. webrtc/modules/audio_coding/neteq4/neteq_impl.cc : 745
assert(sync_buffer_->FutureLength() >= expand_->overlap_length());
3. some other assert triggered.
The above happens not very often and goes away with no assertion.
3. (most common, this CL addresses this)
webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc : 201
payload_data_length = payload_length - rtp_header->header.paddingLength;
There are situations that
payload_length < rtp_header->header.paddingLength;
OLD ACM + NetEq3 can handle this:
a) webrtc/modules/audio_coding/main/source/acm_neteq.cc : 477
int16_t payload_length = static_cast<int16_t>(length_payload);
payload_length becomes negative in this situation
b) webrtc/modules/audio_coding/neteq/recin.c
WebRtcNetEQ_RecInInternal() handles negative payload length
I do not want to touch VoE, so I tried to let ACM2 and NetEq4 handle negative payload length.
This CL does not follow the exact way of OLD ACM + NetEq3. I stopped negative payload length at ACM and did not allow it go to NetEq4.
To try this, apply my uploaded patch : https://webrtc-codereview.appspot.com/2281004/
Let me know if you see better solutions.
R=henrik.lundin@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2292005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4860 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_receiver_audio.cc
|
| dc3fa083318049d3b1c8958a6ed44433f2eac090 |
26-Sep-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove include_dirs from rtp_rtcp.
BUG=1662
TEST=compile on trybots
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4851 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp.gypi
est/bwe_standalone.gypi
|
| 3e7703640fbc3c402f9ae7925dca697714ceddb9 |
26-Sep-2013 |
niklas.enbom@webrtc.org <niklas.enbom@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove unused constants, so chrome can enable a warning for that. Patch from thakis@
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2296006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4844 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver.cc
ource/rtp_rtcp_impl.cc
|
| be63fd644f2506a34c4d8b0239edc50d796884ac |
17-Sep-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Initialize CodecInst structs in test_api_audio.cc.
Fixes errors detected by DrMemory on Windows.
BUG=2382
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2228004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4764 4adac7df-926f-26a2-2b94-8c16560cd09d
est/testAPI/test_api_audio.cc
|
| 28a331eededf17dc3a0860bb1bdf5b2dc3f9e763 |
17-Sep-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add support for multiple report blocks.
Use a weighted average of fraction loss for bandwidth estimation.
TEST=trybots and vie_auto_test --automated
BUG=1811
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2198004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4762 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_help.cc
ource/rtcp_receiver_help.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| e07049f19f7ca7a9ab7bc91acbfa24cbac3f8031 |
10-Sep-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Lock RTPSender statistics.
Suppressing these errors in TSan has become tedious. It's better to just
lock them.
BUG=2349
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2197004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4713 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 59f20bb735562d245357609799578edeed46be32 |
09-Sep-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Break out RTCPSender dependency on ModuleRtpRtcpImpl.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2191004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4706 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_format_remb_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 30e055c4dd708636df46c6d76964c7f984dbec46 |
08-Sep-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Handle empty RTP video packets agnostic to codec.
Sending empty RTP packets caused a crash when using a generic codec
instead of VP8. This fix moves handling of empty RTP packets out of
ReceiveVp8Codec and into ParseRtpPacket.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2185004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4701 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_receiver_video.cc
|
| b2c8a952a7a996b89c6ff2ecdc1364641f2571f6 |
06-Sep-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Improving padding rules and breaking out bw allocation to ViEEncoder.
BUG=1837
TESTS=vie_auto_test --automated, trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2170004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4693 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| 7bb8f02274ecbfa1f7ef134d708369a369a78c83 |
06-Sep-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds support for combining RTX and FEC/RED.
This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX.
Enables retransmissions over RTX by default in the loopback test.
BUG=1811
TESTS=voe/vie_auto_test --automated and trybots.
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2154004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/fec_receiver.h
nterface/receive_statistics.h
nterface/rtp_payload_registry.h
nterface/rtp_receiver.h
nterface/rtp_rtcp_defines.h
ource/fec_receiver_impl.cc
ource/fec_receiver_impl.h
ource/fec_receiver_unittest.cc
ource/nack_rtx_unittest.cc
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/receiver_fec.cc
ource/receiver_fec.h
ource/receiver_fec_unittest.cc
ource/rtcp_sender_unittest.cc
ource/rtp_payload_registry.cc
ource/rtp_receiver_audio.cc
ource/rtp_receiver_impl.cc
ource/rtp_receiver_impl.h
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp.gypi
ource/rtp_sender_unittest.cc
ource/rtp_utility.cc
ource/rtp_utility.h
est/testAPI/test_api.cc
est/testAPI/test_api.h
est/testAPI/test_api_video.cc
|
| 9080518a3928285be9f94684adad064c65d2cdf3 |
05-Sep-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Restore severity precondition to logging.h.
I mistakenly ommitted the checks when logging.h was ported from
libjingle to webrtc. This caused a significant CPU cost for logs which
were later filtered out anyway.
Verified with LS_VERBOSE logging in neteq4, running:
$ out/Release/modules_unittests \
--gtest_filter=NetEqDecodingTest.TestBitExactness \
--gtest_repeat=50 > time.txt
$ grep "case ran" time.txt | grep "[0-9]* ms" -o | sort
Results on a MacBook Retina, averaged over 5 runs:
Verbose logs disabled: 666 ms
Exisiting implementation, verbose logs enabled: 944 ms (1.42x)
New implementation, verbose logs enabled: 673 ms (1.01x)
BUG=2314
R=henrik.lundin@webrtc.org, henrike@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2160005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4682 4adac7df-926f-26a2-2b94-8c16560cd09d
est/BWEStandAlone/BWEStandAlone.cc
|
| b21e528c60f0bfb1dca294baaddb9a274d751516 |
04-Sep-2013 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Protecting Bitrate to avoid data race found by tsan.
TEST=try and vie_auto_test with tsan.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2163004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4673 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/bitrate.cc
ource/bitrate.h
|
| cac7325b84a57ba4d1ab73e1ce58777a5946e4ba |
03-Sep-2013 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding critsect for child_modules_ in ModuleRtpRtcpImpl::Process() to avoid race with ModuleRtpRtcp::RegisterChildModule.
Found with tsan.
TEST=try job and tsan
R=holmer@google.com
Review URL: https://webrtc-codereview.appspot.com/2156004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4661 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| 286fe0b04d97205ac84688bbe613d5749192b2d1 |
21-Aug-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ..."""
...and fixes the RTCP bug.
BUG=2277
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4588 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
nterface/rtp_rtcp_defines.h
ource/nack_rtx_unittest.cc
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_receiver_impl.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_utility.cc
ource/rtp_utility.h
est/testAPI/test_api_rtcp.cc
|
| a0218a84d17a727111e2e24cf5af915b1b91c06e |
21-Aug-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 4582 "Reverts a second set of reverts caused by a bug in ..."
> Reverts a second set of reverts caused by a bug in a dependency.
>
> Revert "Revert r4328"
>
> Revert "Revert r4322 "Support sending multiple report blocks and keeping track
> of statistics on"
>
> BUG=1811
> R=henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/2072004
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2087004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4585 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
nterface/rtp_rtcp_defines.h
ource/nack_rtx_unittest.cc
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_receiver_impl.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_utility.cc
ource/rtp_utility.h
est/testAPI/test_api_rtcp.cc
|
| 1a65d6c36b6a25f9f734176c697c684c3b43ac4b |
21-Aug-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reverts a second set of reverts caused by a bug in a dependency.
Revert "Revert r4328"
Revert "Revert r4322 "Support sending multiple report blocks and keeping track
of statistics on"
BUG=1811
R=henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2072004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4582 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
nterface/rtp_rtcp_defines.h
ource/nack_rtx_unittest.cc
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_receiver_impl.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_utility.cc
ource/rtp_utility.h
est/testAPI/test_api_rtcp.cc
|
| d4f607e70ad85102b77cf0beba8f11e2599e8f99 |
19-Aug-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes to padding when driven by encoder.
- Allow padding to be sent on an ssrc which doesn't produce video, therefore
never having the last_packet_marker_bit_ set.
- Add the random timestamp offset to all padding packets.
- Store the capture time of padding packets to properly create an offset.
BUG=2258
TEST=trybots and a new test which will be committed separately.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2060005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4566 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 822fbd8b68ffdb481b9557e2950ae8d6657c8ce6 |
16-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 50918584.
Together with Stefan's http://review.webrtc.org/1960004/.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2048004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
nterface/rtp_payload_registry.h
nterface/rtp_receiver.h
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/bitrate.cc
ource/bitrate.h
ource/mock/mock_rtp_payload_strategy.h
ource/mock/mock_rtp_receiver_video.h
ource/nack_rtx_unittest.cc
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receiver_fec.cc
ource/receiver_fec.h
ource/receiver_fec_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_payload_registry.cc
ource/rtp_payload_registry.h
ource/rtp_payload_registry_unittest.cc
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_impl.cc
ource/rtp_receiver_impl.h
ource/rtp_receiver_strategy.cc
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp.gypi
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender_video.cc
ource/rtp_utility.cc
ource/rtp_utility.h
ource/rtp_utility_unittest.cc
est/testAPI/test_api.cc
est/testAPI/test_api.h
est/testAPI/test_api_audio.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
|
| a3b740621959d984f81f39f51d7e8a0d2bf2f423 |
09-Aug-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove unused unreferenced code in webrtc/
The code removed here are .c, .cc and .h files that are not referenced
from anywhere else. E.g. if git-grep showed no occurrence of the file
it's removed. This process was repeated until no more unreferenced
files were present.
BUG=
R=andrew@webrtc.org, henrike@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org, turaj@webrtc.org, wu@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1945004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4511 4adac7df-926f-26a2-2b94-8c16560cd09d
est/BWEStandAlone/BWEConvergenceTest.cc
est/BWEStandAlone/BWEConvergenceTest.h
est/BWEStandAlone/BWEStabilityTest.cc
est/BWEStandAlone/BWEStabilityTest.h
est/BWEStandAlone/BWETestBase.cc
est/BWEStandAlone/BWETestBase.h
est/BWEStandAlone/BWETester.cc
est/BWEStandAlone/BWETwoWayLimitFinding.cc
est/BWEStandAlone/BWETwoWayLimitFinding.h
est/testTMMBR/testTMMBR.cc
|
| e2703314816bad1c39736501cbb1a74062890244 |
09-Aug-2013 |
niklas.enbom@webrtc.org <niklas.enbom@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix duplicate code
R=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1993004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4507 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| 12dc1a38ca54a000e4fecfbc6d41138b895c9ca5 |
05-Aug-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Switch C++-style C headers with their C equivalents.
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.
BUG=1833
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1917004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/forward_error_correction.cc
ource/forward_error_correction_internal.cc
ource/receiver_fec.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver_help.cc
ource/rtcp_sender.cc
ource/rtcp_utility.cc
ource/rtcp_utility.h
ource/rtp_format_vp8.cc
ource/rtp_header_extension.cc
ource/rtp_packet_history.cc
ource/rtp_receiver.cc
ource/rtp_receiver_audio.cc
ource/rtp_receiver_strategy.cc
ource/rtp_receiver_video.cc
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_audio.cc
ource/rtp_utility.cc
ource/rtp_utility.h
est/testFec/test_fec.cc
|
| f3e4ceee47d747c8868d919c179ecc640b9541f0 |
31-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix some chromium-style warnings in webrtc/modules/rtp_rtcp/
BUG=163
R=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1904005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4444 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
ource/forward_error_correction.cc
ource/forward_error_correction.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_format_vp8.cc
ource/rtp_format_vp8.h
ource/rtp_header_parser.cc
ource/rtp_payload_registry.cc
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_strategy.cc
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.h
ource/rtp_rtcp_impl.h
ource/rtp_sender.h
|
| 64e2cbf184ecf8f20fb949ea5a1c6e1c1bdd8bc3 |
16-Jul-2013 |
tnakamura@webrtc.org <tnakamura@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
clean up incomplete revert in r4357
Also revert r4319, will follow up with pbos
Reason for recent series of reverts: video freezes when testing with packet loss
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1817004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4359 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_payload_registry.cc
est/testAPI/test_api_video.cc
|
| aa4d96a134a03f998d52fb9699845d9c644eb24b |
16-Jul-2013 |
tnakamura@webrtc.org <tnakamura@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r4301
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
nterface/rtp_payload_registry.h
nterface/rtp_receiver.h
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/bitrate.cc
ource/bitrate.h
ource/mock/mock_rtp_payload_strategy.h
ource/mock/mock_rtp_receiver_video.h
ource/nack_rtx_unittest.cc
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receiver_fec.cc
ource/receiver_fec.h
ource/receiver_fec_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_payload_registry.cc
ource/rtp_payload_registry.h
ource/rtp_payload_registry_unittest.cc
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_impl.cc
ource/rtp_receiver_impl.h
ource/rtp_receiver_strategy.cc
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp.gypi
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender_video.cc
ource/rtp_utility.cc
ource/rtp_utility.h
ource/rtp_utility_unittest.cc
est/testAPI/test_api.cc
est/testAPI/test_api.h
est/testAPI/test_api_audio.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
|
| 9b82dced8dd72b1cba12fac396dfdc0dfc5418b6 |
16-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make sure first RTP packet counts as in-order.
BUG=
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1811004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4350 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_receiver_impl.cc
|
| 4a44ea21d71150b7532325a292faa2a02337f596 |
15-Jul-2013 |
elham@webrtc.org <elham@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r4320 "Fix three uninitialized members in rtp_receiver_impl.cc"
TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1803004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4346 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_receiver_impl.cc
|
| 4888fd48272dc7fde24b21a3a7dfefdc9b4e9466 |
15-Jul-2013 |
elham@webrtc.org <elham@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r4321 "Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered"
R=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1790006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4345 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_payload_registry.cc
|
| b7eda43810125cd01b29671a6beab61ddb48ebdb |
15-Jul-2013 |
elham@webrtc.org <elham@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r4322 "Support sending multiple report blocks and keeping track of statistics on
several SSRCs"
R=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1774006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4344 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
nterface/rtp_rtcp_defines.h
ource/nack_rtx_unittest.cc
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_receiver_impl.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
est/testAPI/test_api_rtcp.cc
|
| 6f5707e184f798db335527d3d7757347cdce3be3 |
15-Jul-2013 |
elham@webrtc.org <elham@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r4328
R=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1774005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4343 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
ource/rtcp_sender.cc
ource/rtp_rtcp_impl.cc
ource/rtp_utility.cc
ource/rtp_utility.h
|
| e4736eee200873837bf66ff757004971f377b712 |
11-Jul-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes a crash when sending SR reports from a sender only module.
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1790004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4328 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
ource/rtcp_sender.cc
ource/rtp_rtcp_impl.cc
ource/rtp_utility.cc
ource/rtp_utility.h
|
| aeba6e87402aea3e6431f2b18c0e5f6f80f71783 |
11-Jul-2013 |
braveyao@webrtc.org <braveyao@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
ModuleRTPRTCP call rtcp_sender_.TMMBR() directly instead of calling its own API.
BUG=2051
TEST=autotest
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1790005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4327 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| 96edd561703ad9e257e91b92e3c1436bef446f13 |
10-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Sorted headers under rtp_rtcp/.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1781005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4325 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_sender_unittest.cc
ource/rtp_receiver_impl.cc
ource/tmmbr_help.h
est/BWEStandAlone/BWETester.cc
est/testAPI/test_api_audio.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
|
| 717d147ebb168ed498fa4777ffaf8646a1dc6d7a |
10-Jul-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Support sending multiple report blocks and keeping track of statistics on several SSRCs.
BUG=1811
TEST=vie_auto_test --automated, voe_auto_test --automated, trybots
R=andresp@webrtc.org, tommi@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1768004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4322 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
nterface/rtp_rtcp_defines.h
ource/nack_rtx_unittest.cc
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receive_statistics_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_receiver_impl.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
est/testAPI/test_api_rtcp.cc
|
| 9de89a6f6bd9e7635c41966a9ab4b8d818521e57 |
10-Jul-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered.
R=pbos@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1782004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4321 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_payload_registry.cc
|
| 452d853c434d002f89a5c0ec5930f607fee05571 |
10-Jul-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix three uninitialized members in rtp_receiver_impl.cc.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1781004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4320 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_receiver_impl.cc
|
| 08933a5dfb2c9d9b55ebd513daece26465b7d3e2 |
10-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Initialize payload-type frequency in channel.cc.
Uninitialized values triggered divide-by-zero crashes in voe_auto_test.
BUG=
R=stefan@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1780004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4319 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_payload_registry.cc
ource/rtp_receiver_impl.cc
|
| f56d612c707374dee18c0b6b5411f87de8854dbb |
09-Jul-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Create gyp target for bwe components.
R=henrikg@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1775004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4311 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp.gypi
|
| 1a7b9b94be10119224c58edcddebf9ad398331ce |
08-Jul-2013 |
hclam@chromium.org <hclam@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Cleanup WebRTC tracing
The goal of this change is to:
1. Remove unused tracing events.
2. Organize tracing events to facilitate measurement of end to end latency.
The major change in this CL is to use ASYNC_STEP such that operation
flow can be traced for the same frame.
R=marpan@webrtc.org, pwestin@webrtc.org, turaj@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1761004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4308 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/receive_statistics_impl.cc
ource/rtcp_receiver.cc
ource/rtcp_sender.cc
ource/rtp_receiver_impl.cc
ource/rtp_sender.cc
ource/rtp_sender_audio.cc
ource/rtp_sender_video.cc
|
| 66b2e5c05a3f2a93d634d1dbbcbb283fb218ca4f |
05-Jul-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
rtp_rtcp implementation.
This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.
With this change the dead-or-alive and packet timeout APIs are removed.
TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1745004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/receive_statistics.h
nterface/rtp_payload_registry.h
nterface/rtp_receiver.h
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/bitrate.cc
ource/bitrate.h
ource/mock/mock_rtp_payload_strategy.h
ource/mock/mock_rtp_receiver_video.h
ource/nack_rtx_unittest.cc
ource/receive_statistics_impl.cc
ource/receive_statistics_impl.h
ource/receiver_fec.cc
ource/receiver_fec.h
ource/receiver_fec_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_payload_registry.cc
ource/rtp_payload_registry.h
ource/rtp_payload_registry_unittest.cc
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_impl.cc
ource/rtp_receiver_impl.h
ource/rtp_receiver_strategy.cc
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp.gypi
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender_video.cc
ource/rtp_utility.cc
ource/rtp_utility.h
ource/rtp_utility_unittest.cc
est/testAPI/test_api.cc
est/testAPI/test_api.h
est/testAPI/test_api_audio.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
|
| d900e8bea84c474696bf0219aed1353ce65ffd8e |
03-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Proper spacing for end-of-namespace comments.
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1760006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ource/H264/bitstream_builder.cc
ource/H264/bitstream_builder.h
ource/H264/bitstream_parser.cc
ource/H264/bitstream_parser.h
ource/H264/h264_information.cc
ource/H264/h264_information.h
ource/H264/rtp_sender_h264.cc
ource/H264/rtp_sender_h264.h
ource/forward_error_correction.h
ource/forward_error_correction_internal.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_help.cc
ource/rtcp_receiver_help.h
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_utility.cc
ource/rtcp_utility.h
ource/rtp_header_extension.cc
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_rtcp_config.h
ource/rtp_sender_audio.cc
ource/rtp_sender_audio.h
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
ource/ssrc_database.cc
ource/ssrc_database.h
ource/tmmbr_help.cc
ource/tmmbr_help.h
ource/video_codec_information.h
est/testFec/average_residual_loss_xor_codes.h
est/testFec/test_fec.cc
|
| a5fd2f1348f7d155293316b4230c688f1ac2448e |
26-Jun-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Do basic parsing of RTCP headers in PcapFileReader to enable log filtering.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1697004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4266 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_utility.cc
ource/rtp_utility.h
|
| 91811e2b0457e091886508894a771f0e12054d0b |
25-Jun-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove unused multi stream bandwidth estimator.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1712004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4264 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
ource/rtcp_receiver.cc
est/testAPI/test_api_rtcp.cc
|
| a4c5abb52a4677ea576c5076ce36df33bb6c9cba |
25-Jun-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make sure padding packets are sent.
BUG=1837
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1717006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4260 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
|
| 2e402ce873f48e0848468345d848bd3fff75dd3e |
20-Jun-2013 |
hclam@chromium.org <hclam@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enqueue packet in pacer if sending fails
If a packet cannot be sent while pacer is in use it should be
queued. This avoid packet loss due to congestion.
BUG=1930
R=pwestin@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1693004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4250 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 8ccb9f9716f306dd1ec284b4f61f0b6c82c08c3c |
19-Jun-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes some pacer/padding issues found while testing.
- A bug was introduced in r4234 causing no paced packets to be sent.
- Only update the sequence number counter if a padding packet is actually going to be sent, to avoid packet loss.
- Have all packets go through the pacer if pacing is enabled to avoid reordering.
- Fix race condition on reading capture_time_ms_/timestamp_ in rtp_sender.cc.
BUG=1837
TEST=trybots and vie_auto_test --automated
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1682004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4246 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_audio.cc
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
|
| 508a84b25597a8d12177eabed002b71f5730d4c8 |
17-Jun-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Wire up pacer-based padding.
This connects the pacer-based padding with the RTP modules, which will
generate padding packets roughly according to what the pacer suggests.
It will only generate padding packets of maximum size to keep the number
off padding packets as small as possible. This also sets a limit of how much
padding + media bitrate which the pacer is allowed to "request" from the
RTP modules.
Padding will for now only be generated by the first sending RTP module.
BUG=1837
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1612005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4234 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_audio.cc
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
|
| 63e988856ee6fe5999980404c2567f3e2cf759da |
14-Jun-2013 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Merge more tests into modules_{unit,integration}tests.
A new test target named 'modules_integrationtests' is created
and the following test targets were merged into it:
* audio_coding_module_test
* test_fec
* video_coding_integrationtests
* vp8_integrationtests
A couple of other targets were merged into modules_unittests:
* audio_coding_unittests
* audioproc_unittest
* common_unittests
* video_coding_unittests
* video_processing_unittests
* vp8_unittests
I wasn't able to merge audio_decoder_unittests and neteq_unittests due to
conflicts with different defines in these tests.
Some tests that have special requirements aren't merged into
modules_integrationtests yet. I took the opportunity to rename them
since the bot configs will need to be update anyway:
* audio_device_test_api -> audio_device_integrationtests
* video_capture_module_test -> video_capture_integrationtests
* video_render_module_test -> video_render_integrationtests
Exclude files were added for modules_integrationtests to make sure
the memcheck and tsan bots doesn't tests that are too slow
(audio_coding_module_test and vp8_integrationtests were previously
disabled on those bots).
Suppressions for AudioCodingModuleTest needed to be added to get
modules_integrationtests to pass memcheck (even if the test is
excluded from execution).
BUG=1843
TEST=local execution on Linux and trybots (passing except the merged tests of course)
R=andrew@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1656004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4228 4adac7df-926f-26a2-2b94-8c16560cd09d
est/testFec/test_fec.gypi
|
| 04996cd5e5f49879a77c03bcbc898bc47fd1b8bb |
12-Jun-2013 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix breakage due to test_fec conversion to gtest.
In my attempt to commit a subset of http://review.webrtc.org/1647005/
instead of all of it, I forgot to add the gtest dependency to the
test_fec.gypi. This CL fixes that.
TEST=local compile + win_rel,mac_rel,linux_rel trybots
BUG=1916
R=marpan
TBR=marpan
Review URL: https://webrtc-codereview.appspot.com/1655004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4219 4adac7df-926f-26a2-2b94-8c16560cd09d
est/testFec/test_fec.gypi
|
| 22bbbdfa6809a2fb543c6c26d022804df06a4f2c |
12-Jun-2013 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Convert test_fec to gtest
All tests needs to be gtest tests in order to be executed
with the upcoming isolate/swarm framework.
TEST=trybots passing
BUG=1916
R=andrew@webrtc.org, marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/1647005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4218 4adac7df-926f-26a2-2b94-8c16560cd09d
est/testFec/test_fec.cc
|
| 6c35e0b0f7ca9ea5c56bfb78cb98268f1ed0f7d9 |
11-Jun-2013 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reorganize test targets in WebRTC
This CL will lower the number of test targets in WebRTC by:
Add common_audio_unittests and merge the following targets into it (copied from http://review.webrtc.org/1584006):
* resampler_unittests
* signal_processing_unittests
* vad_unittests
Merge into modules_unittests:
* bitrate_controller_unittests
* desktop_capture_unittests
* media_file_unittests
* remote_bitrate_estimator_unittests
* rtp_rtcp_unittests
* paced_sender_unittests
Merge into test_support_unittests:
* channel_transport_unittests
channel_transport.gyp was also removed in favor for test.gyp.
I had to remove a main method from rtcp_format_remb_unittest.cc
since it caused the fileutils.h code to not be able to find the
right project root path in ordrer to provide correct paths
to test files.
Buildbot configuration update will be synced with the commit of this CL.
TEST=trybots
BUG=1843
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4213 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_format_remb_unittest.cc
ource/rtp_rtcp_tests.gypi
|
| a817962bab1602a0229cb1d450bae55f22d9bd74 |
04-Jun-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor padding and rtp header functionality.
BUG=1837
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1611004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4172 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
|
| fa64a595adef6beefa07caaf65e2dcde44d0be04 |
03-Jun-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling.
This makes it easier for the users of the interface, i.e. doesn't need to remember the id in order to disable audio level indication later.
BUG=1828
TEST=unit tests
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1598005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4157 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender_audio.cc
ource/rtp_sender_unittest.cc
|
| a048d7cb0a5bad5ca49bbcc5273cb4cca28c1710 |
29-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Include files from webrtc/.. paths in rtp_rtcp/
BUG=1662
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1557004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4135 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/H264/bitstream_builder.h
ource/H264/bitstream_parser.h
ource/bitrate.h
ource/fec_private_tables_bursty.h
ource/fec_private_tables_random.h
ource/mock/mock_rtp_payload_strategy.h
ource/mock/mock_rtp_receiver_video.h
ource/producer_fec.cc
ource/producer_fec.h
ource/producer_fec_unittest.cc
ource/receiver_fec_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_help.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtcp_utility.cc
ource/rtcp_utility.h
ource/rtp_fec_unittest.cc
ource/rtp_format_vp8.cc
ource/rtp_format_vp8.h
ource/rtp_format_vp8_test_helper.cc
ource/rtp_format_vp8_test_helper.h
ource/rtp_format_vp8_unittest.cc
ource/rtp_header_extension.cc
ource/rtp_header_extension.h
ource/rtp_header_extension_unittest.cc
ource/rtp_packet_history.cc
ource/rtp_packet_history.h
ource/rtp_packet_history_unittest.cc
ource/rtp_payload_registry_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_sender_audio.cc
ource/rtp_sender_audio.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
ource/rtp_utility.cc
ource/rtp_utility.h
ource/rtp_utility_unittest.cc
ource/ssrc_database.cc
ource/ssrc_database.h
ource/tmmbr_help.cc
ource/tmmbr_help.h
ource/video_codec_information.h
ource/vp8_partition_aggregator.cc
ource/vp8_partition_aggregator.h
ource/vp8_partition_aggregator_unittest.cc
est/BWEStandAlone/BWEConvergenceTest.cc
est/BWEStandAlone/BWEConvergenceTest.h
est/BWEStandAlone/BWEStabilityTest.cc
est/BWEStandAlone/BWEStabilityTest.h
est/BWEStandAlone/BWEStandAlone.cc
est/BWEStandAlone/BWETestBase.cc
est/BWEStandAlone/BWETestBase.h
est/BWEStandAlone/BWETester.cc
est/BWEStandAlone/BWETwoWayLimitFinding.cc
est/BWEStandAlone/BWETwoWayLimitFinding.h
est/BWEStandAlone/MatlabPlot.cc
est/BWEStandAlone/MatlabPlot.h
est/BWEStandAlone/TestLoadGenerator.cc
est/BWEStandAlone/TestLoadGenerator.h
est/BWEStandAlone/TestSenderReceiver.cc
est/BWEStandAlone/TestSenderReceiver.h
est/testAPI/test_api_audio.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
est/testFec/test_fec.cc
est/testFec/test_packet_masks_metrics.cc
est/testTMMBR/testTMMBR.cc
|
| a5cb98cbbd11e93cb6d0a6232387814aac168c7d |
29-May-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Breaking out RTP header parsing from the RTP module.
This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video.
Moving bandwidth estimation before the RTP module is also required for RTX.
TEST=vie_auto_test, voe_auto_test, trybots.
BUG=1811
R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1545004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_header_parser.h
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/nack_rtx_unittest.cc
ource/receiver_fec.cc
ource/rtcp_sender_unittest.cc
ource/rtp_header_parser.cc
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_receiver_audio.cc
ource/rtp_receiver_video.cc
ource/rtp_rtcp.gypi
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_utility.cc
ource/rtp_utility.h
est/testAPI/test_api.h
est/testAPI/test_api_video.cc
|
| a6ae644e5276bb9dde5c17d0ce48cff784076d10 |
28-May-2013 |
marpan@webrtc.org <marpan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add comment about test_packet_masks_metrics.
R=andrew@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1577004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4124 4adac7df-926f-26a2-2b94-8c16560cd09d
est/testFec/test_fec.gypi
|
| 8665da89267ae370b4f2d20c5e33b4c1960483b3 |
24-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove dead testRateControl.cc
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1556004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4101 4adac7df-926f-26a2-2b94-8c16560cd09d
est/testRateControl/testRateControl.cc
|
| a01f7f6509eaf8aa674f153cec0d167d74ad82a4 |
24-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removed dead testH263Parser.cc
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1555004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4100 4adac7df-926f-26a2-2b94-8c16560cd09d
est/testH263Parser/testH263Parser.cc
|
| c1f0eb2c0374084b26c8720b95ca7b2569ff0303 |
24-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove dead bitstreamTest.cc.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1553004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4099 4adac7df-926f-26a2-2b94-8c16560cd09d
est/bitstreamTest/bitstreamTest.cc
|
| c74c3c244784fc1d6cea53ecb2dccfe353394e6a |
23-May-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds integration test for RTX and fixes bugs found.
BUG=1811
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4096 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
|
| 5c58f63d3fbce3f894a583a438c164b00c0b15dc |
23-May-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix regression where retransmission bitrate is no longer estimated.
BUG=1813
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1530004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4095 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender_unittest.cc
|
| c0352d566a4291cf587c25ca023e44b52ad7484e |
20-May-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix assertions in rtp_header_extension.h caused by not handling the AudioLevel extension. Added unit tests to do basic checks of the AudioLevel extension.
BUG=
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1510004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4069 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_header_extension.h
ource/rtp_sender.cc
ource/rtp_sender_unittest.cc
|
| 9919ad5caf288f5d5dda9cb644fa81492288eeff |
16-May-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Formatted FEC stuff.
Unfortunately I had to pull in quite a bit of stuff due to use of unencapsulated public member variables.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1401004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4047 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/forward_error_correction.cc
ource/forward_error_correction.h
ource/forward_error_correction_internal.cc
ource/forward_error_correction_internal.h
ource/receiver_fec.cc
ource/receiver_fec.h
ource/receiver_fec_unittest.cc
ource/rtp_fec_unittest.cc
est/testFec/test_fec.cc
|
| 7ebbea14a956c87f6f6aebb839486b9f12fcdf52 |
16-May-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add handling of the absolute send time header extension to the rtp_rtcp module.
BUG=
R=asapersson@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1480004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4041 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
ource/rtp_header_extension.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/rtp_utility.cc
|
| 6cfa3907c8b4cff62f13e4fe8beb66f89b6c0912 |
15-May-2013 |
mikhal@webrtc.org <mikhal@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Updating NACK RTX test
BUG=1513
R=holmer@google.com, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1274006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4036 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/nack_rtx_unittest.cc
ource/rtp_rtcp_tests.gypi
est/testAPI/test_api_nack.cc
|
| 29b2219914a059fe5164c312e7cc6d1bf0b4e610 |
14-May-2013 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding a factory to remote bitrate estimator and allow it to be set via config.
Additionally:
- clean api to set remote bitrate estimator mode.
- clean api to set over use detector options.
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1448006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4027 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender_unittest.cc
|
| 527f6c62fc63d1f3829409a7822dc81983d1db86 |
14-May-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reformatted FEC tables.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1400004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4020 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/fec_test_helper.cc
ource/fec_test_helper.h
|
| 7bfb3a322738fdf79a8d2498fd35c00bcc4617a7 |
14-May-2013 |
justinlin@chromium.org <justinlin@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add more tracing for key frames.
R=mallinath@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1428004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4015 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver.cc
ource/rtp_sender.cc
|
| 315d39866e4190d16283398eb044e4a9f420d3a8 |
08-May-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Formatted dtmf_queue.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1398004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3982 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/dtmf_queue.cc
ource/dtmf_queue.h
|
| 3004c79c6ad0ca4b4df27d0ca76c2eb29735e267 |
07-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix clang errors in non-GYP_DEFINES=clang=1 build
BUG=1623
R=stefan@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1368004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3968 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_format_remb_unittest.cc
ource/rtp_header_extension.cc
ource/rtp_header_extension.h
ource/rtp_header_extension_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender_unittest.cc
ource/rtp_utility.cc
est/testAPI/test_api.h
est/testFec/test_fec.cc
|
| 77f6b2175e55c12f5b8c95d719d3ce2070df079c |
03-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..."
> Revert 3933 "Remove traces of deprecated WebRtc_Word types."
>
> > Remove traces of deprecated WebRtc_Word types.
> >
> > BUG=314
> > R=tommi@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/1385004
>
> TBR=pbos@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1386004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1397004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3948 4adac7df-926f-26a2-2b94-8c16560cd09d
ocks/mock_rtp_rtcp.h
|
| 52b4e8871a7c43a12177cb9a717baff3fb2680c0 |
02-May-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding trace and changing pacing constants
BUG=1721,1722
R=mikhal@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1380005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3940 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
|
| 68e5a68f073b43a195ef7a846f3965fa9e6a2356 |
02-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 3933 "Remove traces of deprecated WebRtc_Word types."
> Remove traces of deprecated WebRtc_Word types.
>
> BUG=314
> R=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1385004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1386004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3934 4adac7df-926f-26a2-2b94-8c16560cd09d
ocks/mock_rtp_rtcp.h
|
| 265a5d298aeff39a31defb51a80569844a232831 |
02-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove traces of deprecated WebRtc_Word types.
BUG=314
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1385004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3933 4adac7df-926f-26a2-2b94-8c16560cd09d
ocks/mock_rtp_rtcp.h
|
| b0061f94b23062aa10c45f967dff622287bd68dc |
27-Apr-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enable Nack pacing.
Review URL: https://webrtc-codereview.appspot.com/1357004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3912 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| 8ca8a71de2ab16eaafd9c0e5aac87d28aab490ea |
23-Apr-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode."
This reverts commit aae26db1da5803482b094357c546b8454ab1c26d.
BUG=1613
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1327008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3890 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_rtcp_config.h
ource/rtp_rtcp_impl.cc
|
| ccd4b2aec88c79c531254fd31611ec741c77738f |
23-Apr-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add a default RTT to CallStats and use different values for buffered/real-time mode.
BUG=1613
Review URL: https://webrtc-codereview.appspot.com/1326007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3888 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_rtcp_config.h
ource/rtp_rtcp_impl.cc
|
| 6e788df19ef1e37049717757218fe1e74bbce1c2 |
16-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove vim/emacs modelines from .gypi files
BUG=1655
Review URL: https://webrtc-codereview.appspot.com/1326005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3857 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp.gypi
ource/rtp_rtcp_tests.gypi
est/bwe_standalone.gypi
est/testFec/test_fec.gypi
|
| 9f5ebb525130f207229dfa350ce8c2bdd22163c7 |
12-Apr-2013 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding a payload type for RTX.
BUG=736
TEST=Modified RTP unittests.
Review URL: https://webrtc-codereview.appspot.com/1278004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3843 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/nack_rtx_unittest.cc
ource/rtp_payload_registry.h
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
est/testAPI/test_api.cc
|
| b8e7f4cc9763a473a9abd8e20d832a734881f99d |
12-Apr-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Change capture interface to use NTP capture time.
Move NTP functionality to Clock.
BUG=1563
TEST=trybots and vie_auto_test --automated
Review URL: https://webrtc-codereview.appspot.com/1313005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3842 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver.cc
ource/rtcp_sender.cc
ource/rtp_utility.h
|
| 7bc465bd21b6df643edbb1a8902df12bd8e2b912 |
11-Apr-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix issues with incorrect wrap checks when having big buffers and high bitrate.
Introduces shared functions for timestamp and sequence number wrap checks.
BUG=1607
TESTS=trybots
Review URL: https://webrtc-codereview.appspot.com/1291005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3833 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/forward_error_correction.cc
ource/forward_error_correction.h
ource/rtp_receiver.cc
ource/rtp_utility.cc
|
| 523f93729b8b1295d4ba3826c3ec8acd6835cf99 |
11-Apr-2013 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Re-write the build of the nacklist.
Review URL: https://webrtc-codereview.appspot.com/1304008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3822 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_sender.cc
|
| ab9202b673f85b424169e8071e5f11ef0b72f889 |
10-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removing remaining WebRtc_Word32 not in typedefs.h
BUG=
Review URL: https://webrtc-codereview.appspot.com/1306006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3813 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/H264/rtp_sender_h264.cc
|
| 806dc3b0e62ec68f594e9aadab601b2db7e6c6d5 |
09-Apr-2013 |
hclam@chromium.org <hclam@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
More trace events
The goal of this change is to unify tracing events styles
and add trace events for all RTP traffic.
BUG=1555
Review URL: https://webrtc-codereview.appspot.com/1290007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3806 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver.cc
ource/rtcp_sender.cc
ource/rtp_receiver.cc
ource/rtp_receiver_audio.cc
ource/rtp_receiver_video.cc
ource/rtp_sender.cc
ource/rtp_sender_audio.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
|
| 7da3459b2ac83923c1ccbf11ad24d3f700305feb |
09-Apr-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps."
This reverts commit 4954b3650192d78037714138a5c519ef08f2670e.
Reverts r3799
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1308004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3802 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_receiver.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_utility.cc
ource/rtp_utility.h
|
| afcc6101d01be8c6cd9cf246dcf5b37b31ce0cd0 |
09-Apr-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps.
We should consider making the same change to the render timestamps generated at the receiver.
BUG=1563
Review URL: https://webrtc-codereview.appspot.com/1283005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3799 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_receiver.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_utility.cc
ource/rtp_utility.h
|
| 2f44673d665899ca788ae44247a9a7f4764f5e2b |
08-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 => int32_t for rtp_rtcp/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1279007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/H264/bitstream_builder.cc
ource/H264/bitstream_builder.h
ource/H264/bitstream_parser.cc
ource/H264/bitstream_parser.h
ource/H264/h264_information.cc
ource/H264/h264_information.h
ource/H264/rtp_sender_h264.cc
ource/H264/rtp_sender_h264.h
ource/bitrate.cc
ource/bitrate.h
ource/dtmf_queue.cc
ource/dtmf_queue.h
ource/mock/mock_rtp_payload_strategy.h
ource/mock/mock_rtp_receiver_video.h
ource/nack_rtx_unittest.cc
ource/producer_fec.cc
ource/receiver_fec.cc
ource/receiver_fec.h
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_help.cc
ource/rtcp_receiver_help.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtcp_utility.cc
ource/rtcp_utility.h
ource/rtp_format_vp8.cc
ource/rtp_format_vp8.h
ource/rtp_format_vp8_test_helper.cc
ource/rtp_payload_registry.cc
ource/rtp_payload_registry.h
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_audio.cc
ource/rtp_sender_audio.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
ource/rtp_utility.cc
ource/rtp_utility.h
ource/rtp_utility_unittest.cc
ource/ssrc_database.cc
ource/ssrc_database.h
ource/tmmbr_help.cc
ource/tmmbr_help.h
est/BWEStandAlone/BWEConvergenceTest.cc
est/BWEStandAlone/BWEConvergenceTest.h
est/BWEStandAlone/BWEStabilityTest.cc
est/BWEStandAlone/BWEStabilityTest.h
est/BWEStandAlone/BWEStandAlone.cc
est/BWEStandAlone/BWETestBase.cc
est/BWEStandAlone/BWETestBase.h
est/BWEStandAlone/BWETester.cc
est/BWEStandAlone/BWETwoWayLimitFinding.cc
est/BWEStandAlone/BWETwoWayLimitFinding.h
est/BWEStandAlone/MatlabPlot.cc
est/BWEStandAlone/MatlabPlot.h
est/BWEStandAlone/TestLoadGenerator.cc
est/BWEStandAlone/TestLoadGenerator.h
est/BWEStandAlone/TestSenderReceiver.cc
est/BWEStandAlone/TestSenderReceiver.h
est/bitstreamTest/bitstreamTest.cc
est/testAPI/test_api.cc
est/testAPI/test_api.h
est/testAPI/test_api_audio.cc
est/testAPI/test_api_nack.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
est/testFec/test_fec.cc
est/testRateControl/testRateControl.cc
est/testTMMBR/testTMMBR.cc
|
| 19da719a5febb4baa6e5dcdef8270792f9d31d6d |
05-Apr-2013 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Resolves TSan v2 reports data races in voe_auto_test.
--- Note that I will add more fixes to this CL ---
BUG=1590
Review URL: https://webrtc-codereview.appspot.com/1286005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3770 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_sender.cc
ource/rtp_rtcp_impl.cc
|
| b5bf54c4e7efa1bd37ec42b9150f9746b015cd79 |
05-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Permit arbitrary payload names for kVideoCodecGeneric.
BUG=1575
Review URL: https://webrtc-codereview.appspot.com/1282005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3768 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_payload_registry.cc
ource/rtp_payload_registry_unittest.cc
ource/rtp_sender_video.cc
|
| 79b0289bfc9f425d15442b1ecd73c2ae69646326 |
04-Apr-2013 |
edjee@google.com <edjee@google.com@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds event traces and counters for WebRTC receive side.
Review URL: https://webrtc-codereview.appspot.com/1279005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3766 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_receiver.cc
ource/rtp_receiver_audio.cc
ource/rtp_receiver_video.cc
|
| e1a719386935a72d9489fcd7a078bf8fd76eb39f |
27-Mar-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix flakiness in network up/down event tests when running under memcheck.
TBR=pwestin@webrtc.org
BUG=1524
Review URL: https://webrtc-codereview.appspot.com/1261005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3732 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_sender.cc
|
| bfacda60be5f816a04bd278d4aa4cd3d8fd01e9f |
27-Mar-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add interface to signal a network down event.
- In real-time mode encoding will be paused until the network is back up.
- In buffering mode the encoder will keep encoding, and packets will be
buffered at the sender. When the buffer grows above the target delay
encoding will be paused.
- Fixes a couple of issues related to pacing which was found with the new test.
- Introduces different max bitrates for pacing and for encoding. This allows
the pacer to faster get rid of the queue after a network down event.
(Work based on issue 1237004)
BUG=1524
TESTS=trybots,vie_auto_test
Review URL: https://webrtc-codereview.appspot.com/1258004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3730 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
ource/rtp_sender_video.cc
|
| d8a6e72057ec3ecc16833694f1ff6658f5f66db9 |
26-Mar-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust.
BUG=
Review URL: https://webrtc-codereview.appspot.com/1232005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3726 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_utility.cc
|
| 8911ce46a4c76c09b8c58828532836c9cd95549d |
18-Mar-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Generic video-codec support.
Labels frames as key/delta, also marks the first RTP packet of a frame as such,
to allow proper reconstruction even if packets are received out of order.
BUG=1442
TBR=ajm@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1207004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3680 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_format_video_generic.h
ource/rtp_payload_registry.cc
ource/rtp_receiver_video.cc
ource/rtp_rtcp.gypi
ource/rtp_sender.cc
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
ource/rtp_utility.cc
ource/rtp_utility.h
|
| 41211466d8b67769c8b3837d3401b2c824c6e337 |
18-Mar-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert the deletion of test_api_nack.cc in r3674.
TBR=mflodman@webrtc.org, mikhal@webrtc.org
BUG=1513
Review URL: https://webrtc-codereview.appspot.com/1217004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3677 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/nack_rtx_unittest.cc
ource/rtp_rtcp_tests.gypi
est/testAPI/test_api_nack.cc
|
| bda7f305c5d7d675f1c35813bd2b2a5732775bb9 |
16-Mar-2013 |
mikhal@webrtc.org <mikhal@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding RTX on source
Review URL: https://webrtc-codereview.appspot.com/1190004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3674 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/nack_rtx_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_tests.gypi
ource/rtp_sender.cc
ource/rtp_sender.h
est/testAPI/test_api.cc
est/testAPI/test_api.h
est/testAPI/test_api_nack.cc
|
| b7edd065306329309dac6767fe4914c185f941f8 |
12-Mar-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove DTMF detection. Talk team has been in the loop and there is no need for
DTMF detection at the receiver side.
test=voe_auto_test, VoE extended test DTMF
Review URL: https://webrtc-codereview.appspot.com/1168004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3663 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
est/testAPI/test_api_audio.cc
|
| 03e3117d87e7b70d2658cdd4141b1bc5239ba11d |
12-Mar-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removed redundant VP8 width/height and made sure the generic width/height is set.
Review URL: https://webrtc-codereview.appspot.com/1158005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3656 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_receiver_video.cc
|
| 1dc0aa2de286ba53692a548513c685909cfc0dab |
05-Mar-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix for build error on android introduced with r3609.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1164004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3611 4adac7df-926f-26a2-2b94-8c16560cd09d
est/testAPI/test_api_nack.cc
|
| a27107004d8544c6dbf8eaa231e6079b73c90efe |
05-Mar-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Split the NACK list into multiple RTCPs if it's too big.
TEST=rtp_rtcp_unittests
BUG=1434
Review URL: https://webrtc-codereview.appspot.com/1148006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3609 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_sender.cc
ource/rtp_rtcp_config.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
est/testAPI/test_api_nack.cc
|
| 44f85a49d8baf36a6521bdde7a768179a9266c07 |
04-Mar-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixed coverity defects (CID 14657 and 14656).
BUG=
Review URL: https://webrtc-codereview.appspot.com/1153006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3597 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_receiver_audio.cc
|
| 603ae3ece2d3b4167eb0b88362866b4fa0eb0f4f |
01-Mar-2013 |
bemasc@google.com <bemasc@google.com@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make RtpHeaderExtensionMap::Register and ::Deregister idempotent.
This CL changes the return code of these methods to indicate
success instead of failure when there is nothing to change.
This change appears to resolve an issue where enabling the
timestamp offset extension via SDP would result in a failure if
that extension had already been enabled.
Review URL: https://webrtc-codereview.appspot.com/1118008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3588 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_header_extension.cc
ource/rtp_header_extension_unittest.cc
|
| e1c4ed958da4a269e59263ca0531b64ee1fd95c7 |
27-Feb-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix to send a full NACK list at least roughly once every 1.5 x RTT.
BUG=1434
Review URL: https://webrtc-codereview.appspot.com/1111007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3576 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| 59b2d5fbce3cee1ccaf5e23ce8ece9e315bae2d0 |
20-Feb-2013 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Stop and restart fix.
BUG=1398
TEST=Local stop and start test.
Review URL: https://webrtc-codereview.appspot.com/1115004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3545 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| 0cb48a0a18a5fa40107b83c147101c9cef85e116 |
11-Feb-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Set SingleStream BWE in unittests.
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1094004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3494 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender_unittest.cc
|
| a7303bdfb5c2f16e1c8d7189a2a315a6f0b5373f |
05-Feb-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Lint-cleaned video and audio receivers.
BUG=
TESTED=trybots
Review URL: https://webrtc-codereview.appspot.com/1093004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3471 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
|
| 244251a9cd283575b27b0b4ab3beddb069e6fa9d |
04-Feb-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Moved almost all payload-related stuff to the payload registry.
The big benefit is we no longer have a circular dependency between the media receiver and the payload registry. The payload registry is starting to take a bit more place on the stage, and now knows how to do different things depending on audio or video.
BUG=
TESTED=rtp_rtcp_unittests, vie_auto_test, voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/1078004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3465 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/mock/mock_rtp_payload_strategy.h
ource/rtp_payload_registry.cc
ource/rtp_payload_registry.h
ource/rtp_payload_registry_unittest.cc
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_tests.gypi
|
| fa53d8717cacd3fe82e63d0d96089d8d22034214 |
04-Feb-2013 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixing/disabling Windows x64 warnings
Disabled MSVC #4267 warnings in common.gypi to enable x64 builds
for Windows.
Fixed MSVC #4267 warnings in test/testsupport.
Added third_party/directxsdk to .gitignore.
With http://review.webrtc.org/1070008 landed, this should make it possible
to build for x64 on Windows.
BUG=1348
TEST=Compiling with http://review.webrtc.org/1070008 applied:
set GYP_DEFINES="target_arch=x64"
set GYP_GENERATORS=ninja
gclient sync
ninja -C out\Debug_x64
Review URL: https://webrtc-codereview.appspot.com/1060008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3464 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_tests.gypi
|
| becf9c897c41eea3f021f99d87889c32c78b0de9 |
01-Feb-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix mismatch between different NACK list lengths and packet buffers.
This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors.
BUG=1289
Review URL: https://webrtc-codereview.appspot.com/1065007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtcp_receiver.cc
ource/rtcp_receiver_help.cc
ource/rtcp_receiver_help.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender_unittest.cc
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_rtcp_config.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
est/testAPI/test_api.cc
est/testAPI/test_api_nack.cc
est/testAPI/test_api_video.cc
|
| b5865079868c4dec49571e7aef0aa52124b50c64 |
01-Feb-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages.
Also make sure RTT is computed independently of whether it's time to send RTCP messages or not.
BUG=1298
Review URL: https://webrtc-codereview.appspot.com/1060005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3455 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender_unittest.cc
ource/rtp_rtcp_config.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
est/testAPI/test_api_rtcp.cc
|
| 63e09640392ad55e487da4c6a11ddd6d578a883b |
29-Jan-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix webrtc compilation errors for Chrome Win64
Mostly disabling warnings in the gyp files.
BUG=1348
BUG=http://crbug.com/166496
BUG=http://crbug.com/167187
Review URL: https://webrtc-codereview.appspot.com/1063011
Patch from Justin Schuh <jschuh@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3423 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp.gypi
|
| 43da54a458a7a992c702d85f0327e1d394ec5cf3 |
25-Jan-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reformatted rtp_sender: made lint clean.
TESTED=rtp_rtcp_unittests
BUG=
Review URL: https://webrtc-codereview.appspot.com/1062004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3412 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 5accd370e70b94517a39e622c75b794cc7a28829 |
22-Jan-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
RTP Receiver is now only deals with a receiver strategy. Cleaned up dependencies.
BUG=
TESTED=vie/voe_auto_test, rtp_rtcp_unittests
Review URL: https://webrtc-codereview.appspot.com/1058004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3397 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/mock/mock_rtp_receiver_video.h
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| a678a3baee2e680bd521f3a6caf97707fffd6093 |
21-Jan-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests.
TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots
Review URL: https://webrtc-codereview.appspot.com/1044004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/bitrate.cc
ource/bitrate.h
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_packet_history.cc
ource/rtp_packet_history.h
ource/rtp_packet_history_unittest.cc
ource/rtp_receiver.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender_audio.cc
ource/rtp_sender_audio.h
ource/rtp_sender_unittest.cc
est/testAPI/test_api.cc
est/testAPI/test_api.h
est/testAPI/test_api_audio.cc
est/testAPI/test_api_nack.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
|
| a3c82bf6673a2e0367bcb89a287cdc9ec0c37a53 |
19-Jan-2013 |
wjia@webrtc.org <wjia@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove '<(library)' in gyp files.
This will remove all usage of '<(library)' in all webrtc gyp files.
Review URL: https://webrtc-codereview.appspot.com/1049005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3392 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp.gypi
est/bwe_standalone.gypi
|
| efae5d59016ebdf959bf5970e36edcd31c9d9867 |
17-Jan-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Extracted rtp receiver payload management to its own class, made video receiver depend on that instead.
Eliminated need for video receiver to talk to its parent. Also we will now determine if the packet is the first one already in the rtp general receiver. The possible downside would be that recovered video packets no longer can be flagged as the first packet, but I don't think that can happen. Even if it can happen, maybe the bit was set anyway at an earlier stage. The tests run fine.
BUG=
TEST=rtp_rtcp_unittests, vie_auto_test, voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/1022011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3382 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_payload_registry.cc
ource/rtp_payload_registry.h
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp.gypi
|
| 20ed36dada62ad56ec01263fc0eef0ed198f6476 |
17-Jan-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Break out RtpClock to system_wrappers and make it more generic.
The goal with this new clock interface is to have something which is used all
over WebRTC to make it easier to switch clock implementation depending on where
the components are used. This is a first step in that direction.
Next steps will be to, step by step, move all modules, video engine and voice
engine over to the new interface, effectively deprecating the old clock
interfaces. Long-term my vision is that we should be able to deprecate the clock
of WebRTC and rely on the user providing the implementation.
TEST=vie_auto_test, rtp_rtcp_unittests, trybots
Review URL: https://webrtc-codereview.appspot.com/1041004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ource/bitrate.cc
ource/bitrate.h
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtp_packet_history.cc
ource/rtp_packet_history.h
ource/rtp_packet_history_unittest.cc
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_audio.cc
ource/rtp_sender_audio.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
ource/rtp_utility.cc
ource/rtp_utility.h
est/testAPI/test_api.h
|
| acfdd96ee3d23ad9c77df18523bf6d154deb390e |
16-Jan-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reformatted rtp_rtcp_impl*.
BUG=
TEST=Trybots.
Review URL: https://webrtc-codereview.appspot.com/1035004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3374 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| a22a9bd9ca66e98f2d51ea082dec8481f2f39e6e |
14-Jan-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional.
The audio receiver is now completely independent of rtp_receiver: video will hopefully be too in the next patch.
BUG=
TEST=vie & voe_auto_test full runs
Review URL: https://webrtc-codereview.appspot.com/1014006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3372 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ource/mock/mock_rtp_receiver_video.h
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_strategy.cc
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp_impl.cc
|
| f908011eb422bc077e32209c67e74b2f9a9a8182 |
11-Jan-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove extra line.
TBR=elham
Review URL: https://webrtc-codereview.appspot.com/1024008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3365 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver.cc
|
| 2f225cadde8627a64d2cede283965bac25a2807c |
09-Jan-2013 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add logs when no RTCP RR has been received for three regular RTCP intervals.
BUG=1267
TEST=Unittest added.
Review URL: https://webrtc-codereview.appspot.com/1019006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3346 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_unittest.cc
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| c38eef896a483c5d4a2975d76060c9942031a94a |
07-Jan-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reformatted RTPReceiver.
This is a pure reformat patch, with the exception that I also fixed all comments and moved a constant. I did not change the types in this patch since I
though that is more risky, so I'll do that in a separate patch later (perhaps
we could purge the types from the whole module in one go?)
BUG=
TEST=Trybots, vie_ & voe_auto_test --automated
Review URL: https://webrtc-codereview.appspot.com/998007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3338 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
ource/Bitrate.h
ource/bitrate.cc
ource/bitrate.h
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_receiver_video.h
ource/rtp_rtcp.gypi
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_video.h
|
| 1b6da28047ccc8ac50a2e2b09c142bea7679761a |
21-Dec-2012 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Bugfix for NACK behavior. Current code sends a number of duplicate NACK requests.
Landing of 573005 On behalf of an1kumar@gmail.com
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1002008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3322 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| ad0ed582b5c7c5aae4da924efd584700e21bb78f |
20-Dec-2012 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixed a missed initialization (found by valgrind FYI bot).
http://webrtc-cb-linux-master.cbf.corp.google.com:8011/builders/LinuxLargeTests/builds/327/steps/memory%20test%3A%20memcheck_voe_auto_test/logs/stdio
BUG=
TEST=Reproduced valgrind error, verified gone after fixing.
Review URL: https://webrtc-codereview.appspot.com/1008005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3318 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_receiver.cc
|
| 61f39a317425ece5bbc1a209b794c1ea7c043b32 |
18-Dec-2012 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixed bad header name.
TBR=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/1001008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3307 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp.gypi
|
| 07bf43c67303db4ab64b44f5b849465ec7dfef75 |
18-Dec-2012 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Replaced the _audio parameter with a strategy.
The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches.
In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on.
BUG=
TEST=vie/voe_auto_test, trybots
Review URL: https://webrtc-codereview.appspot.com/1001006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_strategy.cc
ource/rtp_receiver_strategy.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp.gypi
ource/rtp_rtcp_impl.cc
ource/rtp_utility.cc
ource/rtp_utility.h
|
| 3c37354b70e1b4058bf869af97ba3e4f69aef3d5 |
15-Dec-2012 |
fbarchard@google.com <fbarchard@google.com@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Initialize 3 variables which are preventing VS2012 from building.
BUG=1211
TESTED=ninja -C out\Release
Review URL: https://webrtc-codereview.appspot.com/992005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3301 4adac7df-926f-26a2-2b94-8c16560cd09d
est/testFec/test_fec.cc
|
| 7659d914bb201d65d1829ed0f0344adeac2fac49 |
14-Dec-2012 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Decoupled video rtp receiver from rtp receiver.
BUG=
Review URL: https://webrtc-codereview.appspot.com/995005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3292 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/mock/mock_rtp_receiver_video.h
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
|
| 92bb417cb1db62cb762a0af8de52c1514a05fe3e |
13-Dec-2012 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Decoupled RTP audio processor from RTP receiver.
BUG=
TEST=Ran vie_auto_test, rtp_rtcp_unittests, voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3279 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_rtcp_impl.cc
|
| 8d0cd07d0c5e30eacaac4c118f9fd624b11e67ab |
03-Dec-2012 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add test to verify that padding only frames are passing through the RTP module.
Review URL: https://webrtc-codereview.appspot.com/934023
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3224 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_sender.cc
ource/rtp_rtcp_tests.gypi
est/testAPI/test_api.cc
est/testAPI/test_api.gypi
est/testAPI/test_api.h
est/testAPI/test_api_audio.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
|
| f3cefe1104f705d818b9e2a129919c2f757718c3 |
29-Nov-2012 |
marpan@webrtc.org <marpan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Added metrics test code for the FEC packet masks.
The test computes metrics (average residual loss) for each mask type and size,
for a given set of loss models (random and bursty), and verifies various
behaviour of the codes (including relation/comparison to RS code).
http://webrtc-codereview.appspot.com/748008
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/929034
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3189 4adac7df-926f-26a2-2b94-8c16560cd09d
est/testFec/average_residual_loss_xor_codes.h
est/testFec/test_fec.gypi
est/testFec/test_packet_masks_metrics.cc
|
| c244cefe1d66f01ffdfdb588bbb6b2f660b1d4f4 |
28-Nov-2012 |
marpan@webrtc.org <marpan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reverting r3185
TBR=marpan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/933029
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3186 4adac7df-926f-26a2-2b94-8c16560cd09d
est/testFec/average_residual_loss_xor_codes.h
est/testFec/test_fec.gypi
est/testFec/test_packet_masks_metrics.cc
|
| 993494764da9d7e7e45e056b98927e240cbbdf0d |
28-Nov-2012 |
marpan@webrtc.org <marpan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Added metrics test code for the FEC packet masks.
The test computes metrics (average residual loss) for each mask type and size,
for a given set of loss models (random and bursty), and verifies various
behaviour of the codes (including relation/comparison to RS code).
Review URL: https://webrtc-codereview.appspot.com/748008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3185 4adac7df-926f-26a2-2b94-8c16560cd09d
est/testFec/average_residual_loss_xor_codes.h
est/testFec/test_fec.gypi
est/testFec/test_packet_masks_metrics.cc
|
| ef90c3227ebd4008bbcfabd17a9f422965f11a25 |
26-Nov-2012 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Will now correctly identify the first-ever received packet as the first packet in its frame.
We used to flag the _second_ packet in the first frame as the first. Subsequent frames worked as intended.
BUG=1103
TEST=vie_auto_test --automated, rtp_rtcp_unittests
Review URL: https://webrtc-codereview.appspot.com/964020
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3164 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/mock/mock_rtp_receiver_video.h
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
|
| 7c894b7cc718773f32d21985ff33a64f9e13946e |
26-Nov-2012 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Wire up CallStats to provide modules with correct RTT.
BUG=769
TEST=Manual test since there is no ViE APi to get RTT for receive channels.
Review URL: https://webrtc-codereview.appspot.com/937027
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3163 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
|
| 418443c53131d9fba0784f907a2c8599d971d8d6 |
23-Nov-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove operator overloading from RTPFragmentationHeader.
Instead supply a CopyFrom() method.
TEST=vie_auto_test
Review URL: https://webrtc-codereview.appspot.com/972004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3158 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_format_vp8.cc
|
| 1c611960952eaa16358942a6730f976cf381eeeb |
22-Nov-2012 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removed not used include.
TEST=Compiles.
Review URL: https://webrtc-codereview.appspot.com/966025
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3150 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtcp_sender.cc
|
| 4100b0402eea1fdea52e5899ee12e93c1f84b4db |
19-Nov-2012 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move SSRC list to RemoteBitrateEstimator.
BUG=1105
Review URL: https://webrtc-codereview.appspot.com/965027
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3130 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
|
| b2f474e8bb0385ef25b11fb4b75ca17e1f423a66 |
16-Nov-2012 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled.
This CL will be followed by another CL connecting the dots.
BUG=769
TEST=New unittest added.
Review URL: https://webrtc-codereview.appspot.com/968006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3117 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp_defines.h
|
| 571a1c035be6b0afd7f357001bef775c51ec9364 |
13-Nov-2012 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enable paced sender.
Review URL: https://webrtc-codereview.appspot.com/965016
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3089 4adac7df-926f-26a2-2b94-8c16560cd09d
nterface/rtp_rtcp.h
ocks/mock_rtp_rtcp.h
ource/rtp_rtcp.gypi
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_tests.gypi
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
ource/transmission_bucket.cc
ource/transmission_bucket.h
ource/transmission_bucket_unittest.cc
est/testAPI/test_api.cc
|
| 1726661ca26245c4b871d9144b64f605f52862b6 |
13-Nov-2012 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update parsed non ref frame info.
Review URL: https://webrtc-codereview.appspot.com/932015
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3084 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_receiver_video.cc
|
| c66e8b3f31db39d96bec6dc9ee9439455415a2be |
07-Nov-2012 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
pre-factor cleanup pre-work.
Review URL: https://webrtc-codereview.appspot.com/938010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3054 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_rtcp_impl.cc
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_unittest.cc
|
| e5b49a0472b97fa262b641b78cf4230bd824296f |
06-Nov-2012 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update timestamp offset for re-transmitted packets.
BUG=1059
Review URL: https://webrtc-codereview.appspot.com/930011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3046 4adac7df-926f-26a2-2b94-8c16560cd09d
ource/rtp_packet_history.cc
ource/rtp_packet_history.h
ource/rtp_packet_history_unittest.cc
ource/rtp_sender.cc
ource/rtp_sender.h
|
| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
WNERS
nterface/rtp_rtcp.h
nterface/rtp_rtcp_defines.h
ocks/mock_rtp_rtcp.h
ource/Android.mk
ource/Bitrate.h
ource/H264/bitstream_builder.cc
ource/H264/bitstream_builder.h
ource/H264/bitstream_parser.cc
ource/H264/bitstream_parser.h
ource/H264/h264_information.cc
ource/H264/h264_information.h
ource/H264/rtp_sender_h264.cc
ource/H264/rtp_sender_h264.h
ource/bitrate.cc
ource/dtmf_queue.cc
ource/dtmf_queue.h
ource/fec_private_tables_bursty.h
ource/fec_private_tables_random.h
ource/fec_test_helper.cc
ource/fec_test_helper.h
ource/forward_error_correction.cc
ource/forward_error_correction.h
ource/forward_error_correction_internal.cc
ource/forward_error_correction_internal.h
ource/mock/mock_rtp_receiver_video.h
ource/producer_fec.cc
ource/producer_fec.h
ource/producer_fec_unittest.cc
ource/receiver_fec.cc
ource/receiver_fec.h
ource/receiver_fec_unittest.cc
ource/rtcp_format_remb_unittest.cc
ource/rtcp_receiver.cc
ource/rtcp_receiver.h
ource/rtcp_receiver_help.cc
ource/rtcp_receiver_help.h
ource/rtcp_receiver_unittest.cc
ource/rtcp_sender.cc
ource/rtcp_sender.h
ource/rtcp_sender_unittest.cc
ource/rtcp_utility.cc
ource/rtcp_utility.h
ource/rtp_fec_unittest.cc
ource/rtp_format_vp8.cc
ource/rtp_format_vp8.h
ource/rtp_format_vp8_test_helper.cc
ource/rtp_format_vp8_test_helper.h
ource/rtp_format_vp8_unittest.cc
ource/rtp_header_extension.cc
ource/rtp_header_extension.h
ource/rtp_header_extension_unittest.cc
ource/rtp_packet_history.cc
ource/rtp_packet_history.h
ource/rtp_packet_history_unittest.cc
ource/rtp_receiver.cc
ource/rtp_receiver.h
ource/rtp_receiver_audio.cc
ource/rtp_receiver_audio.h
ource/rtp_receiver_video.cc
ource/rtp_receiver_video.h
ource/rtp_rtcp.gypi
ource/rtp_rtcp_config.h
ource/rtp_rtcp_impl.cc
ource/rtp_rtcp_impl.h
ource/rtp_rtcp_tests.gypi
ource/rtp_sender.cc
ource/rtp_sender.h
ource/rtp_sender_audio.cc
ource/rtp_sender_audio.h
ource/rtp_sender_unittest.cc
ource/rtp_sender_video.cc
ource/rtp_sender_video.h
ource/rtp_utility.cc
ource/rtp_utility.h
ource/rtp_utility_unittest.cc
ource/ssrc_database.cc
ource/ssrc_database.h
ource/tmmbr_help.cc
ource/tmmbr_help.h
ource/transmission_bucket.cc
ource/transmission_bucket.h
ource/transmission_bucket_unittest.cc
ource/video_codec_information.h
ource/vp8_partition_aggregator.cc
ource/vp8_partition_aggregator.h
ource/vp8_partition_aggregator_unittest.cc
est/BWEStandAlone/BWEConvergenceTest.cc
est/BWEStandAlone/BWEConvergenceTest.h
est/BWEStandAlone/BWEStabilityTest.cc
est/BWEStandAlone/BWEStabilityTest.h
est/BWEStandAlone/BWEStandAlone.cc
est/BWEStandAlone/BWETestBase.cc
est/BWEStandAlone/BWETestBase.h
est/BWEStandAlone/BWETester.cc
est/BWEStandAlone/BWETwoWayLimitFinding.cc
est/BWEStandAlone/BWETwoWayLimitFinding.h
est/BWEStandAlone/MatlabPlot.cc
est/BWEStandAlone/MatlabPlot.h
est/BWEStandAlone/TestLoadGenerator.cc
est/BWEStandAlone/TestLoadGenerator.h
est/BWEStandAlone/TestSenderReceiver.cc
est/BWEStandAlone/TestSenderReceiver.h
est/bitstreamTest/bitstreamTest.cc
est/bwe_standalone.gypi
est/testAPI/test_api.cc
est/testAPI/test_api.gypi
est/testAPI/test_api.h
est/testAPI/test_api_audio.cc
est/testAPI/test_api_nack.cc
est/testAPI/test_api_rtcp.cc
est/testAPI/test_api_video.cc
est/testFec/test_fec.cc
est/testFec/test_fec.gypi
est/testH263Parser/testH263Parser.cc
est/testRateControl/testRateControl.cc
est/testTMMBR/testTMMBR.cc
|