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Author SHA1 Message Date
kjellander@webrtc.org
c68d046bb1 Fix BUILD.gn to load all Chromium GN configurations. 
After troubleshooting with brettw@chromium.org we
found that this line is needed in order to get the
build/config/BUILD.gn configurations loaded.

This should solve the runhooks failures in
http://build.chromium.org/p/client.webrtc.fyi/waterfall
and make it possible to generate projects with
GYP_DEFINES="disable_glibcxx_debug=1"
(which fails today).

TEST=Ran on Linux:
GYP_DEFINES=build_with_tool=tsan gclient runhooks
without and with this patch applied (it fails without).

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

Review URL: https://webrtc-codereview.appspot.com/8379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5521 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-10 21:28:55 +00:00
michaelbai@google.com
a65abf9d3a Revert "Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file." 
This reverts commit 7686f0ddda.

BUG=

TBR=andrew@webrtc.org, fischman@webrtc.org,

Review URL: https://webrtc-codereview.appspot.com/8369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5520 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-10 19:26:26 +00:00
jiayl@webrtc.org
1f64f06784 Add stats of incoming frame delays for debugging bandwidth estimation. 
BUG=crbug/338380
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5519 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-10 19:12:14 +00:00
fischman@webrtc.org
82387e4608 Add ability to receive calls for iOS 
BUG=2701
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7989005

Patch from Sajid Hussain <shussain@temasys.com.sg>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5518 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-10 18:47:11 +00:00
michaelbai@google.com
7686f0ddda Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file. 
This patch removes generate_asm_header.gypi and uses libvpx's obj_int_extract and
unpack_lib_posix to generate offset header files.

It make the simliar feature's implementation consistent.

R=andrew@webrtc.org, fischman@webrtc.org, fischman@chromium.org
BUG=334447

Review URL: https://webrtc-codereview.appspot.com/7769006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-10 17:42:34 +00:00
pbos@webrtc.org
0a7085ffc2 Thread annotation of talk_base::CriticalSection. 
Also enabling -Wthread-safety in talk/build/common.gypi for clang on
Linux. Thread annotations are compile-time checks that for instance
certain locks are held before accessing a value.

BUG=
TEST=Local GUARDED_BY() annotations.
R=andresp@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5516 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-10 13:58:37 +00:00
kjellander@webrtc.org
9cba2e4802 Exclude libjingle_p2p_unittest tests failing TSan Race verifier. 
These tests have been failing for a long time at
http://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20TsanRV
This bot runs a tool called TSan Race Verifier.
More info about Race Verifiers can be found at
https://code.google.com/p/data-race-test/wiki/RaceVerifier

By excluding the tests like this, they will also be disabled
for the normal TSan bot, where they currently pass. This shouldn't
be a problem since we have a TSan v2 bot running too (which uses
a different set of suppressions), so it should catch any newly
introduced race conditions. The TSan v2 bot will soon replace the
old TSan bot, as v1 is being deprecated.

TBR=mallinath@webrtc.org
BUG=2396
TEST=Passing run on Linux of:
GYP_DEFINES=build_with_tool=tsan gclient runhooks
ninja -C out/Release libjingle_p2p_unittest
tools/valgrind-webrtc/webrtc_tests.sh --tool=tsan_rv -b out/Release -t libjingle_p2p_unittest

Review URL: https://webrtc-codereview.appspot.com/8329005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5515 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-10 12:43:40 +00:00
sprang@webrtc.org
6f0ca57fb2 Add experiment: SkipEncodingUnusedStreams 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5514 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-10 09:20:51 +00:00
kjellander@webrtc.org
4723dc88b3 Revert 5511 "Revert 5510 "Disable failing libjingle_p2p_unittest..." 
So, the test apparently failed right away at 

http://build.chromium.org/p/client.webrtc/builders/Linux64%20Debug/builds/1224/steps/libjingle_p2p_unittest/logs/stdio


> Revert 5510 "Disable failing libjingle_p2p_unittest test on Linux"
> 
> According to https://code.google.com/p/webrtc/issues/detail?id=2907#c2
> r5505 was committed to resolve exactly these flakes.
> Let's revert the disabling and see.
> 
> BUG=2907
> TBR=mallinath@webrtc.org
> 
> > Disable failing libjingle_p2p_unittest test on Linux
> > 
> > I realize this diables 84 test cases and for all platforms, which
> > I'm not really comfortable with. I tried finding a better way but
> > couldn't without doing significant changes to the file.
> > I think the tests either needs to be fixed or otherwise refactored
> > in order to make more fine-grained disabling possible.
> > 
> > Another (too) large disabling was done by holmer@ in
> > https://webrtc-codereview.appspot.com/2227004 where he should only have
> > disabled them on Windows, if the failures in webrtc:2383 was all that
> > caused those flakes.
> > 
> > BUG=2907
> > TEST=Verified this ran 0 tests:
> > out/Release/libjingle_p2p_unittest --gtest_filter=P2PTransportChannelTest.TestNAT_ADDR_RESTRICTEDToNAT_PORT_RESTRICTEDAsGiceBothSharedUfragWithMinimumStepDelay
> > TBR=wu@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/8309004
> 
> TBR=kjellander@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/8329004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5513 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-09 18:48:56 +00:00
kjellander@webrtc.org
607c805b87 Roll chromium_revision 245382:249215 
The find_depot_tools.py is needed to workaround the import
error we get from gyp_chromium when importing it in
webrtc/build/gyp_webrtc (to avoid code duplication).
gyp_chromium introduced a dependency on it in
http://crrev.com/245412 but as we cannot sync all of Chrome's
src/tools (it's quite big), we'll work around this by
adding an empty find_depot_tools module.

The removal of the Cygwin relates to
http://crrev.com/248802 which is a step on the way to remove
Cygwin in Chromium. We seem to already be able to remove it
entirely for WebRTC though.

Changes in the isolate framework required us to update our
copies of the isolate.gypi files.

BUG=none
TEST=trybots passing on all platforms
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5512 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-09 18:38:31 +00:00
kjellander@webrtc.org
ce2b44532e Revert 5510 "Disable failing libjingle_p2p_unittest test on Linux" 
According to https://code.google.com/p/webrtc/issues/detail?id=2907#c2
r5505 was committed to resolve exactly these flakes.
Let's revert the disabling and see.

BUG=2907
TBR=mallinath@webrtc.org

> Disable failing libjingle_p2p_unittest test on Linux
> 
> I realize this diables 84 test cases and for all platforms, which
> I'm not really comfortable with. I tried finding a better way but
> couldn't without doing significant changes to the file.
> I think the tests either needs to be fixed or otherwise refactored
> in order to make more fine-grained disabling possible.
> 
> Another (too) large disabling was done by holmer@ in
> https://webrtc-codereview.appspot.com/2227004 where he should only have
> disabled them on Windows, if the failures in webrtc:2383 was all that
> caused those flakes.
> 
> BUG=2907
> TEST=Verified this ran 0 tests:
> out/Release/libjingle_p2p_unittest --gtest_filter=P2PTransportChannelTest.TestNAT_ADDR_RESTRICTEDToNAT_PORT_RESTRICTEDAsGiceBothSharedUfragWithMinimumStepDelay
> TBR=wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/8309004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5511 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-09 18:21:00 +00:00
kjellander@webrtc.org
8d2ddd00f1 Disable failing libjingle_p2p_unittest test on Linux 
I realize this diables 84 test cases and for all platforms, which
I'm not really comfortable with. I tried finding a better way but
couldn't without doing significant changes to the file.
I think the tests either needs to be fixed or otherwise refactored
in order to make more fine-grained disabling possible.

Another (too) large disabling was done by holmer@ in
https://webrtc-codereview.appspot.com/2227004 where he should only have
disabled them on Windows, if the failures in webrtc:2383 was all that
caused those flakes.

BUG=2907
TEST=Verified this ran 0 tests:
out/Release/libjingle_p2p_unittest --gtest_filter=P2PTransportChannelTest.TestNAT_ADDR_RESTRICTEDToNAT_PORT_RESTRICTEDAsGiceBothSharedUfragWithMinimumStepDelay
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5510 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-08 21:35:20 +00:00
kjellander@webrtc.org
6f8acdc76f Suppressions for libjingle_unittest after roll in r5502 
New errors arrived when rolling libjingle in r5502.
These suppressions are needed to green up the Memcheck and
TSan bots.

BUG=1976,2080
TEST=local runs on Linux:
tools/valgrind-webrtc/webrtc_tests.sh --tool=tsan -b out/Release -t libjingle_unittest
tools/valgrind-webrtc/webrtc_tests.sh --tool=memcheck -b out/Release -t libjingle_unittest
and trybot:
git try --bot=linux_memcheck,linux_tsan -t libjingle_unittest
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5509 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-08 21:15:59 +00:00
sergeyu@chromium.org
cc685acbdf Disable AsyncInvokeTest.CancelInvoker test 
Test is flaky.

BUG=b/12944358
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5508 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-08 19:59:00 +00:00
sergeyu@chromium.org
0178810659 Don't use LOG() in callback.h 
Because chromium is compiled with a different version of logging macros
defined in logging.h that header cannot be used in headers that can
also included from chromium code. Removed LOG_F(LS_WARNING) from
callback.h . That issue would block this code from being rolled in
chromium.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5507 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-08 03:18:03 +00:00
fischman@webrtc.org
bfc26dcc10 gitignore: ignore webrtc android demo apps build artifacts, and sort list 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5506 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-08 01:05:45 +00:00
mallinath@webrtc.org
5a59ccbb6d Switching to NSS random number generator and adding init method to unittests. 
R=jiayl@webrtc.org, sergeuy@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5505 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 23:22:00 +00:00
sergeyu@chromium.org
ad3035fc9e Fix WindowCapturerWin to unselect bitmap before destroying DC. 
BUG=https://code.google.com/p/webrtc/issues/detail?id=2901
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/8229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5504 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 21:24:04 +00:00
vikasmarwaha@webrtc.org
c5a839c3a9 Updated demos so they work on FF, the createOffer api cannot have null parameters according to spec. Same applies to createAnswer. 
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/8219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5503 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 19:08:38 +00:00
sergeyu@chromium.org
9cf037b831 Update libjingle to 61168196 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5502 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 19:03:26 +00:00
sprang@webrtc.org
9510e53cc0 Make VideoReceiveStream::GetStats() const. 
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5501 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 15:32:45 +00:00
kjellander@webrtc.org
f7d4993940 Remove Coverity script. 
Chromium has decided to drop Coverity so
we don't have any reason for maintaining this code.

Personally, I think that from a quality perspective other tools,
like all the new compiler warnings that are constantly being added
to the Clang compiler is a better way to address dangers in the code.
The maintenance cost and overhead of such advanced tools like Coverity
is simply too high.

BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7669007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5500 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 13:50:40 +00:00
sprang@webrtc.org
09315705b9 Wire up statistics in video receive stream of new API 
This CL includes Call tests that test both send and receive sides.

BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5499 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 12:06:29 +00:00
kjellander@webrtc.org
90ce73a0d5 Remove svn:ignore for third_party/cygwin 
In preparation for landing DEPS removal of Cygwin
in https://webrtc-codereview.appspot.com/8099004/



git-svn-id: http://webrtc.googlecode.com/svn/trunk@5498 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 10:04:21 +00:00
vikasmarwaha@webrtc.org
b307e86076 Updated demos to use the sucess and failure callback in addIceCandidate api. 
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/7969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5497 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-06 22:38:37 +00:00
fbarchard@google.com
60de116687 libyuv.gyp fix for ios sim which is intel not neon, fixing a link error. 
BUG=none
TEST=try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5496 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-06 21:17:16 +00:00
marpan@webrtc.org
dfef7ba971 Roll libvpx 241571:248011 
TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/8129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5495 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-06 18:40:35 +00:00
stefan@webrtc.org
77c917a6ee Plot the capacity of a trace-based delivery filter. 
Breaks out the instantaneous rate counters to its own class.

R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7999005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5494 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-06 16:34:47 +00:00
pbos@webrtc.org
ea1c5ad58f Fix gunit compilation on VS2012. 
In VS2012 compiling gunit or its dependencies triggers a lot of
"'std::tuple' : too many template arguments" warnings. The workaround
for this, done for gtest already, is to define _VARIADIC_MAX=10.

BUG=2616
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5493 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-06 13:17:20 +00:00
michaelbai@google.com
f928f5c87c Use system's cpu_features library 
Remove the copied cpu_featrues.c/h
Use the NDK's cpu_features.a or the one build from android source.
This issue blocked libvpx roll.

BUG=334447
R=andrew@webrtc.org, fischman@webrtc.org, henrike@webrtc.org, wjia@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5492 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-05 18:43:46 +00:00
stefan@webrtc.org
c88d3368d5 Add delay and send/receive throughput plots to BWE simulation. 
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5491 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-05 15:57:14 +00:00
henrik.lundin@webrtc.org
75642fcd9a Implementing replacement audio support in neteq_rtpplay 
This CL makes it possible to replace the payload in an RTP stream
with audio from another (PCM) file. The new payload will be encoded as
PCM16b. The RTP headers will be updated to reflect this change of
payload type.

BUG=2834
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5490 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-05 08:49:13 +00:00
henrik.lundin@webrtc.org
e6ab21b9ca Fixing a bug in DummyRTPpacket 
This bug caused writing outside allocated memory when RTP header
extensions were used.

BUG=2834
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8009005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-05 08:46:46 +00:00
andrew@webrtc.org
54744918ef Update AudioProcessing::Create docs. 
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/8039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5488 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-05 06:30:29 +00:00
jiayl@webrtc.org
20a60ea39d Fix a cursor capturing issue on Windows. 
The input position to WindowFromPoint should be relative to the desktop, not
relative to the window; if the result from WindowFromPoint is a child window
of the shared top window, it should be captured.

BUG=
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5487 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-04 17:49:12 +00:00
stefan@webrtc.org
0e5a2b5de6 Handle the invalid case of setting multiple stream_bitrates if there is only a single send stream registered. 
This can happen when switching between multiple streams and a single while getting feedback from the receiver.

BUG=2881
TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5486 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-04 14:38:25 +00:00
pbos@webrtc.org
3e6c41c48f Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents" 
This reverts commit r5479.

R=henrika@webrtc.org
BUG=2880

Review URL: https://webrtc-codereview.appspot.com/7989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5485 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-04 10:45:14 +00:00
pbos@webrtc.org
064b32acbb Fix locking in LoopBackTransport::StorePacket. 
The critical section in StorePacket was unnamed and only existed in
expression scope. Added GUARDED_BY annotations (which caught the bug),
then fixed it by naming the variable.

BUG=2880
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5484 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-04 09:42:02 +00:00
andrew@webrtc.org
36291da197 Pull Chromium's clang-format binaries. 
This gets 'git cl format' working again in a standalone webrtc checkout.
It started failing after this depot_tools change:
https://codereview.chromium.org/134313007

Depends on this change:
https://codereview.chromium.org/135653014/

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5483 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-04 01:45:10 +00:00
andrew@webrtc.org
f6a638e001 Trivial rename of non-compile time consts. 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7669006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5482 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-04 01:31:28 +00:00
marpan@webrtc.org
e88c186dbe Revert r5480 
TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/7959005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5481 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-04 00:02:35 +00:00
marpan@webrtc.org
e35ecb476b Roll libvpx 241571:248011 
R=andrew@webrtc.org
TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/7949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5480 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 22:53:10 +00:00
marpan@webrtc.org
f6b8f496ee Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents 
Issue: https://code.google.com/p/webrtc/issues/detail?id=2880

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

Review URL: https://webrtc-codereview.appspot.com/7939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 21:34:35 +00:00
fischman@webrtc.org
6e08228525 PeerConnectionTest(java): remove the obsolete magical names of streams & tracks. 
BUG=1253
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7929005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5478 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 19:15:44 +00:00
fischman@webrtc.org
a06ebab1e1 PeerConnectionTest(java): test SCTP DataChannels. 
BUG=1408,2253,2626
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5477 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 19:11:29 +00:00
mallinath@webrtc.org
ecd622eec3 Updating libjingle.gyp after addition new files yuvframescapturer.cc. 
TBR=pbos@webrc.org

Review URL: https://webrtc-codereview.appspot.com/7919006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5476 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 17:17:05 +00:00
mallinath@webrtc.org
67ee6b9a62 Update talk to 60923971 
Review URL: https://webrtc-codereview.appspot.com/7909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5475 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 16:57:16 +00:00
stefan@webrtc.org
422fdbf502 Wire up feedback to VideoSender. 
BUG=
R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 16:33:50 +00:00
aluebs@webrtc.org
c9ee412070 Re-enabling audio processing tests 
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5473 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 14:41:57 +00:00
xians@webrtc.org
c1e28038ba Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base 
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-02 15:30:20 +00:00