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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
2014-09-02Remove deprecated RTCVideoRenderer constructor.tkchin@webrtc.org
Removes -[RTCVideoRenderer initWithView]. Also, fix potential issue where we hold on to a video frame longer than the lifetime of its associated track. BUG=3341 R=glaznev@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16099004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7032 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29win: Replace custom assert() macro with regular assert.hthakis@chromium.org
The current code got added in libjingle r103; I don't see a good reason for it. Things still build with plain old assert.h. The custom assert was wrong: __debugbreak() is documented to return void, so doing `cond ? true : __debugbreak()` shouldn't build (and it doesn't in clang-cl). It's possible to make it build by writing `cond ? true : (__debugbreak(), true)`, but just using the regular header seems like a much better fix. BUG=chromium:82385 Review URL: https://webrtc-codereview.appspot.com/19139004/ git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7007 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-22(Auto)update libjingle 73927658-> 73927775buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6958 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21Making sure muc members get recorded.phoglund@webrtc.org
This is an upstream of a change I made; will describe in a separate email thread. Essentially, the members map wasn't getting populated in the callclient example, so it was always empty. Now it will be populated correctly. R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13189004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6950 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
2014-08-13(Auto)update libjingle 73222930-> 73226398buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-07Add new OWNERS file to talk/examples.tkchin@webrtc.org
R=juberti@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/15039004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6851 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29(Auto)update libjingle 72097588-> 72159069buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-28(Auto)update libjingle 72016417-> 72097588buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6792 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-24(Auto)update libjingle 71829282-> 71834788buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6773 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23(Auto)update libjingle 71766184-> 71775619buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6768 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18Revert "Reland r6707 with the fix for callclient.cc."jiayl@webrtc.org
Breaking pulse build again. This reverts commit 3e0bb9b5bf7f616000399e24f1d9622ad6b612f9. TBR=wu@webrtc.org BUG=3310 Review URL: https://webrtc-codereview.appspot.com/17979004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6740 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18Reland r6707 with the fix for callclient.cc.jiayl@webrtc.org
TBR=mallinath@webrtc.org BUG=3310 Review URL: https://webrtc-codereview.appspot.com/13039004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6737 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15Handle the case if an unusually long peer name is provided in the ↵tommi@webrtc.org
peerconnection example. R=xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6687 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01Implement command line flags for peerconnection client example on Windowskjellander@webrtc.org
Adding the flags and functionality for 'autoconnect', 'autocall', 'server', 'port', and 'help' like in the linux example. BUG=3459 R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13609004 Patch from Vicken Simonian <vsimon@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6573 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06APPRTCDemo(objc): Remove regex parsing in favor of JSON struct.tkchin@webrtc.org
Also some cleanup/refactoring of APPRTCAppClient. R=fischman@webrtc.org, glaznev@webrtc.org BUG=3407 Review URL: https://webrtc-codereview.appspot.com/18499004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6362 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
2014-06-05AppRTCDemo(android): remove HTML/regex hackery in favor of JSON struct.fischman@webrtc.org
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
2014-06-04Fix retain cycle in RTCEAGLVideoView.tkchin@webrtc.org
CADisplayLink increases its target's refcount. In order to break retain cycle, we wrap CADisplayLink in a new RTCDisplayLinkTimer class and use that instead. R=fischman@webrtc.org, noahric@chromium.org BUG=3391 Review URL: https://webrtc-codereview.appspot.com/16599006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6331 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-03talk/ios: Fixes source after corrupt sync in r6305 (which corrupted r6291).henrike@webrtc.org
BUG=N/A R=tkchin@webrtc.org TBR=tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6318 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-02(Auto)update libjingle 68275107-> 68379861buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6305 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30Implement mac version of AppRTCDemo.tkchin@webrtc.org
- Refactored and moved AppRTCDemo to support sharing AppRTC connection code between iOS and mac counterparts. - Refactored OpenGL rendering code to be shared between iOS and mac counterparts. - iOS AppRTCDemo now respects video aspect ratio. BUG=2168 R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6291 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
2014-05-28AppRTCDemo(android): README updates for a shrinking envsetup.sh world.fischman@webrtc.org
There was duplicated (and out of date!) information in README relative to getting-started so de-duped to point to getting-started as the canonical reference. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15589006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6265 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23(Auto)update libjingle 67872893-> 67873348buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6244 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23(Auto)update libjingle 67869540-> 67872893buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6243 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19Add a UIView for rendering a video track.tkchin@webrtc.org
RTCEAGLVideoView provides functionality to render a supplied RTCVideoTrack using OpenGLES2. R=fischman@webrtc.org BUG=3188 Review URL: https://webrtc-codereview.appspot.com/12489004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6192 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
2014-04-30Implement ObjC DataChannel wrappertkchin@webrtc.org
R=fischman@webrtc.org BUG=3112 Review URL: https://webrtc-codereview.appspot.com/16369004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6031 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-30AppRTCDemo was blocking the main thread for network requests. This fixes it ↵fischman@webrtc.org
by making the background queue serial instead of using @synchronize to make the background operations serial. R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16379004 Patch from Bridger Maxwell <bridgeyman@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6028 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
2014-04-26Use CreatePeerConnection method which accepts port_allocator.mallinath@webrtc.org
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/talk@5987 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22Provide GetStats method in RTCPeerConnectiontkchin@webrtc.org
BUG=3144 R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12069006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5960 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-21Fix typo by renaming RTCSessionDescriptonDelegate -> ↵tkchin@webrtc.org
RTCSessionsDescriptionDelegate R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5946 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
2014-03-31AppRTCDemo(iOS): now works in the iOS Simulator!fischman@webrtc.org
...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/talk@5815 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-31AppRTCDemo(iOS): remote-video reliability fixesfischman@webrtc.org
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/talk@5814 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
2014-03-26Fix after auto update in r5787. APPRTCVideoView.h/m was removed incorrectly.henrike@webrtc.org
BUG=3121 R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10739004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5793 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-26(Auto)update libjingle 63738002-> 63773382henrike@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5787 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25Fix after auto update: removed files were brought back.henrike@webrtc.org
BUG=N/A R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5782 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25AppRTCDemo(iOS): allow rooms with no incoming audio.fischman@webrtc.org
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/talk@5780 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25(Auto)update libjingle 63648983-> 63738002henrike@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5779 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25PeerConnection(iOS): make ARC-clean talk/.../objc* and talk/examples/iosfischman@webrtc.org
- 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/talk@5771 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25AppRTCDemo(ios): style/cleanup fixes following cr/62871616-p10fischman@webrtc.org
BUG=2168 R=noahric@google.com Review URL: https://webrtc-codereview.appspot.com/9709004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5768 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-10(Auto)update libjingle 62865357-> 62871616henrike@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@5674 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-07 04:45:04 +0000