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Author SHA1 Message Date
kwiberg@webrtc.org
0552356fda isacfix: Refactor big-endian reading and writing 
Make subroutines for encoding and decoding arrays of 16-bit big-endian
integers, and in the process fix a bug: When decoding an odd number of
bytes from be16, the least significant byte of the last int16 in the
array was properly taken to be zero instead of actually being read
(since it's outside the array). However, when encoding an odd number
of bytes, the least significant byte of the last int16 in the array
was written to the output as-is instead of being taken to be zero;
thus, we encoded one byte more than we should. This was probably not
harmful, and the value was dropped at decoding anyway; nevertheless,
writing a constant zero is the safe thing to do, and this patch does
so.

R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7527 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 11:25:37 +00:00
pbos@webrtc.org
9fed099208 Increase max trace message size to 1024 characters. 
A recent CL by pbos:
https://code.google.com/p/webrtc/source/detail?r=7518

added long log messages and triggered errors on the DrMemory bot due to
WEBRTC_TRACE. The trace mechanism _should_ truncate the log strings
but something appears to be going awry.

This sweeps the problem under the rug, but given that WEBRTC_TRACE
should die fairly soon, seems to be a reasonable tradeoff.

TEST=passing try on DrMemory.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27849004

Patch from Andrew MacDonald <andrew@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7526 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 09:31:05 +00:00
pbos@webrtc.org
c86ec3e3bc Fix ::~LogMessage to print as a string. 
R=andrew@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/26949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7525 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 09:22:03 +00:00
braveyao@webrtc.org
1732df6129 Use flags set by the port allocator. 
Currently, port allocator flags are ignored. This is inconvenient if you
want to have your own PortAllocatorFactory subclass.

BUG=webrtc:3958
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7524 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 03:01:37 +00:00
kjellander@webrtc.org
3b839d008f PRESUBMIT: Add linux_msan to default trybots. 
Will commit as soon it's online.

BUG=
R=pbos@webrtc.org
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7523 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-24 21:41:24 +00:00
buildbot@webrtc.org
3f7bcc126d (Auto)update libjingle 78430441-> 78445452 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7522 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-24 17:26:28 +00:00
buildbot@webrtc.org
c7ed8db7fd (Auto)update libjingle 78427027-> 78430441 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7521 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-24 12:59:08 +00:00
perkj@webrtc.org
470988742a Add HD support to Android if we detect a hardware video encoder that can be used. This Change the internal class MediaCodecVideoEncoder to have a one public method for checking if the platform is supported. It also adds &hd=true to the reqest url a hardware encoder is detected. 
BUG=3934
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7520 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-24 11:38:19 +00:00
houssainy@google.com
39b1743116 Adding the subtool rtcBot report visualizer 
This tool for visualize the output reports of rtcBot by calculating
the average and max of a specific stats and plot the output.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7519 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-24 09:26:16 +00:00
pbos@webrtc.org
ad3b5a5c16 Move min transmit bitrate to VideoEncoderConfig. 
min_transmit_bitrate_bps needs to be reconfigurable during a call (since
this is currently set only for screensharing through libjingle and can't
be set once and for all for the entire Call.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7518 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-24 09:23:21 +00:00
pthatcher@webrtc.org
c9d6d14020 patch from issue 25469004 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 23:37:22 +00:00
buildbot@webrtc.org
8fe75ee234 (Auto)update libjingle 78381351-> 78389679 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7516 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 23:07:23 +00:00
buildbot@webrtc.org
fb5e9fc44e (Auto)update libjingle 78344087-> 78381351 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7515 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 21:36:17 +00:00
aluebs@webrtc.org
7e19a11a71 Break out WebRtcNs_ComputeDdUpdate function in ns_core 
This is done in order to make the code more readible and maintainable.
It generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7514 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 19:54:33 +00:00
aluebs@webrtc.org
f8ea0d5518 Break out WebRtcNs_UpdateNoise function in ns_core 
This is done in order to make the code more readible and maintainable.
It generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7513 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 19:49:42 +00:00
aluebs@webrtc.org
799e88ae19 Break out FFT function in ns_core 
This is done in order to make the code more readible and maintainable.
This introduces an error of only +1 and -1.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7512 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 19:36:42 +00:00
aluebs@webrtc.org
8454ad88ed Break out ComputeSnr function in ns_core 
This is done in order to make the code more readible and maintainable.
The output is bit-exact.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7511 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 19:34:14 +00:00
houssainy@google.com
0d3e254c89 Adding three video conference bots test 
A video conference between three bots, each bot creating two
peerConnections, and each peer connected to one of the other bots.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7510 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 16:45:07 +00:00
houssainy@google.com
0e19d0c2aa Adding file from test.webrtc.org domain to be downloaded 
This has been configured to allow cross domain to access this generated
file:
https://test.webrtc.org/test-download-file/9000KB.data

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7509 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 15:41:30 +00:00
asapersson@webrtc.org
580d367b14 Add macros and APIs for webrtc histograms. 
BUG=crbug/419657

Code that links system_wrappers.gyp:system_wrappers should either:
- provide implementations for the APIs, or
- link with default implementations in system_wrappers.gyp:system_wrappers_default.

R=andresp@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7508 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 12:57:56 +00:00
buildbot@webrtc.org
9d446f2e16 (Auto)update libjingle 78296920-> 78342456 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7507 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 12:22:06 +00:00
kjellander@webrtc.org
8539bd0184 Download full Chromium checkouts by default 
This changes sync_chromium.py to download a full Chromium
checkout instead of one with no history. It has been noticed
that the download of the no-history checkout is very slow, even
when on high-speed internet connections, due to current limitations
in the Git backend serving these clones.
Switching to a full checkout is faster, but requires more bandwidth
and disk space.

To keep the old behavior, users must set the CHROMIUM_NO_HISTORY
environment variable to 1.

Using a full checkout also enables the use of the Chromium
infrastructure teams' Git cache functionality, that speeds up
the initial download and also heavily reduces the traffic when
setting up multiple checkouts on the same machine.
This is not enabled by default, but is supported if the user is
setting the cache_dir variable in his checkout's .gclient file to
point at a directory on local disk.

BUG=3882
TESTED=
* Ran gclient sync and verified chromium/src now contained a Git
repo with full history.
* Tested rolling chromium_revision in DEPS forward + sync.
* Tested rolling it back again + sync.
* Tested with an existing no-history checkout:
  CHROMIUM_NO_HISTORY=1 gclient sync
  No change was performed.
* Tested with a .gclient that had cache_dir configured.
* Verified error message is displayed when .gclient has cache_dir
  configured and CHROMIUM_NO_HISTORY=1.

R=iannucci@chromium.org

Review URL: https://webrtc-codereview.appspot.com/22869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7506 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 12:17:58 +00:00
stefan@webrtc.org
82462aade0 Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate. 
Also wires up a finch experiment to control this.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7505 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 11:57:05 +00:00
houssainy@google.com
2192701135 Using the Unused turn configuration in two way test 
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7504 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 08:40:53 +00:00
pbos@webrtc.org
ad553a2731 Let video_loopback use internal VCM capturers. 
R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7503 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 08:24:02 +00:00
andrew@webrtc.org
15c717beee Add a memcheck exclusion for EndToEndTest.CanSwitchToUseAllSsrcs. 
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7502 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 05:37:37 +00:00
buildbot@webrtc.org
a9f0898e7d (Auto)update libjingle 78273470-> 78296920 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7501 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 22:02:00 +00:00
glaznev@webrtc.org
7bb4a9881d Merging Henrik's and Peter's changes for AppRTCDemo 
from https://github.com/hkjellander/AppRTCDemo.

Description of changes:
- Add connect screen with an option to enter room number or select loopback mode.
- Add 'hangup' and 'WebRTC statistics' buttons to AppRTCDemo activity.

BUG=3938
R=kjellander@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7500 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 17:43:37 +00:00
houssainy@google.com
fce8f5d319 NOTE: This code review based on the running issue: 
https://webrtc-codereview.appspot.com/24939004/

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7499 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 17:24:20 +00:00
houssainy@google.com
3382059e55 Adding Two way video and audio streaming test to RtcBot 
NOTE: This code review based on this running issue:
https://review.webrtc.org/24939004/

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7498 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 17:17:15 +00:00
houssainy@google.com
e9b7d03db6 HTTPS Server used instead of HTTP for loading the bots to avoid the media permission pop-up clicks every time running the test. 
This code review based on the running issue:
https://webrtc-codereview.appspot.com/24939004/

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7497 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 16:34:25 +00:00
buildbot@webrtc.org
fb5410a8b7 (Auto)update libjingle 78262388-> 78262615 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7496 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 15:45:17 +00:00
pbos@webrtc.org
eacc6e4657 Remove some disabled tests in WebRtcVideoEngine2. 
Removes some tests that shouldn't have to be implemented or have already
been through other tests.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/25929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7495 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 15:36:54 +00:00
kjellander@webrtc.org
82e430c316 Suppress libyuv uninitialized read in CopyRow_AVX 
BUG=libyuv:377
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7494 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 13:51:49 +00:00
pbos@webrtc.org
32452b20b8 Make ReconfigureVideoEncoder use current bitrate. 
Prevents bitrate drops when changing resolution etc.

R=stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/24069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7493 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 12:15:24 +00:00
kjellander@webrtc.org
860ccc9407 Tighten up MSan blacklist.txt owners. 
To avoid people adding stuff to the blacklist unless
it's really valid to do so.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7492 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 11:20:07 +00:00
pbos@webrtc.org
3f8f5554a0 Disable TestVp8Impl.BaseUnitTest on MSan. 
MemorySanitizer uses generic (non-optimized) libvpx which is not bit
exact. This may be a bug in upstream libvpx, but we're forced to disable
it now as it blocks launching the MSan bot.

R=stefan@webrtc.org
TBR=marpan@webrtc.org
BUG=3904

Review URL: https://webrtc-codereview.appspot.com/24089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7491 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 10:30:30 +00:00
stefan@webrtc.org
76960d5f74 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
 2014-10-22 09:47:14 +00:00
kjellander@webrtc.org
1d9af96c06 Roll chromium_revision de13cf4..28d1981 (299488:300483) 
Mainly to pick up https://codereview.chromium.org/656293004/
to fix some MSan issues.

Summary of changes (de13cf4..28d1981/DEPS):
* third_party/android_tools d2b8620..36bf7ac
* third_party/libyuv 455c66b..5a09c3e (1038:1130)
* third_party/usrsctp/usrsctplib a11b3c5..7accb99
* tools/gyp 1977:1990
* tools/swarming_client c28b74f..a57d7db

Clang updated 217949:218707 (git diff de13cf4..28d1981 tools/clang/scripts/update.sh)

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 06:43:29 +00:00
aluebs@webrtc.org
67cf1d742b Break out WebRtcNs_Windowing function in ns_core 
This is done in order to make the code more readible and maintainable.
This introduces only +1 and -1 errors.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7488 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 22:35:40 +00:00
aluebs@webrtc.org
0e7099244c Break out WebRtcNs_Energy function in ns_core 
This is done in order to make the code more readible and maintainable.
This generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7487 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 22:14:10 +00:00
aluebs@webrtc.org
7634c09406 Break out WebRtcNs_IFFT function in ns_core 
This is done in order to make the code more readible and maintainable.
This creates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7486 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 21:27:00 +00:00
buildbot@webrtc.org
a5c36b397a (Auto)update libjingle 78193292-> 78199328 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7485 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 20:44:16 +00:00
guoweis@webrtc.org
b6173abe59 Fix local address leakage when IceTransportsType is relay 
As part of implementing IceTransportsType constraint, we should hide the raddr which is the mapped address to prevent local address leakage.

BUG=1179
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7484 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 20:40:21 +00:00
aluebs@webrtc.org
333e2556ed Break out WebRtcNs_UpdateBuffer function in ns_core 
This is done in order to make the code more readible and maintainable.
It generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7483 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 20:33:09 +00:00
buildbot@webrtc.org
1288cbb704 (Auto)update libjingle 78106439-> 78193292 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7482 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 19:29:16 +00:00
henrik.lundin@webrtc.org
def1e97ed2 Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests 
BUG=3926
R=kjellander@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7481 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 12:48:29 +00:00
bjornv@webrtc.org
78ea06dd34 audio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
Removed usage of trivial macro.

BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7480 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 07:17:24 +00:00
henrik.lundin@webrtc.org
913f7b8d5e Fix for glitches in ACM when switching desired output sample rate 
The problem was that if the output sample rate is changed such from one
where no resampling is needed to a rate that requires resampling, the
first output from the resampler will contain an onset period. The
solution provided in this CL is to keep a copy of the last output frame
in ACM, and if the resampler is engaged, it will be primed with this
old frame before resampling the current frame.

BUG=3919
R=bjornv@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 06:54:23 +00:00
glaznev@webrtc.org
a8c0edd29f Avoid using EGLContext class for Android 4.1 and below. 
Support for this class was added in Android 4.2, so
disable surface decoding for lower Android versions.

BUG=3901
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7478 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 19:08:05 +00:00