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15 Commits

Author SHA1 Message Date
kjellander@webrtc.org
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro 
Clang version changed 223108:230914
Details: 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
 2015-03-04 13:04:54 +00:00
pkasting@chromium.org
4591fbd09f 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
 2014-11-20 22:28:14 +00:00
stefan@webrtc.org
6071b0636d 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
 2014-09-12 07:42:33 +00:00
wu@webrtc.org
822fbd8b68 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
 2013-08-15 23:38:54 +00:00
tnakamura@webrtc.org
aa4d96a134 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
 2013-07-16 19:25:04 +00:00
pbos@webrtc.org
a4407329d4 Include files from webrtc/.. paths in video_coding/. 
BUG=1662
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1783006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 12:32:05 +00:00
stefan@webrtc.org
66b2e5c05a 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
 2013-07-05 14:30:48 +00:00
pbos@webrtc.org
77f6b2175e 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
 2013-05-03 12:02:11 +00:00
pbos@webrtc.org
68e5a68f07 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
 2013-05-02 09:30:12 +00:00
pbos@webrtc.org
265a5d298a 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
 2013-05-02 09:11:20 +00:00
solenberg@webrtc.org
56b5f77a2b Add support for multiple streams to RtpPlayer: 
- Tests video_rtp_play.cc, video_rtp_play_mt.cc, decode_from_storage.cc rewritten
 - rtp_player.cc/.h rewritten; added interfaces for externally setting up sinks
 - Support for reading .rtp files pulled out into rtp_file_reader namespace
 - Added support for reading .pcap (libpcap/wireshark/tcpdump) files, see pcap_file_reader

BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/1201009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3856 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 10:31:56 +00:00
pbos@webrtc.org
7b859cc1e9 Webrtc_Word32 => int32_t in video_coding/main/ 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3753 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 15:54:38 +00:00
stefan@webrtc.org
a4b58860b7 Add a counter to the video rtp play output filename. 
Review URL: https://webrtc-codereview.appspot.com/1040004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-17 09:27:17 +00:00
mikhal@webrtc.org
9fedff7c17 Switching to I420VideoFrame 
Review URL: https://webrtc-codereview.appspot.com/922004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2983 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-24 18:33:04 +00:00
andrew@webrtc.org
14b43beb7c 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
 2012-10-22 18:19:23 +00:00