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85 Commits

Author SHA1 Message Date
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