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
|