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path: root/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
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2014-09-15HW video decoding optimization to better support HD resolution:glaznev@webrtc.org
- Change hw video decoder wrapper to allow to feed multiple input and query for an output every 10 ms. - Add an option to decode video frame into an Android surface object. Create shared with video renderer EGL context and external texture on video decoder thread. - Support external texture rendering in Android renderer. - Support TextureVideoFrame in Java and use it to pass texture from video decoder to renderer. - Fix HW encoder and decoder detection code to avoid query codec capabilities from sw codecs. BUG= R=tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18299004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7185 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-19Active connection stats [LocalAddress,RemoteAddress,LocalCandidateType...etc]houssainy@google.com
is now printed in the head-up display in Android appRTC. This printing will be usefull in debugging switching ICE candidates. R=andresp@webrtc.org, glaznev@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13189005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6927 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06Add OpenGL Android video renderer which can display multipleglaznev@webrtc.org
yuv420 images in a single GLSurfaceView. Start using new video renderer in AppRTC demo app. BUG= R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6360 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06AppRTCDemo(android): support app (UI) & capture rotation.fischman@webrtc.org
Now app UI rotates as the device orientation changes, and the captured stream tries to maintain real-world-up, matching Chrome/Android and Hangouts/Android behavior. BUG=2432 R=glaznev@webrtc.org, henrike@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15689005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6354 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-29AppRTCDemo(android): run in full-screen & immersive mode.fischman@webrtc.org
Also: - Only show stats HUD on demand - Only collect stats when HUD is showing - Don't render solid green frame when video is not present in either direction R=glaznev@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6275 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14PeerConnection(android): allow initializing either (or neither) of ↵fischman@webrtc.org
{Voice,Video}Engine. Enables applications that don't want to pay the init/startup cost or request extra permissions (e.g. audio-only app, or DataChannel-only app). BUG=3234 Review URL: https://webrtc-codereview.appspot.com/15489004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6164 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-01AppRTCDemo(android): added a Heads-Up Display for bandwidth estimation.fischman@webrtc.org
- tap display to toggle visibility - increased getStats frequency to 1hz. R=glaznev@google.com Review URL: https://webrtc-codereview.appspot.com/19419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6039 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-01AppRTCDemo(android): send the created SDP, not the local description after ↵fischman@webrtc.org
setting it This is required to allow explicit filtering of ICE candidates. BUG=3288 R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6038 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28AppRTCDemo(android): don't initialize process-globals more than once.fischman@webrtc.org
BUG=3257 R=braveyao@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19369004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6001 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-04AppRTCDemo(android): fix a couple of SDP-related regressions.fischman@webrtc.org
- 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/talk@5847 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-28AppRTCDemo(android): specify DtlsSrtpKeyAgreement:true in ↵fischman@webrtc.org
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/talk@5809 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
2014-02-13PeerConnection(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
2014-01-13PeerConnection(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
2013-12-11PeerConnection(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
2013-10-17AppRTCDemo(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
2013-09-03AppRTCDemo(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
2013-08-12PeerConnection 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
2013-07-12Update 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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