Age | Commit message (Collapse) | Author |
|
- 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
|
|
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
|
|
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
|
|
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
|
|
BUG=3407
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16619006
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6345 4adac7df-926f-26a2-2b94-8c16560cd09d
|
|
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
|
|
{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
|
|
- 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
|
|
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
|
|
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
|
|
- 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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
- 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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
trunk/talk
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
|