generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
7,261 Commits 1 Branch 0 Tags 108 MiB
d95435c17a
Commit Graph

140 Commits

Author SHA1 Message Date
buildbot@webrtc.org
3740d74106 (Auto)update libjingle 73927658-> 73927775 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6958 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-22 22:27:04 +00:00
phoglund@webrtc.org
7bd5fefb17 Making sure muc members get recorded. 
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@6950 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-21 09:53:28 +00:00
houssainy@google.com
d5b292e450 Active connection stats [LocalAddress,RemoteAddress,LocalCandidateType...etc] 
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@6927 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-19 11:43:32 +00:00
buildbot@webrtc.org
a09a99950e (Auto)update libjingle 73222930-> 73226398 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 17:26:08 +00:00
tkchin@webrtc.org
cb46de24fb Add new OWNERS file to talk/examples. 
R=juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/15039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6851 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 20:01:34 +00:00
buildbot@webrtc.org
d4e598d57a (Auto)update libjingle 72097588-> 72159069 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-29 17:36:52 +00:00
buildbot@webrtc.org
51c5508bf1 (Auto)update libjingle 72016417-> 72097588 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6792 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-28 22:26:15 +00:00
buildbot@webrtc.org
45304ff0a7 (Auto)update libjingle 71829282-> 71834788 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6773 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-24 16:06:35 +00:00
buildbot@webrtc.org
e2da234e27 (Auto)update libjingle 71766184-> 71775619 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 21:09:01 +00:00
jiayl@webrtc.org
a0b929b63c Revert "Reland r6707 with the fix for callclient.cc." 
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@6740 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 22:28:36 +00:00
jiayl@webrtc.org
a6e8cf8fb7 Reland r6707 with the fix for callclient.cc. 
TBR=mallinath@webrtc.org
BUG=3310

Review URL: https://webrtc-codereview.appspot.com/13039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6737 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 21:34:11 +00:00
tommi@webrtc.org
2adc51c86e Handle the case if an unusually long peer name is provided in the peerconnection example. 
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6687 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 08:56:07 +00:00
kjellander@webrtc.org
0402515d35 Implement command line flags for peerconnection client example on Windows 
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@6573 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 16:28:13 +00:00
tkchin@webrtc.org
013bdf802a APPRTCDemo(objc): Remove regex parsing in favor of JSON struct. 
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@6362 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 22:29:10 +00:00
glaznev@webrtc.org
c3288c130d Add OpenGL Android video renderer which can display multiple 
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@6360 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 21:57:46 +00:00
fischman@webrtc.org
9512719569 AppRTCDemo(android): support app (UI) & capture rotation. 
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@6354 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 18:40:44 +00:00
fischman@webrtc.org
130fa64d4c AppRTCDemo(android): remove HTML/regex hackery in favor of JSON struct. 
BUG=3407
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16619006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6345 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 20:31:41 +00:00
tkchin@webrtc.org
738df8913d Fix retain cycle in RTCEAGLVideoView. 
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@6331 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 20:19:39 +00:00
henrike@webrtc.org
09a71cd9ce talk/ios: Fixes source after corrupt sync in r6305 (which corrupted r6291). 
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@6318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 22:46:23 +00:00
buildbot@webrtc.org
34a08b4fb8 (Auto)update libjingle 68275107-> 68379861 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6305 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 15:48:10 +00:00
tkchin@webrtc.org
acca675bcf Implement mac version of AppRTCDemo. 
- 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@6291 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 22:26:06 +00:00
fischman@webrtc.org
abe01dd634 AppRTCDemo(android): run in full-screen & immersive mode. 
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@6275 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 21:46:52 +00:00
fischman@webrtc.org
43a1395370 AppRTCDemo(android): README updates for a shrinking envsetup.sh world. 
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@6265 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 17:29:09 +00:00
buildbot@webrtc.org
727ff69829 (Auto)update libjingle 67872893-> 67873348 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6244 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 23:20:53 +00:00
buildbot@webrtc.org
75cb3dc5f2 (Auto)update libjingle 67869540-> 67872893 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 23:13:35 +00:00
tkchin@webrtc.org
1732a591e7 Add a UIView for rendering a video track. 
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@6192 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 23:26:01 +00:00
fischman@webrtc.org
a150bc9bbf PeerConnection(android): allow initializing either (or neither) of {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@6164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 22:00:50 +00:00
fischman@webrtc.org
14ea7e8922 AppRTCDemo(android): added a Heads-Up Display for bandwidth estimation. 
- 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@6039 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 20:57:55 +00:00
fischman@webrtc.org
dd92feb6dd AppRTCDemo(android): send the created SDP, not the local description after 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@6038 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 19:06:18 +00:00
tkchin@webrtc.org
ff2733204d Implement ObjC DataChannel wrapper 
R=fischman@webrtc.org
BUG=3112

Review URL: https://webrtc-codereview.appspot.com/16369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6031 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 18:32:33 +00:00
fischman@webrtc.org
7c82adae61 AppRTCDemo was blocking the main thread for network requests. This fixes it 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@6028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 00:17:47 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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