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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
fbarchard@google.com
08564546cb roll libyuv to r1025 for mips n32 support, arm nacl port, psnr tool jpeg support. 
BUG=none
TESTED=untested
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6576 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 18:17:21 +00:00
buildbot@webrtc.org
6335645400 (Auto)update libjingle 70329914-> 70330023 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 16:46:01 +00:00
henrike@webrtc.org
37b4e1bbcb webrtc/base: add dependent setting for gtest include directory that was missed when creating base_tests.gyp. Same as https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle_tests.gyp?r=6484#39 
BUG=N/A
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6574 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 16:39:17 +00:00
kjellander@webrtc.org
0402515d35 Implement command line flags for peerconnection client example on Windows 
Adding the flags and functionality for 'autoconnect', 'autocall', 'server',
'port', and 'help' like in the linux example.

BUG=3459
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13609004

Patch from Vicken Simonian <vsimon@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6573 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 16:28:13 +00:00
stefan@webrtc.org
9138eb649b Fix compile error introduced with r6571. 
TBR=mflodman@webrtc.org

BUG=3527

Review URL: https://webrtc-codereview.appspot.com/20799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6572 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 12:44:05 +00:00
stefan@webrtc.org
5779ca478d Fixes a potential BWE clock mismatch bug. 
Since libjingle provides a packet arrival timestamp to webrtc, and the clock in remote bitrate estimator and the clock used for packet arrival timestamp can be different. This can cause the bandwidth estimator to malfunction.

This CL changes the remote bitrate estimator so that packet arrival timestamps never are compared to the time taken from the internal clock.

BUG=3527
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6571 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 12:14:49 +00:00
bjornv@webrtc.org
6d21ddca5f audio_processing/aec: Refactors NonLinearProcessing to prepare for NEON optimizations 
Puts functionality necessary to calculate sub-band coherences into a function.

BUG=3131
TESTED=trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6570 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 10:03:42 +00:00
henrike@webrtc.org
d5a0506e84 Use X509_NAME, not struct X509_name_st. 
Also include openssl/x509.h explicitly since we're using functions and types
from it.

BUG=none
R=henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6569 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 20:38:56 +00:00
bjornv@webrtc.org
59adb1dcd7 Neon version of cftmdl_128() 
The performance gain on a Nexus 7 reported by audioproc is ~2.3

The output is bit exact.

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19829004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6568 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 19:34:33 +00:00
aluebs@webrtc.org
9825afc3bd Add ExperimentalNs support in Config 
R=andrew@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6567 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 17:39:53 +00:00
pbos@webrtc.org
2be53a3b00 Disable CanSwitchToUseAllSsrcs on DrMemory. 
Test times out.

R=mflodman@webrtc.org
BUG=3159

Review URL: https://webrtc-codereview.appspot.com/13799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6566 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 15:53:07 +00:00
pbos@webrtc.org
be9d2a4549 Reserve RTP/RTCP modules in SetSSRC. 
Allows setting SSRCs for future simulcast layers even though no set send
codec uses them.

Also re-enabling CanSwitchToUseAllSsrcs as an end-to-end test, required
for bitrate ramp-up, instead of send-side only (resolving issue 3078).
This test was used to verify reserved modules' SSRCs are preserved
correctly.

To enable a multiple-stream end-to-end test test::CallTest was modified
to work on a vector of receive streams instead of just one.

BUG=3078
R=kjellander@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15859005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6565 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 13:19:09 +00:00
bjornv@webrtc.org
cd9b90ab53 Neon version of cft1st_128() 
The performance gain on a Nexus 7 reported by audioproc is ~2%

See comments regarding the output.

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

Review URL: https://webrtc-codereview.appspot.com/21679004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6564 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 12:05:18 +00:00
phoglund@webrtc.org
e9b9ec5ced Removing W3C conformance tests after move to web-platform-tests. 
BUG=webrtc:3455
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6563 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 09:54:27 +00:00
wuchengli@chromium.org
ae7cfd7bc8 Make MediaOptimization thread-safe. 
HW encoder posts the encode callback to libjingle worker
thread. It accesses MediaOptimization and is not protected
by the critial section of VideoSender. Make MediaOptimization
thread-safe to fix it.

BUG=chromium:367691
TEST=Run apprtc loopback with SW or HW encoders.
     Run module_unittests.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6562 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 08:01:47 +00:00
kjellander@webrtc.org
62711f8227 GN: Fix build by disabling compiler warning in base. 
It seems like it is not possible to disabled the -Wall
warnings that are enabled in build/config/compiler/BUILD.gn
with -Wno-all.

According to the documentation at
https://code.google.com/p/chromium/wiki/GNCookbook
the proper way is to disable the chromium_code config instead.

System wrappers also needed some minor fixes for Android.

TBR=henrike@webrtc.org
BUG=3441
TEST=Passing our GN trybots.

Review URL: https://webrtc-codereview.appspot.com/18649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6561 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-29 13:37:08 +00:00
kjellander@webrtc.org
7497fa74e4 GN: Refactor base/BUILD.gn and fix dbus-glib error. 
Refactor webrtc/base/BUILD.gn to not have any subtracted
source entries.

Also fix an error in webrtc/BUILD.gn that occurs when running
on Chormium trybots as a part of enabling WebRTC for GN in
https://codereview.chromium.org/321313006/
The error is that pkg-config for dbus-glib fails. Workaround
this by putting the pkg-config entry within the proper condition.

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 clang_use_chrome_plugins=false" && ninja -C out/Default

I built successfully from a Chromium checkout (with
https://codereview.chromium.org/321313006/ applied) using:
gn gen out/Default && ninja -C out/Default webrtc

R=brettw@chromium.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6560 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-28 18:05:22 +00:00
andrew@webrtc.org
b3c188f27b Use the libvpx rev from Chromium's DEPS, not the Chromium rev. 
R=kjellander@webrtc.org, marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/18639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6559 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-28 17:49:31 +00:00
marpan@webrtc.org
ee4e466661 Roll libvpx: follow the Chromium revision. 
R=andrew@webrtc.org
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6558 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-27 21:00:22 +00:00
henrike@webrtc.org
6f833c332c Rebase webrtc/base with r6555 version of talk/base: 
cd webrtc/base
svn diff -r 6521:6555 http://webrtc.googlecode.com/svn/trunk/talk/base >
6555.diff
patch -p0 -i 6555.diff

BUG=3379
TBR=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6556 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-27 16:21:49 +00:00
buildbot@webrtc.org
bfa758a54c (Auto)update libjingle 70004190-> 70103367 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6555 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-27 16:04:43 +00:00
henrike@webrtc.org
680555f923 constructormagic.h macros are duplicated in several repositories. undef them in webrtc to prevent conflict for some build configurations. 
BUG=N/A
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6554 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-27 15:49:02 +00:00
aluebs@webrtc.org
f4d6d7c27e Add DrMemory suppression for AsyncWriteTest 
BUG=webrtc:3490
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6553 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-27 13:55:36 +00:00
kjellander@webrtc.org
767d98ebff TSan: Move suppressions to source file. 
Chromium has deprecated text-file based suppressions for
TSan (v2) and is about to remove the support for it in the
test toolchain in https://codereview.chromium.org/357673002/

This CL moves our suppressions to a source file (based on the
Chromium copy).
It also moves the sanitizer_options.gyp into webrtc/build.

BUG=chromium:302040
TEST=Locally executing all the standalone tests under TSan v2.
R=niklas.enbom@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6552 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-27 09:18:51 +00:00
pbos@webrtc.org
994d0b7229 Refactor Call-based tests. 
Greatly reduces duplication of constants and setup code for tests based
on the new webrtc::Call APIs. It also makes it significantly easier to
convert sender-only to end-to-end tests as they share more code.

BUG=3035
R=kjellander@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6551 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-27 08:47:52 +00:00
kjellander@webrtc.org
35d46fbe1a Roll chromium_revision 277350:280149 
This fixes an error for GN (http://crrev.com/278107)

Overview of changes in Chrome DEPS:
$ svn diff http://src.chromium.org/chrome/trunk/src/DEPS -r 277350:280149

which can be compared with the output of:
$ svn cat http://webrtc.googlecode.com/svn/trunk/DEPS | grep chromium_deps | sed 's/^ *//' | sort | uniq

in a WebRTC checkout, gives the following relevant changes:
* buildtools 5d8997:fb782d
* third_party/android_tools c6e658:fbd420
* tools/gyp 1927:1944
* tools/swarming_client ae8085:aea506

BUG=3441
TEST=Local compile on most platforms (since trybots currently cannot detect DEPS-changes properly).
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6550 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-27 07:54:02 +00:00
henrik.lundin@webrtc.org
c8e98187d1 Receiver bit-exactness test for AudioCoding Module 
This CL introduces a bit-exactness test for the receive-side of the
AudioCoding Module. The main part of the test is done in the helper
class AcmReceiveTest. The test is executed from the test fixture
AcmReceiverBitExactness.

The test inserts packets from a pre-encoded RTP file. The output is
summed up into a checksum, which is verified versus a reference at the
end of the test. Alternatively, if the flag --generate_output is given,
the output is written to a file for subjective verification.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6549 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-26 19:07:04 +00:00
henrike@webrtc.org
7ea71de396 clock.h: Removed GUARDED_BY annotation as it breaks som builds. 
BUG=N/A
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6547 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-26 16:13:58 +00:00