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Author SHA1 Message Date
jiayl@webrtc.org
bac5f0fb56 Fix an invalid memory access due to typo in win/cursor.cc. 
BUG=crbug/391468
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/19949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 20:32:03 +00:00
tkchin@webrtc.org
122caa51b1 After an audio interruption the audio unit no longer invokes its render callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue. 
CL also replaces deprecated AudioSession calls with equivalent AVAudioSession ones.

BUG=3487
R=glaznev@webrtc.org, noahric@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6697 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 20:20:47 +00:00
tommi@webrtc.org
47218956fc Minor refactoring of StatsCollector. 
* Make GetTimeNow a static method in the cc file.
* Make GetTransportIdFromProxy a static method as well and not a class method.

The second change is in preparation of removing the proxy_to_transport_ member variable which isn't needed and is just a copy from the session stats.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6696 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 19:22:37 +00:00
tkchin@webrtc.org
42fe4350fe Remove Thread::RunningForChannelManager(). 
I haven't heard of this failing, so it should be safe to remove. Let me know if this isn't the case.

BUG=3388
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6695 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 17:52:43 +00:00
stefan@webrtc.org
89fd1e8e99 Improvements to the pacer where it lost some budget due to truncation errors. 
With this CL the resolution is increased to microseconds and proper rounding
is done in the Process() function. This means that we will be allowed to send
more than prior to r6664 as we previously truncated away parts of our budget.

We will also not lose budget due to inaccurate calculations in
TimeUntilNextProcess(), which was a regression in r6664.

BUG=cr/393950
TEST=out/Debug/webrtc_perf_tests --gtest_filter=RampUpTest.Simulcast
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6694 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 16:40:38 +00:00
pbos@webrtc.org
376b4ea93f Fix breakage introduced by r6691. 
ModuleRtpRtcpImpl returned incorrectly on RemoteNTP as the
RTCPReceiver::NTP changed return type.

BUG=
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6693 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 15:51:33 +00:00
pbos@webrtc.org
2f4b14e3f3 Make RTCP sender report send media bytes. 
r6654 changed RtpSender::Bytes() to return the number of bytes sent
instead of number of media bytes. This is used by VideoEngine for stats.
This change broke RTCP which sends this same count as the number of
payload bytes sent (excluding headers and padding).

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6691 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 15:25:39 +00:00
kwiberg@webrtc.org
ffa8dcab1e Eliminate unnecessary #include 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6690 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 12:50:13 +00:00
kwiberg@webrtc.org
324f63ca38 rtc::Fatal output: Print space between # and message 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6689 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 11:41:05 +00:00
pbos@webrtc.org
bc73871251 Remove the VPM denoiser. 
The VPM denoiser give bad results, is slow and has not been used in
practice. Instead we use the VP8 denoiser. Testing this denoiser takes
up a lot of runtime on linux_memcheck (about 4 minutes) which we can do
without.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6688 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 09:50:40 +00:00
tommi@webrtc.org
2adc51c86e Handle the case if an unusually long peer name is provided in the peerconnection example. 
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6687 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 08:56:07 +00:00
pbos@webrtc.org
cb859ecd3b Replace strcpy with talk_base::strcpyn. 
Cpplint reports error 'Almost always, snprintf is better than strcpy'
when checking code styles. The function talk_base::strcpyn() is a better
option than strcpy().

BUG=1788
R=pbos@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12919004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6686 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 08:28:20 +00:00
fbarchard@google.com
6823479ad3 Roll libyuv from 1033 to 1035 to get cpuid fix for AVX2 that avoids misdetect causing a crash in AVX2 code on cpus that do not have AVX2. 
BUG=libyuv:343
TESTED=libyuv try bots pass
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6685 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 23:27:05 +00:00
fgalligan@google.com
d873540101 Roll chromium 282462:282879. 
Pick up the libvpx roll:
https://codereview.chromium.org/387003005/

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6684 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 23:14:48 +00:00
henrike@webrtc.org
92a9bacf9a Rebase webrtc/base with r6682 version of talk/base: 
cls ported: r6671, r6672, r6679 (reverts and unreverts in r6680, r6682).
svn diff -r 6656:6682 http://webrtc.googlecode.com/svn/trunk/talk/base >
6682.diff
sed -i.bak "s/talk_base/rtc/g" 6682.diff
sed -i.bak "s/#ifdef WIN32/#if defined(WEBRTC_WIN)/g" 6682.diff
sed -i.bak "s/#if defined(WIN32)/#if defined(WEBRTC_WIN)/g" 6682.diff
patch -p0 -i 6682.diff

BUG=3379
TBR=tommi@webrtc.org,jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6683 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 22:03:57 +00:00
henrike@webrtc.org
1b84116417 Add a facility to the Thread class to catch blocking regressions. 
This facility should be used in methods that run on known threads
(e.g. signaling, worker) and do not have blocking thread syncronization
operations via the Thread class such as Invoke, Sleep, etc.

This is a reland of an already reviewed cl (r6679) that got reverted by mistake.

TBR=xians@google.com,tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6682 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 21:42:39 +00:00
tkchin@webrtc.org
b038c72369 Enable SCTP compile for iOS. 
Chromium's been updated to pull a version of usrsctplib that will compile correctly. This update DEPS to point at new revision and turn on the compile time flags for iOS sctp.

BUG=3211
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6681 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 20:24:09 +00:00
buildbot@webrtc.org
aac14973aa (Auto)update libjingle 71116846-> 71117224 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6680 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 20:22:21 +00:00
tommi@webrtc.org
5be649fcfc Add a facility to the Thread class to catch blocking regressions. 
This facility should be used in methods that run on known threads
(e.g. signaling, worker) and do not have blocking thread syncronization
operations via the Thread class such as Invoke, Sleep, etc.

This is a reland of an already reviewed cl that got reverted by mistake.

TBR=xians@google.com

Review URL: https://webrtc-codereview.appspot.com/12999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6679 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 20:21:36 +00:00
tommi@webrtc.org
242068d58c A step towards changing StatsReport::Value::name to an enum. 
The stats reporting code does a lot of unnecessary string copying.
This is a step in the direction of removing that and forcing use of only known constants.

This is a reland of an already reviewed cl that got reverted by mistake.

TBR=xians@google.com

Review URL: https://webrtc-codereview.appspot.com/12989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6678 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 20:19:56 +00:00
tommi@webrtc.org
03505bcb7a Make StatsCollector depend on always having a valid session pointer. 
This is required since the session pointer is currently used on multiple threads but there's no synchronization code to guard it.
I'm removing the set_session() method and session() getter since they would cause problems if used without synchronization.

This is a reland of an already reviewed cl that got reverted by mistake.

TBR=xians@google.com

Review URL: https://webrtc-codereview.appspot.com/13959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6677 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 20:15:26 +00:00
tommi@webrtc.org
b5348c64bb Minor refactoring of the session classes. 
Make member variables that never change and are touched on multiple threads, const.
Move implementations of setters/getters of variables that can change, into the cc file in preparation of adding thread correctness checks.

This is a relanding of a cl already reviewed but got reverted by mistake.

TBR=xians@google.com

Review URL: https://webrtc-codereview.appspot.com/12979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6676 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 20:11:49 +00:00
buildbot@webrtc.org
d8524348bb (Auto)update libjingle 71107853-> 71115715 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6675 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 20:05:09 +00:00
buildbot@webrtc.org
b92f6f9371 (Auto)update libjingle 71099685-> 71107853 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 18:22:37 +00:00
glaznev@webrtc.org
a4da771914 Fix deadlock in Android stopCapture() call. 
BUG=3467
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6673 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 17:01:53 +00:00
jiayl@webrtc.org
5f43ce6784 Fix a type cast issue for compiling webrtc with BoringSSL. 
BUG=
R=juberti@google.com, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6672 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 16:42:46 +00:00
buildbot@webrtc.org
e04cb0eb81 (Auto)update libjingle 70948025-> 70959275 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6671 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 14:54:16 +00:00
kjellander@webrtc.org
9bef551ba1 GN: Fix include paths for WebRTC in Chromium build. 
Most WebRTC source files are using full paths for includes which
requires the root to be in the include path.

This is currently handled in the common_inherited_config config in
webrtc/BUILD.gn: the .. include_dir.

However, when built from Chromium, the include
paths are not inherited in the same way when building the all target.
Building the 'webrtc' target of Chrome works without the changes
in this CL, but the default target fails.

BUG=3441
TEST=Built the default target from a Chromium checkout with
https://codereview.chromium.org/321313006/ applied and
src/third_party/webrtc linked to the webrtc folder of the WebRTC
workspace.

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/15989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6670 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-13 09:02:54 +00:00
tommi@webrtc.org
9e1acc8728 Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 . 
A few places were relying on temporalIdx being signed. Fix to explicitly check
for kNoTemporalIdx.

TBR=pbos,stefan

Review URL: https://webrtc-codereview.appspot.com/13939005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6669 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 20:33:39 +00:00
tommi@webrtc.org
dd6780d85d Remove always-true expression. 
TBR=pbos

Review URL: https://webrtc-codereview.appspot.com/16059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 19:34:54 +00:00
tommi@webrtc.org
eec6ecdb1e Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. 
---

Fixes for re-enabling more MSVC level 4 warnings: webrtc/ edition

This contains fixes for the following sorts of issues:
* Possibly-uninitialized local variable
* Signedness mismatch
* Assignment inside conditional

This also contains a small number of other cleanups to nearby code. In
particular several warning-disables for MSVC are removed because they don't seem
to be necessary (either that warning is not enabled or the code does not trigger
it).

BUG=crbug.com/81439
TEST=none
R=henrika@webrtc.org, pkasting@chromium.org

Review URL: https://webrtc-codereview.appspot.com/18769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6667 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 19:09:59 +00:00
pbos@webrtc.org
180e516bef Thread annotate RTCPSender. 
Also fixes data races in RTCPSender::SetCSRCStatus() and
RTCPSender::SetStartTimestamp().

BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6666 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 15:36:26 +00:00
pbos@webrtc.org
336e8e8f50 Fixing memcheck leak suppressions for XMPPClient tests. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6665 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 13:44:45 +00:00
stefan@webrtc.org
168f23faa5 Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. 
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.

R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21869005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 13:44:02 +00:00
pbos@webrtc.org
ccbed3b3c4 Implement unittest SetRecvCodecsAcceptDefaultCodecs. 
BUG=1788
R=pbos@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14869004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 13:02:54 +00:00
pbos@webrtc.org
a1bfcad3a3 Cast payload types to int for logging. 
uint8_t gets interpreted as char and printed as such, instead of being
printed in decimal, casting them to int allows us to read what payload
types are actually used without converting them from ASCII first.

BUG=chromium:390874
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6662 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 12:33:45 +00:00
aluebs@webrtc.org
fb2e7c22a0 Document that channels are stored contiguously in AudioBuffer 
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6661 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 11:40:48 +00:00
tommi@webrtc.org
d212ffcfc6 Remove unnecessary build message. 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 11:15:35 +00:00
stefan@webrtc.org
4ef438e2de Remove the send-side cname getter APIs from voice and video engine. 
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.

R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 09:55:30 +00:00
henrikg@webrtc.org
0f426685e1 Roll chromium_revision 280876:282462 
No significant DEPS changes in this roll, only some changes in how clang_format is downloaded.

clang_format changes based on https://webrtc-codereview.appspot.com/20829004 which was reverted.

R=henrika@webrtc.org
TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6658 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 08:10:19 +00:00
fbarchard@google.com
cb973686e8 roll libyuv to r1033 for clang-cl support on windows. 
BUG=chromium:391927
TESTED=manual testing libyuv compiles with clang-cl
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6657 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 23:40:15 +00:00
henrike@webrtc.org
b614d0626f Rebase webrtc/base with r6655 version of talk/base: 
cls to port: r6633,r6639 (there is no cl in between that affects base and all other talk/base cls took care of webrtc/base as well (see r6569, r6624)):
svn diff -r 6632:6639 http://webrtc.googlecode.com/svn/trunk/talk/base > 6655.diff
sed -i.bak "s/talk_base/rtc/g" 6655.diff
patch -p0 -i 6555.diff

BUG=3379
TBR=tommi@webrtc.org,jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 22:47:02 +00:00
pbos@webrtc.org
72491b9a90 Count total bytes sent in RTPSender::Bytes(). 
Previously only media bytes were included, this adds header bytes and
padding bytes to the calculation.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 16:24:54 +00:00
pbos@webrtc.org
0422100818 Fix data race in VCMTiming::ResetDecodeTime. 
Also thread annotating class.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6653 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 15:25:37 +00:00
pbos@webrtc.org
bd9c0920ec Skip encoding in fake VP8 encoder. 
Broke memcheck, FakeEncoder::Encode doesn't produce valid VP8 frames.

BUG=3424
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6652 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 13:21:40 +00:00
andresp@webrtc.org
7ae9108b60 Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6651 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 10:35:12 +00:00
pbos@webrtc.org
91f1752f2d Support VP8 encoder settings in VideoSendStream. 
Stop-gap solution to support VP8 codec settings in the new API until
encoder settings can be passed on to the VideoEncoder without requiring
explicit support for the codec.

BUG=3424
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6650 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 10:13:37 +00:00
andresp@webrtc.org
8f1512140e Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 09:39:23 +00:00
bjornv@webrtc.org
5bde66e913 audio_processing: Updates aec_core_sse2.c with changes made to aec_common.h 
The change of definitions moved to aec_common.h was done in CL17839005.

BUG=3131
TBR=kwiberg@webrtc.org
TESTED=builds locally

Review URL: https://webrtc-codereview.appspot.com/16859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6648 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 08:09:50 +00:00
bjornv@webrtc.org
555fc78f27 Neon version of SubbandCoherence() 
The performance gain on a Nexus 7 reported by audioproc is ~1.4%

The output is NOT bit exact.  Any difference seen is +-1.

BUG=3131
R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17839005

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6647 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 08:03:11 +00:00