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17 Commits

Author SHA1 Message Date
mflodman@webrtc.org
19fc09efba Adding missing break in media_file_utility.cc. 
There has been no reports of problems, but adding this to get it correct.

Review URL: https://webrtc-codereview.appspot.com/19599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 05:21:56 +00:00
fischman@webrtc.org
2c89b5cb27 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
 2014-04-14 20:08:03 +00:00
henrike@webrtc.org
79cf3acc79 Removes usage of ListWrapper from several files. 
BUG=2164
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 15:21:30 +00:00
andrew@webrtc.org
de7c9e8884 Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0. 
Move the logic to common.gypi to reduce the chance of the define being
unprotected in the future.

BUG=b/12018143
TESTED=git try, and local Linux build with -Denable_video=0
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5244 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-09 16:23:00 +00:00
stefan@webrtc.org
b082ade3db Hook up audio/video sync to Call. 
Adds an end-to-end audio/video sync test.

BUG=2530, 2608
TEST=trybots
R=henrika@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5128 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 11:45:11 +00:00
andrew@webrtc.org
eda189be14 Remove redundant STR_CASE_CMP macro definitions. 
R=minyue@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2187005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4711 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 17:50:10 +00:00
fischman@webrtc.org
f696f253b2 Invert dependency between webrtc_utility and media_file targets to reflect reality. 
BUG=2166
R=henrike@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1953004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4488 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 18:45:19 +00:00
pbos@webrtc.org
0c4e05afbb Include files from webrtc/.. paths in media_file/. 
BUG=1662
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1784005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4351 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 13:05:40 +00:00
henrike@webrtc.org
a950300b0e Disables unit tests that don't work on Android for Android. 
BUG=N/A
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1747004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4306 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-08 18:53:54 +00:00
pbos@webrtc.org
d900e8bea8 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
 2013-07-03 15:12:26 +00:00
kjellander@webrtc.org
6c35e0b0f7 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
 2013-06-11 08:29:17 +00:00
pbos@webrtc.org
6e788df19e 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
 2013-04-16 12:40:34 +00:00
pbos@webrtc.org
0ea11c1768 WebRtc_Word32 -> int32_t in media_file/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1304005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3796 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:31:37 +00:00
andrew@webrtc.org
63e0964039 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
 2013-01-29 06:45:22 +00:00
wjia@webrtc.org
a3c82bf667 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
 2013-01-18 23:42:21 +00:00
andrew@webrtc.org
00c7c4315b Replace voice engine utility functions with system wrapper variants. 
SLEEP -> SleepMs
GET_TIME_IN_MS -> TickTime::MillisecondTimestamp

These could cause unused function errors on some compilers.

BUG=1228

Review URL: https://webrtc-codereview.appspot.com/1013004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3326 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-02 16:06:39 +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