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Author SHA1 Message Date
sergeyu@chromium.org
894e6fe9ea Add DesktopCaptureOptions class. 
The new class is used to pass configuration parameters to screen/window
capturers. It also allows to share X Window connection between multiple
objects.

R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2374004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4952 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-12 22:40:05 +00:00
henrike@webrtc.org
f53622d42e WebRTCDemo: Fixes warning for devices with pre-17 API level. Also fixes broken build build.xml and project.properties. 
BUG=2083
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2375004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4951 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-11 21:28:26 +00:00
kjellander@webrtc.org
4c61792600 Add SyzyASan to DEPS 
This will make it possible to run our tests under ASan
on Windows.

BUG=2491
TEST=local builds with this DEPS added makes it possible to use
the buildbot code available out-of-the-box.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2381004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4950 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-10 11:56:09 +00:00
kjellander@webrtc.org
5b3b6b1784 Reorganize GYP targets to make webrtc.gyp more usable. 
When WebRTC is built as a part of Chromium, some of
the stuff in webrtc.gyp will not be found. This CL
fixes this.

TEST=trybots passing. I also did some manual builds for Android with the android_builder_webrtc target in build/all_android.gyp of a Chromium checkout.
BUG=chromium:304143
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2353004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4949 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-10 08:48:16 +00:00
wu@webrtc.org
40dfbc4d3d Update talk to 53984350. 
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/2376004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4947 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-09 17:58:06 +00:00
wu@webrtc.org
4551b793de Update libjingle to 53920541. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2371004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4945 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-09 15:37:36 +00:00
andrew@webrtc.org
13b2d46593 clang-format audio_processing/aec/* 
TBR=bjornv
TESTED=trybots

Review URL: https://webrtc-codereview.appspot.com/2373004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4944 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-08 23:41:42 +00:00
wu@webrtc.org
d241718e17 Increase base Chromium revision to get an update to libnss. 
The function signature of SSL_PeerCertificateChain in libnss
was changed by https://codereview.chromium.org/25107004/ ,
and webrtc now uses that function when linked to libnss.

TBR=bemasc

A clone of https://webrtc-codereview.appspot.com/2372004/. Tried by Ben.

Review URL: https://webrtc-codereview.appspot.com/2372005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4943 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-08 22:11:40 +00:00
wu@webrtc.org
ff7b360314 * Remove suppressions that are fixed. 
* Remove duplicated suppression bug_1205_21.

TESTED=try with tsan
BUG=1205
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/2368004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4942 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-08 17:32:39 +00:00
wu@webrtc.org
7818752566 Update libjingle to 53856368. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2366004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4941 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 23:32:02 +00:00
wu@webrtc.org
e0d55a0782 Removing suppressions that has been fixed, i.e. r4661. 
Rename suppressions to match the correct issue.

TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/2357004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4940 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 16:44:38 +00:00
andrew@webrtc.org
ca764ab22d Add a parameter to audioproc for overriding the delay. 
Rename the parameter for adding to the input delay to "add_delay".

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2345007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4939 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 16:44:32 +00:00
elham@webrtc.org
11e9cbc399 Updated WebRTC version to 3.44 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2365004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4937 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 16:18:35 +00:00
stefan@webrtc.org
f5d7c5891c Revert r4934 "Add a tool for parsing an RTP file and outputting the BWE relevant fields." 
Revert r4935 "Fix build error in r4934."

TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2364004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4936 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 08:42:46 +00:00
stefan@webrtc.org
611e5141cb Fix build error in r4934. 
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2363004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4935 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 08:35:36 +00:00
stefan@webrtc.org
bc99bcfa6f Add a tool for parsing an RTP file and outputting the BWE relevant fields. 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2237005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4934 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 08:21:24 +00:00
turaj@webrtc.org
6d5d248075 Wrap ACM2 code inside acm2 namespace. This gurantees that one ACM would not use components of others by accident. 
BUG=
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2344004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4933 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-06 04:47:28 +00:00
turaj@webrtc.org
f31639612d Accounting for wrap-around of timestamps. 
BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2340006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4932 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-06 02:21:24 +00:00
andrew@webrtc.org
20078e2f9b Support video constraints and use key/value pairs. 
- Remove the minre and maxre parameters in favour of setting video
constraints directly.
- In order to support non-boolean values, have constraints passed as
key/value pairs, rather than the leading "-" syntax used earlier to
specify false.

TESTED=Verified that setting various audio and video constraints has
the desired effect, including "true" and "false". Verified that the "hd"
parameter still works.

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2360005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4931 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-05 02:26:50 +00:00
mikhal@webrtc.org
35e4dd3067 VPM: Fixing namespace 
R=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2355004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4930 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 20:21:30 +00:00
fischman@webrtc.org
4598380860 Android: enable camera video stabilization when available. 
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2347005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4929 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 20:14:19 +00:00
kjellander@webrtc.org
7fca2ce097 Add owners to [webrtc,talk]/build and *.isolate (take 2) 
After fischman@'s comments in http://review.webrtc.org/2347006/ here's another CL to clean up the redundancies and add wu@ to webrtc/build/

TEST=none
BUG=none
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2348006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4928 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 19:36:45 +00:00
kjellander@webrtc.org
495f29ef94 Remove unused Android dummy APK 
This is a leftover from our initial Android efforts.
It is not used anywhere and is only confusing to keep around.

The Android precompiled tools in http://review.webrtc.org/2353004/
still have some use when testing Android devices on Mac, so we'll
keep them around by request from henrike@

TEST=none
BUG=none
R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2344008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4927 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 19:33:48 +00:00
kjellander@webrtc.org
e6938185a5 Add isolate targets for libjingle 
Add .isolate file for libjingle tests and and the necessary isolate.gypi file, similar to the change in
http://review.webrtc.org/2338004/

TEST=trybots passing.
I also ran build/gyp_chromium in a Chromium checkout
with third_party/libjingle/source/talk having this patch
applied to ensure GYP processing was still working.

BUG=1916
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2353005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4926 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 19:31:27 +00:00
kjellander@webrtc.org
3f9288f987 Add APK and isolate target for video_engine_tests 
Add .isolate file and _run target for video_engine_tests.

Move tools/swarm_client to be untracked in all .isolate file,
so refactorings in swarm_client doesn't require us updating
all our .isolate files (similar to the changes for the
Chromium tests done in:
https://src.chromium.org/viewvc/chrome?view=rev&revision=218844)

Update modules_unittests.isolate with new NetEq4 reference files
needed.

TEST=trybots passing
I also setup a Chromium workspace where I patched third_party/webrtc
with the changes in this CL, followed by compiling with the settings
described in
https://code.google.com/p/webrtc/issues/detail?id=1882#c11
I then verified that the video_engine_tests_apk dir was created
in the output folder.
BUG=1916,2462
R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2344007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4925 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 18:20:38 +00:00
andrew@webrtc.org
6c264cc92e Clean up AudioProcessing defaults and errors. 
- Remove unneeded #defines and switch the remainder to consts.
- All AudioProcessing components are disabled by default, so remove
explicit disables.
- AudioProcessing uses a rational 16 kHz mono default, so no need to
explictly initialize.
- Add assert(false) to real-time errors which should not occur.

TESTED=trybots
R=bjornv@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2253005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4924 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 17:54:09 +00:00
kjellander@webrtc.org
83b9e5b328 Add owners to [webrtc,talk]/build and *.isolate 
BUG=none
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2347006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4923 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 17:35:26 +00:00
andrew@webrtc.org
acb00505b6 Only declare kDelayDiffOffset when used. 
And remove the redundant Windows block.

R=hans@chromium.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2351004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4922 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 16:59:17 +00:00
henrike@webrtc.org
ad2eb6f67d Unbreaks Android build after r4915. 
TBR=ajm@webrtc.org

BUG=Not filed

Review URL: https://webrtc-codereview.appspot.com/2348005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4921 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 14:21:23 +00:00
andresp@webrtc.org
be9c560aab Revert r4913 that reverts r4911. Original CL description: 
"Adding temporal layer strategy that keeps base layer framerate at an acceptable value."

R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2351006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4920 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 13:11:31 +00:00
andrew@webrtc.org
bab2aa5113 Add audio and video parameters for setting media constraints. 
- These replace the media parameter, now removed.
- Organize the parameter getting a bit.

To describe the new parameters, I'll just copy the code comments here:
Use "audio" and "video" to set the media stream constraints. "true" and
"false" are recognized and interpreted as bools, for example:
  "?audio=true&video=false" (start an audio-only call).
  "?audio=false" (start a video-only call)
If unspecified, the constraint defaults to True.

audio-specific parsing:
To set certain constraints, pass in a comma-separated list of audio
constraint strings. If preceded by a "-", the constraint will be set to
False, and otherwise to True. There is no validation of constraint
strings. Examples:
  "?audio=googEchoCancellation" (enables echo cancellation)
  "?audio=-googEchoCancellation,googAutoGainControl" (disables echo
      cancellation and enables gain control)

TESTED=Verified that passing true, false and various audio constraints
has the desired effect in apprtc.

R=vikasmarwaha@google.com

Review URL: https://webrtc-codereview.appspot.com/2345004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4919 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 22:37:29 +00:00
fischman@webrtc.org
4446134757 AppRTCDemo(android): support boolean value for MediaStreamConstraints.{audio,video}. 
Previously it was assumed that these values were always MediaTrackConstraints but
http://dev.w3.org/2011/webrtc/editor/getusermedia.html#idl-def-MediaStreamConstraints
allows them to be boolean, too.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2352004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4918 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 22:34:10 +00:00
fischman@webrtc.org
a7266ca134 Fix clang build break 
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2350004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4917 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 19:04:18 +00:00
fischman@webrtc.org
6c82e04cee Android standalone: remove some usages of deprecated APIs and prevent further regressions. 
Also:
- Fixed WebRTCDemo UI to say "SwitchToBack" at startup since default camera is front
- Rebuild WebRTCDemo APK when resources/layout/strings change

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2337004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4916 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 18:57:48 +00:00
fischman@webrtc.org
4e65e07e41 VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. 
Besides being ~40% the size of the previous implementation, this makes it so
that VideoCaptureAndroid can stop and restart capture, which is necessary to
support onPause/onResume reasonably on Android.

BUG=1407
R=henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2334004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4915 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 18:23:13 +00:00
fischman@webrtc.org
ddc5a19ce9 AppRTCDemo(android): uncaught exceptions now display a modal dialog box before killing the app. 
BUG=2458
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2348004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4914 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 18:09:40 +00:00
turaj@webrtc.org
44db9d1a57 Revert 4911 "Adding temporal layer strategy that keeps base laye..." 
> Adding temporal layer strategy that keeps base layer framerate at an acceptable value.
> 
> R=stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/2272005

TBR=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4913 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 17:42:07 +00:00
mikhal@webrtc.org
b43d8078a1 Reformatting VPM: First step - No functional changes. 
R=marpan@google.com, marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2333004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4912 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 16:42:41 +00:00
andresp@webrtc.org
26f78f7ecb Adding temporal layer strategy that keeps base layer framerate at an acceptable value. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2272005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4911 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 14:06:14 +00:00
henrik.lundin@webrtc.org
70df305760 Minor fix to avoid breakage 
Related to AutoMute feature. Fixed a lint nit, too.

TBR=mflodman@webrtc.org
BUG=2436

Review URL: https://webrtc-codereview.appspot.com/2347004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4910 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 13:38:59 +00:00
turaj@webrtc.org
7ee3efb0d8 Disable Receiver unittests on Android. 
BUG=
TBR=minyue@google.com

Review URL: https://webrtc-codereview.appspot.com/2344005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4909 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 00:05:15 +00:00
turaj@webrtc.org
6ea3d1cc9e ACM test are modified to run with both ACM1 and ACM2. 
Beside the changes in test files. acm2/acm_generic_codec.cc and acm2/audio_coding_module_impl.cc are modified to fix a bug.

Also, nack{.cc, .h, _unittest.cc} are removed form main/sourc as nack files in both ACM1 and ACM2 are essentially identical.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2192005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4908 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-02 21:44:33 +00:00
kjellander@webrtc.org
2a97317953 Fix include of isolate.gypi 
Recent changes in GYP seem to have broken our previous
"hack" for getting the GYP rule for .isolate files
imported from the Chromium build/isolate.gypi.

The best solution for now is to remove the hack
and check in a copy of Chromium's src/build/isolate.gypi
in WebRTC's build/ dir instead. A similar approach is
used for our build/protoc.gypi file.

TEST=On Linux, I successfully ran:
gclient runhooks
ninja -C out/Release
and verified a bunch of .isolated files were created in
out/Release (which didn't happen before this patch).

I also renamed the build/isolate.gypi from Chromium to
ensure that our own is used and not that one (in case any
paths would be incorrect).

I also ran build/gyp_chromium in a Chromium checkout
with WebRTC in third_party/webrtc having this patch applied
to ensure GYP processing was still working.

Finally, I verified that the same project generation and
compilation from a Chromium checkout worked the way we build
our Android native tests, using:
. build/android/envsetup.sh
GYP_DEFINES="$GYP_DEFINES include_tests=1 enable_tracing=1" gclient runhooks
ninja -C out/Release android_builder_webrtc

BUG=1916
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2338004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4907 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-02 19:31:16 +00:00
henrike@webrtc.org
f8f78b1316 Android OpenSL: Fixes faulty assertion in jni-code. 
BUG=2452
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2342004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4906 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-02 18:41:06 +00:00
pbos@webrtc.org
9b5c807272 Remove ReturnTrace from DeregisterCallback(). 
Should fix deadlock on build bots. Before, TraceImpl called
TraceDispatcher::Print, while TraceDispatcher::Deregister called
TraceImpl through VideoEngine::SetTraceCallback. This violates locking
order as both take their own locks.

BUG=2421
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2340005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4905 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-02 16:22:18 +00:00
henrik.lundin@webrtc.org
4887114af7 Remove templatization of the AudioVector test 
This CL converts the unit tests for AudioVector from typed tests to
regular tests. It is in preparation for removing templatization for
AudioVector in an upcoming CL.

BUG=1363
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2319005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4903 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-02 15:07:28 +00:00
kjellander@webrtc.org
c0b4c4a3c4 Workaround issue with stdin on Windows. 
On Windows, sometimes the compare_videos.py script fails because
the inherited stdin handle from the parent process fails.
It seems we can work around this by passing null to stdin to
the subprocesses, since we're not using it anyway.

The errors looked like this:
Traceback (most recent call last):
  File "C:\b\build\slave\Win7_Tester\build\src\third_party/webrtc/tools/compare_videos.py", line 116, in <module>
    sys.exit(main())
  File "C:\b\build\slave\Win7_Tester\build\src\third_party/webrtc/tools/compare_videos.py", line 91, in main
    barcode_decoder = subprocess.Popen(cmd, stdout=sys.stdout, stderr=sys.stderr)
  File "C:\b\depot_tools\python_bin\lib\subprocess.py", line 588, in __init__
    errread, errwrite) = self._get_handles(stdin, stdout, stderr)
  File "C:\b\depot_tools\python_bin\lib\subprocess.py", line 686, in _get_handles
    p2cread = GetStdHandle(STD_INPUT_HANDLE)
WindowsError: [Error 6] The handle is invalid

Example from http://build.chromium.org/p/chromium.webrtc/builders/Win7%20Tester/builds/4498/steps/webrtc_manual_browser_tests_test/logs/stdio

BUG=302915
TEST=successful runs on Windows and Linux.
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2334005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4902 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-02 15:04:45 +00:00
henrike@webrtc.org
1fdc51ae2a APK for opensl loopback. 
BUG=N/A
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2212004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4901 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-02 14:58:19 +00:00
pbos@webrtc.org
de74b64184 Implement TraceCallbacks in Call. 
Uses a global TraceDispatcher in Call. Lazy initialization of it misses
an atomic compare and exchange to be correct. This is expected to work
fine so long as no Calls are created concurrently.

BUG=2421
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2321005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4900 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-02 13:36:09 +00:00
henrik.lundin@webrtc.org
7ea4f24ea5 Piping AutoMuter interface through to ViE API 
This is a piece of the AutoMuter effort. A second CL will follow containing modifications to the new API, and tests.

BUG=2436
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2331004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4899 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-02 13:34:26 +00:00