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Author SHA1 Message Date
kjellander@webrtc.org
d1cfa7149e TSan v2 suppressions and exclusions for libjingle tests. 
Add suppressions for libjingle tests so they pass under TSan v2.
Disable the following tests for TSan v2 (only) since they're failing:
* StunServerTest.TestGood
* JsepPeerConnectionP2PTestClient.*

See build logs at:
http://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20Tsan%20v2/
for more details.

BUG=1205,2078,2079,2080,2517
TEST=Ran a successful run of each test locally on Linux using:
GYP_DEFINES='tsan=1 linux_use_tcmalloc=0 release_extra_cflags="-gline-tables-only"' gclient runhooks
ninja -C out/Release
For each test, run standing in trunk/:
TSAN_OPTIONS="suppressions=tools/valgrind-webrtc/tsan_v2/suppressions.txt print_suppressions=1 report_signal_unsafe=0 report_thread_leaks=0 history_size=7" out/Release/[libjingle_testname]
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2411004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4977 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 16:51:52 +00:00
mallinath@webrtc.org
6fa456f928 Disabling the DTLS renegotiation test case for PeerConnection. 
Currently it's failing on Linux memcheck, most likely due to timing issues.

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2394006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4962 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-15 00:11:54 +00:00
mallinath@webrtc.org
19f27e6a24 Update talk to 54527154. 
TBR=wu

Review URL: https://webrtc-codereview.appspot.com/2389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4954 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-13 17:18:27 +00:00
wu@webrtc.org
40dfbc4d3d Update talk to 53984350. 
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/2376004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4947 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-09 17:58:06 +00:00
wu@webrtc.org
4551b793de Update libjingle to 53920541. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2371004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4945 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-09 15:37:36 +00:00
wu@webrtc.org
7818752566 Update libjingle to 53856368. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2366004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4941 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 23:32:02 +00:00
kjellander@webrtc.org
7fca2ce097 Add owners to [webrtc,talk]/build and *.isolate (take 2) 
After fischman@'s comments in http://review.webrtc.org/2347006/ here's another CL to clean up the redundancies and add wu@ to webrtc/build/

TEST=none
BUG=none
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2348006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4928 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 19:36:45 +00:00
kjellander@webrtc.org
e6938185a5 Add isolate targets for libjingle 
Add .isolate file for libjingle tests and and the necessary isolate.gypi file, similar to the change in
http://review.webrtc.org/2338004/

TEST=trybots passing.
I also ran build/gyp_chromium in a Chromium checkout
with third_party/libjingle/source/talk having this patch
applied to ensure GYP processing was still working.

BUG=1916
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2353005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4926 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 19:31:27 +00:00
kjellander@webrtc.org
83b9e5b328 Add owners to [webrtc,talk]/build and *.isolate 
BUG=none
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2347006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4923 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 17:35:26 +00:00
fischman@webrtc.org
4446134757 AppRTCDemo(android): support boolean value for MediaStreamConstraints.{audio,video}. 
Previously it was assumed that these values were always MediaTrackConstraints but
http://dev.w3.org/2011/webrtc/editor/getusermedia.html#idl-def-MediaStreamConstraints
allows them to be boolean, too.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2352004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4918 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 22:34:10 +00:00
fischman@webrtc.org
a7266ca134 Fix clang build break 
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2350004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4917 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 19:04:18 +00:00
fischman@webrtc.org
6c82e04cee Android standalone: remove some usages of deprecated APIs and prevent further regressions. 
Also:
- Fixed WebRTCDemo UI to say "SwitchToBack" at startup since default camera is front
- Rebuild WebRTCDemo APK when resources/layout/strings change

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2337004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4916 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 18:57:48 +00:00
fischman@webrtc.org
4e65e07e41 VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. 
Besides being ~40% the size of the previous implementation, this makes it so
that VideoCaptureAndroid can stop and restart capture, which is necessary to
support onPause/onResume reasonably on Android.

BUG=1407
R=henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2334004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4915 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 18:23:13 +00:00
fischman@webrtc.org
ddc5a19ce9 AppRTCDemo(android): uncaught exceptions now display a modal dialog box before killing the app. 
BUG=2458
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2348004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4914 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 18:09:40 +00:00
fischman@webrtc.org
7e4d0df8ee PeerConnection(Android): enable tracing to logcat. 
BUG=1295
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2258007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4888 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-01 02:40:43 +00:00
mallinath@webrtc.org
7e809c323a Update libjingle to CL 53496343. 
Review URL: https://webrtc-codereview.appspot.com/2323005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4882 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-30 18:59:08 +00:00
mallinath@webrtc.org
ad81ab8861 Suppress SSL error strings on mac_asan to unbreak that build 
Example borkedness:
http://chromegw/i/client.webrtc/builders/Mac%20Asan/builds/642/steps/libjingl...

Original CL for this issue is here
https://webrtc-codereview.appspot.com/2263004/

and this got reverted in here
https://code.google.com/p/webrtc/source/diff?spec=svn4874&r=4872&format=side&path=/trunk/talk/base/openssladapter.cc&old_path=/trunk/talk/base/openssladapter.cc&old=4798

Trying to land it again now.

TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2318005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4875 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-28 07:24:39 +00:00
mallinath@webrtc.org
a27be8e4a1 Update libjingle to CL 53398036. 
Review URL: https://webrtc-codereview.appspot.com/2323004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4872 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-27 23:04:10 +00:00
andrew@webrtc.org
4475905613 Disable flaky RapidSpeakerChange test. 
Example:
chromegw/i/internal.client.webrtc/builders/Win32%20Debug/builds/762/steps/libjingle_p2p_unittest/logs/stdio

e:\b\build\slave\win32_debug\build\src\talk\session\media\currentspeakermonitor_unittest.cc(144):
error: Value of: kSsrc2
  Actual: 1002
Expected: current_speaker_
Which is: 1001
e:\b\build\slave\win32_debug\build\src\talk\session\media\currentspeakermonitor_unittest.cc(145):
error: Value of: 1
Expected: num_changes_
Which is: 2
[  FAILED  ] CurrentSpeakerMonitorTest.RapidSpeakerChange (16 ms)

TBR=wu@webrtc.org
BUG=2409

Review URL: https://webrtc-codereview.appspot.com/2318004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-27 18:24:40 +00:00
andrew@webrtc.org
2f240b43f5 Disable some flaky libjingle base tests. 
ThreadTest.Main and VirtualSocketServerTest.delay_v6

Example:
http://build.chromium.org/p/tryserver.webrtc/builders/win/builds/1234

TBR=wu@webrtc.org
BUG=2409

Review URL: https://webrtc-codereview.appspot.com/2297004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4838 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-25 02:33:50 +00:00
andrew@webrtc.org
f832a551cc Disable flaky TestPartialFrameHeader. 
Example failure:
[http://chromegw/i/internal.client.webrtc/builders/Linux%20Asan/builds/657]

TBR=wu@webrtc.org
BUG=2409

Review URL: https://webrtc-codereview.appspot.com/2286004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4832 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-24 20:09:30 +00:00
andrew@webrtc.org
f0f92fae12 Disable flaky SendDataMultipleClocks. 
Example failure:
[http://chromegw/i/internal.client.webrtc/builders/Linux32%20Debug/builds/719]

TBR=mallinath
BUG=2409

Review URL: https://webrtc-codereview.appspot.com/2270005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4828 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-24 16:26:41 +00:00
mallinath@webrtc.org
1112c30e1e Update libjingle to 53057474. 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2274004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4818 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 20:34:45 +00:00
asapersson@webrtc.org
b533a82bf9 Disabled flaky tests. 
BUG=2409
R=andrew@webrtc.org, mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2267005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4815 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 19:47:49 +00:00
fischman@webrtc.org
d29ab4e17c Suppress SSL error strings on mac_asan to unbreak that build 
Example borkedness: http://chromegw/i/client.webrtc/builders/Mac%20Asan/builds/642/steps/libjingle_p2p_unittest/logs/stdio

R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/2263004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4798 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 23:54:56 +00:00
fischman@webrtc.org
76fe9309b9 Use link_settings instead of all_dependent_settings to pacify xcode gyp generator 
Should unbreak e.g. http://chromegw/i/chromium.webrtc.fyi/builders/Mac%20%5Blatest%20WebRTC%20trunk%5D/builds/2396

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2261004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4796 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 21:11:08 +00:00
fischman@webrtc.org
ccddd0a941 Roll webrtc's chromium_revision 217707:224141 
Also adds -lm for executables depending on isac since the newer clang in the
newer chromium revision requires it, and -lstdc++ for dependencies of the objc lib because newer gyp links with gcc instead of g++ for non-C++-containing libs.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2177007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 20:27:32 +00:00
wu@webrtc.org
967bfff54d Update talk to 52534915. 
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2251004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4786 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 05:49:50 +00:00
wu@webrtc.org
8d1e4d6149 Increase the dtmfsender test toleration to 100ms to avoid flaky. 
BUG=2391
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/2248004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4780 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-18 18:01:07 +00:00
stefan@webrtc.org
da79008ab4 Disabling crashing or flaky tests in peerconnection_unittest. 
R=kjellander@webrtc.org
TBR=wu@webrtc.org
TESTS=trybots
BUG=2378

Review URL: https://webrtc-codereview.appspot.com/2227004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4767 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-17 13:11:38 +00:00
mallinath@webrtc.org
b3af8aea3e Verify local and remote transport description before 
negotiation.

TBR=sergeyu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2221004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4756 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-17 00:11:05 +00:00
sergeyu@chromium.org
8a1448950c Disable WebRtcSessionTest.TestCreateOfferWithSctpEnabledWithoutStreams 
BUG=2374
TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2214004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4747 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-14 00:30:51 +00:00
sergeyu@chromium.org
a59696b2a5 Update libjingle to 52300956 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2213004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4744 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-13 23:48:58 +00:00
henrike@webrtc.org
82f014aa0b OpenSL (not default): Enables low latency audio on Android. 
BUG=1669
R=andrew@webrtc.org, fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2032004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 18:24:07 +00:00
mallinath@webrtc.org
1b476d9a56 Disabling channelmanager unittest. This test is causing 
TSAN error. The problem could be in thread Invoke method.

TBR=wu@webrtc.org
BUG=https://code.google.com/p/webrtc/issues/detail?id=2355

Review URL: https://webrtc-codereview.appspot.com/2190004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4700 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-07 18:59:12 +00:00
mallinath@webrtc.org
ab5a0912a3 Fixing the build error on Windows. 
Problem is in coversion from size_t to int.

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-07 00:12:57 +00:00
mallinath@webrtc.org
1b15f4226f Update talk to 51960985. 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2188004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4696 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-06 22:56:28 +00:00
fischman@webrtc.org
016eec0983 Unbreak build by adding new mandatory ICE username param. 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2182004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4689 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-05 23:11:55 +00:00
fischman@webrtc.org
c31d4d0324 AppRTCDemo(iOS): prefer ISAC as audio codec 
This makes audio flow well bidirectionally to an iPod Touch (5th gen).
Also:
- Update to new turnserver JSON style:
  - separate username field
  - multiple URLs for the same server (e.g. both UDP & TCP)
- Added more explicit logging for ICE Connected since it's useful for debugging
- Give focus to the input field on app launch since that's the only useful
  thing to have focus on, anyway.
- Fix minor typos
- Cleaned up trailing whitespace and hard tabs

BUG=2191
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2127004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4687 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-05 21:49:58 +00:00
andrew@webrtc.org
9080518a39 Restore severity precondition to logging.h. 
I mistakenly ommitted the checks when logging.h was ported from
libjingle to webrtc. This caused a significant CPU cost for logs which
were later filtered out anyway.

Verified with LS_VERBOSE logging in neteq4, running:
$ out/Release/modules_unittests \
--gtest_filter=NetEqDecodingTest.TestBitExactness \
--gtest_repeat=50 > time.txt
$ grep "case ran" time.txt | grep "[0-9]* ms" -o | sort

Results on a MacBook Retina, averaged over 5 runs:
Verbose logs disabled:                          666 ms
Exisiting implementation, verbose logs enabled: 944 ms (1.42x)
New implementation, verbose logs enabled:       673 ms (1.01x)

BUG=2314
R=henrik.lundin@webrtc.org, henrike@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2160005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4682 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-05 16:40:43 +00:00
fischman@webrtc.org
ccf8b56670 AppRTCDemo(android): prefer ISAC for audio codec. 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2126004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4666 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-03 19:02:11 +00:00
fischman@webrtc.org
8788167b9b PeerConnection Java: explicitly cast DataChannel* to jlong for Java. 
Otherwise on 32-bit ARM Android the nativeDataChannel param the Java ctor sees
is a 64-bit value whose low 32 bits are the pointer, and whose high 32-bits are
garbage.

BUG=2302
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2114004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4665 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-03 18:58:12 +00:00
wu@webrtc.org
cadf9040cb Update talk to 51664136. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2148004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-30 21:24:16 +00:00
sergeyu@chromium.org
80b56a71e7 Revert part of libjingle roll that caused flakiness of WebRTC tests. 
BUG=crbug.com/279270
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4631 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-27 22:16:21 +00:00
elham@webrtc.org
d6fef9d380 Fixing SetDecodeErrorMode build error 
- got introduced when reverting r4562

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2118004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4624 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 23:59:38 +00:00
fischman@webrtc.org
e3de6b1e90 Enable ObjC build by default and reenable 64-bit mac libjingle build 
BUG=2124
TESTED=trybots & building for mac, mac64, ios-sim, and ios-device on my MBP all build everything in out/Debug.
R=niklas.enbom@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2080004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4620 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 19:31:21 +00:00
mallinath@webrtc.org
af84d782f0 Initialize ssl_role_ to the default role in FakeTransportChannel 
constructor.
This is needed as BaseSession tests can query the transport channel
without creating dtlstransportchannel ( as they are unaware of the
underlying implementation).

This will also fix the memcheck error in webrtc bots.

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2110004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4615 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 17:14:13 +00:00
sergeyu@chromium.org
f1fd9d0c5c Fix compilation on windows after libjingle updated. 
For some reason MSVC doesn't use implicit char[]->std::string 
conversion when comparing char[] and std::string in EXPECT_EQ.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2104004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4611 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-24 01:02:36 +00:00
sergeyu@chromium.org
492e315400 Update gyp file after libjingle roll. 
TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2103004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4609 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-24 00:06:43 +00:00
sergeyu@chromium.org
0be6aa0665 Update talk to 51314459 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2100004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4608 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-23 23:21:25 +00:00
henrike@webrtc.org
c0b1a280ab Some tests were not disabled correctly as it should be DISABLED_* not DISABLE_*. 
TBR=wu@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/2095005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4602 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-23 14:32:21 +00:00
fischman@webrtc.org
d26f791273 AppRTCDemo(android): allow audio-only calls to test iOS interop 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2091004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4598 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-22 23:50:48 +00:00
henrike@webrtc.org
61b262c427 Disable tests according to issues: 1205,2272,2288,2290,2291 
BUG=1205,2272,2288,2290,2291
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2069005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-22 20:27:49 +00:00
henrike@webrtc.org
7666db79fa Update talk to 51242664. 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2090005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4594 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-22 14:45:42 +00:00
fischman@webrtc.org
d0f4c2185b iOS: unbreak the build following r4546 
BUG=2255
R=niklas.enbom@webrtc.org, sjlee@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2078004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4577 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-20 22:16:55 +00:00
wu@webrtc.org
ebe68aad44 Fix memory leak in portallocatorsessionproxy_unittest. 
Remove the suppressions that have been fixed.

BUG=1972,2263
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2062005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4576 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-20 21:14:39 +00:00
fischman@webrtc.org
28ff3ee6aa Fix invalid cricket::SrtpStat::FailureKey::operator<() implementation. 
If operator<(a, b) returns true, then it must not be the case that
operator<(b, a) is true as well, but the old implementation would do exactly
that if a={1, 0, 0} and b={0, 0, 1}, for example.

Should fix e.g.:
[004:555] Error(unittest_main.cc:40): c:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\xtree(1746) : Assertion failed: invalid operator<
from http://chromegw/i/client.libjingle/builders/Win32%20Debug/builds/245/steps/libjingle_p2p_unittest/logs/stdio

R=juberti@webrtc.org, mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2054005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4561 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-16 19:12:26 +00:00
mallinath@webrtc.org
4d3e8b8c1b Update srtp error value in channel unittests. 
TBR=ronghuawu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2053004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4557 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-16 00:31:17 +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
wu@webrtc.org
97d1a988b6 Remove suppressions for the cases that's already fixed. 
Rename some of the suppressions to new issue.
Fix leaks in virtualsocket_unittest.

BUG=1972,1976,2100
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2010005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4536 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 00:13:26 +00:00
wu@webrtc.org
6603736038 PeerConnection::RemoveStream now removes the local stream even when it's closed. Updated the unit test accordingly. 
RISK=P3
TESTED=PeerConnectionInterfaceTest.CloseAndTestMethods
TBR=fischman_webrtc

Review URL: https://webrtc-codereview.appspot.com/2018005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4535 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 00:09:35 +00:00
fischman@webrtc.org
32001ef124 PeerConnection shutdown-time fixes 
- TCPPort::~TCPPort() was leaking incoming_ sockets; now they are deleted.
- PeerConnection::RemoveStream() now removes streams even if the
  PeerConnection::IsClosed().  Previously such streams would never get removed.
- Gave MediaStreamTrackInterface a virtual destructor to ensure deletes on base
  pointers are dispatched virtually.
- VideoTrack.dispose() delegates to super.dispose() (instead of leaking)
- PeerConnection.dispose() now removes streams before disposing of them.
- MediaStream.dispose() now removes tracks before disposing of them.
- VideoCapturer.dispose() only unowned VideoCapturers (mirroring C++ API)
- AppRTCDemo.disconnectAndExit() now correctly .dispose()s its
  VideoSource and PeerConnectionFactory.
- CHECK that Release()d objects are deleted when expected to (i.e. no ref-cycles
  or missing .dispose() calls) in the Java API.
- Create & Return webrtc::Traces at factory birth/death to be able to assert
  that _all_ threads started during the test are collected by the end.
- Name threads attached to the JVM more informatively for debugging.
- Removed a bunch of unnecessary scoped_refptr instances in
  peerconnection_jni.cc whose only job was messing with refcounts.

RISK=P2
TESTED=AppRTCDemo can be ended and restarted just fine instead of crashing on camera unavailability.  No more post-app-exit logcat lines.  PCTest.java now asserts that all threads are collected before exit.

BUG=2183
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2005004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4534 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 23:26:21 +00:00
mallinath@webrtc.org
a5506690b4 Update libjingle to 50733053. 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2017004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4532 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 21:18:15 +00:00
fischman@webrtc.org
dd14b2add1 libjingle gyp: signal errors during gyp time to avoid cryptic failures during build time. 
- $JAVA_HOME / java_home missing or not pointing to a JDK
- Multiple or zero mac codesigning identities

BUG=2206
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2012004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4527 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 18:06:29 +00:00
wu@webrtc.org
91053e7c5a Update libjingle to 50654631. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2000006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-10 07:18:04 +00:00
fischman@webrtc.org
825e9b0a9b talk/objc/README: s/libjingle/webrtc/ in repository path. 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1985004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4501 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-07 16:52:03 +00:00
fischman@webrtc.org
c883fdc273 PeerConnection.java: enable setting trace & log levels from Java 
Replaces the hard-coded scheme that was there before and lets apps decide what
to log and to where.

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4498 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-06 19:00:53 +00:00
wu@webrtc.org
9dba525627 * Update libjingle to 50389769. 
* Together with "Add texture support for i420 video frame." from
wuchengli@chromium.org.
https://webrtc-codereview.appspot.com/1413004

RISK=P1
TESTED=try bots
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1967004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 20:36:57 +00:00
fischman@webrtc.org
c3d93c6921 talk/PRESUBMIT: Accept copyright years going back to 2004. 
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1956004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4485 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 15:01:33 +00:00
wu@webrtc.org
a054569c15 Fix memory leak in datachannel and its test. 
RISK=P3
TESTED=memcheck build
tools/valgrind-webrtc/webrtc_tests.sh --tool memcheck --test out/Debug/libjingle_peerconnection_unittest  --gtest_filter=SctpDataChannelTest*

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1941005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4470 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 22:08:14 +00:00
wu@webrtc.org
0dc0f172a3 sscanf isn't safe with strings that aren't null-terminated. In such case, create a local copy that is null-terminated first. 
TESTED=GYP_DEFINES=build_for_tool=memcheck gclient runhooks
ninja -C out/Debug/ libjingle_unittest
tools/valgrind-webrtc/webrtc_tests.sh --tool memcheck --test out/Debug/libjingle_unittest  --gtest_filter=Http*

R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/1941004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4469 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 21:20:46 +00:00
fischman@webrtc.org
86d7a198ec ObjC PeerConnection README: note workaround needed for crbug.com/248168 
BUG=2106
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1940004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 19:27:54 +00:00
fischman@webrtc.org
1bc1954174 AppRTCDemo: builds using ninja on iOS for simulator and device! 
Things included in this CL:
- updated READMEs to provide an exact/reproable set of steps for getting the app
  running.
- gyp changes to build the iOS AppRTCDemo sample app using gyp+ninja instead of
  the hand-crafted Xcode project (which has never worked in its checked-in
  form), including a gyp action to sign the sample app for deployment to an iOS
  device (the app can also be used in the simulator)
- deleted the busted hand-crafted Xcode project for the sample app
- updated the sample app to match the PeerConnection API that ended up landing
  (in a surprising twist of fate, the API landed quite a bit later than the
  sample app and this is the first time the CR-time changes in the API are
  reflected in the sample app)
- updated the sample app to reflect apprtc.appspot.com HTML/JS changes (equiv to
  the AppRTCClient.java changes in http://s10/47299162)
- picked up the iossim DEPS to enable launching the sample app in the simulator
  from the command-line.
- renamed some files to match capitalization of the classes they contain (Ice ->
  ICE) per ObjC naming guidelines.
- ran the files involved in this CL through clang-format to deal with xcode
  formatting craxy.

BUG=2106
RISK=P2
TESTED=unittest builds with ninja and passes on OS=mac; sample app builds with ninja and runs on simulator and device, though no audio flows from simulator/device (will fix in a follow-up CL)
R=andrew@webrtc.org, justincohen@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1874005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4466 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 18:29:45 +00:00
wu@webrtc.org
d64719d895 Update libjingle to 50191337. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1885005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4461 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 00:00:07 +00:00
wu@webrtc.org
7fdbb1c832 We don't need to link with libssl.so when we already depend on openssl. 
This fixes the hidden-symbol linker warnings.

BUG=2149
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1927004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 22:41:36 +00:00
fischman@webrtc.org
caa7024b86 PeerConnectionTest.java: build on android bots as well as linux ones. 
BUG=1796
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1921005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 21:56:30 +00:00
fischman@webrtc.org
85f07f59ee PeerConnectionTest.java: use java_home gyp var instead of hardcoding /usr. 
BUG=1796
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1899005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4433 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 18:11:07 +00:00
fischman@webrtc.org
3d496fb046 Roll chromium_revision 205140:214260 to pick up build fixes for ninja iOS device build. 
TESTED=git try
BUG=2106
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1888005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4431 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 17:14:35 +00:00
henrike@webrtc.org
9638564340 Adds no parent to talk folder. 
BUG=1933
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1896004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4430 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 15:51:54 +00:00
fischman@webrtc.org
d6134c7cfd PeerConnectionTest.java: make the test work for the bots' v4l2loopback. 
- Make the test agnostic to the actual resolution used, since v4l2_file_player
  is playing a non-640x480 file (go/httfw)
- Teach DeviceInfoLinux::FillCapabilityMap() about I420 since that's what
  v4l2_file_player is feeding.

Requires https://gist.github.com/fischman/2e9a9b2efd2ad363ef82 be applied to the
v4l2loopback driver code.

BUG=1796
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1891004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 20:43:15 +00:00
fischman@webrtc.org
147d44a450 AppRTCDemo: replace the use of query-string parameters for pre-JB devices. 
Replaces the use of a query-string parameter with a (once-per-session)
JS-to-Java function call, because query-string parameters on file:// URLs are
busted on ICS and earlier Android releases
(https://code.google.com/p/android/issues/detail?id=17535).

Also added channel.html to the list of inputs to cause edits to it to cause a
rebuild of the .apk.

BUG=1949
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1890004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 19:07:33 +00:00
henrike@webrtc.org
ea40bd0cc8 Presubmit script for preventing changes to protected files and add the full list of those files. 
BUG=2090
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1855004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4419 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 18:20:07 +00:00
henrike@webrtc.org
1e09a71126 Update talk folder to revision=49952949 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4413 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 19:17:59 +00:00
turaj@webrtc.org
c46967dc53 Revert 4391 "Roll chromium 205140:212975 to support ninja iOS ar..." 
r4391 results in Mac Release Bot fail: http://chromegw/i/internal.client.webrtc/builders/Mac32%20Release/builds/334/steps/modules_integrationtests


> Roll chromium 205140:212975 to support ninja iOS armv7 build.
> 
> In particular, picks up new clang, libvpx, libsrtp, yasm, and gyp.
> 
> TESTED=git try on patchset #1
> BUG=2106
> R=henrike@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1849005

TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1874004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4399 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 17:48:24 +00:00
fischman@webrtc.org
33584f942c Introduce a bit of sanity to talk/PRESUBMIT.py's license checking. 
The comma this allows is a very common variant of the license header (3:1
preferred over the no-comma variant in talk/).

Also pacify pylint a bit, and correct a flagrantly incorrect header I happened
to come across.

BUG=2098,2133
R=henrike@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1866004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4396 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 16:43:30 +00:00
fischman@webrtc.org
9fbc558dd4 talk/OWNERS: add libjingle team members from internal webrtc/files/OWNERS 
BUG=1933
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1867004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 16:22:37 +00:00
fischman@webrtc.org
880c842627 AppRTCDemo: don't render frames that are already outdated. 
BUG=2121
R=henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1850004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4392 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-24 20:40:03 +00:00
fischman@webrtc.org
87f8a7eb67 Roll chromium 205140:212975 to support ninja iOS armv7 build. 
In particular, picks up new clang, libvpx, libsrtp, yasm, and gyp.

TESTED=git try on patchset #1
BUG=2106
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1849005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4391 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-24 20:15:43 +00:00
henrike@webrtc.org
8d27a1c723 Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle 
BUG=1932
TESTED=git try
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1851004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-23 18:15:11 +00:00
henrike@webrtc.org
5c280ecd57 Revert 4382 "Makes webrtc and libjingle build from the same gyp-..." 
Failures: breaks build bots. Will have to disable Android NDK build for libjingle. The TSAN issues are in webrtc which should be unaffected. Flakey? Here are the failing tests:
 http://chromegw/i/internal.client.webrtc/builders/Android%20NDK/builds/303 and http://chromegw/i/internal.client.webrtc/builders/Linux%20Tsan/builds/284

> Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle
> 
> BUG=1932
> TESTED=git try
> R=andrew@webrtc.org, fischman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1836004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1834005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-23 03:30:32 +00:00
henrike@webrtc.org
5fcddf2334 Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle 
BUG=1932
TESTED=git try
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1836004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4382 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-23 00:27:43 +00:00
henrike@webrtc.org
390fcb7a20 Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder. 
BUG=2091
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1833004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4381 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-22 22:32:50 +00:00
henrike@webrtc.org
28654cbc22 Update talk folder to revision=49713299. 
TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1848004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4380 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-22 21:07:49 +00:00
henrike@webrtc.org
0df5b8dfa6 Revert 4372 "Makes webrtc and libjingle build from the same gyp-..." 
> Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches.
> 
> TESTED=git try
> BUG=1932
> R=andrew@webrtc.org, fischman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1804004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1835004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-18 18:58:29 +00:00
henrike@webrtc.org
4e4bf4db8b Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. 
TESTED=git try
BUG=1932
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1804004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-18 18:33:55 +00:00
henrike@webrtc.org
8c7347124c talk: DataChannel.java repeated contents. This removes the duplicate. 
TBR=ajm

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1825004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-17 16:28:37 +00:00
henrike@webrtc.org
9de257d00f Update talk folder to revision=49470012. Same as 375 in libjingle's google code repository. 
TBR=wu@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1824004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4364 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-17 14:42:53 +00:00
henrike@webrtc.org
7b2f955e56 Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README. 
BUG=1935
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1805005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4356 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 18:04:56 +00:00
henrike@webrtc.org
723d683ecb Update talk folder to revision=49260075. Same as 369 in libjingle's google code repository. 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1797004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4338 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-12 16:04:50 +00:00
henrike@webrtc.org
28e2075280 Adds trunk/talk folder of revision 359 from libjingles google code to 
trunk/talk


git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 00:45:36 +00:00