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Author SHA1 Message Date
jiayl@webrtc.org
e398954658 Update usrsctp to r8875 
TBR=pthatcher@webrt.org
BUG=2749

Review URL: https://webrtc-codereview.appspot.com/16739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 18:16:08 +00:00
fgalligan@google.com
32196decd6 Update generated asm offsets scripts. 
Libvpx updated the unpack scripts to fix building dependencies.

Roll libvpx 269083:277778
See https://codereview.chromium.org/295313002/
https://codereview.chromium.org/298063002/
https://codereview.chromium.org/305533008/
https://codereview.chromium.org/305703002/
https://codereview.chromium.org/298383003/
https://codereview.chromium.org/302863004/
https://codereview.chromium.org/320923003/
https://codereview.chromium.org/325313007/
for the libvpx changes.

See https://codereview.chromium.org/313243004/
for the WebView changes.

*NOTE* This CL will break the Android bots as they are built in a
Chromium checkout, which will pull in old libvpx DEPS. They will
cycle to green when we roll libvpx into Chromium. We must do the
rolls in this order because we have to land webrtc and libvpx at
the same time into Chromium.

BUG=377062
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6473 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 17:55:23 +00:00
stefan@webrtc.org
a15fbfdcde Add round-robin selection of send stream to pad on. 
BUG=1812
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 17:32:05 +00:00
niklas.enbom@webrtc.org
9c09e6ee2b Add high perf mode to VP8 
R=marpan@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6470 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 16:32:08 +00:00
andrew@webrtc.org
26eaf7c7f7 Add a check to all.gyp to respect the include_tests variable. 
When include_tests==0, tests should be excluded from the build. This
ensures libjingle_tests.gyp is excluded appropriately.

BUG=b/15673188
R=tnakamura@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16729005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6469 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 16:10:20 +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
henrike@webrtc.org
a685c9df62 base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/ 
BUG=N/A
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 14:48:44 +00:00
henrike@webrtc.org
5654b305e5 Rebase webrtc/base with r6464 version of talk/base: 
cd webrtc/base
svn diff -r 6463:6464 http://webrtc.googlecode.com/svn/trunk/talk/base >
6464.diff
patch -p0 -i 6464.diff

BUG=3379
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12749005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6466 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 14:37:05 +00:00
tina.legrand@webrtc.org
d469443959 Rolling new version of opus.gyp 
This roll includes changes that enables FIXED_POINT and -O3 for Opus when building for ARM, for higher speed.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6465 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 13:24:48 +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
minyue@webrtc.org
d42da54768 Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..." 
> Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling.
> 
> TEST=passed_all_trybots
> R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/16619005

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6462 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 09:50:12 +00:00
kjellander@webrtc.org
851a09e71a Initial GN work for WebRTC 
This CL makes it possible to build the 'webrtc_base'
target using GN.
The majority of our GYP stuff in webrtc/build/common.gypi has been
translated into the configs of webrtc/BUILD.gn.
The webrtc/base/base.gyp file is translated into webrtc/base/BUILD.gn.

This CL depends on https://codereview.chromium.org/322373002/ for the
jsoncpp BUILD.gn file and the ssl config.
To build inside Chromium, https://codereview.chromium.org/321313006/
needs to be landed first.

BUG=webrtc:3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true" && ninja -C out/Default
I also ran:
gn gen out/Default --args="build_with_chromium=false have_dbus_glib=true"
but it fails to compile: something is probably wrong with with pkg-config for that.

For Chromium, I symlinked src/third_party/webrtc to the webrtc subfolder of the
WebRTC checkout and applied the following patches:
https://codereview.chromium.org/322373002 (for jsoncpp and ssl config)
https://codereview.chromium.org/321313006 (enable building WebRTC)
Then I built successfully using:
gn gen out/Default && ninja -C out/Default webrtc_base

R=brettw@chromium.org
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6461 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 08:54:03 +00:00
henrik.lundin@webrtc.org
2ca2188906 Restore ptypes.txt file 
The file was lost when the neteq folders where moved and renamed.

BUG=2996
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6460 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 08:51:01 +00:00
phoglund@webrtc.org
6b061425c2 Updated W3C getusermedia tests to the latest version of the spec. 
BUG=webrtc:3455
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 08:46:58 +00:00
minyue@webrtc.org
8f8503d947 Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling. 
TEST=passed_all_trybots
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6458 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 08:02:05 +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
henrike@webrtc.org
9f36c087f1 Makes it possible to prevent some third party libraries (jsoncpp and openssl) from being linked. This makes it possible to link webrtc with external implementations of those libraries in case the project depending on webrtc requires another version of those libraries. 
BUG=3379
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17699005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 21:35:20 +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
asapersson@webrtc.org
d980307197 Add max limit of number for overuses. When limit is reached always apply the rampup delay. 
BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6451 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 14:27:19 +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
asapersson@webrtc.org
4b12d40008 Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class. 
BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6449 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 14:09:28 +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
kjellander@webrtc.org
d6e2213edd Remove ivinnichenko from webrtc/test/OWNERS 
Apparently, We're doing a poor job of cleaning out
really old OWNERS.

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6447 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 11:42:27 +00:00
henrik.lundin@webrtc.org
1e3c5c248a Importing ThreadChecker class from Chromium 
The ThreadChecker class is imported/re-implemented from Chromium.
The implementation is changed to depend on WebRTC primitives.

R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6446 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 11:34:44 +00:00
bjornv@webrtc.org
b099a6f9ab Adds aluebs@webrtc.org as owner to audio_processing 
BUG=N/A
TESTED=trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6445 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 10:30:58 +00:00
bjornv@webrtc.org
721f970cba common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16 
We should avoid macros in general (see style guide) and the shift ones are particular dangerous since they assume that the user apply a non-negative shift.

Related CL: https://webrtc-codereview.appspot.com/16669004

BUG=3348,3353
TESTED=trybots and manually on linux
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6444 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 10:30:14 +00:00
mflodman@webrtc.org
eb16b811fb Implements start bitrate for new video API. 
Added  a new rampup test.

BUG=2879
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6443 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 08:57:39 +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
henrik.lundin@webrtc.org
63e46077a3 Add thread annotations to parts of ACMGenericCodec 
This change adds annotations to all member variables that could be
annotated without acquiring any new locks, or changing the lock
structure in any other way.

BUG=3041
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6441 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 08:02:25 +00:00
asapersson@webrtc.org
249211eec6 Disable flaky test (WebRtcVideoMediaChannelTest.GetStats) on DrMemory Full. 
BUG=3482
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6440 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 07:56:17 +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
kjellander@webrtc.org
2bae3211b1 Add missing sources to webrtc/base/base.gyp 
During my work on setting up a GN build for
WebRTC, I discovered that the base.gyp is trying
to remove source files (for the Chromium build)
that are not added in the initial set.
I assume these files should be listed and that
GYP just doesn't complain when it's trying to
remove a file that is not present in the sources
list.

natserver_main.cc is also removed, since it's not used anywhere.

There are also a couple of other header files that are
used in other code that probably also should be listed in
base.gyp (please do this in another CL):
* compile_assert.h
* dscp.h
* move.h
* template_util.h

BUG=None
TEST=Trybots passing clobber compile step.
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6438 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 07:11:19 +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
bjornv@webrtc.org
af6f02f7bd Neon version of OverdriveAndSuppress() 
audioproc reports the average frame time going from 279us to 255us with the test data used.

the output does not match the c version, but the difference seen is +-1.

Performance gain on Nexus7: 8.8%

BUG=3131
TESTED=trybots and manually
R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19539004

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6433 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 14:50:23 +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
wu@webrtc.org
8e256eec4f Revert 6415 "Update generated asm offsets scripts." 
> Update generated asm offsets scripts.
> 
> This is the same CL as https://webrtc-codereview.appspot.com/16629004/
> Relanding and TBR from previous lgtm.
> 
> Libvpx updated the unpack scripts to fix building dependencies.
> 
> Roll libvpx 269083:275816
> See https://codereview.chromium.org/295313002/
> https://codereview.chromium.org/298063002/
> https://codereview.chromium.org/305533008/
> https://codereview.chromium.org/305703002/
> https://codereview.chromium.org/298383003/
> https://codereview.chromium.org/302863004/
> https://codereview.chromium.org/320923003/
> for the libvpx changes.
> 
> See https://codereview.chromium.org/313243004/
> for the WebView changes.
> 
> *NOTE* This CL will break the Android bots as they are built in a
> Chromium checkout, which will pull in old libvpx DEPS. They will
> cycle to green when we roll libvpx into Chromium. We must do the
> rolls in this order becuase we have to land webrtc and libvpx at
> the same time into Chromium.
> 
> BUG=377062
> TBR=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/17689004

TBR=fgalligan@google.com

Review URL: https://webrtc-codereview.appspot.com/13709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6423 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 23:03:17 +00:00