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507 Commits

Author SHA1 Message Date
buildbot@webrtc.org
7e71b77f8a (Auto)update libjingle 69102234-> 69116997 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6424 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 01:14:01 +00:00
jiayl@webrtc.org
1a6c6281ca Revert r6420 'Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck' 
Failing tests are disabled for memcheck.

TBR=wu@webrtc.org
BUG=2626

Review URL: https://webrtc-codereview.appspot.com/13699004

Review URL: https://webrtc-codereview.appspot.com/13699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 21:59:29 +00:00
jiayl@webrtc.org
ddeec048c0 Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck 
This reverts commit c3272a942f04f9dd0db3f6bf0d201bcf47c3fa08.

TBR=wu@webrtc.org
BUG=2626

Review URL: https://webrtc-codereview.appspot.com/13689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6420 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 21:42:46 +00:00
buildbot@webrtc.org
3f3f428d2b (Auto)update libjingle 69097619-> 69099564 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6419 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 21:36:26 +00:00
jiayl@webrtc.org
6c6f33b5bb Fix the flaky RTP DataChannel test. 
BUG=2891
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6418 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 21:05:19 +00:00
buildbot@webrtc.org
18dfa8d574 (Auto)update libjingle 69069003-> 69082899 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6417 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 18:11:02 +00:00
xians@webrtc.org
4cb012858f Fixed GetStats when local and remote track are using the same ssrc. 
R=hta@chromium.org, kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6414 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 14:57:05 +00:00
buildbot@webrtc.org
b90619c07f (Auto)update libjingle 69049090-> 69054765 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6412 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 09:19:08 +00:00
buildbot@webrtc.org
d41eaeb7cd (Auto)update libjingle 69005149-> 69049090 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 07:13:26 +00:00
buildbot@webrtc.org
e9e8007ab4 (Auto)update libjingle 68985065-> 69005149 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 18:41:17 +00:00
pbos@webrtc.org
9e65a3b013 Re-land webrtcmediaengine.cc part of r6397. 
webrtcvideoengine.cc un-reverted by a bot roll in r6399 so half of r6397
is still applied. The applied fix (diff between r6397) is to put
WebRtcVideoEngine2 in ifdefs and only build for WEBRTC_CHROMIUM_BUILDs
corresponding to webrtcmediaengine.h.

BUG=
R=minyue@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19719005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 13:42:37 +00:00
buildbot@webrtc.org
5d223a7d2d (Auto)update libjingle 68982444-> 68983526 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6399 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 13:05:05 +00:00
minyue@webrtc.org
6604c6df26 Revert 6397 "(Auto)update libjingle 68949184-> 68982444" 
> (Auto)update libjingle 68949184-> 68982444

TBR=buildbot@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6398 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 13:02:36 +00:00
buildbot@webrtc.org
af214d804f (Auto)update libjingle 68949184-> 68982444 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 12:46:49 +00:00
jiayl@webrtc.org
e61b8e32d8 Adds end to end DataChannel tests. 
BUG=2626
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 23:54:13 +00:00
glaznev@webrtc.org
a40210aee2 Add support for NVidia VP8 HW encoder. 
- Some changes in HW VP8 encoder search logic to detect HW codec
with supported color space format.
- Support yuv420 and nv12 formants for encoder input.
- Add some extra logging and encoder frame drop statistics.

BUG=3176
R=fischman@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6389 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 23:48:29 +00:00
kjellander@webrtc.org
1014101470 Revert 6380 "Replace libjingle_root with talk_root variable." 
It turns out this doesn't fix the problem we're trying to solve...

> Replace libjingle_root with talk_root variable.
> 
> This CL is similar to https://review.webrtc.org/9019004/
> It is needed in order to be able to build with different
> copies of libjingle. Having the libjingle_root variable didn't
> make this possible, since relative paths in the .isolate files
> ended up at the wrong directory level and .isolate files doesn't
> support all the normal GYP variables like <(DEPTH).
> 
> BUG=chromium:343106
> TEST=trybots passing compile step with clobber.
> R=tommi@webrtc.org, wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/15709004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 10:13:38 +00:00
buildbot@webrtc.org
3eb2c2f4c3 (Auto)update libjingle 68891947-> 68893961 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 09:39:06 +00:00
pbos@webrtc.org
86f613d6b8 Move WebRtcVideoEngine2 fakes to unittest header. 
BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6382 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 08:53:05 +00:00
kjellander@webrtc.org
0238682984 Replace libjingle_root with talk_root variable. 
This CL is similar to https://review.webrtc.org/9019004/
It is needed in order to be able to build with different
copies of libjingle. Having the libjingle_root variable didn't
make this possible, since relative paths in the .isolate files
ended up at the wrong directory level and .isolate files doesn't
support all the normal GYP variables like <(DEPTH).

BUG=chromium:343106
TEST=trybots passing compile step with clobber.
R=tommi@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6380 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 05:46:31 +00:00
kjellander@webrtc.org
6b6e58d632 Remove unused test_env.py from isolate files + fix nss path. 
This is not necessary for executing tests for WebRTC.
It probably appeared in our .isolate files because of code
copied from Chromium.

BUG=
TEST=All non-baremetal trybots passing.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 14:35:09 +00:00
stefan@webrtc.org
85d2794e5b Adds support for the "apt" format parameter and turns on the RTX feature. 
BUG=1811,1095
R=henrike@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12579009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 12:51:39 +00:00
jiayl@webrtc.org
e3cdd9959e Revert "Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio." 
This reverts commit 56631a14bdae24aa0bfaceeb2b57df729fee1227.

TBR=henrike@webrtc.org
BUG=3235

Review URL: https://webrtc-codereview.appspot.com/19669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 22:32:57 +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
jiayl@webrtc.org
745a39cced Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio. 
BUG=3235
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6356 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 19:24:02 +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
buildbot@webrtc.org
91c910469f (Auto)update libjingle 68701339-> 68703656 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6352 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 16:29:00 +00:00
pbos@webrtc.org
910473b31a Fix C++11 -Wnarrowing in channel_unittest.cc. 
Implicit conversion from int to unsigned char inside {} initializers is
ill-formed C++11 and triggers a warning in clang when building it as
such.

BUG=
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6351 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 15:44:00 +00:00
buildbot@webrtc.org
7b6cbb3aa0 (Auto)update libjingle 68689052-> 68689059 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6350 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 10:54:08 +00:00
pbos@webrtc.org
6ae48c6609 Make VideoSendStream/VideoReceiveStream configs const. 
Benefits of this is that the send config previously had unclear locking
requirements, a lock was used to lock parts parts of it while
reconfiguring the VideoEncoder. Primary work was splitting out video
streams from config as well as encoder_settings as these change on
ReconfigureVideoEncoder. Now threading requirements for both member
configs are clear (as they are read-only), and encoder_settings doesn't
stay in the config as a stale pointer.

CreateVideoSendStream now takes video streams separately as well as the
encoder_settings pointer, analogous to ReconfigureVideoEncoder.

This change required changing so that pacing is silently enabled when
using suspend_below_min_bitrate rather than silently setting it.

R=henrik.lundin@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
BUG=3260

Review URL: https://webrtc-codereview.appspot.com/20409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 10:49:19 +00:00
buildbot@webrtc.org
4b83a471de (Auto)update libjingle 68646004-> 68648993 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 21:11:28 +00:00
wu@webrtc.org
94454b71ad Fix the chain that propagates the audio frame's rtp and ntp timestamp including: 
* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.

Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.

Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.

BUG=3111
R=henrik.lundin@webrtc.org, turaj@webrtc.org, xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc

Review URL: https://webrtc-codereview.appspot.com/14559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 20:34:08 +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
pbos@webrtc.org
0d523eea83 Remove static initializer from WebRtcVideoEngine2. 
BUG=
R=pliard@google.com, pthatcher@webrtc.org, pliard@chromium.org

Review URL: https://webrtc-codereview.appspot.com/15679005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6338 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 09:10:55 +00:00
buildbot@webrtc.org
f1adbeedb4 (Auto)update libjingle 68562943-> 68571194 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6333 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 21:57:16 +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
buildbot@webrtc.org
6f237769b3 (Auto)update libjingle 68507189-> 68543735 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6329 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 16:23:10 +00:00
buildbot@webrtc.org
40b45fc07a (Auto)update libjingle 68506654-> 68507189 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 14:48:33 +00:00
buildbot@webrtc.org
0cdcd23a03 (Auto)update libjingle 68501302-> 68506654 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6321 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 01:31:14 +00:00
buildbot@webrtc.org
af81b9bffd (Auto)update libjingle 68499439-> 68501302 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6320 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 00:08:54 +00:00
buildbot@webrtc.org
251fdf64cb (Auto)update libjingle 68495561-> 68499439 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6319 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 23:43:48 +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
53217848b2 (Auto)update libjingle 68465410-> 68487517 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 21:09:11 +00:00
fischman@webrtc.org
83eb7dff5c PeerConnection(java): disable wait for flaky ICEConnection.COMPLETED. 
This should be reverted when COMPLETED is delivered reliably.

BUG=3021
TESTED=without this patch the test fails in Debug mode after a handful of runs.  With this patch 100 runs passed in a row on my desktop.
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6315 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 16:38:08 +00:00
pbos@webrtc.org
289a35c56d Add empty webrtcmediaengine.cc. 
Should contain CreateWebRtcMediaEngine as soon as
libjingle/libjingle.gyp in Chromium builds this file. This file is added
ahead of time to get a smoother rolling process.

BUG=1788
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6313 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 14:51:34 +00:00
buildbot@webrtc.org
b525a9d790 (Auto)update libjingle 68379861-> 68445177 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6309 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 09:42:15 +00:00
pbos@webrtc.org
044bdacfef Remove kMaxWaitForStatsMs from tsanv2 compilation. 
As some tests are #ifdef'd out on THREAD_SANITIZER this constant
triggers an unused-const-variable warning which breaks the build.

BUG=1205,3220
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 09:40:01 +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
pbos@webrtc.org
174a67439b Enable -Wall, -Wextra and -Wunused-variable for talk/ on clang. 
Also removes one case of unused-variable.

BUG=3220
R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6297 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 07:58:30 +00:00
jiayl@webrtc.org
8a09af3f67 Fix the build error from OpenSSLStreamAdapter::SSLVerifyCallback 
TBR=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/17639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 23:24:08 +00:00
jiayl@webrtc.org
0163674f99 Make OpenSSLStreamAdapter verify the leaf certificate digest for chained certificates. 
It used to compre a parent certificate's digest against the SDP fingerprint and caused connection failure.

BUG=3383
R=bemasc@webrtc.org, juberti@webrtc.org, rsleevi@chromium.org

Review URL: https://webrtc-codereview.appspot.com/17589005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 23:14:08 +00:00
tkchin@webrtc.org
56d114627b Fix AppRTC target configuration in libjingle_examples.gyp. 
libjingle_peerconnection_objc doesn't exist as a target in 32bit, so AppRTCDemo
needs that guard as well.

R=andrew@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/18489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6292 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 23:04:39 +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
jiayl@webrtc.org
9f8164c060 Fix two bugs in DataChannel state transition. 
1. OnStateChange should not be fired if state is not changed.
2. RemotePeerRequestClose should be a no-op if it's already closed.

TBR=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/21559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 21:53:17 +00:00
buildbot@webrtc.org
1678db9df6 (Auto)update libjingle 68230113-> 68244456 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6287 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 14:02:09 +00:00
buildbot@webrtc.org
540a2251aa (Auto)update libjingle 68230011-> 68230113 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:40:35 +00:00
pbos@webrtc.org
35efb839ed Implement new-API test RecvStreamWithoutRtx. 
R=pthatcher@google.com, pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/20449005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:40:04 +00:00
pbos@webrtc.org
c34bb3a886 Log default receive stream creation. 
Log when receiving a packet that doesn't have a receiver, this way you
can tell from logs where the AddRecvStream call came from.

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

Review URL: https://webrtc-codereview.appspot.com/17459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6279 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:38:43 +00:00
pbos@webrtc.org
198647473b Implement and fix new-API NackIsEnabled test. 
Required enabling NACK on receiver side which was apparently missed.

BUG=1788
R=pthatcher@google.com, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16499007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6278 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:35:47 +00:00
buildbot@webrtc.org
1d66be22c8 (Auto)update libjingle 68203780-> 68206793 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 22:54:24 +00:00
jiayl@webrtc.org
8dcd43c4f7 Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF. 
This is the first step toward switching completely to UDP/TLS/RTP/SAVPF.

BUG=2796
R=juberti@webrtc.org, pthatcher@google.com

Review URL: https://webrtc-codereview.appspot.com/13439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6276 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 22:07:59 +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
jiayl@webrtc.org
5dc51fbe50 Closes the DataChannel when the send buffer is full or on transport errors. 
As stated in the spec.

BUG=2645
R=pthatcher@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6270 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 15:33:54 +00:00
jiayl@webrtc.org
001fd2d503 Fire OnRenegotiationNeeded only for the first SCTP DataChannel. 
Subsequent DataChannels do not need renegotiation since SCTP data streams are not negotiated through SDP.

BUG=2431
R=pthatcher@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6268 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 15:31:11 +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
jiayl@webrtc.org
b364016cbb Revert r6161 "Drop the DataChannel message if it's received when the channel is not open." 
The spec does not say the DataChannel has to be open to receive a message.

TBR=pthatcher@google.com
BUG=crbug/363005

Review URL: https://webrtc-codereview.appspot.com/16569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6264 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 16:37:25 +00:00
phoglund@webrtc.org
f666ecc60d Disabling flaky libjingle tests after fixit week. 
BUG=webrtc:3316,webrtc:3317,webrtc:3318
TBR=fischman@google.com

Review URL: https://webrtc-codereview.appspot.com/12569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-27 08:08:00 +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
mallinath@webrtc.org
b445f26f24 Fixing correct UMA metric for PeerConnection enabled with IPv4 Vs IPv6. 
BUG=N/A
TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21499007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6242 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 22:19:37 +00:00
fischman@webrtc.org
39eccefbde Disable ChannelManagerTest.StartupShutdownOnUnstartedThread 
The test is testing a scenario that shouldn't happen.

BUG=3388
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6238 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 17:50:38 +00:00
buildbot@webrtc.org
7aa1a4767f (Auto)update libjingle 67848628-> 67848776 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6237 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 17:33:05 +00:00
fischman@webrtc.org
e5063b1733 Thread: delete racy API (Release()) and fix racy code (started()). 
- Thread::Release() wrote a local variable on the calling thread but read it on
  another thread, with no synchronization.  Happily it has no non-test callers
  so deleting it instead of trying to fix it (see bug for details).
- Thread::started_ similarly was racily being written to; replaced with a
  running_ Event, and hid the accessor except for tests & legacy callers,
  with a note about why it's a bad idea.

webrtc/base patched with:
git diff origin --relative=talk/base | patch -p1 -dwebrtc/base
followed by manual merge of 3 thunks that ran afoul of naming differences
between talk/base and webrtc/base.

BUG=3388
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14589005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6236 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 17:28:50 +00:00
fischman@webrtc.org
18f41b8eb4 PRESUBMIT.py: accept variants on the copyright message that are present in the codebase. 
Example files that this makes ok instead of flagging include:
  talk/base/signalthread_unittest.cc
  talk/base/thread_unittest.cc
  webrtc/base/signalthread_unittest.cc
  webrtc/base/thread.cc
  webrtc/base/thread.h
  webrtc/base/thread_unittest.cc

BUG=1027
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19539006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6235 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 17:27:18 +00:00
pbos@webrtc.org
706152dcc9 Fix uninitialized reads in IsDefaultBrowserFirefox 
BUG=
TEST=Local DrMemory.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19529006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6232 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 14:35:48 +00:00
mallinath@webrtc.org
8e755c1ad2 Connect SignalDestroyed in AllocationSequence after TURN ports are destroyed 
when TURN ports are using shared socket with UDP port.

This is required as AllocationSequence maintains a map of turn ports. If the
ports are destroyed without the knowledge of AllocationSequence, sequence will
try to deliver packets to the destoyed ports.

R=jiayl@webrtc.org
BUG=https://code.google.com/p/chromium/issues/detail?id=368877

Review URL: https://webrtc-codereview.appspot.com/14569007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6219 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 23:00:46 +00:00
buildbot@webrtc.org
f9f1bfbdae (Auto)update libjingle 67686255-> 67689476 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6216 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 17:02:15 +00:00
buildbot@webrtc.org
ce4201df52 (Auto)update libjingle 67643194-> 67686255 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6214 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:22:51 +00:00
henrike@webrtc.org
000658a138 Revert of 6211 as it was committed despite of PRESUBMIT.py warning. The commit breaks the sync bot. 
BUG=N/A
TBR=mcasas@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21519006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6212 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:01:13 +00:00
mcasas@webrtc.org
3b7e282caa Disabling systematically failing 
WebRtcVideoMediaChannelTest.SendVp8HdAndReceiveAdaptedVp8Vga

TBR= pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14569006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 14:25:20 +00:00
buildbot@webrtc.org
49a6a27bf0 (Auto)update libjingle 67555838-> 67643194 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6206 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 00:24:54 +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
40bc7779aa talk_base: remove lock inversion between MessageQueue and MessageQueueManager. 
Removes the concept of a MessageQueue being "active" in favor of considering all
live MQ's to be active.
(previously a MQ was active starting from the first Post to it and stopped being
active in its dtor).

BUG=3230
R=sriniv@google.com

Review URL: https://webrtc-codereview.appspot.com/21489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6190 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 17:58:04 +00:00
wu@webrtc.org
cb711f77d2 Add interface to propagate audio capture timestamp to the renderer. 
BUG=3111
R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 17:39:11 +00:00
pbos@webrtc.org
1e019d10b8 Fix delivery error-checking missed in r6151. 
Gets rid of quite a bit of false-warning logging in WebRtcVideoEngine2.

BUG=3228
R=perkj@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6183 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 11:38:45 +00:00
buildbot@webrtc.org
6bfd6196ff (Auto)update libjingle 67052073-> 67134648 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6174 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 16:15:59 +00:00
mallinath@webrtc.org
bb6201ae4b TCP remote socket address should have both server hostname and IP address. 
Hostname is necessary when we are creating TLS based socket, for certificate
verification.

BUG=https://code.google.com/p/chromium/issues/detail?id=306285
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6165 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 22:43:05 +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
buildbot@webrtc.org
ef5a752c29 (Auto)update libjingle 67043374-> 67044055 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6163 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 21:35:19 +00:00
buildbot@webrtc.org
3e924683d4 (Auto)update libjingle 67037200-> 67043374 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 21:29:04 +00:00
jiayl@webrtc.org
4f5801494d Drop the DataChannel message if it's received when the channel is not open. 
It may happen when the JS has closed the channel on the signaling thread while messages are received on the worker thread and posted before the state change is pushed to the worker thread.

BUG=crbug/363005
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19469005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6161 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 20:32:35 +00:00
buildbot@webrtc.org
372701a872 (Auto)update libjingle 67023528-> 67036361 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6160 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 20:27:59 +00:00
buildbot@webrtc.org
688ed699e0 (Auto)update libjingle 67017551-> 67023528 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6158 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 18:26:09 +00:00
fischman@webrtc.org
2c98af7935 PeerConnection(Java): auto-WrapCurrentThread() when creating PeerConnectionFactory. 
Various pieces of talk/ assume that the current Thread is ThreadManager'd
without checking this, so unconditionally wrap the caller's thread in case it
was created by Java code unbeknownst to ThreadManager.

BUG=2947
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6154 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 17:33:32 +00:00
pbos@webrtc.org
4e545cc244 Update webrtcvideoengine2.cc to use DeliveryStatus. 
talk/ changes corresponding to https://review.webrtc.org/12289005/.

BUG=3228
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 13:58:13 +00:00
andresp@webrtc.org
581e2172af Fix libjingle to provide a field_trial implementation. 
This completes https://webrtc-codereview.appspot.com/14489004/ by updating libjingle rules.

BUG=crbug/367114
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6149 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 13:12:45 +00:00
buildbot@webrtc.org
cd846dd374 (Auto)update libjingle 66924241-> 66927231 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6134 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 22:58:27 +00:00
buildbot@webrtc.org
da510c5de6 (Auto)update libjingle 66923202-> 66924241 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6132 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 22:30:56 +00:00
fischman@webrtc.org
d8af5b51c0 Deallocate the result of mach_host_self() when done with it, fixing a 
port leak.

The port rights obtained by mach_host_self() and mach_thread_self() need
to be deallocated with mach_port_deallocate(). They consume finite
system-wide resources. This is in contrast to mach_task_self(), which is
a macro that wraps an extern global variable, and must not be
deallocated.

http://crbug.com/105513 shows the sorts of problems that can occur when
these aren't properly deallocated.

R=fischman@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15469004

Patch from Mark Mentovai <mark@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6131 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 22:18:48 +00:00