webrtc

Author	SHA1	Message	Date
fischman@webrtc.org	f27fdeb9c9	AppRTCDemo(android): don't initialize process-globals more than once. BUG=3257 R=braveyao@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19369004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6001 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-28 16:32:38 +00:00
kjellander@webrtc.org	7d825e9b2c	Revert "libjingle_unittest now compiles and passes on iOS!" This reverts commit r5986 as it fails compilation on Mac (non-iOS). The failure was not discovered on the commitbots since they don't clobber their builds. BUG=3241 TBR=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19399004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5997 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-28 12:50:47 +00:00
mallinath@webrtc.org	a0d3067575	Use CreatePeerConnection method which accepts port_allocator. Other method will be removed, in a different CL. R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20369006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5987 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-26 00:00:15 +00:00
fischman@webrtc.org	95cd1551f8	libjingle_unittest now compiles and passes on iOS! Example run from cmd-line: ninja -C out_ios/Debug-iphoneos libjingle_unittest && ~/src/ios-deploy/ios-deploy -d -u -v -b ~/src/wr/trunk/out_ios/Debug-iphoneos/libjingle_unittest.app Note that the test's use of signals means that lldb will break in the middle of the suite. To ignore these signals tell lldb: pro hand -p true -s false -n false SIGINT pro hand -p true -s false -n false SIGTERM continue BUG=3241 R=noahric@google.com, tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12229004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5986 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-25 23:59:56 +00:00
buildbot@webrtc.org	658a94595d	(Auto)update libjingle 65619249-> 65622932 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5984 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-25 22:03:30 +00:00
buildbot@webrtc.org	ff90ed6e96	(Auto)update libjingle 65561104-> 65619249 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5983 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-25 21:12:10 +00:00
buildbot@webrtc.org	2b93402e36	(Auto)update libjingle 65484212-> 65561104 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5978 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-25 00:18:27 +00:00
buildbot@webrtc.org	3f1aa24078	(Auto)update libjingle 65469804-> 65484212 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5967 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-24 00:00:12 +00:00
jiayl@webrtc.org	0d915ff603	Fix the return value of DtlsTransportChannelWrapper::SendPacket in the case of invalid RTP packet. R=juberti@webrtc.org, mallinath@webrtc.org BUG=3244 Review URL: https://webrtc-codereview.appspot.com/12299006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5966 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-23 20:46:29 +00:00
buildbot@webrtc.org	504fc89f36	(Auto)update libjingle 65394435-> 65417850 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5961 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-23 03:23:19 +00:00
tkchin@webrtc.org	19b1be159e	Provide GetStats method in RTCPeerConnection BUG=3144 R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12069006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5960 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-22 21:05:38 +00:00
tkchin@webrtc.org	ec3d8ecdcc	Fix typo by renaming RTCSessionDescriptonDelegate -> RTCSessionsDescriptionDelegate R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5946 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-21 18:47:24 +00:00
mallinath@webrtc.org	54fd70046d	Remove ASSERT in TransportChannelProxy::SetImplementation, when proxy already set to same transport channel impl. Since session can call SetImplementation multiple times with or without BUNDLE, there are cases when SetImplementation is called with same impl (OnRemoteCandidates/PushdownTransportDescription/SetupMux). Also variables in cricket::TransportProxy like \|connecting_\| and \|negotiated_\| are accessed both between worker thread and signaling threads (which calls for bigger change on how session interacts with Transport and TransportChannelProxy). I have a created a separate bug to address later issue. Also if single thread used as worker and signaling thread, we can end up calling SetLocalDescription and OnRemoteCandidates in same call sequence, which will end up calling SetImplementation twice. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12019007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5944 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-19 01:03:33 +00:00
buildbot@webrtc.org	8e5ec52e76	(Auto)update libjingle 65152644-> 65219629 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5941 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-19 00:00:31 +00:00
fischman@webrtc.org	29540b1879	Revert "PeerConnectionFactory: delay deletion of owned threads." This reverts r5933 because it broke http://build.chromium.org/p/client.webrtc/builders/Win64%20Release/builds/1598 BUG=3100 Review URL: https://webrtc-codereview.appspot.com/12159004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5935 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-17 22:54:30 +00:00
buildbot@webrtc.org	1a87f529a2	(Auto)update libjingle 65151416-> 65151642 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5934 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-17 22:41:30 +00:00
fischman@webrtc.org	cea024d672	PeerConnectionFactory: delay deletion of owned threads. Since PeerConnection holds a ref to its creating PeerConnectionFactory, it's possible for ~PeerConnectionFactory() to be run on its signaling thread. Deleting a thread from within that thread is sad times, so don't do it. It would be nicer to avoid having PeerConnection hold a ref to the factory, and instead require the user to keep the factory alive. Unfortunately that changes the contract on PeerConnection{,Factory} and it's unclear how to vet existing callers for safety. BUG=3100 R=juberti@webrtc.org, noahric@google.com Review URL: https://webrtc-codereview.appspot.com/11289004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5933 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-17 22:36:00 +00:00
henrike@webrtc.org	aeb0c28193	Update PRESUBMIT.py's list of "DO_NOT_SUBMIT_FILES". BUG=N/A R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12029006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5931 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-17 14:15:43 +00:00
buildbot@webrtc.org	0b3c6c3191	(Auto)update libjingle 65086785-> 65104022 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5925 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-17 10:03:57 +00:00
buildbot@webrtc.org	39b868bad3	(Auto)update libjingle 65055925-> 65086785 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5921 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-17 00:04:39 +00:00
jiayl@webrtc.org	8f88f20af2	Expand the test max wait time from 1000ms to 2000ms. The createOffer/createAnswer methods sometimes times out due to slow identity generation under memcheck. BUG=2838 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5920 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-16 17:14:21 +00:00
wu@webrtc.org	36eda7cf0e	Workaround for https://bugzilla.mozilla.org/show_bug.cgi?id=996329 , where the m line from firefox have a space at the end. For example: "m=application 38233 DTLS/SCTP 5000 " BUG=3212 TEST=manually try to use DataChannel between FF 28 and Chrome with rtccopy.com R=jiayl@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12029005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5915 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-15 20:37:30 +00:00
buildbot@webrtc.org	1fd5b45a0e	(Auto)update libjingle 64956819-> 64982143 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5910 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-15 17:39:43 +00:00
kjellander@webrtc.org	190b72a350	Make libjingle Android example build without sourcing envsetup.sh See https://webrtc-codereview.appspot.com/11799004 for full details (separate to avoid webrtc+talk changes in same CL). BUG=chromium:346198 TEST=Local builds using: . build/android/envsetup.sh unset ANDROID_SDK_ROOT webrtc/build/gyp_webrtc ninja -C out/Debug ninja -C out/Release + trybots passing: git try --bot=android,android_rel,android_clang R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5908 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-15 08:35:49 +00:00
mallinath@webrtc.org	ad4440a64e	In shared socket mode, use udp port as default receiver even if stun server address is not set. This can happen in a loopback scenarios where clients do not need to provide any server information. R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5906 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-15 01:10:58 +00:00
buildbot@webrtc.org	505f400f27	(Auto)update libjingle 64909599-> 64919255 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5905 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-14 20:33:47 +00:00
fischman@webrtc.org	e98598d3f0	Make everyone an OWNER for .gyp/.gypi add/delete purposes, talk/ edition. This CL brought to you by: $ for d in $(for f in $(git ls-files 'gyp' 'gypi'); do dirname $f; done\|sort\|uniq\|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file .gyp=\nper-file .gypi=" >> $d/OWNERS; done $ for d in $(for f in $(git ls-files 'gyp' 'gypi'); do dirname $f; done\|sort\|uniq\|grep -v '^\.$'); do git add $d/OWNERS; done (and then removed the non-talk/ impact) R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11979004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5904 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-14 20:31:16 +00:00
buildbot@webrtc.org	1da6047132	(Auto)update libjingle 64813990-> 64909599 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5900 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-14 18:15:15 +00:00
fischman@webrtc.org	cf0b46c762	iosdeviceinfo.cc: remove unnecessary file The do-nothing implementation in this file is already present in mobiledevicemanager.cc (shared with Android) so this isn't adding value, and causes duplicate-symbol errors under some compilers. BUG=3201 R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5899 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-14 18:12:32 +00:00
buildbot@webrtc.org	f875f15afb	(Auto)update libjingle 64709629-> 64813990 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5897 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-14 16:06:21 +00:00
wu@webrtc.org	b884eb6118	(Auto)update libjingle 64630087-> 64709629 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5884 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-10 16:59:16 +00:00
henrike@webrtc.org	8dce41b3c6	Remove erronuous commit message from auto sync. BUG=N/A TBR=kjellander@webrtc.org http://webrtc-codereview.appspot.com/11639004/ git-svn-id: http://webrtc.googlecode.com/svn/trunk@5883 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-10 14:39:38 +00:00
buildbot@webrtc.org	15192f909e	(Auto)update libjingle 64594651-> 64630087 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5878 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-10 06:34:32 +00:00
henrike@webrtc.org	8f89497949	Remove erronuous commit message. BUG=N/A TBR=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11549004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5875 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-09 14:43:43 +00:00
buildbot@webrtc.org	61c1b8ea32	(Auto)update libjingle 64585415-> 64594651 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5870 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-09 06:06:38 +00:00
henrike@webrtc.org	f824fde36f	(Auto)update libjingle 64326665-> 64585415 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5864 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-08 22:13:01 +00:00
henrike@webrtc.org	74a7c482b9	Removes unused thread causing compiler warnings. BUG=N/A R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11369004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5859 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-07 20:49:34 +00:00
wu@webrtc.org	4e393070be	Compare the answer's media type against offer to make sure they are match. Otherwise we should return failure. BUG=2687 R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11079005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5858 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-07 17:04:35 +00:00
henrike@webrtc.org	09b0c10eed	Talk: fixes warning: local variable is initialized but not referenced due to only using the variable in question for asserts. BUG=N/A R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11249004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5848 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-04 22:33:34 +00:00
fischman@webrtc.org	d1fe6b728e	AppRTCDemo(android): fix a couple of SDP-related regressions. - r5834 made it so that empty fields are a fatal SDP parsing error, exposing opportunities for improvement in the preferISAC; changed split/join to use \r\n instead of \n and now omitting the trailing space on the m=audio line that triggered the new failure. - DTLS requires a different role for each endpoint so conflicts with loopback calling. apprtc.py suppresses DTLS for that reason in loopback calls, so the android demo app now only enables DTLS by default if it is not suppressed by a constraint (matching Chrome). BUG=3164,3165,2507 R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11229004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5847 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-04 21:40:46 +00:00
henrike@webrtc.org	f5bebd40f3	(Auto)update libjingle 64247466-> 64326665 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5845 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-04 18:39:07 +00:00
wu@webrtc.org	148149138d	(Auto)update libjingle 64147530-> 64247466 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5835 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-02 23:25:15 +00:00
wu@webrtc.org	5e760e7b94	Check the return value of the FromString call and return failure when then value is invalid. I.e. uses bool FromString(const std::string& s, T* t) instead of T FromString(const std::string& str) Before this change we will silently continue the parsing and take whatever default value returned by FromString. TEST=new tests BUG=2507 R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11069004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5834 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-02 23:19:09 +00:00
wu@webrtc.org	e387771b98	Remove webrtc_unittest.cc from talk presubmit script. BUG= R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5833 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-02 22:23:16 +00:00
wu@webrtc.org	05e7b44b83	(Auto)update libjingle 63948945-> 64147530 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5825 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-04-01 17:44:24 +00:00
fischman@webrtc.org	49c5ba32bb	AppRTCDemo(iOS): now works in the iOS Simulator! ...which has no camera device emulation or pass-through, so no local video view. R=noahric@google.com Review URL: https://webrtc-codereview.appspot.com/10919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5815 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-31 20:22:19 +00:00
fischman@webrtc.org	61e78fca6c	AppRTCDemo(iOS): remote-video reliability fixes Previously GAE Channel callbacks would be handled by JS string-encoding the payload into a URL. Unfortunately this is limited to the (undocumented, silently problematic) maximum URL length UIWebView supports. Replaced this scheme by a notification from JS to ObjC and a getter from ObjC to JS (which happens out-of-line to avoid worrying about UIWebView's re-entrancy, or lack thereof). Part of this change also moved from a combination of: JSON, URL-escaping, and ad-hoc :-separated values to simply JSON. Also incidentally: - Removed outdated TODO about onRenegotiationNeeded, which is unneeded - Move handling of PeerConnection callbacks to the main queue to avoid having to think about concurrency too hard. - Replaced a bunch of NSOrderedSame with isEqualToString for clearer code and not having to worry about the fact that [nil compare:@"foo"]==NSOrderedSame is always true (yay ObjC!). - Auto-scroll messages view. BUG=3117 R=noahric@google.com Review URL: https://webrtc-codereview.appspot.com/10899006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5814 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-31 20:16:49 +00:00
fischman@webrtc.org	fe16488184	AppRTCDemo(android): specify DtlsSrtpKeyAgreement:true in CreatePeerConnection's constraints. This is required to interop with Chrome now that SDES is disabled in Chrome (as of r5640). BUG=2774 R=jiayl@chromium.org Review URL: https://webrtc-codereview.appspot.com/10749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5809 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-28 19:58:03 +00:00
fischman@webrtc.org	4f2bd68744	Silence pointless LS_WARNING about port 0 for active-only candidates. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5808 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-28 18:13:34 +00:00
wu@webrtc.org	987f2c9aae	(Auto)update libjingle 63913264-> 63948945 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5807 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-28 16:22:19 +00:00
wu@webrtc.org	f7d501d48a	(Auto)update libjingle 63884381-> 63913264 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5805 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-27 23:48:25 +00:00
andrew@webrtc.org	a5586b50e5	Protect ENABLE_PROFILING to fix profiling=1. Chromium defines ENABLE_PROFILING under the gyp flag profiling=1. This corrects the resulting mulitple defintion error: ../../talk/base/profiler.h:61:9: error: 'ENABLE_PROFILING' macro redefined [-Werror] #define ENABLE_PROFILING and allows us to use profiling=1 in standalone builds. TESTED=build passes with profiling=1 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5804 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-27 22:44:13 +00:00
wu@webrtc.org	cfe5e9c894	(Auto)update libjingle 63837929-> 63884381 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5800 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-27 17:03:58 +00:00
henrike@webrtc.org	b0ecc1c6fb	(Auto)update libjingle 63777286-> 63837929 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5797 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-26 22:44:28 +00:00
fischman@webrtc.org	b25576a75b	talk/: enable _DEBUG in Debug for all posix Chromium's build/common.gypi defines _DEBUG for Debug builds _except_ on (OS=="mac" OS=="ios"). But libjingle uses _DEBUG on all platforms so define it on all posix (chromium covers non-posix separately and fine). BUG=webrtc:3101 R=juberti@google.com Review URL: https://webrtc-codereview.appspot.com/10699004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5795 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-26 21:53:47 +00:00
henrike@webrtc.org	1ca08f65e3	Fix after auto update in r5787. APPRTCVideoView.h/m was removed incorrectly. BUG=3121 R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10739004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5793 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-26 16:42:14 +00:00
henrike@webrtc.org	5fb7428496	(Auto)update libjingle 63775799-> 63776369 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5789 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-26 02:00:10 +00:00
henrike@webrtc.org	a92fd74f40	(Auto)update libjingle 63773382-> 63775799 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5788 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-26 01:46:18 +00:00
henrike@webrtc.org	dce3feb0b0	(Auto)update libjingle 63738002-> 63773382 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5787 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-26 01:17:30 +00:00
henrike@webrtc.org	ae3347a546	Fix after auto update: removed files were brought back. BUG=N/A R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5782 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-25 18:17:02 +00:00
fischman@webrtc.org	76d4f389bb	AppRTCDemo(iOS): allow rooms with no incoming audio. Also fix a compile-time warning for a leftover unimplemented method (RTCVideoRenderer:setTransform). R=noahric@google.com Review URL: https://webrtc-codereview.appspot.com/10629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5780 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-25 17:40:38 +00:00
henrike@webrtc.org	6e3dbc2a77	(Auto)update libjingle 63648983-> 63738002 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5779 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-25 17:09:47 +00:00
fischman@webrtc.org	385a722646	PeerConnection(iOS): make ARC-clean talk/.../objc* and talk/examples/ios - Removes a strong-reference cycle between RTCPeerConnection and RTCPeerConnectionObserver - Gives RTCPeerConnectionObserver a virtual dtor - Ensures RTCPeerConnectionTest tears down correctly - Ensures AppRTCDemo tears down correctly This is the talk/ half; the webrtc/ half is in https://webrtc-codereview.appspot.com/10539005 BUG=3054,3055,3100 R=noahric@google.com Review URL: https://webrtc-codereview.appspot.com/10499005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5771 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-25 05:16:29 +00:00
sergeyu@chromium.org	e42b8ab129	Cleanups in libjingle to make it compile with chromium_code=1 Fixed all warnings that show up when compiling libjingle in chromium with compiling with chromium_code=1. chromium_code=1 enables various warnings that are off by default. Most changes are for unused variables and consts. R=pthatcher@google.com, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9699004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5769 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-25 00:31:35 +00:00
fischman@webrtc.org	7fa1fcb72c	AppRTCDemo(ios): style/cleanup fixes following cr/62871616-p10 BUG=2168 R=noahric@google.com Review URL: https://webrtc-codereview.appspot.com/9709004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5768 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-25 00:11:56 +00:00
fischman@webrtc.org	c693a2a624	PeerConnection(iOS): fix case in #import statements. We've been skating by on OS/X's default case-insensitive filesystem, but this is a bit silly. This change brought to you by: sed -i '' 's/\+internal\.h/+Internal.h/g' $(git grep -l '+internal.h') BUG=3088 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10559004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5764 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-24 18:56:37 +00:00
wu@webrtc.org	1e6cb2c5d2	(Auto)update libjingle 63560528-> 63648983 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5762 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-24 17:01:50 +00:00
henrike@webrtc.org	152208adeb	(Auto)update libjingle 63547048-> 63560528 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5753 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-21 21:43:26 +00:00
henrike@webrtc.org	be7e26d229	(Auto)update libjingle 63503990-> 63547048 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5751 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-21 16:40:18 +00:00
mallinath@webrtc.org	0c6f0f94f1	Revert 5737 "Add system wrapper dependency to libjingle targets." Adding additional dependency is not required for libjingle targets. > Add system wrapper dependency to libjingle targets. > This is necessary to handle usage of STR_CASE_CMP in > common_types.h ( as in https://webrtc-codereview.appspot.com/10099005/) > > TBR=wu@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/10309004 TBR=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10379004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5744 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-20 23:07:03 +00:00
henrike@webrtc.org	5e83c65aee	(Auto)update libjingle 63493960-> 63503990 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5743 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-20 22:33:30 +00:00
henrike@webrtc.org	a8ebdb71e3	Revert "(Auto)update libjingle 63363208-> 63493960" (r5740) BUG=N/A R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5741 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-20 20:27:25 +00:00
henrike@webrtc.org	5f768adc27	(Auto)update libjingle 63363208-> 63493960 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5740 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-20 20:18:06 +00:00
mallinath@webrtc.org	979f1f8235	Add system wrapper dependency to libjingle targets. This is necessary to handle usage of STR_CASE_CMP in common_types.h ( as in https://webrtc-codereview.appspot.com/10099005/) TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10309004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5737 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-20 15:09:09 +00:00
henrike@webrtc.org	ffe2620c97	(Auto)update libjingle 63352036-> 63363208 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5731 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-19 22:20:10 +00:00
henrike@webrtc.org	8b61011b6f	(Auto)update libjingle 63293120-> 63352036 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5720 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-18 21:39:10 +00:00
henrike@webrtc.org	e9793ab8b8	(Auto)update libjingle 63111035-> 63293120 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5717 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-18 14:36:23 +00:00
henrike@webrtc.org	18e5911d92	(Auto)update libjingle 63089643-> 63111035 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5705 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-14 17:19:38 +00:00
henrike@webrtc.org	f45a55083f	(Auto)update libjingle 63019975-> 63089643 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5699 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-13 18:51:34 +00:00
mallinath@webrtc.org	827faae0ec	Fixing incorrect memset. Found when ENABLE_EXTERNAL_AUTH is enabled in chrome. TBR=ronghuawu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5691 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-13 02:15:12 +00:00
henrike@webrtc.org	c7bec8484b	(Auto)update libjingle 62948689-> 63019975 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5689 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-12 19:53:43 +00:00
henrike@webrtc.org	10bd88e2b5	(Auto)update libjingle 62871616-> 62948689 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5683 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-11 21:07:25 +00:00
henrike@webrtc.org	d3d6bce9ed	(Auto)update libjingle 62865357-> 62871616 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5674 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-10 20:41:22 +00:00
henrike@webrtc.org	0537634154	(Auto)update libjingle 62713454-> 62865357 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5670 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-10 15:53:12 +00:00
fischman@webrtc.org	a01daf0359	RTCPeerConnectionTest(objc): deflake by ignoring ICECompleted. Delivery of the state seems intermittent at best on OS/X so ignore it until we can make it reliable. BUG=1414,2993,chromium:348982 TBR=bemasc@chromium.org Review URL: https://webrtc-codereview.appspot.com/9609004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5664 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-08 03:17:55 +00:00
fischman@webrtc.org	13320ea3d3	PeerConnectionTest(objc): expect ICE Completed state post 61460797-p10 Also a few trivial cleanups: - No need to use STUN for a loopback test - Reduce test call duration 10s->2s for faster iteration - Remove obviously-irrelevant Info.plist entries (copy/pasta from iOS) BUG=1414,2993 R=noahric@google.com Review URL: https://webrtc-codereview.appspot.com/9369004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5663 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-07 22:15:30 +00:00
jiayl@webrtc.org	11aab0edc2	Populate VoiceReceiverInfo::delay_estimate_ms, jitter_buffer_ms, and jitter_buffer_preferred_ms to getStats. BUG=2665 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9579004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5661 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-07 18:56:26 +00:00
pbos@webrtc.org	371243dfa3	Remove std:: prefixes from C functions in talk/. std::memcpy -> memcpy for instance. This change was motivated by a compile report complaining that std::rand() was used instead of rand(), probably with a stdlib.h include instead of cstdlib. Use of C functions without the std:: prefix is a lot more common, so removing std:: to address this. BUG= R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9559004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5657 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-07 15:22:04 +00:00
henrike@webrtc.org	79047f99c1	(Auto)update libjingle 62691533-> 62713454 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5653 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-06 23:46:59 +00:00
henrike@webrtc.org	2d213e450c	(Auto)update libjingle 62550414-> 62691533 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5652 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-06 18:51:21 +00:00
pbos@webrtc.org	f714e7faea	Remove abs() use in PseudoTcp::process. Squelches a clang 3.5 compile error for using abs() with a long instead of labs(). Updated affected code to use uint32:s to match the sign of m_rx_srtt. BUG= R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9409004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5651 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-06 18:31:08 +00:00
kjellander@webrtc.org	cf85f1cf3c	Reorganize libjingle path variables. BUG=chromium:343106 TEST=Trybots passing. I also successfully ran build/gyp_chromium and built Chromium with the talk/build/common.gypi modification in the checkout. R=andrew@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9019004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5644 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-05 00:40:33 +00:00
henrike@webrtc.org	b90991dade	Update libjingle 62472237->62550414 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5640 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-04 19:54:57 +00:00
pbos@webrtc.org	24dae9419a	Add pthatcher@webrtc.org to talk/OWNERS. pthatcher@ is a new member of the team with good libjingle knowledge. BUG= R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5636 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-04 05:58:25 +00:00
jiayl@webrtc.org	db41b4dbcd	Remove the deprecated GetStats method from PeerConnectionInterface. R=fischman@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8879004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5634 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-03 21:30:06 +00:00
jiayl@webrtc.org	80bbf4c312	Enable test SSLStreamAdapterTestDTLS.TestDTLSConnectWithSmallMtu since it does not fail anymore. BUG=2712 R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5633 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-03 18:41:27 +00:00
henrike@webrtc.org	40b3b68cdf	Update libjingle 62364298->62472237 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5632 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-03 18:30:11 +00:00
henrike@webrtc.org	1bbfb57d71	Rollback of r5629 "(Auto)update libjingle 62364298-> 62368661". BUG=N/A R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9329004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5631 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-03 17:37:52 +00:00
henrike@webrtc.org	31413dc635	(Auto)update libjingle 62364298-> 62368661 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5629 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-03 16:47:01 +00:00
mallinath@webrtc.org	d3dc424fe5	Remove posting of ICE messages from WebRTCSession in PeerConnection to signaling thread. These callbacks are called from signal thread already. There is no point in posting messages on the same thread again. BUG=2922 R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9219004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5626 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-01 00:05:52 +00:00
fischman@webrtc.org	bcfc1670d6	AppRTCDemo(android): don't send local SDP until it's set. This fixes a race condition where the remote participant could receive the offer, create & set its answer locally, send it back, and then try to set the answer before the local set completed. Observed intermittently in loopback calls when setLocalDescription is intentionally delayed (debugging something else). R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9249004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5625 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-03-01 00:02:27 +00:00
henrike@webrtc.org	b8395ebe14	(Auto)update libjingle 62293974-> 62364298 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5623 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-28 21:57:22 +00:00
henrike@webrtc.org	806768a6ca	(Auto)update libjingle 62281784-> 62293974 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5619 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-27 21:03:09 +00:00
henrike@webrtc.org	704bf9ebec	(Auto)update libjingle 62063505-> 62278774 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5617 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-27 17:52:04 +00:00
braveyao@webrtc.org	eaadecaf98	iOS, AppRTCDemo: Fixes exception due to JSON for ice using "urls" instead of "url", which is introduced by r5599. BUG=2962 TEST= R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9109005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5610 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-26 04:16:02 +00:00
henrike@webrtc.org	79a1cff65a	Android, AppRTCDemo: Fixes java exception due to JSON for ice using "urls" instead of "url". BUG=2952 TEST=Manual TBR=braveyao Review URL: https://webrtc-codereview.appspot.com/9099004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5605 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-24 23:22:18 +00:00
henrike@webrtc.org	91cbaa477c	(Auto)update libjingle 61966318-> 62063505 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5602 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-24 16:51:32 +00:00
henrike@webrtc.org	d43aa9de7a	Update libjingle 61901702->61966318 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5596 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-21 23:43:24 +00:00
henrike@webrtc.org	a7b981843f	Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702). BUG=N/A R=xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5595 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-21 15:51:43 +00:00
xians@webrtc.org	ef2215110c	Revert 5590 "description" > description TBR=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8949006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5593 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-21 10:31:29 +00:00
henrike@webrtc.org	2643805a20	description git-svn-id: http://webrtc.googlecode.com/svn/trunk@5590 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-20 22:32:53 +00:00
henrike@webrtc.org	571df2dca9	Update libjingle 61759961->61834300 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5580 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-19 23:04:26 +00:00
henrike@webrtc.org	5cf3e8f0f0	(Auto)update libjingle $LAST_P10_REVISION-> $NEW_P10_REVISION git-svn-id: http://webrtc.googlecode.com/svn/trunk@5572 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-18 22:28:52 +00:00
fischman@webrtc.org	358e3367a3	PeerConnection(java): enable HW encoder on N5 for standalone build. Now that bug 2899 is fixed (r5562) packet-loss is recoverable. Yay. BUG=2575 R=noahric@google.com Review URL: https://webrtc-codereview.appspot.com/8869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5568 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-18 17:29:37 +00:00
fischman@webrtc.org	c2d75e0708	PeerConnection(java): account for thread shutdown vagaries. Android's JVM requires threads to detach before they exit, but ONLY if they needed to AttachCurrentThread. Conversly, threads that were attached by the JVM (e.g. the result of making a native call from Java) must NOT be detached by the application. This is bug 2441. The fix for the above is to only pthread_setspecific() for threads that Attach(), not for already-attached threads. To ensure that we only detach Attached threads, added a GetEnv() call to ThreadDestructor(), which revealed that Oracle's JVM can overly-eagerly clear TLS accounting data, effectively detaching threads without their consent at shutdown. Work around this with a specific check. To guard against (some) regression, added a variant of PeerConnectionTest that runs on a non-main thread. This revealed a bug in LinuxDeviceManager which implicitly assumes its talk_base::Thread has already been initialized. Fixed that here too. BUG=2441 R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8759004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5567 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-18 16:57:36 +00:00
mallinath@webrtc.org	92fdfebedd	Update talk to 61699344. TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5560 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-17 18:49:41 +00:00
henrike@webrtc.org	b8c254abd6	(Auto)update libjingle 61549749-> 61608469 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5555 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-14 23:38:45 +00:00
fischman@webrtc.org	c5d506a106	AppRTCDemo(android): clarified README on how to launch app using adb. TBR=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8689005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5553 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-14 17:55:13 +00:00
fischman@webrtc.org	a3708ecdfe	PeerConnectionTest(java): unbreak following 61460797-p10 BUG=1414 R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5550 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-14 01:51:33 +00:00
mallinath@webrtc.org	385857dfd4	Update talk to 61549749. TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8709004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5549 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-14 00:56:12 +00:00
wu@webrtc.org	b9a088b920	Update talk to 61538839. TBR=mallinath Review URL: https://webrtc-codereview.appspot.com/8669005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5548 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-13 23:18:49 +00:00
wu@webrtc.org	0de29504ab	Revert 5545 "Update libjingle to 61514460" > Update libjingle to 61514460 > > TBR=tommi@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/8649004 TBR=xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8669004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5547 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-13 19:54:28 +00:00
xians@webrtc.org	e749c9ebdb	Update libjingle to 61514460 TBR=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5545 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-13 15:09:40 +00:00
fischman@webrtc.org	3eda643a91	PeerConnection(java): added MediaConstraints support to AudioSource, now fed to AudioTrack. BUG=2912 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5540 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-13 04:01:04 +00:00
fischman@webrtc.org	540acde5b3	PeerConnection(java): use MediaCodec for HW-accelerated video encode where available. Still disabled by default until https://code.google.com/p/webrtc/issues/detail?id=2899 is resolved. Also (because I needed them during development): - make AppRTCDemo "debuggable" for extra JNI checks - honor audio constraints served by apprtc.appspot.com - don't "restart" video when it hasn't been stopped (affects running with the screen off) BUG=2575 R=noahric@google.com Review URL: https://webrtc-codereview.appspot.com/8269004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5539 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-13 03:56:14 +00:00
jiayl@webrtc.org	14d80793a8	PeerConnectionClient needs to initialize SSL. BUG=2911 R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8499004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5531 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-12 00:41:59 +00:00
wjia@webrtc.org	dd82fa726c	Revert 5516 "Thread annotation of talk_base::CriticalSection." r5516 failed compilation on builds with enable_webrtc=0. > Thread annotation of talk_base::CriticalSection. > > Also enabling -Wthread-safety in talk/build/common.gypi for clang on > Linux. Thread annotations are compile-time checks that for instance > certain locks are held before accessing a value. > > BUG= > TEST=Local GUARDED_BY() annotations. > R=andresp@webrtc.org, fischman@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/8189004 TBR=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8409004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5523 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-10 23:20:15 +00:00
fischman@webrtc.org	82387e4608	Add ability to receive calls for iOS BUG=2701 R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7989005 Patch from Sajid Hussain <shussain@temasys.com.sg>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@5518 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-10 18:47:11 +00:00
pbos@webrtc.org	0a7085ffc2	Thread annotation of talk_base::CriticalSection. Also enabling -Wthread-safety in talk/build/common.gypi for clang on Linux. Thread annotations are compile-time checks that for instance certain locks are held before accessing a value. BUG= TEST=Local GUARDED_BY() annotations. R=andresp@webrtc.org, fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8189004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5516 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-10 13:58:37 +00:00
kjellander@webrtc.org	4723dc88b3	Revert 5511 "Revert 5510 "Disable failing libjingle_p2p_unittest..." So, the test apparently failed right away at http://build.chromium.org/p/client.webrtc/builders/Linux64%20Debug/builds/1224/steps/libjingle_p2p_unittest/logs/stdio > Revert 5510 "Disable failing libjingle_p2p_unittest test on Linux" > > According to https://code.google.com/p/webrtc/issues/detail?id=2907#c2 > r5505 was committed to resolve exactly these flakes. > Let's revert the disabling and see. > > BUG=2907 > TBR=mallinath@webrtc.org > > > Disable failing libjingle_p2p_unittest test on Linux > > > > I realize this diables 84 test cases and for all platforms, which > > I'm not really comfortable with. I tried finding a better way but > > couldn't without doing significant changes to the file. > > I think the tests either needs to be fixed or otherwise refactored > > in order to make more fine-grained disabling possible. > > > > Another (too) large disabling was done by holmer@ in > > https://webrtc-codereview.appspot.com/2227004 where he should only have > > disabled them on Windows, if the failures in webrtc:2383 was all that > > caused those flakes. > > > > BUG=2907 > > TEST=Verified this ran 0 tests: > > out/Release/libjingle_p2p_unittest --gtest_filter=P2PTransportChannelTest.TestNAT_ADDR_RESTRICTEDToNAT_PORT_RESTRICTEDAsGiceBothSharedUfragWithMinimumStepDelay > > TBR=wu@webrtc.org > > > > Review URL: https://webrtc-codereview.appspot.com/8309004 > > TBR=kjellander@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/8329004 TBR=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8339004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5513 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-09 18:48:56 +00:00
kjellander@webrtc.org	607c805b87	Roll chromium_revision 245382:249215 The find_depot_tools.py is needed to workaround the import error we get from gyp_chromium when importing it in webrtc/build/gyp_webrtc (to avoid code duplication). gyp_chromium introduced a dependency on it in http://crrev.com/245412 but as we cannot sync all of Chrome's src/tools (it's quite big), we'll work around this by adding an empty find_depot_tools module. The removal of the Cygwin relates to http://crrev.com/248802 which is a step on the way to remove Cygwin in Chromium. We seem to already be able to remove it entirely for WebRTC though. Changes in the isolate framework required us to update our copies of the isolate.gypi files. BUG=none TEST=trybots passing on all platforms R=andrew@webrtc.org, fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8099004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5512 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-09 18:38:31 +00:00
kjellander@webrtc.org	ce2b44532e	Revert 5510 "Disable failing libjingle_p2p_unittest test on Linux" According to https://code.google.com/p/webrtc/issues/detail?id=2907#c2 r5505 was committed to resolve exactly these flakes. Let's revert the disabling and see. BUG=2907 TBR=mallinath@webrtc.org > Disable failing libjingle_p2p_unittest test on Linux > > I realize this diables 84 test cases and for all platforms, which > I'm not really comfortable with. I tried finding a better way but > couldn't without doing significant changes to the file. > I think the tests either needs to be fixed or otherwise refactored > in order to make more fine-grained disabling possible. > > Another (too) large disabling was done by holmer@ in > https://webrtc-codereview.appspot.com/2227004 where he should only have > disabled them on Windows, if the failures in webrtc:2383 was all that > caused those flakes. > > BUG=2907 > TEST=Verified this ran 0 tests: > out/Release/libjingle_p2p_unittest --gtest_filter=P2PTransportChannelTest.TestNAT_ADDR_RESTRICTEDToNAT_PORT_RESTRICTEDAsGiceBothSharedUfragWithMinimumStepDelay > TBR=wu@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/8309004 TBR=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8329004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5511 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-09 18:21:00 +00:00
kjellander@webrtc.org	8d2ddd00f1	Disable failing libjingle_p2p_unittest test on Linux I realize this diables 84 test cases and for all platforms, which I'm not really comfortable with. I tried finding a better way but couldn't without doing significant changes to the file. I think the tests either needs to be fixed or otherwise refactored in order to make more fine-grained disabling possible. Another (too) large disabling was done by holmer@ in https://webrtc-codereview.appspot.com/2227004 where he should only have disabled them on Windows, if the failures in webrtc:2383 was all that caused those flakes. BUG=2907 TEST=Verified this ran 0 tests: out/Release/libjingle_p2p_unittest --gtest_filter=P2PTransportChannelTest.TestNAT_ADDR_RESTRICTEDToNAT_PORT_RESTRICTEDAsGiceBothSharedUfragWithMinimumStepDelay TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8309004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5510 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-08 21:35:20 +00:00
sergeyu@chromium.org	cc685acbdf	Disable AsyncInvokeTest.CancelInvoker test Test is flaky. BUG=b/12944358 TBR=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8289004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5508 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-08 19:59:00 +00:00
sergeyu@chromium.org	0178810659	Don't use LOG() in callback.h Because chromium is compiled with a different version of logging macros defined in logging.h that header cannot be used in headers that can also included from chromium code. Removed LOG_F(LS_WARNING) from callback.h . That issue would block this code from being rolled in chromium. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5507 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-08 03:18:03 +00:00
mallinath@webrtc.org	5a59ccbb6d	Switching to NSS random number generator and adding init method to unittests. R=jiayl@webrtc.org, sergeuy@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8259004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5505 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-07 23:22:00 +00:00
sergeyu@chromium.org	9cf037b831	Update libjingle to 61168196 R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8139004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5502 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-07 19:03:26 +00:00
pbos@webrtc.org	ea1c5ad58f	Fix gunit compilation on VS2012. In VS2012 compiling gunit or its dependencies triggers a lot of "'std::tuple' : too many template arguments" warnings. The workaround for this, done for gtest already, is to define _VARIADIC_MAX=10. BUG=2616 R=perkj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5493 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-06 13:17:20 +00:00
fischman@webrtc.org	6e08228525	PeerConnectionTest(java): remove the obsolete magical names of streams & tracks. BUG=1253 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7929005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5478 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-03 19:15:44 +00:00
fischman@webrtc.org	a06ebab1e1	PeerConnectionTest(java): test SCTP DataChannels. BUG=1408,2253,2626 R=jiayl@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7929004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5477 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-03 19:11:29 +00:00
mallinath@webrtc.org	ecd622eec3	Updating libjingle.gyp after addition new files yuvframescapturer.cc. TBR=pbos@webrc.org Review URL: https://webrtc-codereview.appspot.com/7919006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5476 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-03 17:17:05 +00:00
mallinath@webrtc.org	67ee6b9a62	Update talk to 60923971 Review URL: https://webrtc-codereview.appspot.com/7909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5475 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-02-03 16:57:16 +00:00
jiayl@webrtc.org	808b99b111	Disable a test assert which fails due to usrsctp not cleaned up in SctpDataEngine.cc BUG=2749 R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7739005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5460 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-29 19:44:40 +00:00
jiayl@webrtc.org	a576faf82a	Enable SCTP and use OPENSSL on Anroid and NSS on other platforms. It includes unit test fixes to properly initialize SSL if DTLS or SSL random number generator is used in the tests. The private key and certificate constant strings used in some tests are updated to be compatible with NSS. A few potentially overflow type conversions caused compiling warning on Windows and they are fixed by importing and using Chromium's checked_cast, which aborts the program if overflow occurs. It also fixes a leak in nssstreamadapter.cc by releasing the PRFileDesc* in StreamClose. BUG=2253 R=fischman@webrtc.org, juberti@google.com, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4679005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5459 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-29 17:45:53 +00:00
mallinath@webrtc.org	7433a088d2	Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..." We reverted the r5421 to allow us roll webrtc to chrome without any modifications to libjingle. Since webrtc is rolled with r5444, we can add back the original CL and changes to libjingle will be upstreamed in the next roll. TBR=andresp@webrtc.org > Revert 5421 "Fix deadlock on register/unregister observer while ..." > > Failure to compile on Chromium Internal bots, because of API changes. > > http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio > > You need to follow the steps mentioned in > https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer. > > Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs > as mentioned in the doc. > > > Fix deadlock on register/unregister observer while there is a an going callback. > > > > BUG=2835 > > R=mallinath@webrtc.org > > > > Review URL: https://webrtc-codereview.appspot.com/7119005 > > TBR=andresp@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/7679004 TBR=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7729005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5453 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-29 00:56:02 +00:00
mallinath@webrtc.org	0dac5378e5	Revert 5447 "Update talk to 60420316." > Update talk to 60420316. > > TBR=wu@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/7719005 TBR=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5448 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-28 06:58:42 +00:00
mallinath@webrtc.org	752a017809	Update talk to 60420316. TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7719005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5447 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-28 06:45:52 +00:00
mallinath@webrtc.org	18586d38bc	Revert 5421 "Fix deadlock on register/unregister observer while ..." Failure to compile on Chromium Internal bots, because of API changes. http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio You need to follow the steps mentioned in https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer. Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs as mentioned in the doc. > Fix deadlock on register/unregister observer while there is a an going callback. > > BUG=2835 > R=mallinath@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/7119005 TBR=andresp@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7679004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5444 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-27 22:00:57 +00:00
wu@webrtc.org	256d0ada35	Remove the check for audio codec num in WebRtcVoiceEngineTest.HasCorrectCodecs. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5430 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-24 18:58:51 +00:00
wu@webrtc.org	ca5ff9972e	Re-enable webrtcvoice/videoengine unittests. TEST=try bots BUG= R=mallinath@webrtc.org Committed: https://code.google.com/p/webrtc/source/detail?r=5387 Review URL: https://webrtc-codereview.appspot.com/7149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5427 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-24 17:37:46 +00:00
andresp@webrtc.org	8d375c95b7	Fix deadlock on register/unregister observer while there is a an going callback. BUG=2835 R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7119005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5421 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-23 23:09:25 +00:00
wu@webrtc.org	a8910d2f88	Update talk to 60094938. Review URL: https://webrtc-codereview.appspot.com/7489005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5420 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-23 22:12:45 +00:00
mallinath@webrtc.org	0d92ef67c4	Libjingle source code has some spelling mistakes and one of them is "renegotation", which should be "renegotiation". This CL is attempting to correct those. BUG=2810 TBR=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7439004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5411 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-22 02:21:22 +00:00
mallinath@webrtc.org	68cbd01216	enabling disabled data channels tests on win32. The real culprit was that ice candidates not included in SDP when there were failure causing transport channels never becoming writable. BUG=2799 R=jiayl@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7369004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5410 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-22 00:16:46 +00:00
henrike@webrtc.org	28da47c52f	Android example apps: fixes issue where useful failure information was suppressed. BUG=2808 R=andrew@webrtc.org, fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7309004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5408 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-21 19:03:51 +00:00
henrike@webrtc.org	2ce9a64b75	Talk: Removes deprecated example apps and moves the server apps to trunk/talk/examples. BUG=12545067 R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7159004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5397 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-16 16:49:53 +00:00
sergeyu@chromium.org	4b26e2eee3	Update libjingle to 59676287 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7229004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5390 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-15 23:15:54 +00:00
wu@webrtc.org	8f19cb9fbc	Revert 5387 "Re-enable webrtcvoice/videoengine unittests." Missed the result from the last try bot. > Re-enable webrtcvoice/videoengine unittests. > > TEST=try bots > BUG= > R=mallinath@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/7149004 TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5388 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-14 22:31:11 +00:00
wu@webrtc.org	eda6823397	Re-enable webrtcvoice/videoengine unittests. TEST=try bots BUG= R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5387 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-14 22:15:09 +00:00
wjia@webrtc.org	03cfde2d10	Roll Chromium 238260 -> 243863 R=andrew@webrtc.org, henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6939004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5385 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-14 17:48:34 +00:00
henrika@webrtc.org	aebb1ade9d	pRevert 5371 "Revert 5367 "Update talk to 59410372."" > Revert 5367 "Update talk to 59410372." > > > Update talk to 59410372. > > > > R=jiayl@webrtc.org, wu@webrtc.org > > > > Review URL: https://webrtc-codereview.appspot.com/6929004 > > TBR=mallinath@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/6999004 TBR=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7109004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5381 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-14 10:00:58 +00:00
fischman@webrtc.org	d7568a08c3	PeerConnection(java): Add OnRenegotiationNeeded support Also: - Make PeerConnectionObserver::OnRenegotiationNeeded() pure virtual to avoid this sort of mistake in the future. - Sprinkle @Override annotations on some callback definitions that were missing them. - Fix a JNI method-signature-lookup typo (s/(V)V/()V/) in PCOJava::OnError() - Add an explicit ScopedLocalFrameRef to PCOJava::OnError() to match all other C++-fired callbacks, for consistency. BUG=2771 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5376 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-13 22:04:12 +00:00
henrika@webrtc.org	44461fa5cb	Revert 5367 "Update talk to 59410372." > Update talk to 59410372. > > R=jiayl@webrtc.org, wu@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/6929004 TBR=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5371 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-13 09:35:02 +00:00
mallinath@webrtc.org	0f3356e20b	Update talk to 59410372. R=jiayl@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6929004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5367 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-11 01:26:23 +00:00
sergeyu@chromium.org	4625df3e3e	Fix NaCl compilation nethelpers.cc was using LOG() but didn't include logging.h R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6829005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5360 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-09 21:26:50 +00:00
fischman@webrtc.org	4177615e87	PeerConnection(java): replace ScopedLocalRef with ScopedLocalRefFrame and fix a local reference leak in OnMessage. Hopefully the approach of pushing/popping frames will be easier to avoid messing up than remembering to annotate every single local reference with a ScopedLocalRef. BUG=2761 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5355 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-09 00:31:17 +00:00
fischman@webrtc.org	1794693ec8	AppRTCDemo(android): close() the throw-away DataChannel. Otherwise, the PeerConnection remembers the channel enough to include an m=application line in its offer SDP, causing connection to chrome to fail, since apprtc.appspot.com doesn't include an RtpDataChannels:true constraint in its RTCPeerConnection constructor call. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6729005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5354 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-08 18:29:34 +00:00
wu@webrtc.org	e00265ed49	Fix a compile error on Android on sctpdataengine.cc. TEST=try bots BUG= R=jiayl@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5350 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-07 19:32:40 +00:00
wu@webrtc.org	f6d6ed0c66	Update talk to 59039880. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5339 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-01-03 22:08:47 +00:00
fischman@webrtc.org	000dde99c8	Android build: make it quiet on success and not overly noisy on failure. - OpenSLDemo and WebRTCDemo get the sauce that AppRTCDemo got in r5271 - libjingle_peerconnection_jar is now silent on success - Fix a bug introduced by r5271 which caused ant logs to be emitted to a subdir of talk/examples instead of in the gyp output directory. R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6199005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5332 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-20 22:49:35 +00:00
fischman@webrtc.org	af320fd2f7	The designated initializer method declaration in the Objective-C headers for RTCICEServer does't match its implementation. R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6019004 Patch from Rafael Lopez Diez <rafalopezdiez@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@5309 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-17 21:33:27 +00:00
fischman@webrtc.org	5b3c67ef25	objc/README: Remove outdated advice about target_os. BUG=chromium:248168 TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5979005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5302 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-16 17:15:19 +00:00
wu@webrtc.org	24301a67c6	Update talk to 58174641 together with http://review.webrtc.org/4319005/. R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5287 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-13 19:17:43 +00:00
mallinath@webrtc.org	62451dcba0	Update talk to 58157731. R=wu@webrtc.org TBR=wu@webrc.org Review URL: https://webrtc-codereview.appspot.com/5339005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5282 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-13 12:29:34 +00:00
wu@webrtc.org	a9890800e0	Update talk to 58127566 together with https://webrtc-codereview.appspot.com/5309005/. R=mallinath@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5277 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-13 00:21:03 +00:00
wu@webrtc.org	2018269dc3	Revert 5274 "Update talk to 58113193 together with https://webrt ..." > Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. > > R=mallinath@webrtc.org, niklas.enbom@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/5719004 TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5275 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-12 22:54:25 +00:00
wu@webrtc.org	a129b6cd13	Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. R=mallinath@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-12 22:40:39 +00:00
fischman@webrtc.org	df7b1d6e39	AppRTCDemo(android): make ant be quiet on success and not overly noisy on failure. Also silence a 'cd' that would otherwise emit the path/to/talk. R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5539004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5271 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-11 22:36:22 +00:00
henrike@webrtc.org	9ee75e9c77	Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency). BUG=N/A R=fischman@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4189004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5270 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-11 21:42:44 +00:00
fischman@webrtc.org	f41f06b916	PeerConnection(java): rationalize pointer-to-jlong conversion. In r4665 I went a bit crazy with the manual reinterpretation of a pointer to a jlong (to avoid undefined behavior) but that's what reinterpret_cast<> is for. So use it directly now. Added a do-nothing DataChannel to AppRTCDemo to regression test this, since the only repro I've found of the original bug requires ARM ABI (PeerConnectionTest on ia32 fails to repro). BUG=2302 R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5489004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5269 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-11 21:07:18 +00:00
wu@webrtc.org	9caf2765b2	Update talk to 58037405. R=sergeyu@chromium.org Review URL: https://webrtc-codereview.appspot.com/5579004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5267 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-11 18:25:07 +00:00
turaj@webrtc.org	4c3faa9d73	Disable a libjingle unittest which is failing after a chromium roll out. TBR=kjellander@google.com BUG= Review URL: https://webrtc-codereview.appspot.com/5559007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5264 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-11 16:12:31 +00:00
kjellander@webrtc.org	f9bdbe3619	Roll chromium_revision 232627:238260 This brings us the updated swarming_client that has moved out from Chromium into a standalone project. Because of this, all .isolate files needed to be updated as well, similar to the changes in https://codereview.chromium.org/29993003 TEST=trybots passing BUG=none R=andrew@webrtc.org, perkj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4859004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5260 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-11 13:37:12 +00:00
andrew@webrtc.org	77507eff4f	Correctly define OVERRIDE when building with g++ 4.7 and C++11 support g++ 4.7 and later support explicit virtual overrides when building with C++11 support enabled. However, libjingle does not detect that and makes OVERRIDE a no-op. This CL updates base/common.h to define OVERRIDE properly when g++ 4.7 is used with C++11 support enabled. See this page for GCC support of C++11 features: http://gcc.gnu.org/projects/cxx0x.html R=fischman@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5159004 Patch from Chris Dumez <ch.dumez@samsung.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@5255 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-11 00:07:11 +00:00
fischman@webrtc.org	eb7def234e	Fix compilation errors on Fedora 20. peerconnection_jni.cc: syscall() comes from <unistd.h> RTPtimeshift.cc: char* being compared to 0 instead of the atoi() of it rtp_payload_registry_unittest.cc: avoid narrowing int to uint32. BUG=2700 R=andrew@webrtc.org, fischman@webrtc.org, henrik.lundin@webrtc.org, henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5019004 Patch from Victor Costan <costan@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@5248 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-09 21:34:30 +00:00
sergeyu@chromium.org	32f485b16a	Fix constants.[h\|cc] to avoid static initializer in webrtcvideoengine.cc. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5233 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-05 22:36:21 +00:00
sergeyu@chromium.org	57a5f64264	revert r5230 r5230 broke windows build. TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5169004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5232 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-05 22:14:46 +00:00
sergeyu@chromium.org	a1b21cd777	Fix constants.[h\|cc] to avoid static initializer in webrtcvideoengine.cc. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5139004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5230 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-05 21:28:34 +00:00
sergeyu@chromium.org	5bc25c41fc	Update libjingle to 57692857 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5217 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-05 00:24:06 +00:00
fischman@webrtc.org	e0034557a7	RTCPeerConnection(objc): avoid leaking ICE candidate on addition. BUG=2670 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4379004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5199 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-12-02 18:49:54 +00:00
wu@webrtc.org	b43202d839	Disable PeerConnectionEndToEndTest for tsanv2 build. BUG=1205 TEST=try R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5162 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-11-22 19:14:25 +00:00
fischman@webrtc.org	1977960866	AppRTCDemo(ios): remove codesigning hack now that gyp signs by default. R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4119005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5155 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-11-21 16:48:51 +00:00
wu@webrtc.org	364f204d16	Update talk to 56698267. TBR=sergeyu@chromium.org Review URL: https://webrtc-codereview.appspot.com/4119004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5143 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-11-20 21:49:41 +00:00
sergeyu@chromium.org	183c727bca	Disable datachannel_unittest.cc the test fails to compile because it uses incorrect gmock path (as some other tests). TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3849004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5121 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-11-13 22:59:20 +00:00
sergeyu@chromium.org	a23f0ca4ba	Update talk to 56619788 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3839005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5120 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-11-13 22:48:52 +00:00
wu@webrtc.org	16d6254e8c	Update talk to 56183333. TEST=try bots R=sheu@chromium.org Review URL: https://webrtc-codereview.appspot.com/3469004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5087 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-11-05 23:45:14 +00:00
fischman@webrtc.org	7b273a545d	PeerConnection iOS: update README instructions This is needed to account for https://codereview.chromium.org/25535004/ R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5079 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-11-04 18:48:12 +00:00
wu@webrtc.org	07a6fbe83d	Update talk to 56092586. R=jiayl@webrtc.org, mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5078 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-11-04 18:41:34 +00:00
wu@webrtc.org	de305014c6	Update talk to 55906045. Review URL: https://webrtc-codereview.appspot.com/3159005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5065 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-10-31 15:40:38 +00:00
wu@webrtc.org	f424cb8e13	Update talk to 55863981. TBR=mallinath Review URL: https://webrtc-codereview.appspot.com/3089006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5056 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-10-30 17:57:33 +00:00
wu@webrtc.org	cecfd1832d	Update talk to 55821645. TEST=try bots R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3139004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5053 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-10-30 05:18:12 +00:00
fischman@webrtc.org	9ca93a8b8e	Explicitly @synthesize ObjC @properties This is required after https://code.google.com/p/gyp/source/detail?r=1768 turned on -Wobjc-missing-property-synthesis for ninja builds (until then it was only enabled for xcode builds) to allow chromium_deps to roll in webrtc/DEPS. BUG=2560 R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5047 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-10-29 00:14:15 +00:00
pbos@webrtc.org	850bcbe855	Remove frame_callback.h include in webrtcvie.h. This file is about to be moved and it's not really needed. The class I420FrameCallback is forward declared inside vie_image_process.h and only used in talk/ for a no-op implementation that doesn't access the pointer. BUG= R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3039004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5041 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-10-28 15:41:17 +00:00
wu@webrtc.org	97077a3ab2	Update libjingle to 55618622. Update libyuv to r826. TEST=try bots R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2889004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5038 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-10-25 21:18:33 +00:00
wu@webrtc.org	d371a29227	Fix tsan failures for libjingle_unittest. 1) Change AsyncSocket's SignalReadEvent and SignalWriteEvent's thread mode to multi_threaded_local as they can be accessed from different threads. 2) Protect NATServer::TransEntry::whitelist. 3) Protect PhysicalSocket:error_. Detail failures can be seen from issue 2080, comment #5. TBR=fischman@webrtc.org RISK=P1 TEST=try bots and tsanv2 BUG=2080 Review URL: https://webrtc-codereview.appspot.com/2669005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5026 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-10-23 23:56:09 +00:00
wu@webrtc.org	8804a29951	Add CriticalSection to fakeaudiocapturemodule to protect the variables which will be accessed from process_thread_ and the main thread. TEST=try bots BUG=1205 R=henrike@webrtc.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5019 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-10-22 23:09:20 +00:00
wu@webrtc.org	4d7116be7a	Fix tsan failures on filevideocapturer.cc. 1) init start_time_ns_ before the file_read_thread_ is started to avoid data racing as the start_time_ns_ will also be read by the file_read_thread_. 2) add CriticalSection to protect \|finished_\| that is accessed by FileReadThread and the main thread Also remove the suppression for filemediaengine.cc, which has already been fixed in other cl. TBR=henrike@webrtc.org TEST=try bots and manual tsan v2 test BUG=2078 Review URL: https://webrtc-codereview.appspot.com/2509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5018 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-10-22 18:41:17 +00:00
andrew@webrtc.org	31628aae7e	Upgrade scoped_ptr to Chromium's latest version. Analogous to the recent libjingle change: http://cl/54929753-p10. This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather than scoped_array and scoped_ptr_malloc respectively. - Add Chromium's template-based COMPILE_ASSERT. We didn't have this previously in order to support the macro in C. Instead, move the existing macro to compile_assert_c.h. - Additionally copy the move.h and template_util.h depedencies and add the WARN_UNUSED_RESULT macro. - Leave scoped_array and scoped_ptr_malloc for now, but mark as deprecated. - Remove scoped_ptr foo(NULL) use. The default constructor handles it. - Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc. - Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove some repeated code. TESTED=trybots R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2449005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5015 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-10-22 12:50:00 +00:00
mallinath@webrtc.org	50bc553852	Reenable DTLS renegotiation unittest in libjingle. This test is failing on memcheck bots. After investigation problem per say is not in this particular unittest and rather is in suite. So this test is added to memcheck exclude list. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2689004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5011 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-10-21 17:58:35 +00:00
wu@webrtc.org	3c5d2b43ec	Thread::Stop() must be called before any subclass's destructor completes. Update Thread documentation, fix all subclasses that had a problem. This is to avoid a data racing between the destructor modifying the vtable, and Thread::PreRun calling virtual method Run at the same time. For example: [ RUN ] FileMediaEngineTest.TestGetCapabilities ================== WARNING: ThreadSanitizer: data race on vptr (ctor/dtor vs virtual call) (pid=2967) Read of size 8 at 0x7d480000bd00 by thread T1: #0 talk_base::Thread::PreRun(void*) /mnt/data/b/build/slave/Linux_Tsan_v2/build/src/out/Release/../../talk/base/thread.cc:353 (libjingle_media_unittest+0x000000234da8) Previous write of size 8 at 0x7d480000bd00 by main thread: #0 talk_base::Thread::~Thread() /mnt/data/b/build/slave/Linux_Tsan_v2/build/src/out/Release/../../talk/base/thread.cc:158 (libjingle_media_unittest+0x00000023478c) #1 ~RtpSenderReceiver /mnt/data/b/build/slave/Linux_Tsan_v2/build/src/out/Release/../../talk/media/base/filemediaengine.cc:122 (libjingle_media_unittest+0x0000001b551f) ... RISK=P2 TESTED=try bots and tsan BUG=2078,2080 R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2428004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4999 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-10-18 16:27:26 +00:00
fischman@webrtc.org	1c82037494	AppRTCDemo(android): remove vestigial mentions of PowerManager R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2402004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4995 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-10-17 20:53:12 +00:00
wu@webrtc.org	1d1ffc9ad2	Update talk to 54898858. TEST=try bots TBR=mallinath Review URL: https://webrtc-codereview.appspot.com/2414004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4979 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-10-16 18:12:02 +00:00
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

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812 Commits