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2014-03-07Remove std:: prefixes from C functions in talk/.pbos@webrtc.org
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/talk@5657 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-03Update libjingle 62364298->62472237henrike@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5632 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-03Rollback of r5629 "(Auto)update libjingle 62364298-> 62368661".henrike@webrtc.org
BUG=N/A R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9329004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5631 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-03(Auto)update libjingle 62364298-> 62368661henrike@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5629 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-01AppRTCDemo(android): don't send local SDP until it's set.fischman@webrtc.org
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/talk@5625 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-28(Auto)update libjingle 62293974-> 62364298henrike@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-26iOS, AppRTCDemo: Fixes exception due to JSON for ice using "urls" instead of ↵braveyao@webrtc.org
"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/talk@5610 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-24Android, AppRTCDemo: Fixes java exception due to JSON for ice using "urls" ↵henrike@webrtc.org
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/talk@5605 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14AppRTCDemo(android): clarified README on how to launch app using adb.fischman@webrtc.org
TBR=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8689005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5553 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13PeerConnection(java): added MediaConstraints support to AudioSource, now fed ↵fischman@webrtc.org
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/talk@5540 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13PeerConnection(java): use MediaCodec for HW-accelerated video encode where ↵fischman@webrtc.org
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/talk@5539 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12PeerConnectionClient needs to initialize SSL.jiayl@webrtc.org
BUG=2911 R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8499004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5531 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10Add ability to receive calls for iOSfischman@webrtc.org
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/talk@5518 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16Talk: Removes deprecated example apps and moves the server apps to ↵henrike@webrtc.org
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/talk@5397 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-15Update libjingle to 59676287sergeyu@chromium.org
R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7229004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5390 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-13PeerConnection(java): Add OnRenegotiationNeeded supportfischman@webrtc.org
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/talk@5376 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-08AppRTCDemo(android): close() the throw-away DataChannel.fischman@webrtc.org
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/talk@5354 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12Revert 5274 "Update talk to 58113193 together with https://webrt..."wu@webrtc.org
> 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/talk@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12Update talk to 58113193 together with ↵wu@webrtc.org
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/talk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11PeerConnection(java): rationalize pointer-to-jlong conversion.fischman@webrtc.org
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/talk@5269 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11Roll chromium_revision 232627:238260kjellander@webrtc.org
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/talk@5260 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05Update libjingle to 57692857sergeyu@chromium.org
R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5217 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13Update talk to 56619788sergeyu@chromium.org
R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3839005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5120 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-29Explicitly @synthesize ObjC @propertiesfischman@webrtc.org
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/talk@5047 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-25Update libjingle to 55618622.wu@webrtc.org
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/talk@5038 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-17AppRTCDemo(android): remove vestigial mentions of PowerManagerfischman@webrtc.org
R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2402004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@4995 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03AppRTCDemo(android): support boolean value for ↵fischman@webrtc.org
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/talk@4918 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03Android standalone: remove some usages of deprecated APIs and prevent ↵fischman@webrtc.org
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/talk@4916 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03VideoCaptureAndroid: rewrote the (standalone) implementation of video ↵fischman@webrtc.org
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/talk@4915 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03AppRTCDemo(android): uncaught exceptions now display a modal dialog box ↵fischman@webrtc.org
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/talk@4914 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-01PeerConnection(Android): enable tracing to logcat.fischman@webrtc.org
BUG=1295 R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2258007 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@4888 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05AppRTCDemo(iOS): prefer ISAC as audio codecfischman@webrtc.org
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/talk@4687 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03AppRTCDemo(android): prefer ISAC for audio codec.fischman@webrtc.org
R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2126004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@4666 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-22AppRTCDemo(android): allow audio-only calls to test iOS interopfischman@webrtc.org
R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2091004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@4598 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-20iOS: unbreak the build following r4546fischman@webrtc.org
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/talk@4577 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-12PeerConnection shutdown-time fixesfischman@webrtc.org
- 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/talk@4534 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-06PeerConnection.java: enable setting trace & log levels from Javafischman@webrtc.org
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/talk@4498 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-01AppRTCDemo: builds using ninja on iOS for simulator and device!fischman@webrtc.org
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/talk@4466 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-29AppRTCDemo: replace the use of query-string parameters for pre-JB devices.fischman@webrtc.org
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/talk@4421 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-24AppRTCDemo: don't render frames that are already outdated.fischman@webrtc.org
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/talk@4392 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-22Update talk folder to revision=49713299.henrike@webrtc.org
TBR=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1848004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@4380 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12Update talk folder to revision=49260075. Same as 369 in libjingle's google ↵henrike@webrtc.org
code repository. TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1797004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@4338 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10Adds trunk/talk folder of revision 359 from libjingles google code tohenrike@webrtc.org
trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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