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Author SHA1 Message Date
fischman@webrtc.org
18f41b8eb4 PRESUBMIT.py: accept variants on the copyright message that are present in the codebase. 
Example files that this makes ok instead of flagging include:
  talk/base/signalthread_unittest.cc
  talk/base/thread_unittest.cc
  webrtc/base/signalthread_unittest.cc
  webrtc/base/thread.cc
  webrtc/base/thread.h
  webrtc/base/thread_unittest.cc

BUG=1027
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19539006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6235 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 17:27:18 +00:00
turaj@webrtc.org
546961a9d3 Avoid reading uninitialized values (outside baundary) in DFT arithmatic decoder of iSAC-fix. 
Arithmetic encoder does not right the last 2 or 3 bytes of |streamval| when terminating the bit-stream. Perhaps the last bytes makes no difference in decoding the stream. However, the decoder reads full |streamval| (int16_t) going out of boundary and reading uninitialized values. This avoids this problem. by inserting zero-bytes whenever decoder intends to read outside boundary.

BUG=1353,chrome373312,b/13468260
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 17:14:29 +00:00
minyue@webrtc.org
aa5ea1c0f9 1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED 
2. Add two new APIs to configure codec internal FEC

3. Add a test and listened to results. This is based modifying EncodeDecodeTest and deriving a new class from it.

New ACM gives good result.
Old ACM does not use NetEq 4, so FEC won't be decoded.

BUG=
R=tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6233 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 15:16:51 +00:00
pbos@webrtc.org
706152dcc9 Fix uninitialized reads in IsDefaultBrowserFirefox 
BUG=
TEST=Local DrMemory.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19529006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6232 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 14:35:48 +00:00
pbos@webrtc.org
1566ee2893 Revert "Revert "Remove VideoSendStreamInput::PutFrame."" 
This reverts commit r6230 to re-land r6229.

ViECapturer::SwapFrame now resets timestamps.

BUG=
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 13:03:45 +00:00
pbos@webrtc.org
2cdd433edf Revert "Remove VideoSendStreamInput::PutFrame." 
This reverts r6229.

Test WebRtcVideoChannel2BaseTest.MuteStream fails after r6229.

BUG=
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6230 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 10:43:26 +00:00
pbos@webrtc.org
f3085e43ab Remove VideoSendStreamInput::PutFrame. 
PutFrame just copies the frame before swapping it, if it's required that
can easily be done outside this API before swapping the frame.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14529006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 09:41:45 +00:00
pbos@webrtc.org
6e98ef4b35 Fix deadlock in RegisterPreDecodeImageCallback. 
Fixes lock-order inversion between ViEChannel::callback_cs_ and
VideoReceiver::_receiveCritSect detected on DrMemory Full which
exhibited different timing behavior.

Also removes most of the suppressions on DrMemory Full as they're able
to run again without deadlocking.

BUG=3336,3375
TEST=Run DrMemory Full trybots.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6228 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 09:41:07 +00:00
pbos@webrtc.org
bc524ae41a Added mirror of gtest-parallel. 
gtest-parallel is a Python script that runs gtest binaries in parallel.

R=andrew@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/11309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6227 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 09:37:29 +00:00
stefan@webrtc.org
b60bfe4759 Suppress webrtc trace races detected by tsan. 
BUG=3372
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 07:29:37 +00:00
wu@webrtc.org
10f871f29b Remove the restriction to allow having both webrtc and talk changes in the same cl. 
This restriction is no longer needed as the auto sync script can handle changes to both folder in same commit correctly.

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

Review URL: https://webrtc-codereview.appspot.com/15479006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6225 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-22 22:35:46 +00:00
tnakamura@webrtc.org
0720758f9f Bump WebRTC version number to 3.54 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6222 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-22 17:51:18 +00:00
henrike@webrtc.org
1bb5da04fe Adds missing include of assert header. 
BUG=3380
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/14569008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6221 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-22 14:31:14 +00:00
braveyao@webrtc.org
21f7d6d2fe WebRTCDemo: move the deletion of CritSect to end of the dtor to fix a crash in Android video renderer. 
BUG=3368
TEST=Manual Test

Review URL: https://webrtc-codereview.appspot.com/21519005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6220 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-22 02:57:55 +00:00
mallinath@webrtc.org
8e755c1ad2 Connect SignalDestroyed in AllocationSequence after TURN ports are destroyed 
when TURN ports are using shared socket with UDP port.

This is required as AllocationSequence maintains a map of turn ports. If the
ports are destroyed without the knowledge of AllocationSequence, sequence will
try to deliver packets to the destoyed ports.

R=jiayl@webrtc.org
BUG=https://code.google.com/p/chromium/issues/detail?id=368877

Review URL: https://webrtc-codereview.appspot.com/14569007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6219 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 23:00:46 +00:00
henrike@webrtc.org
88fbb2d86b Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. 
Same as https://webrtc-codereview.appspot.com/19519004. The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing...
(tested locally).

BUG=3380
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 21:18:46 +00:00
henrike@webrtc.org
99b4162ccf Rebase webrtc/base 6163:6216 (svn diff -r 6163:6216 http://webrtc.googlecode.com/svn/trunk/talk/base, apply diff manually) 
BUG=3379
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6217 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 20:42:17 +00:00
buildbot@webrtc.org
f9f1bfbdae (Auto)update libjingle 67686255-> 67689476 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6216 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 17:02:15 +00:00
henrike@webrtc.org
a148704b4b Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h. 
BUG=3380
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6215 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:52:14 +00:00
buildbot@webrtc.org
ce4201df52 (Auto)update libjingle 67643194-> 67686255 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6214 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:22:51 +00:00
jiayl@webrtc.org
7ca277b574 Initializes WINDOWPLACEMENT::length in GetCroppedWindowRect. 
BUG=https://code.google.com/p/webrtc/issues/detail?id=3196
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6213 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:02:31 +00:00
henrike@webrtc.org
000658a138 Revert of 6211 as it was committed despite of PRESUBMIT.py warning. The commit breaks the sync bot. 
BUG=N/A
TBR=mcasas@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21519006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6212 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:01:13 +00:00
mcasas@webrtc.org
3b7e282caa Disabling systematically failing 
WebRtcVideoMediaChannelTest.SendVp8HdAndReceiveAdaptedVp8Vga

TBR= pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14569006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 14:25:20 +00:00
mcasas@webrtc.org
2fa7f79094 Revert 6202 "Switch to using base/constructormagic.h and remove ..." 
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
> 
> BUG=N/A
> R=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/19519004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 11:07:29 +00:00
mcasas@webrtc.org
c2213b6a0f Revert 6208 "Patch from henrike@webrtc.org" 
Wasn't enough. I'll have to revert the whole rev 6202.

> Patch from henrike@webrtc.org
> https://code.google.com/p/webrtc/source/detail?r=6202
> didn't work for at least one file and broke most of 
> the compile steps in the FYI bots. The file is reverted
> here.
> 
> TBR= henrike@webrtc.org, sergeyu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/17609004

TBR=mcasas@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6209 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 10:03:09 +00:00
mcasas@webrtc.org
86df8acc92 Patch from henrike@webrtc.org 
https://code.google.com/p/webrtc/source/detail?r=6202
didn't work for at least one file and broke most of 
the compile steps in the FYI bots. The file is reverted
here.

TBR= henrike@webrtc.org, sergeyu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6208 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 08:40:56 +00:00
braveyao@webrtc.org
1a79bb8d30 WebRTCDemo: clean the error message due to API clean up and add ability to route the audio through all three outputs, headset/earpiece/loudspeaker 
BUG=3366
TEST=Manual Test
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6207 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 03:37:45 +00:00
buildbot@webrtc.org
49a6a27bf0 (Auto)update libjingle 67555838-> 67643194 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6206 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 00:24:54 +00:00
wu@webrtc.org
82c4b8531c Calculate capture ntp timestamp in local timebase for decoded audio frame. 
BUG=3111
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19449005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6205 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 22:55:01 +00:00
henrik.lundin@webrtc.org
48438c2c90 Enabling NetEq bit-exactness test for Win x64 
A new reference file (neteq4_universal_ref_win_64.pcm) was generated and
uploaded.

Also removing the old hack to have different reference files
for different version of Visual Studio. The test is now only supporting
VS 2012 and later (_MSC_VER >= 1700). This makes the windows 32-bit
output identical to the generic reference file
(neteq4_universal_ref.pcm), so the specialized one
(neteq4_universal_ref_win_32.pcm) could have been removed. However,
since the resources sync mechanism does not include removing of old
files, a client could pick up the old reference and fail. Therefore,
this cl also updates neteq4_universal_ref_win_32.pcm to be identical to
neteq4_universal_ref.pcm.

BUG=1458
R=kjellander@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6204 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 16:07:43 +00:00
henrik.lundin@webrtc.org
aed31fe8ab Modifying WATCHLISTS 
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6203 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 16:05:47 +00:00
henrike@webrtc.org
125ffd709d Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 15:20:44 +00:00
stefan@webrtc.org
4059c2f579 Disable CallPerfTest.CaptureNtpTimeWithNetworkDelay due to being flaky. 
TBR=wu@webrtc.org
BUG=3374

Review URL: https://webrtc-codereview.appspot.com/14579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6201 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 15:12:29 +00:00
stefan@webrtc.org
70bb2d5755 Revert r6198 "Expose the original packet length in in the RTP play tools." 
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6200 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 13:25:48 +00:00
stefan@webrtc.org
83599cba77 Reenable WebRtcVideoEngineTestFake.SendReceiveBitratesStats under DrMemory. 
The uninitialized read has been fixed. Suppressing CL: https://code.google.com/p/webrtc/source/detail?r=6073

BUG=11288120
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6199 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 13:16:35 +00:00
stefan@webrtc.org
e208458643 Expose the original packet length in in the RTP play tools. 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6198 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 13:09:16 +00:00
stefan@webrtc.org
be4ab99a53 Disabling RealFFTTest.RealAndComplexMatch and AudioProcessingTest.Formats as they currently are broken with gcc 4.8. 
BUG=3370
R=bjornv@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6197 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 12:42:01 +00:00
henrik.lundin@webrtc.org
a36db970bd Suppress GMOCK printouts from TestVideoSenderWithVp8 
Adding a missing EXPECT_CALL.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6196 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 11:16:10 +00:00
bjornv@webrtc.org
f3e1341da7 VoEVolumeTest: Enabled Linux flaky tests 
Fixed error checks only on Linux to be able to turn on flaky tests. The cause of flaky failures is due to late values in pulse audio.

Related (deleted) CLs:
https://webrtc-codereview.appspot.com/19469007/
https://webrtc-codereview.appspot.com/19469004/

BUG=367
TESTED=trybots, voe_auto_test repeated
R=henrikg@webrtc.org, tina.legrand@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6195 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 10:43:42 +00:00
asapersson@webrtc.org
a826006132 Add NACK and RPSI packet types to RTCP packet builder. 
Fixes bug found when parsing received RPSI packet.

BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6194 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 09:53:51 +00:00
minyue@webrtc.org
2db9f45038 Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size 
BUG=webrtc:2925

TEST=passed_all_trybots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18479005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6193 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 08:33:30 +00:00
tkchin@webrtc.org
1732a591e7 Add a UIView for rendering a video track. 
RTCEAGLVideoView provides functionality to render a supplied RTCVideoTrack using OpenGLES2.

R=fischman@webrtc.org
BUG=3188

Review URL: https://webrtc-codereview.appspot.com/12489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6192 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 23:26:01 +00:00
tkchin@webrtc.org
7ca1edb31d Remove IOKit linkage from iOS builds. 
IOKit has been removed in iOS7, so link fails. iOS build succeeds after removing this setting and the corresponding one in talk/libjingle.gyp. Presubmit script tells me that CLs aren't allowed to touch both talk/ and webrtc/ at the same time so doing this separately.

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

Review URL: https://webrtc-codereview.appspot.com/20509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6191 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 21:05:10 +00:00
fischman@webrtc.org
40bc7779aa talk_base: remove lock inversion between MessageQueue and MessageQueueManager. 
Removes the concept of a MessageQueue being "active" in favor of considering all
live MQ's to be active.
(previously a MQ was active starting from the first Post to it and stopped being
active in its dtor).

BUG=3230
R=sriniv@google.com

Review URL: https://webrtc-codereview.appspot.com/21489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6190 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 17:58:04 +00:00
wu@webrtc.org
cb711f77d2 Add interface to propagate audio capture timestamp to the renderer. 
BUG=3111
R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 17:39:11 +00:00
pbos@webrtc.org
ebb467fdc8 Avoid NACK-list flush error on keyframe packets. 
Receiver code used to indicate a flush error even if the incoming packet
is a keyframe, forcing a request of a keyframe. Now it takes this
keyframe into account and doesn't error as the stream is decodable from
this point.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15549005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 15:28:02 +00:00
stefan@webrtc.org
64339a7069 Don't crash if a frame returned from the decoder is too old. 
BUG=crbug/371805
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6187 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 13:31:35 +00:00
michaelbai@google.com
725e582461 Use the new gyp_var_prefix local variable set by gyp instead of the 
global GYP_VAR_PREFIX set by the makefiles, since the latter is not
guaranteed to still be the same value at the time the command is
executed. Also, use abspath instead of realpath to convert paths to
absolute, since realpath expands to the empty string if the target file
doesn't exist, complicating build debugging.

BUG=
R=andrew@webrtc.org, torne@chromium.org

Review URL: https://webrtc-codereview.appspot.com/12559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6186 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 17:56:10 +00:00
henrike@webrtc.org
14abcc7322 libvpx's UNUSED macro conflicts with webrtc/base's. Added missing include of assert.h. Globally defined function "Unused" in talk/base and its copy (webrtc/base) is causing a conflict. 
libvpx macro (UNUSED) can be found here:
http://src.chromium.org/viewvc/chrome/trunk/deps/third_party/libvpx/source/libvpx/vpx/vpx_codec.h

BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6185 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 16:54:44 +00:00
bjornv@webrtc.org
a3b5673879 common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16 
This macro was only used on two lines in iSACfix and I replaced those with the operations the macro performed.

BUG=3348
TESTED=trybots, manual unittests
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 12:11:20 +00:00