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Author SHA1 Message Date
kjellander@webrtc.org
5ed4f46af1 Remove TSan v2 disabled test in condition_variable_unittest.cc 
When we rolled our chromium_revision 226126:228675 in r4966
we picked up Clang r191856, which fixes the problem we've
seen earlier in condition_variable_unittest.cc.
Because of this, I'm now re-enabling this test.

TEST=trybots passing
BUG=2259
R=hta@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2404004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4971 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 10:50:41 +00:00
kjellander@webrtc.org
662b1c59c1 Add suppressions for DrMemory and TSan on Windows. 
BUG=2515,2516
TEST=locally running the test as described in the bug.
The TSan excluded test was verified to be the only remaining
test that is too slow for TSan on Windows.

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2405004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4970 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 10:46:37 +00:00
pbos@webrtc.org
b44c2a3193 Open file in binary in CreateFromYuvFile(). 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2399005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4969 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 08:46:06 +00:00
sergeyu@chromium.org
e6e749da38 Add MouseCursorRenderer. 
The new class acts as a wrapper for DesktopCapturer interface. It takes
mouse shape and position from MouseCursorCapturer and renders it on the
frames produced by underlying DesktopCapturer.

BUG=crbug.com/173265
R=wez@chromium.org
TBR=andrew@webrtc.org (modules.gyp)

Review URL: https://webrtc-codereview.appspot.com/2387004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 02:48:41 +00:00
sergeyu@chromium.org
2767b53f66 Add MouseCursorCapturer interface with implementation for X11. 
The new interface will be used to capture cursor shape and position and
blend it into the image captured with desktop capturers.

BUG=crbug.com/173265
R=wez@chromium.org
TBR=andrew@webrtc.org (modules.gyp)

Review URL: https://webrtc-codereview.appspot.com/2386005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4967 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 02:42:38 +00:00
kjellander@webrtc.org
3555303cb0 Roll chromium_revision 226126:228675 and fix clang warnings 
By request from thakis@chromium.org, I disabled the
-Wno-unused-const-variable setting that is set in Chromium's
common.gypi so we can prepare our code for it's removal.

This required some cleanup in order to get the code to compile
with Clang having the -Wunused-const-variable warning enabled.

TEST=all trybots passing
BUG=none
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2400004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-15 20:10:17 +00:00
stefan@webrtc.org
e5021fe590 Make RtpData and RtpFeedback destructors public. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4965 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-15 10:38:40 +00:00
pbos@webrtc.org
266c7b330a Move ChromaGenerator to common_video/. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2394004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4964 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-15 09:15:47 +00:00
andrew@webrtc.org
c2e471d8b3 Compile out unused kMinTrustedDelayMs. 
TBR=niklas.enbom@webrtc.org
TESTED=trybots

Review URL: https://webrtc-codereview.appspot.com/2398004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4963 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-15 02:11:21 +00:00
mallinath@webrtc.org
6fa456f928 Disabling the DTLS renegotiation test case for PeerConnection. 
Currently it's failing on Linux memcheck, most likely due to timing issues.

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2394006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4962 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-15 00:11:54 +00:00
henrike@webrtc.org
901ae77618 Android: Fixes WebRTCDemo build (missing Java code). 
TBR=ajm@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/2395005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4961 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-14 21:46:53 +00:00
henrik.lundin@webrtc.org
1871dd2fb7 NetEq4: Removing templatization for AudioVector 
This is the last CL for removing templates in Audio(Multi)Vector.

BUG=1363
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2341004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4960 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-14 20:33:25 +00:00
kjellander@webrtc.org
fc89ba580b Fix build dir flag in webrtc_test.py as passed by runtests.py 
It seems we were hit by the changes in
https://codereview.chromium.org/26184003/
in how we roll with our own wrapper script for the
memory tools.
The build dir flag was changed from --build_dir to
--build-dir, which caused our script to break.

BUG=none
TEST=verified the exact command line executed by the bot succeeds
in my local checkout
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2394005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4959 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-14 18:05:52 +00:00
sergeyu@chromium.org
30792987b8 Remove empty line in SharedXDisplay::RemoveEventHandler. 
TBR=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2397004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4958 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-14 17:58:46 +00:00
kjellander@webrtc.org
09418c3320 Add support for --target flag to webrtc_tests.py. 
In https://codereview.chromium.org/26190002 Chromium
started a cleanup of their wrapper script, by adding
the --target flag and start passing build dir similar
to how the bots are setting it (i.e. pass out and not
out/Release).
This CL adds --target support for our wrapper script,
without changing the existing behavior (I'll do a
larger update at a later stage).

BUG=none
TEST=Ran the following successfully:
tools/valgrind-webrtc/webrtc_tests.sh --build_dir out/Release --target Release --test test_support_unittests
tools/valgrind-webrtc/webrtc_tests.sh --build_dir out --target Release --test test_support_unittests
tools/valgrind-webrtc/webrtc_tests.sh --target Release --test  out/Release/test_support_unittests
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2396004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4957 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-14 17:34:38 +00:00
henrike@webrtc.org
05773e5a70 Android OpenSlDemo: remove some usages of deprecated APIs that is breaking the bots. 
TBR=fischman@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/2395004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4956 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-14 16:25:11 +00:00
mallinath@webrtc.org
19f27e6a24 Update talk to 54527154. 
TBR=wu

Review URL: https://webrtc-codereview.appspot.com/2389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4954 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-13 17:18:27 +00:00
sergeyu@chromium.org
7419a72383 Add event handling in SharedXDisplay. 
SharedXDisplay has to handle X events because the events may belong to
different clients of that class.

R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2386004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4953 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-13 00:44:09 +00:00
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