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35 Commits

Author SHA1 Message Date
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
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
mallinath@webrtc.org
ecd622eec3 Updating libjingle.gyp after addition new files yuvframescapturer.cc. 
TBR=pbos@webrc.org

Review URL: https://webrtc-codereview.appspot.com/7919006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5476 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 17:17:05 +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
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
henrika@webrtc.org
aebb1ade9d pRevert 5371 "Revert 5367 "Update talk to 59410372."" 
> Revert 5367 "Update talk to 59410372."
> 
> > Update talk to 59410372.
> > 
> > R=jiayl@webrtc.org, wu@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/6929004
> 
> TBR=mallinath@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/6999004

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5381 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 10:00:58 +00:00
henrika@webrtc.org
44461fa5cb Revert 5367 "Update talk to 59410372." 
> Update talk to 59410372.
> 
> R=jiayl@webrtc.org, wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/6929004

TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5371 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 09:35:02 +00:00
mallinath@webrtc.org
0f3356e20b Update talk to 59410372. 
R=jiayl@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-11 01:26:23 +00:00
fischman@webrtc.org
000dde99c8 Android build: make it quiet on success and not overly noisy on failure. 
- OpenSLDemo and WebRTCDemo get the sauce that AppRTCDemo got in r5271
- libjingle_peerconnection_jar is now silent on success
- Fix a bug introduced by r5271 which caused ant logs to be emitted to a subdir of talk/examples instead of in the gyp output directory.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6199005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-20 22:49:35 +00:00
henrike@webrtc.org
9ee75e9c77 Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency). 
BUG=N/A
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5270 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-11 21:42:44 +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
wu@webrtc.org
1d1ffc9ad2 Update talk to 54898858. 
TEST=try bots
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/2414004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4979 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 18:12:02 +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
76fe9309b9 Use link_settings instead of all_dependent_settings to pacify xcode gyp generator 
Should unbreak e.g. http://chromegw/i/chromium.webrtc.fyi/builders/Mac%20%5Blatest%20WebRTC%20trunk%5D/builds/2396

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2261004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4796 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 21:11:08 +00:00
fischman@webrtc.org
ccddd0a941 Roll webrtc's chromium_revision 217707:224141 
Also adds -lm for executables depending on isac since the newer clang in the
newer chromium revision requires it, and -lstdc++ for dependencies of the objc lib because newer gyp links with gcc instead of g++ for non-C++-containing libs.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2177007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 20:27:32 +00:00
wu@webrtc.org
967bfff54d Update talk to 52534915. 
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2251004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4786 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 05:49:50 +00:00
henrike@webrtc.org
82f014aa0b OpenSL (not default): Enables low latency audio on Android. 
BUG=1669
R=andrew@webrtc.org, fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2032004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 18:24:07 +00:00
fischman@webrtc.org
e3de6b1e90 Enable ObjC build by default and reenable 64-bit mac libjingle build 
BUG=2124
TESTED=trybots & building for mac, mac64, ios-sim, and ios-device on my MBP all build everything in out/Debug.
R=niklas.enbom@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2080004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4620 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 19:31:21 +00:00
sergeyu@chromium.org
492e315400 Update gyp file after libjingle roll. 
TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2103004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4609 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-24 00:06:43 +00:00
wu@webrtc.org
91053e7c5a Update libjingle to 50654631. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2000006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-10 07:18:04 +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
wu@webrtc.org
9dba525627 * Update libjingle to 50389769. 
* Together with "Add texture support for i420 video frame." from
wuchengli@chromium.org.
https://webrtc-codereview.appspot.com/1413004

RISK=P1
TESTED=try bots
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1967004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 20:36:57 +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
wu@webrtc.org
d64719d895 Update libjingle to 50191337. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1885005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4461 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 00:00:07 +00:00
wu@webrtc.org
7fdbb1c832 We don't need to link with libssl.so when we already depend on openssl. 
This fixes the hidden-symbol linker warnings.

BUG=2149
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1927004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 22:41:36 +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
9de257d00f Update talk folder to revision=49470012. Same as 375 in libjingle's google code repository. 
TBR=wu@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1824004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4364 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-17 14:42:53 +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