| 6db6cdc604f9a866991ecf8454eb7f7aa69918ea |
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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}
/external/webrtc/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
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| 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/fec_receiver_impl.h
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| 14665ff7d4024d07e58622f498b23fd980001871 |
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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
/external/webrtc/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
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| 00b8f6b3643332cce1ee711715f7fbb824d793ca |
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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
/external/webrtc/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
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| 0800db74b991dec8ef750c428eb611360a1286f4 |
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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
/external/webrtc/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
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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/fec_receiver_impl.h
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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/fec_receiver_impl.h
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| 7bb8f02274ecbfa1f7ef134d708369a369a78c83 |
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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
/external/webrtc/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
|