generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
6,030 Commits 1 Branch 0 Tags 108 MiB
e6f84ae8a6
Commit Graph

6030 Commits

This Branch
This Branch
All Branches
Author SHA1 Message Date
andresp@webrtc.org
d11bec40b2 Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6626 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 14:32:58 +00:00
stefan@webrtc.org
3d7da88e06 Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable. 
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6625 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 13:59:46 +00:00
tommi@webrtc.org
ecb8723402 Change Timing::WallTimeNow to be static. 
There's no need to construct a Timing object to call this method.
On Windows we were unnecessarily calling CreateWaitableTimer + CloseHandle but never actually using that waitable timer.

There's otherwise no change in functionality.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6624 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 12:48:29 +00:00
pbos@webrtc.org
62bafae661 Some refactoring inside rtp_rtcp/. 
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 12:10:51 +00:00
phoglund@webrtc.org
241a9b0b65 Fixing compile error. 
Made a mistake in https://webrtc-codereview.appspot.com/13849004/,
fixing that here.

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6622 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 11:48:37 +00:00
phoglund@webrtc.org
22292df53b Adding explicit check for using dummy file devices. 
Calling into the file device factory without being compiled with file
devices makes no sense and would cause hard-to-debug errors. Therefore
I'm adding an explicit check so this isn't allowed.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6621 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 11:39:19 +00:00
andresp@webrtc.org
33d110d8ea Tight data race suppressions around thread_posix. 
BUG=3372,3549
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6620 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 10:36:39 +00:00
pbos@webrtc.org
af38f4e511 Extract RTP-header SSRC inline in Call. 
Prevents unknown-RTP-header-extension warnings to be flooding from the
RTP-header parsing as there's no way to register RTP extensions for the
parser in Call as they're allowed to differ between RTP streams.

RTP-header parsing should instead be done separately in every
VideoReceiveStream.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6619 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 07:38:12 +00:00
mallinath@webrtc.org
a70be68f65 Disabling shared socket mode for TURN ports. This is done as currently when 
TURN server also used as STUN server, binding responses will be handed over
to TURN port, which simply discard these messages, as requests are originated
from StunPort.

Until we find the right solution for this problem, it's better we disable this
feature.

BUG=https://code.google.com/p/webrtc/issues/detail?id=3537
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6618 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 20:47:24 +00:00
andresp@webrtc.org
3c637cdaa5 Clean data races from system_wrappers_unittests. 
- Remove unittest_utilities that are not used.
 - Remove SetLevelFilter that does not seems necessary and anyhow was racy.

BUG=3549
R=henrike@webrtc.org, henrike

Review URL: https://webrtc-codereview.appspot.com/16819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6617 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 20:37:39 +00:00
andresp@webrtc.org
285e9bc84d Fix potential deadlock in webrtc/system_wrappers/source/logging_unittest.cc. 
crit_ should not be held while calling Trace.

BUG=3003
R=henrike@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17859005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6616 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 20:27:33 +00:00
henrike@webrtc.org
5f2c81c17f webrtc/base: Fixes miss in base.gyp for windows. See https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle.gyp?r=6503#764 for the corresponding condition. 
BUG=3379
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6615 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 17:42:45 +00:00
henrike@webrtc.org
ba93f9a986 drmemory flaky: EndToEndTest.RestartingSendStreamPreservesRtpState[WithRtx] suppressed on drMemory. 
BUG=3552
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6614 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 16:52:19 +00:00
pbos@webrtc.org
161f808500 Add test for VideoEncoder setup/teardown. 
Verifies that InitEncode and RegisterEncodeCompleteCallback gets
called before Encode is called. Also verifies that teardown is correctly
done during DestroyVideoSendStream().

BUG=2339
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6613 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 14:22:35 +00:00
pbos@webrtc.org
2bb1bdab8d Preserve RTP states for restarted VideoSendStreams. 
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.

Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.

BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 13:06:48 +00:00
stefan@webrtc.org
73823cafa4 Add initial gn build files for video_coding and video_processing. 
BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 11:46:43 +00:00
pbos@webrtc.org
03c817e405 Fix pacer to accept duplicate sequence numbers on different SSRCs. 
BUG=3550
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6610 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 10:20:35 +00:00
andresp@webrtc.org
b941fe8098 Fix data races related with traces in bitrate estimator test. 
BUG=3549
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6609 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 08:50:48 +00:00
pbos@webrtc.org
bd249bc711 Remove GetDefaultConfigs() from Call. 
Defaults for configs are instead placed in the Config constructors.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6608 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 04:45:15 +00:00
stefan@webrtc.org
7832648824 Add missing break introduced in r6603. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6607 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 17:04:00 +00:00
stefan@webrtc.org
bee164a214 Fix test issues and a win compile error introduced with r6605. 
Also changes the name of a variable which has been hijacked by windef.h (included by windows.h), which forces #define near and #define far upon us. This issue was introduced via the following inclusion chain:
bwe_test_framework_unittest.cc includes
  paced_sender.h
    tick_util.h
      windows.h
        windef.h

And causes EXPECT_NEAR(foo, bar, near); to expand to EXPECT_NEAR(foo, bar,); generating a very confusing compile error.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6606 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 17:00:06 +00:00
stefan@webrtc.org
875ad49dee Revert conversion from TickTime to int64_t in paced sender. 
Introduced with r6600, causing flakes in SuspendBelowMinBitrate. The reason for this flake is currently unknown.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6605 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 15:27:55 +00:00
pbos@webrtc.org
8faa5db992 Add pbos@webrtc.org as owner for webrtc/test/. 
BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 13:49:17 +00:00
stefan@webrtc.org
b9f5453e29 Add boilerplate code for H.264. 
R=mflodman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17849005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 12:42:07 +00:00
tina.legrand@webrtc.org
d8440f7c45 Have Opus follow Chromium revisions 
Before this change, a pinned version of Opus was used in WebRTC. This could lead to WebRTC running a different version of Opus compared to the version used with the corresponding Chromium revision.
This CL pulls in the Opus version the Chromium uses.

BUG=3546
R=kjellander@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15979005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6602 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 12:13:08 +00:00
pbos@webrtc.org
20c1f56992 Configure RTX send status on new modules. 
Fixes bug where newly-allocated modules wouldn't send payload-based
padding (or probably not send over RTX at all).

As the newly-added test exposed lock-inversions shown on tsan in
VideoReceiver, VideoReceiver was thread-annotated and locks taken less.
BUG=chromium:391085
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6601 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 10:58:12 +00:00
stefan@webrtc.org
88e0dda475 Introduces PacedVideoSender to test framework and moves the Pacer to use Clock. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 09:20:42 +00:00
mflodman@webrtc.org
614000d638 Adding pbos as video/ owner and removing persons never working with this folder. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6599 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 08:50:14 +00:00
kjellander@webrtc.org
c5e53dde71 Revert 6597 "Roll chromium_revision 280876:281094" 
Breaks GN on Linux with errors like this:
[133/485 | 4.471] CC obj/third_party/libjpeg_turbo/simd/simd.jsimd_x86_64.o
FAILED: g++ -MMD -MF obj/out/Debug/gen/library_loaders/libspeechd.libspeechd.o.d -DCHROMIUM_BUILD -DENABLE_ONE_CLICK_SIGNIN -DENABLE_NOTIFICATIONS -DENABLE_EGLIMAGE=1 -DENABLE_BACKGROUND=1 -DUSE_MOJO=1 -DV8_DEPRECATION_WARNINGS -DBLINK_SCALE_FILTERS_AT_RECORD_TIME -DCLD_VERSION=2 -DENABLE_MDNS=1 -DENABLE_PEPPER_CDMS -DENABLE_PLUGINS=1 -DENABLE_PRINTING=1 -DENABLE_FULL_PRINTING=1 -DENABLE_SPELLCHECK=1 -DUSE_UDEV -DTOOLKIT_VIEWS=1 -DUI_COMPOSITOR_IMAGE_TRANSPORT -DUSE_AURA=1 -DUSE_CAIRO=1 -DUSE_CLIPBOARD_AURAX11=1 -DUSE_DEFAULT_RENDER_THEME=1 -DUSE_GLIB=1 -DUSE_NSS=1 -DUSE_X11=1 -DUSE_XI2_MT=2 -DENABLE_WEBRTC=1 -DENABLE_EXTENSIONS=1 -DENABLE_CONFIGURATION_POLICY -DENABLE_TASK_MANAGER=1 -DENABLE_THEMES=1 -DENABLE_CAPTIVE_PORTAL_DETECTION=1 -DENABLE_SESSION_SERVICE=1 -DENABLE_APP_LIST=1 -DENABLE_SETTINGS_APP=1 -DENABLE_MANAGED_USERS=1 -DENABLE_SERVICE_DISCOVERY=1 -DENABLE_AUTOFILL_DIALOG=1 -DENABLE_REMOTING=1 -DENABLE_GOOGLE_NOW=1 -D_FILE_OFFSET_BITS=64 -D__STDC_CONSTANT_MACROS -D__STDC_FORMAT_MACROS -D_DEBUG -DDYNAMIC_ANNOTATIONS_ENABLED=1 -DWTF_USE_DYNAMIC_ANNOTATIONS=1 -D_GLIBCXX_DEBUG=1 -I../.. -Igen -fno-strict-aliasing -fstack-protector --param=ssp-buffer-size=4 -m64 -funwind-tables -fPIC -pipe -pthread -Wall -Werror -Wsign-compare -Wendif-labels -Wno-missing-field-initializers -Wno-unused-parameter -fvisibility=hidden -O0 -g2 -fno-threadsafe-statics -fvisibility-inlines-hidden -fno-rtti -fno-exceptions -c gen/library_loaders/libspeechd.cc -o obj/out/Debug/gen/library_loaders/libspeechd.libspeechd.o
In file included from gen/library_loaders/libspeechd.cc:4:0:
../../out/Debug/gen/library_loaders/libspeechd.h:7:54: fatalerror: third_party/speech-dispatcher/libspeechd.h: No such file or directory
compilation terminated.
ninja: build stopped: subcommand failed.

> Roll chromium_revision 280876:281094
> 
> No significant DEPS changes in this roll, only some changes
> in how clang_format is downloaded.
> 
> BUG=
> TEST=Local testing as trybots currently cannot handle DEPS changes properly.
> R=niklas.enbom@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/20829004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6598 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 07:53:46 +00:00
kjellander@webrtc.org
cb1df98093 Roll chromium_revision 280876:281094 
No significant DEPS changes in this roll, only some changes
in how clang_format is downloaded.

BUG=
TEST=Local testing as trybots currently cannot handle DEPS changes properly.
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6597 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 07:31:19 +00:00
marpan@webrtc.org
720964faac Fix memcheck error in r6594. 
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 21:14:07 +00:00
kjellander@webrtc.org
11bea8977e GN: Implement BUILD.gn for common_video. 
This adds copying of Chromium's third_party/BUILD.gn
to acommondate libyuv's BUILD.gn that imports the 'jpeg'
config from that file.

BUG=3441
TEST=trybots + local compile passing with:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false build_libyuv=false" && ninja -C out/Default

R=niklas.enbom@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6595 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 17:04:12 +00:00
marpan@webrtc.org
c8364539d3 Fix for FEC decoding with sequence number wrap-around. 
BUG=3507
R=stefan@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6594 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 16:49:30 +00:00
bjornv@webrtc.org
69ef9911e4 delay_estimator: Allows dynamically used history sizes 
Gives the user a possibility to dynamically change the history size. The main advantage is, for example, that you now can start with a wide delay range and over time decrease the search window to lower complexity.

Adds
- two new APIs.
- and updates unit tests.
- a history_size member variable to BinaryDelayEstimator.
- two help function re-allocating buffer memory.

One thing that makes this a little complicated is that you are allowed to have multiple delay estimators with the same reference, so changing the buffer sizes at one place will automatically give you a mismatch at other places.

BUG=3532, 3504
TESTED=trybots and manually
R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6593 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 14:59:03 +00:00
aluebs@webrtc.org
224a140339 Make experimental NS API not purely virtual 
Because not all subclasses will want to bother overriding these methods.

R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6592 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 13:41:39 +00:00
bjornv@webrtc.org
c0ba4392f1 common_audio: Removes macro WEBRTC_SPL_SHIFT_W16 
We should avoid macros in general (see style guide). This shift macro is not a severe one, since there is a check for negativity.

BUG=3348,3353
TESTED=trybots and manually
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6591 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 13:38:53 +00:00
kwiberg@webrtc.org
38214d53db EchoCancellationImpl::ProcessRenderAudio: Use float samples directly 
This patch lets EchoCancellationImpl::ProcessRenderAudio ask the given
AudioBuffer for float sample data directly, instead of asking for
int16 samples and then converting manually.

Since EchoCancellationImpl::ProcessRenderAudio takes a const
AudioBuffer*, it was necessary to add some const accessors for float
data to AudioBuffer.

R=aluebs@webrtc.org, andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6590 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 09:47:33 +00:00
andresp@webrtc.org
a82f9a243d Add Tsan2 to .gitignore 
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6589 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 08:23:58 +00:00
asapersson@webrtc.org
dfdaeb92d8 Removed old code and default implementations. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6588 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 07:35:21 +00:00
braveyao@webrtc.org
9c89e932c9 WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering. 
(Including another fix here, https://review.webrtc.org/16779004/, to make the test run)

BUG=3500
TEST=Manual Test
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6587 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 05:59:22 +00:00
buildbot@webrtc.org
3ffa1f917e (Auto)update libjingle 70422491-> 70424781 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6586 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-02 19:51:26 +00:00
kjellander@webrtc.org
b25b08b302 Remove tools/resources 
The script to update the resources before we used the .sha1 files
was moved out in r4277 and later deleted in r5099.
This dir serves no purpose, so let's remove it.

BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6585 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-02 15:48:29 +00:00
jiayl@webrtc.org
93426cd2ff Implement BUILD.gn for desktop_capture. 
BUG=3441
R=brettw@chromium.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6584 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-02 15:47:12 +00:00
andresp@webrtc.org
33586c83b1 Make deadlock suppressions less generic. 
Previously they were enabled on all webrtc and talk primitives directly when TSAN config changed to enable deadlock detections.

BUG=3509
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-02 14:19:05 +00:00
andresp@webrtc.org
1295dc6a23 Possibly fix deadlock happening due to unregister/register modules as switching between AST and TSO estimators. 
I think this does not introduces any contention or new deadlocks. But that is hard to verify at the moment.

BUG=chromium:388191
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6582 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-02 13:23:19 +00:00
buildbot@webrtc.org
0bb9fac98c (Auto)update libjingle 70343444-> 70394475 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6581 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-02 11:54:09 +00:00
marpan@webrtc.org
895698067c Roll chromium 280149:280876. 
Pick up the libvpx roll:
https://codereview.chromium.org/367733002/

R=andrew@webrtc.org
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6580 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-02 02:50:05 +00:00
buildbot@webrtc.org
d8a9069080 (Auto)update libjingle 70340027-> 70343444 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6579 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 19:26:43 +00:00
tkchin@webrtc.org
74bf7a6523 Add tkchin@ to OWNERS. 
Adding myself to OWNERS of subdirectories containing iOS bits.  Added niklas.enbom@ for audio_device and wu@ for everything else.

R=niklas.enbom@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6578 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 18:38:28 +00:00
jiayl@webrtc.org
974bbbb352 Fix uninitialized value in DtlsTransport and TransportDescription. 
BUG=crbug/390304
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6577 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 18:33:07 +00:00