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Author SHA1 Message Date
buildbot@webrtc.org
3e01e0b16c (Auto)update libjingle 66867790-> 66887616 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6128 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 17:54:10 +00:00
pbos@webrtc.org
b5a22b1464 Revert r6110 and r6109. 
Effectively re-landing r6104 as well as r6108 which includes a fix to a
compile error that caused r6104 to be reverted in r6110.

BUG=
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6119 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 11:07:01 +00:00
mallinath@webrtc.org
d37bcfa882 Changed enums to less generic names. 
IPv4/IPv6 will be sent when RegisterUMAObserver is called. This is done
as Initialize is not called through interface.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14469006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6112 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 23:10:18 +00:00
buildbot@webrtc.org
17911dca80 (Auto)update libjingle 66798415-> 66813165 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6110 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 18:42:49 +00:00
pbos@webrtc.org
d266a2020f Initial wiring of new webrtc API in libjingle. 
BUG=1788
R=pthatcher@google.com, pthatcher@webrtc.org
TBR=juberti@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8549005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6104 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 14:32:01 +00:00
mallinath@webrtc.org
0f2a22b3fa Removed sending metrics from PeerConnection about IPv4 and IPv6. 
Reasons: 1: There is memcheck failure.
         2: DoInitialize is called before RegisterUMAObserver,
            which means this will be never triggered in real cases.

BUG=3326
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6097 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 21:15:06 +00:00
buildbot@webrtc.org
1567b8cf8c (Auto)update libjingle 66540208-> 66541346 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 19:54:16 +00:00
buildbot@webrtc.org
ff6a3d920a (Auto)update libjingle 66523887-> 66524760 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6080 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 16:16:41 +00:00
buildbot@webrtc.org
ed97bb0eb4 (Auto)update libjingle 66340694-> 66388864 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6071 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 11:15:20 +00:00
buildbot@webrtc.org
0581f0ba0a (Auto)update libjingle 66303009-> 66322380 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6065 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 21:36:31 +00:00
buildbot@webrtc.org
41451d4e55 (Auto)update libjingle 66106643-> 66138442 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-03 05:39:45 +00:00
jiayl@webrtc.org
9c16c39e61 Sets the SCTP port codec in the native SessionDescription. 
Previously it's only set when a SDP string is parsed into SessionDescription, causing failuring for native client.

BUG=3141
R=juberti@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 18:30:30 +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
f04a6ea733 MediaCodecVideoEncoder: limit MediaCodec bitrate to 95% of requested to avoid overshoot. 
BUG=3194
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/17379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6021 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 17:53:30 +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
29540b1879 Revert "PeerConnectionFactory: delay deletion of owned threads." 
This reverts r5933 because it broke
http://build.chromium.org/p/client.webrtc/builders/Win64%20Release/builds/1598

BUG=3100

Review URL: https://webrtc-codereview.appspot.com/12159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5935 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-17 22:54:30 +00:00
fischman@webrtc.org
cea024d672 PeerConnectionFactory: delay deletion of owned threads. 
Since PeerConnection holds a ref to its creating PeerConnectionFactory, it's
possible for ~PeerConnectionFactory() to be run on its signaling thread.
Deleting a thread from within that thread is sad times, so don't do it.

It would be nicer to avoid having PeerConnection hold a ref to the factory,
and instead require the user to keep the factory alive.  Unfortunately that
changes the contract on PeerConnection{,Factory} and it's unclear how to vet
existing callers for safety.

BUG=3100
R=juberti@webrtc.org, noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/11289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5933 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-17 22:36:00 +00:00
jiayl@webrtc.org
8f88f20af2 Expand the test max wait time from 1000ms to 2000ms. 
The createOffer/createAnswer methods sometimes times out due to slow identity generation under memcheck.

BUG=2838
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5920 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-16 17:14:21 +00:00
wu@webrtc.org
36eda7cf0e Workaround for https://bugzilla.mozilla.org/show_bug.cgi?id=996329, where the m line from firefox have a space at the end. 
For example:
"m=application 38233 DTLS/SCTP 5000 "

BUG=3212
TEST=manually try to use DataChannel between FF 28 and Chrome with rtccopy.com
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12029005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5915 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 20:37:30 +00:00
buildbot@webrtc.org
f875f15afb (Auto)update libjingle 64709629-> 64813990 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5897 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 16:06:21 +00:00
buildbot@webrtc.org
61c1b8ea32 (Auto)update libjingle 64585415-> 64594651 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5870 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-09 06:06:38 +00:00
wu@webrtc.org
4e393070be Compare the answer's media type against offer to make sure they are match. Otherwise we should return failure. 
BUG=2687
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11079005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5858 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-07 17:04:35 +00:00
wu@webrtc.org
148149138d (Auto)update libjingle 64147530-> 64247466 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 23:25:15 +00:00
wu@webrtc.org
5e760e7b94 Check the return value of the FromString call and return failure when then value is invalid. I.e. uses 
bool FromString(const std::string& s, T* t)
instead of
T FromString(const std::string& str)

Before this change we will silently continue the parsing and take whatever default value returned by FromString.

TEST=new tests
BUG=2507
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5834 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 23:19:09 +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
4f2bd68744 Silence pointless LS_WARNING about port 0 for active-only candidates. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5808 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-28 18:13:34 +00:00
wu@webrtc.org
987f2c9aae (Auto)update libjingle 63913264-> 63948945 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-28 16:22:19 +00:00
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