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73 Commits

Author SHA1 Message Date
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