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Author SHA1 Message Date
asapersson@webrtc.org
f8723d666a Add unit tests to rtcp_receiver_test. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6994 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-28 07:35:06 +00:00
marpan@webrtc.org
2dbb47abb4 Roll chromium_revision b1748b:681cc8 
Pick the libvpx roll: https://codereview.chromium.org/513593002

BUG=3747
R=andrew@webrtc.org
TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/14229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6993 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-28 02:32:45 +00:00
pbos@webrtc.org
956f281d2f Re-enable all VideoAdapterTests on DrMemory. 
These bugs should've been resolved as of r6991.

BUG=3655,3671
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6992 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-27 18:41:58 +00:00
pbos@webrtc.org
75c3ec1763 Fix data races during VideoAdapterTest tear-down. 
Explicitly disconnect the VideoCapturer to avoid frames being
delivered during listener destruction. This manifested only on DrMemory
Full on Windows which I was able to repro locally.

BUG=3671
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6991 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-27 18:16:13 +00:00
buildbot@webrtc.org
573a1eef3d (Auto)update libjingle 74202294-> 74230205 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6990 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-27 17:21:19 +00:00
henrik.lundin@webrtc.org
18584fcde4 Move end of namespace inside #ifdef 
The code did not compile unless WEBRTC_ANDROID was defined.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6989 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-27 10:17:22 +00:00
andresp@webrtc.org
c3c29113d1 Expose setPayloadType on the rtp_sender. Thus allowing other users of this module 
to set the payload type to be used without having to call SendOutgoingData.

BUG=3694
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6988 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-27 09:39:43 +00:00
solenberg@webrtc.org
00f11f5e24 - Make local constant non-static. 
- Remove spammy log line.

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

Review URL: https://webrtc-codereview.appspot.com/21339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6987 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-27 08:52:17 +00:00
henrike@webrtc.org
66a3582170 Create a copy of talk/sound under webrtc/sound. 
BUG=3379
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6986 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 22:04:04 +00:00
guoweis@webrtc.org
7087857afd implement handling ALTERNATE-SERVER response from turn protocol as 
specified in RFC 5766, also created 2 test cases for both the normal
redirection case as well as when a pingpong situation happens, the
allocation should fail

BUG=1986 TURN ALTERNATE-SERVER support
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6985 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 21:37:49 +00:00
kjellander@webrtc.org
dc926a000e Avoid syncing unnecessary Chromium deps for WebRTC. 
This should save several gigabytes of traffic and disk space.

On Linux this is about 2.6 GB:
346M	src/chrome/tools/test/reference_build
340M	src/native_client
170M	src/third_party/ffmpeg
1.5G	src/third_party/WebKit
196M	src/v8

BUG=2863
TESTED=Removed the directories locally, ran a sync and verified they didn't reappear (or fail because of platform-specific ones).
R=iannucci@chromium.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6984 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 19:22:03 +00:00
buildbot@webrtc.org
3533bfcb94 (Auto)update libjingle 74132319-> 74133664 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6983 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 15:50:23 +00:00
buildbot@webrtc.org
4470d78c9b (Auto)update libjingle 74128148-> 74132319 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6982 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 15:24:54 +00:00
aluebs@webrtc.org
b623c5c251 Disable EndToEndTest.RestartingSendStreamPreservesRtpState in video_engine_tests because it is flaky 
BUG=webrtc:3745
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6981 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 14:22:51 +00:00
pbos@webrtc.org
f21ac1fd46 Fix Win64 compile of videoadapter_unittest.cc. 
Missed an typecast in videoadapter_unittest.cc in r6979 due to
tryservers being clogged and me waiting for a windows, linux, mac and
tsanv2 bot to finish was not enough. Committing fix straight away to
un-break tree.

TBR=tommi@webrtc.org
BUG=3671

Review URL: https://webrtc-codereview.appspot.com/18279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6980 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 12:46:57 +00:00
pbos@webrtc.org
c9b3f77e65 Fix data races in VideoAdapterTest. 
Adressing clear races between the test thread and capturer thread shown
as heap-use-after-free in vpx_codec_destroy in
WebRtcVideoMediaChannelTest.SetSend (way later in the rest run).

When capturing a frame the test copied it to a separate frame that would
then be read by the test without synchronization, if the test didn't
manage to examine the frame in between captures the adapted frame would
be overwritten by the following frame during accesses to it.

The actual races are suppressed by race:webrtc/base/messagequeue.cc and
race:webrtc/base/thread.cc. These fixes reduce the suppression count
locally from around 3000 to 30 for VideoAdapterTest.*.

Also removing tsan suppressions for talk/base as it's been moved to
webrtc/base.

R=tommi@webrtc.org
BUG=3671

Review URL: https://webrtc-codereview.appspot.com/22169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6979 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 12:33:18 +00:00
kjellander@webrtc.org
8940ce7112 Updating svn:ignore entries 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6978 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 11:22:54 +00:00
pbos@webrtc.org
b648b9d85c Remove test constructor in WebRtcVideoEngine2. 
Removes the need for ::Construct().

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6977 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 11:08:06 +00:00
bjornv@webrtc.org
4f71e22bf9 Refactoring common_audio/signal_processing: Remove macro WEBRTC_SPL_UDIV 
This macro is a direct use of the division operator without checking for division by zero. Hence, it is dangerous to use.
This CL replaces the macro with '/' at place.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6976 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 10:25:10 +00:00
bjornv@webrtc.org
1de0cc4079 common_audio: Re-enable WebRtcSpl_AddSatW32() and WebRtcSpl_SubSatW32() optimizations on armv7 
According to the issue, common_audio_unittests failed on armv7. It currently pass, so we should turn it on again. There is no print out in the issue, so the cause of failure is unknown.

BUG=740
TESTED=locally on N7
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6975 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 09:36:25 +00:00
pbos@webrtc.org
047a46f8b4 Remove Android.mk build files. 
These files are generally not maintained and break, some contain files
that don't exist anymore and do not build anymore. If we need to add
some of these back we should really set up a bot for them.

R=andrew@webrtc.org, glaznev@webrtc.org, henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/15249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 08:48:51 +00:00
kjellander@webrtc.org
b96ea2aab5 Remove former team members from OWNERS and WATCHLISTS 
Remove the following (CCed) former team members from all
OWNERS files and the WATCHLISTS file:
* fischman@
* leozwang@
* mikhal@
* pwestin@
* wu@

BUG=
R=henrike@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6973 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 06:12:08 +00:00
buildbot@webrtc.org
204cd56007 (Auto)update libjingle 74064646-> 74072040 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6972 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 21:10:18 +00:00
kjellander@webrtc.org
e9bfed0648 Move constant so it is not stripped out for TSAN bots. 
BUG=
R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6971 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 19:46:26 +00:00
buildbot@webrtc.org
857130fd5b (Auto)update libjingle 74039473-> 74044292 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6970 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 16:07:12 +00:00
kjellander@webrtc.org
79ad37eac2 Update root OWNERS file 
Add kjellander to owner for the new way of
syncing Chromium deps.
Remove redundant webrtc_examples.gyp entry.
Convert the file from Win to Unix line endings.

BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6969 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 14:41:41 +00:00
solenberg@webrtc.org
6556a59db1 As expected, r6569 (https://code.google.com/p/webrtc/source/detail?r=6965) caused memcheck bots to complain. Adding expections for that, in line with outher peerconnection tests. 
Also, caused some issues with other peerconnection_unittest tests, so changed the design of those.

BUG=
R=kjellander@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 14:35:40 +00:00
kjellander@webrtc.org
c23923447c Roll chromium_revision 289723:291647 
To pick up recent fixes after the Chromium Git switch.

Relevant changes pulled in by this roll:
* r291168 refactor sanitizer_options (we can now remove some hacks)
* r291647 roll of nss.gyp (its paths work with how we build for iOS).

BUG=2863,3731
R=iannucci@chromium.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6967 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 14:16:32 +00:00
kjellander@webrtc.org
42ee5b54b5 GN: Disable Chromium clang plugins for standalone build. 
Now that WebRTC has rolled the chromium_revision past
http://crrev.com/284372 in r6784, clang has become the
default compiler. Since WebRTC standalone code doesn't
yet compile the Chromium Clang plugins enabled, this CL
disables them for the parts of the code that doesn't yet pass
compilation with them enabled.

The buildbots are using Goma which is not yet switched
over to Clang by default. That's why they're not red yet.

BUG=163
TEST=Passing compile locally on Linux using:
gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja
-C out/Debug
gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja
-C out/Release
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/16279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 14:15:35 +00:00
buildbot@webrtc.org
b4c7b09c13 (Auto)update libjingle 73927775-> 74032598 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6965 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 12:11:58 +00:00
bjornv@webrtc.org
926707b167 Refactoring common_audio: Replace trivial multiplication macro 
This multiplication macro literally use the '*' operator, so there is no need for it.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6964 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 11:42:42 +00:00
bjornv@webrtc.org
d32c4389ac Re-landing r6961 
common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8

This macro is nothing but memcpy() and further used at one single place in webrtc, so it makes no sense to keep it. Replaced the operation where it is used.

BUG=3348,3353
TESTED=locally on linux
TBR=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6963 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 11:19:05 +00:00
bjornv@webrtc.org
4a616be12b Revert 6961 "common_audio/signal_processing: Remove macro WEBRTC..." 
> common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8
> 
> This macro is nothing but memcpy() and further used at one single place in webrtc, so it makes no sense to keep it. Replaced the operation where it is used.
> 
> BUG=3348,3353
> TESTED=locally on linux and trybots
> R=kwiberg@webrtc.org, tina.legrand@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/16359004

TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6962 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 10:32:22 +00:00
bjornv@webrtc.org
4f01017e2d common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8 
This macro is nothing but memcpy() and further used at one single place in webrtc, so it makes no sense to keep it. Replaced the operation where it is used.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6961 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 10:23:22 +00:00
bjornv@webrtc.org
6e71d17bc9 Refactoring common_audio/signal_processing: Replaces trivial macros 
The macros WEBRTC_SPL_ADD_SAT_W16 and WEBRTC_SPL_ADD_SAT_W32 make direct use of the corresponding functions WebRtcSpl_AddSatW16() and WebRtcSpl_AddSatW32().
This CL replaces these macros in the code.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6960 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 07:44:52 +00:00
kwiberg@webrtc.org
584cd8da4b Fix WEBRTC_AEC_DEBUG_DUMP (broken by int16->float conversion) 
And in the process, make it dump WAV files instead of raw PCM.

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

Review URL: https://webrtc-codereview.appspot.com/19089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6959 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 06:26:04 +00:00
buildbot@webrtc.org
3740d74106 (Auto)update libjingle 73927658-> 73927775 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6958 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-22 22:27:04 +00:00
buildbot@webrtc.org
309a611670 (Auto)update libjingle 73891518-> 73927658 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6957 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-22 22:24:54 +00:00
buildbot@webrtc.org
2b0554f0e7 (Auto)update libjingle 73794259-> 73891518 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6955 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-22 14:08:15 +00:00
pbos@webrtc.org
97fdeb8329 Remove static initializer in WebRtcVideoEngine2. 
Blocks import into chromium.

R=tommi@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/18249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6954 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-22 10:36:23 +00:00
kjellander@webrtc.org
374d39b7ae Increment sync_chromium.py version to force re-sync 
This should make the remaining red Windows bots cycle green.
Currently, some of them are in a bad state for the Chromium
checkout.

BUG=webrtc:2863
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6953 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-21 19:51:06 +00:00
iannucci@chromium.org
161363808b Make the last_sync_chromium file a bit more comprehensive. 
Adds a SCRIPT_VERSION and the target_os_list to the flag file content. The
script version is so that we can arbitrarially make all slaves/devs re-sync (in
case we change the implementation but don't want to roll chromium), and the
target_os_list is so that devs who change the target_os_list in their .gclient
file don't mysteriously fail to get the new deps.

R=kjellander@webrtc.org, agable@chromium.org, szager@chromium.org
BUG=2863, chromium:339647

Review URL: https://webrtc-codereview.appspot.com/17189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6952 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-21 15:48:23 +00:00
niklas.enbom@webrtc.org
153c6162d2 Landing issue 15189004 
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6951 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-21 14:49:28 +00:00
phoglund@webrtc.org
7bd5fefb17 Making sure muc members get recorded. 
This is an upstream of a change I made; will describe in a separate
email thread.

Essentially, the members map wasn't getting populated in the callclient
example, so it was always empty. Now it will be populated correctly.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6950 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-21 09:53:28 +00:00
henrik.lundin@webrtc.org
038cee2401 Add send-side bit-exactness test for AudioCoding Module 
This test verifies bit exactness for the send-side of ACM. The test
setup is a chain of three different test classes:

test::AcmSendTest -> AcmSenderBitExactness -> test::AcmReceiveTest

The receiver side is driving the test by requesting new packets from
AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the
packet from test::AcmSendTest::NextPacket, which inserts audio from the
input file until one packet is produced. (The input file loops
indefinitely.)  Before passing the packet to the receiver, the
AcmSenderBitExactness class verifies the packet header and updates a
payload checksum with the new payload. The decoded output from the
receiver is also verified with a (separate) checksum.

The current CL only adds tests for 30 ms and 60 ms iSAC. More codecs
will be added in coming changes.

BUG=3521
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6949 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-21 08:59:14 +00:00
henrik.lundin@webrtc.org
9b8102cf0e Use a deterministic input in NetEqBgnTest 
This test has been failing every now and then. This is likely due to the
random input that was used. With this change, the input is now read from
an audio file, making it identical on each run.

The encoding is moved to inside the main test loop, so that new data is
added with each packet. (Before this change, the same payload was added
over and over again; only the RTP header was updated.)

BUG=3715
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6948 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-21 08:27:44 +00:00
bjornv@webrtc.org
6b2659c660 Refactoring common_audio/signal_processing: Remove unused macro WEBRTC_SPL_MUL_32_32_RSFT32BI 
The WEBRTC_SPL_MUL_32_32_RSFT32BI macro was removed in r6169, since it was unused. This CL removes the arm and mips optimizations of it.

BUG=3348, 3353
TESTED=locally and trybots
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6947 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-21 06:13:57 +00:00
thakis@chromium.org
905f9efbae Fix clang -Wformat warnings. 
An unsigned int was passed through %lu instead of %u (harmless on 32bit).
More seriously, a wide string was passed through %s, which means only the
first byte in the string got printed (since the 2nd byte is likely 0 in
UCS-2). Use %ls to include the whole string, even though it might not be
renderable in the target 8bit buffer.

BUG=chromium:82385
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6946 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-21 02:23:30 +00:00
thakis@chromium.org
add54ad770 Convert nsx_core_neon.S to unified syntax. 
That way, it builds with both gcc and clang's integrated assembler.
No intentional behavior change.

BUG=chromium:124610
R=andrew@webrtc.org, johannkoenig@google.com

Review URL: https://webrtc-codereview.appspot.com/15199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6945 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-21 02:23:26 +00:00
iannucci@chromium.org
286210d3ec Use --gclientfile instead of --spec, because windows is THE WORST. 
--spec contains newlines, which are interpreted as actual newlines in the
command line, which causes gclient to fall apart at the seams.

TBR=agable@chromium.org, kjellander@webrtc.org, szager@chromium.org
BUG=2863, chromium:339647

Review URL: https://webrtc-codereview.appspot.com/22429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6944 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-21 02:14:11 +00:00