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641 Commits

Author SHA1 Message Date
buildbot@webrtc.org
4c25c67146 (Auto)update libjingle 69589535-> 69600065 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6504 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-20 04:42:34 +00:00
buildbot@webrtc.org
58e7c8660c (Auto)update libjingle 69588980-> 69589535 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6503 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-20 00:26:50 +00:00
buildbot@webrtc.org
0970dd8767 (Auto)update libjingle 69588608-> 69588980 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6502 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-20 00:18:36 +00:00
buildbot@webrtc.org
8563ef448a (Auto)update libjingle 69587333-> 69588608 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6501 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-20 00:13:01 +00:00
buildbot@webrtc.org
1ef789d455 (Auto)update libjingle 69568113-> 69587333 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6500 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-19 23:54:12 +00:00
buildbot@webrtc.org
df9bbbee56 (Auto)update libjingle 69567902-> 69568113 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6498 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-19 19:54:33 +00:00
buildbot@webrtc.org
fbd13286dc (Auto)update libjingle 69555283-> 69567902 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6497 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-19 19:50:55 +00:00
buildbot@webrtc.org
21794f9862 (Auto)update libjingle 69543894-> 69555283 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6496 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-19 17:14:19 +00:00
buildbot@webrtc.org
d27d9ae644 (Auto)update libjingle 69506154-> 69515138 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6488 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-19 01:56:46 +00:00
jiayl@webrtc.org
acede34aea Fix a memory leak in SctpDataMediaChannelTest. 
BUG=3492
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6486 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-18 23:36:16 +00:00
jiayl@webrtc.org
f8063d34de Properly shut down the SCTP stack. 
TBR phoglund@webrtc.org for the tsan_v2/suppressions.txt change.
R=ldixon@webrtc.org, pthatcher@webrtc.org
TBR=phoglund@webrtc.org
BUG=2749

Review URL: https://webrtc-codereview.appspot.com/12739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6484 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-18 21:30:40 +00:00
jiayl@webrtc.org
2eaac188bb Makes the sid of a closed DataChannel available to reuse per the spec. 
BUG=2646
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6468 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 16:02:46 +00:00
phoglund@webrtc.org
ed3e0d8f0d Increasing tolerances quite a bit to fight flakes. 
From these errors:

[----------] 3 tests from ProfilerTest
[ RUN      ] ProfilerTest.TestFunction
../../talk/base/profiler_unittest.cc:56: Failure
The difference between kWaitSec and event->mean() is 0.13612610600000002, which exceeds kTolerance, where
kWaitSec evaluates to 0.25,
event->mean() evaluates to 0.38612610600000002, and
kTolerance evaluates to 0.10000000000000001.
[  FAILED  ] ProfilerTest.TestFunction (655 ms)
[ RUN      ] ProfilerTest.TestScopedEvents
../../talk/base/profiler_unittest.cc:98: Failure
The difference between kEvent2WaitSec and event2->mean() is 0.33170768900000003, which exceeds kTolerance, where
kEvent2WaitSec evaluates to 0.14999999999999999,
event2->mean() evaluates to 0.48170768899999999, and
kTolerance evaluates to 0.10000000000000001.

I didn't spend time understanding why; I reckon the test had too tight
tolerances to start with so I'm just adjusting them a bit. That's
probably better than disabling the test, now it still has some value.

R=aluebs@webrtc.org
TBR=aluebs@webrtc.org
BUG=None

Review URL: https://webrtc-codereview.appspot.com/13729005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 11:09:00 +00:00
buildbot@webrtc.org
ae740dd94c (Auto)update libjingle 69359922-> 69365993 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6463 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 10:56:41 +00:00
buildbot@webrtc.org
44a317a698 (Auto)update libjingle 69337301-> 69359922 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6457 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 07:49:15 +00:00
buildbot@webrtc.org
53f57936c1 (Auto)update libjingle 69306183-> 69323802 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6454 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 21:08:51 +00:00
pbos@webrtc.org
587ef60056 Implement RTP extension support in WebRtcVideoEngine2. 
BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6453 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 17:32:02 +00:00
buildbot@webrtc.org
d054bff3b9 (Auto)update libjingle 69292418-> 69293749 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6452 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 14:37:41 +00:00
buildbot@webrtc.org
88d9fa63df (Auto)update libjingle 69291002-> 69292418 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 14:11:32 +00:00
buildbot@webrtc.org
27626a6256 (Auto)update libjingle 69278008-> 69291002 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6448 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 13:39:40 +00:00
buildbot@webrtc.org
0a1e7e0b00 (Auto)update libjingle 69276003-> 69278008 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6442 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 08:34:09 +00:00
buildbot@webrtc.org
d159140965 (Auto)update libjingle 69260070-> 69276003 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6439 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 07:49:00 +00:00
buildbot@webrtc.org
117afeec91 (Auto)update libjingle 69188577-> 69260070 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 07:11:01 +00:00
glaznev@webrtc.org
ab23d493e0 Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc. 
Review URL: https://webrtc-codereview.appspot.com/20659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6436 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 23:31:35 +00:00
glaznev@webrtc.org
c6c1dfd7ea Add extra logging and latency restriction to VP8 HW encoder. 
- Do not allow encoder to accumulate more than 100 ms of
data in input buffers.
- Add optional extra logging (disabled by default) to track
encoder buffers timing.

BUG=
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6435 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 22:59:08 +00:00
buildbot@webrtc.org
a6764ab869 (Auto)update libjingle 69144530-> 69164179 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6434 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 18:24:39 +00:00
buildbot@webrtc.org
db56390f7e (Auto)update libjingle 69143161-> 69144530 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 13:05:48 +00:00
pbos@webrtc.org
f99c2f2dbc Add NACK feedback parameter to WebRtcVideoEngine2. 
Also fixing enabling/disabling of NACK. Previous implementation was made
under the assumption that NACK should always be enabled which caused
both missing NACK settings in SDP as well as broken interop between old
and new WebRtcVideoEngines.

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6431 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 12:27:38 +00:00
pbos@webrtc.org
e322a175f6 Implement RTX tests+fixes in WebRtcVideoEngine2. 
BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6430 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 11:47:28 +00:00
pbos@webrtc.org
9fbb717aca Remove engine_codecs_ cache from unittests. 
Used interchangably with engine_.codecs() becomes confusing and it's not
really used that much.

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

Review URL: https://webrtc-codereview.appspot.com/17689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6429 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 09:34:13 +00:00
kjellander@webrtc.org
d54ec1256c Fix GYP DEPTH for libjingle isolate files 
In https://review.webrtc.org/13679004/ the libjingle isolate
files in patch set #2 were not tested, which caused a failure when
6427 was committed. This fixes the talk/build/isolate.gypi with a
similar change.

BUG=343106
TEST=Successful local compile on Linux
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6428 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 09:16:23 +00:00
kjellander@webrtc.org
a1bfc50a72 Pass GYP DEPTH variable to isolate. 
Similar change to https://codereview.chromium.org/322403003/
This will make it possible to handle different
directory levels for special builds of WebRTC, without
breaking GYP when the .isolate files are processed and
their contents is verified.

Also update all our .isolate files to use the <(DEPTH)
variable.

BUG=343106
TEST=Successful compile+test on Linux using:
ninja -C out/Release
tools/swarming_client/isolate.py run -s out/Release/tools_unittests.isolated
Also trybots passing all tests.

R=pbos@webrtc.org
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6427 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 09:02:15 +00:00
buildbot@webrtc.org
c800c1cc40 (Auto)update libjingle 69131548-> 69132244 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6426 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 07:56:17 +00:00
pbos@webrtc.org
1c8223c590 Initial owners file for talk/media/webrtc/. 
Including pthatcher@webrtc.org (already root owner) and
mflodman@webrtc.org.

BUG=
R=juberti@google.com, juberti@webrtc.org
TBR=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15679008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6425 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 07:29:26 +00:00
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
buildbot@webrtc.org
c14f521b1b (Auto)update libjingle 66887616-> 66900106 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6130 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 18:52:57 +00:00
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
buildbot@webrtc.org
eaf2bd916b (Auto)update libjingle 66813165-> 66836233 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6113 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 23:12:19 +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
henrike@webrtc.org
0df2ea064f Rollback of r6108 
BUG=N/A
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6109 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 18:41:12 +00:00
pbos@webrtc.org
a7f70a487f Initialize bitrates in ValidateCodecFormat. 
Attempt to un-break a Visual Studio build (unknown version) that
incorrectly reports that these are potentially uninitialized.

BUG=
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15469005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6108 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 18:20:40 +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
8a54844333 (Auto)update libjingle 66624678-> 66643715 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6095 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 18:10:55 +00:00
buildbot@webrtc.org
1cd14a4502 (Auto)update libjingle 66556498-> 66624678 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6093 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 15:01:40 +00:00
buildbot@webrtc.org
ca27236272 (Auto)update libjingle 66541346-> 66556498 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6088 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 23:10:23 +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
073dfdd10a (Auto)update libjingle 66539128-> 66540208 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 19:36:21 +00:00
buildbot@webrtc.org
d1ae89fae1 (Auto)update libjingle 66524760-> 66539128 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6083 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 19:19:26 +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
jiayl@webrtc.org
f7026cd7c8 Check SCTP_EWOULDBLOCK instead of EWOULDBLOCK in SctpDataMediaChannel. 
usrsctp.h redefines EWOULDBLOCK to WSAEWOULDBLOCK on Windows, but usrsctp_sendv still returns the BSD EWOULDBLOCK (i.e. SCTP_EWOURLBLOCK) when sending data fails due to congestion.
We will need to revert this change when usersctp is fixed.

BUG=2866
R=juberti@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 16:02:23 +00:00
buildbot@webrtc.org
c5bb22395c (Auto)update libjingle 66424806-> 66523513 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6078 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 16:00:58 +00:00
buildbot@webrtc.org
2219037e5e (Auto)update libjingle 66406192-> 66424806 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6075 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 17:52:33 +00:00
buildbot@webrtc.org
dd4742a9ef (Auto)update libjingle 66388864-> 66406192 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6072 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 14:50:35 +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
f9277a9381 (Auto)update libjingle 66326258-> 66340694 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 00:29:05 +00:00
buildbot@webrtc.org
861d4b0de9 (Auto)update libjingle 66322380-> 66326258 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6067 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 22:11:02 +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
a18b4c96af (Auto)update libjingle 66301332-> 66303009 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 17:48:14 +00:00
buildbot@webrtc.org
e65c9a6e67 (Auto)update libjingle 66299810-> 66301332 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 17:28:28 +00:00
buildbot@webrtc.org
0b53bd29af (Auto)update libjingle 66294299-> 66299810 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6062 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 17:12:36 +00:00
buildbot@webrtc.org
150835ea34 (Auto)update libjingle 66236292-> 66294299 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 15:54:38 +00:00
buildbot@webrtc.org
5ee0f05d5f (Auto)update libjingle 66138442-> 66236292 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 20:18:08 +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
buildbot@webrtc.org
cc06c75f28 (Auto)update libjingle 66100938-> 66106643 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6046 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 18:51:11 +00:00
buildbot@webrtc.org
13d6776c46 (Auto)update libjingle 66098243-> 66100938 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 17:33:29 +00:00
buildbot@webrtc.org
0d34f1446a (Auto)update libjingle 66033941-> 66098243 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6044 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 16:54:25 +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
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
jiayl@webrtc.org
53d82350c5 Ignore identical remote fingerprint in DtlsTransportChannelWrapper::SetRemoteFingerprint. 
Trying to set the same remote fingerprint could happen during renegotiation and should not fail.

BUG=crbug/362431
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12449005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6035 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 00:00:19 +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
buildbot@webrtc.org
740e6b339a (Auto)update libjingle 65843899-> 65880186 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6029 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 15:33:45 +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
a86c42c424 libjingle_unittest now compiles and passes on iOS! (reland of r5986) 
Example run from cmd-line:
ninja -C out_ios/Debug-iphoneos libjingle_unittest && \
  ~/src/ios-deploy/ios-deploy -d -u -v -b \
    ~/src/wr/trunk/out_ios/Debug-iphoneos/libjingle_unittest.app

Note that the test's use of signals means that lldb will break in the middle
of the suite.  To ignore these signals tell lldb:

pro hand -p true -s false -n false SIGINT
pro hand -p true -s false -n false SIGTERM
continue

BUG=3241
R=kjellander@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6025 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 18:37:29 +00:00
buildbot@webrtc.org
681f787cc4 (Auto)update libjingle 65752960-> 65813736 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6023 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 17:55:26 +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
buildbot@webrtc.org
af6640fce7 (Auto)update libjingle 65729829-> 65752960 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6004 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 21:31:51 +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
kjellander@webrtc.org
7d825e9b2c Revert "libjingle_unittest now compiles and passes on iOS!" 
This reverts commit r5986 as it fails compilation on Mac
(non-iOS). The failure was not discovered on the commitbots
since they don't clobber their builds.

BUG=3241
TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5997 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 12:50:47 +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
fischman@webrtc.org
95cd1551f8 libjingle_unittest now compiles and passes on iOS! 
Example run from cmd-line:
ninja -C out_ios/Debug-iphoneos libjingle_unittest && ~/src/ios-deploy/ios-deploy -d -u -v -b ~/src/wr/trunk/out_ios/Debug-iphoneos/libjingle_unittest.app
Note that the test's use of signals means that lldb will break in the middle of the suite.  To ignore these signals tell lldb:

pro hand -p true -s false -n false SIGINT
pro hand -p true -s false -n false SIGTERM
continue

BUG=3241
R=noahric@google.com, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5986 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 23:59:56 +00:00
buildbot@webrtc.org
658a94595d (Auto)update libjingle 65619249-> 65622932 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5984 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 22:03:30 +00:00
buildbot@webrtc.org
ff90ed6e96 (Auto)update libjingle 65561104-> 65619249 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5983 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 21:12:10 +00:00
buildbot@webrtc.org
2b93402e36 (Auto)update libjingle 65484212-> 65561104 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5978 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 00:18:27 +00:00
buildbot@webrtc.org
3f1aa24078 (Auto)update libjingle 65469804-> 65484212 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5967 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 00:00:12 +00:00
jiayl@webrtc.org
0d915ff603 Fix the return value of DtlsTransportChannelWrapper::SendPacket in the case of invalid RTP packet. 
R=juberti@webrtc.org, mallinath@webrtc.org

BUG=3244

Review URL: https://webrtc-codereview.appspot.com/12299006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-23 20:46:29 +00:00
buildbot@webrtc.org
504fc89f36 (Auto)update libjingle 65394435-> 65417850 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5961 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-23 03:23:19 +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
mallinath@webrtc.org
54fd70046d Remove ASSERT in TransportChannelProxy::SetImplementation, when 
proxy already set to same transport channel impl.

Since session can call SetImplementation multiple times with or without BUNDLE, there are cases when SetImplementation is called with same impl (OnRemoteCandidates/PushdownTransportDescription/SetupMux). Also variables in
cricket::TransportProxy like |connecting_| and |negotiated_| are accessed
both between worker thread and signaling threads (which calls for bigger change
on how session interacts with Transport and TransportChannelProxy). I have a created a separate bug to address later issue.

Also if single thread used as worker and signaling thread, we can end up
calling SetLocalDescription and OnRemoteCandidates in same call sequence, which
will end up calling SetImplementation twice.

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12019007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5944 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-19 01:03:33 +00:00
buildbot@webrtc.org
8e5ec52e76 (Auto)update libjingle 65152644-> 65219629 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5941 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-19 00:00:31 +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
buildbot@webrtc.org
1a87f529a2 (Auto)update libjingle 65151416-> 65151642 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5934 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-17 22:41:30 +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
henrike@webrtc.org
aeb0c28193 Update PRESUBMIT.py's list of "DO_NOT_SUBMIT_FILES". 
BUG=N/A
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12029006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5931 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-17 14:15:43 +00:00
buildbot@webrtc.org
0b3c6c3191 (Auto)update libjingle 65086785-> 65104022 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5925 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-17 10:03:57 +00:00
buildbot@webrtc.org
39b868bad3 (Auto)update libjingle 65055925-> 65086785 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5921 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-17 00:04:39 +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