c4a1c370aa7e4ec467ff16162ca0eef0cfaf53b0 |
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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}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
|
ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
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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}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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98f53510b222f71fdd8b799b2f33737ceeb28c61 |
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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}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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affa39cb39c77408109fef691533021533d969e1 |
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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}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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e23e737177cf5d131a6d4a4d229aa513c5270a59 |
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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}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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c32d2db69bc94480ecb312268c6e6769d4a1cac6 |
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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}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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c957ffc6dc36879e5ad72d7f0af2a014707d70fa |
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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
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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43c883954f5edc84bd8e0e901ef770fead218ed5 |
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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
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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4591fbd09f9cb6e83433c49a12dd8524c2806502 |
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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
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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420b2567f38241099907d30d8bece1c4db50262d |
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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
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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dc80bae2a62a1bdbe0d342b3260a7e5b2cb958df |
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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
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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79b63206b99912d9a5f97a35b546409886a8fed2 |
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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
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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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
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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a048d7cb0a5bad5ca49bbcc5273cb4cca28c1710 |
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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
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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a678a3baee2e680bd521f3a6caf97707fffd6093 |
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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
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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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
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_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
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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14b43beb7ce4440b30dcea31196de5b4a529cb6b |
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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
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
|