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129 Commits

Author SHA1 Message Date
wu@webrtc.org
f7d501d48a (Auto)update libjingle 63884381-> 63913264 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 23:48:25 +00:00
wu@webrtc.org
cfe5e9c894 (Auto)update libjingle 63837929-> 63884381 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5800 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 17:03:58 +00:00
henrike@webrtc.org
b0ecc1c6fb (Auto)update libjingle 63777286-> 63837929 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 22:44:28 +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
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
sergeyu@chromium.org
e42b8ab129 Cleanups in libjingle to make it compile with chromium_code=1 
Fixed all warnings that show up when compiling libjingle
in chromium with compiling with chromium_code=1.
chromium_code=1 enables various warnings that are off by
default. Most changes are for unused variables and consts.

R=pthatcher@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5769 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 00:31:35 +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
fischman@webrtc.org
c693a2a624 PeerConnection(iOS): fix case in #import statements. 
We've been skating by on OS/X's default case-insensitive filesystem, but this
is a bit silly.

This change brought to you by:
sed -i '' 's/\+internal\.h/+Internal.h/g' $(git grep -l '+internal.h')

BUG=3088
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5764 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-24 18:56:37 +00:00
henrike@webrtc.org
ffe2620c97 (Auto)update libjingle 63352036-> 63363208 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-19 22:20:10 +00:00
henrike@webrtc.org
e9793ab8b8 (Auto)update libjingle 63111035-> 63293120 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5717 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-18 14:36:23 +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
henrike@webrtc.org
0537634154 (Auto)update libjingle 62713454-> 62865357 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5670 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-10 15:53:12 +00:00
fischman@webrtc.org
a01daf0359 RTCPeerConnectionTest(objc): deflake by ignoring ICECompleted. 
Delivery of the state seems intermittent at best on OS/X so
ignore it until we can make it reliable.

BUG=1414,2993,chromium:348982
TBR=bemasc@chromium.org

Review URL: https://webrtc-codereview.appspot.com/9609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5664 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-08 03:17:55 +00:00
fischman@webrtc.org
13320ea3d3 PeerConnectionTest(objc): expect ICE Completed state post 61460797-p10 
Also a few trivial cleanups:
- No need to use STUN for a loopback test
- Reduce test call duration 10s->2s for faster iteration
- Remove obviously-irrelevant Info.plist entries (copy/pasta from iOS)

BUG=1414,2993
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/9369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-07 22:15:30 +00:00
jiayl@webrtc.org
11aab0edc2 Populate VoiceReceiverInfo::delay_estimate_ms, jitter_buffer_ms, and jitter_buffer_preferred_ms to getStats. 
BUG=2665
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5661 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-07 18:56:26 +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
b90991dade Update libjingle 62472237->62550414 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5640 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-04 19:54:57 +00:00
jiayl@webrtc.org
db41b4dbcd Remove the deprecated GetStats method from PeerConnectionInterface. 
R=fischman@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5634 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-03 21:30:06 +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
mallinath@webrtc.org
d3dc424fe5 Remove posting of ICE messages from WebRTCSession in PeerConnection to signaling thread. 
These callbacks are called from signal thread already. There is no point
in posting messages on the same thread again.

BUG=2922
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5626 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-01 00:05:52 +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
henrike@webrtc.org
704bf9ebec (Auto)update libjingle 62063505-> 62278774 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5617 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-27 17:52:04 +00:00
henrike@webrtc.org
a7b981843f Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702). 
BUG=N/A
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5595 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-21 15:51:43 +00:00
xians@webrtc.org
ef2215110c Revert 5590 "description" 
> description

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8949006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5593 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-21 10:31:29 +00:00
henrike@webrtc.org
2643805a20 description 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5590 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-20 22:32:53 +00:00
henrike@webrtc.org
571df2dca9 Update libjingle 61759961->61834300 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5580 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 23:04:26 +00:00
fischman@webrtc.org
358e3367a3 PeerConnection(java): enable HW encoder on N5 for standalone build. 
Now that bug 2899 is fixed (r5562) packet-loss is recoverable.  Yay.

BUG=2575
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/8869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5568 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 17:29:37 +00:00
fischman@webrtc.org
c2d75e0708 PeerConnection(java): account for thread shutdown vagaries. 
Android's JVM requires threads to detach before they exit, but ONLY if
they needed to AttachCurrentThread.  Conversly, threads that were
attached by the JVM (e.g. the result of making a native call from Java)
must NOT be detached by the application.  This is bug 2441.

The fix for the above is to only pthread_setspecific() for threads that
Attach(), not for already-attached threads.  To ensure that we only
detach Attached threads, added a GetEnv() call to ThreadDestructor(),
which revealed that Oracle's JVM can overly-eagerly clear TLS accounting
data, effectively detaching threads without their consent at shutdown.
Work around this with a specific check.

To guard against (some) regression, added a variant of PeerConnectionTest
that runs on a non-main thread.  This revealed a bug in LinuxDeviceManager
which implicitly assumes its talk_base::Thread has already been
initialized.  Fixed that here too.

BUG=2441
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5567 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 16:57:36 +00:00
fischman@webrtc.org
a3708ecdfe PeerConnectionTest(java): unbreak following 61460797-p10 
BUG=1414
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5550 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-14 01:51:33 +00:00
mallinath@webrtc.org
385857dfd4 Update talk to 61549749. 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5549 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-14 00:56:12 +00:00
wu@webrtc.org
b9a088b920 Update talk to 61538839. 
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/8669005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5548 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-13 23:18:49 +00:00
wu@webrtc.org
0de29504ab Revert 5545 "Update libjingle to 61514460" 
> Update libjingle to 61514460
> 
> TBR=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/8649004

TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5547 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-13 19:54:28 +00:00
xians@webrtc.org
e749c9ebdb Update libjingle to 61514460 
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5545 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-13 15:09:40 +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
sergeyu@chromium.org
9cf037b831 Update libjingle to 61168196 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5502 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 19:03:26 +00:00
fischman@webrtc.org
6e08228525 PeerConnectionTest(java): remove the obsolete magical names of streams & tracks. 
BUG=1253
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7929005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5478 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 19:15:44 +00:00
fischman@webrtc.org
a06ebab1e1 PeerConnectionTest(java): test SCTP DataChannels. 
BUG=1408,2253,2626
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5477 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 19:11:29 +00:00
mallinath@webrtc.org
67ee6b9a62 Update talk to 60923971 
Review URL: https://webrtc-codereview.appspot.com/7909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5475 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 16:57:16 +00:00
jiayl@webrtc.org
808b99b111 Disable a test assert which fails due to usrsctp not cleaned up in SctpDataEngine.cc 
BUG=2749
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7739005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5460 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-29 19:44:40 +00:00
jiayl@webrtc.org
a576faf82a Enable SCTP and use OPENSSL on Anroid and NSS on other platforms. 
It includes unit test fixes to properly initialize SSL if DTLS or SSL random number generator is used in the tests.
The private key and certificate constant strings used in some tests are updated to be compatible with NSS.
A few potentially overflow type conversions caused compiling warning on Windows and they are fixed by importing and using Chromium's checked_cast, which aborts the program if overflow occurs.
It also fixes a leak in nssstreamadapter.cc by releasing the PRFileDesc* in StreamClose.

BUG=2253
R=fischman@webrtc.org, juberti@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4679005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-29 17:45:53 +00:00
mallinath@webrtc.org
0dac5378e5 Revert 5447 "Update talk to 60420316." 
> Update talk to 60420316.
> 
> TBR=wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/7719005

TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5448 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-28 06:58:42 +00:00
mallinath@webrtc.org
752a017809 Update talk to 60420316. 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7719005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5447 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-28 06:45:52 +00:00
wu@webrtc.org
a8910d2f88 Update talk to 60094938. 
Review URL: https://webrtc-codereview.appspot.com/7489005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5420 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 22:12:45 +00:00
mallinath@webrtc.org
0d92ef67c4 Libjingle source code has some spelling mistakes and one of them is "renegotation", which should be "renegotiation". 
This CL is attempting to correct those.

BUG=2810
TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5411 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-22 02:21:22 +00:00
mallinath@webrtc.org
68cbd01216 enabling disabled data channels tests on win32. The real culprit was that ice candidates not included in SDP when there were failure causing transport channels never becoming writable. 
BUG=2799
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-22 00:16:46 +00:00