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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
ac800c8004
Neon version of rftbsub_128()
...
The performance gain on a Nexus 7 reported by audioproc is ~4.5%
The output is bit exact.
BUG=3131
TESTED=trybots and manually
R=bjornv@webrtc.org , cd@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19919005
Patch from Scott LaVarnway <slavarnw@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6646 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 07:53:13 +00:00
5ac876bae0
Revert "Remove remains of WEBRTC_NO_STL." (rev 6641).
...
Reason breaks linux_memcheck.
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6645 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 07:41:59 +00:00
47d1c98a4e
Remove remains of WEBRTC_NO_STL.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6641 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 20:18:28 +00:00
10ef8fe611
Create FullScreenChromeWindowDetector in DesktopConfigurationOptions::CreateDefault.
...
BUG=crbug/385294
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/18759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6640 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 19:41:32 +00:00
b753762ce6
delay_estimator: Increases test coverage and makes input spectrum const
...
Noticed lack in tests verifying initial state is not left if we have zero input spectra. This CL adds such a test and change input spectra to const at affected places.
BUG=N/A
TESTED=trybots and manually
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6631 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 06:40:09 +00:00
12b4efefdd
Implement a work around for Chrome full-screen tab switch on Mac.
...
Chrome creates a new window in full-screen and minimizes the old window when a tab is switched to full-screen.
We try to find the new window to continue capturing for window sharing.
BUG=crbug/385294
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/19839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6629 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 22:05:24 +00:00
e55641d4f7
Neon version of rftfsub_128()
...
The performance gain on a Nexus 7 reported by audioproc is ~3.3%
The output is bit exact.
BUG=3131
TESTED=trybots and manually on N7
R=bjornv@webrtc.org , cd@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14819004
Patch from Scott LaVarnway <slavarnw@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6628 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 21:12:23 +00:00
d11bec40b2
Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6626 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 14:32:58 +00:00
62bafae661
Some refactoring inside rtp_rtcp/.
...
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.
BUG=
R=stefan@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 12:10:51 +00:00
241a9b0b65
Fixing compile error.
...
Made a mistake in https://webrtc-codereview.appspot.com/13849004/ ,
fixing that here.
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6622 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 11:48:37 +00:00
22292df53b
Adding explicit check for using dummy file devices.
...
Calling into the file device factory without being compiled with file
devices makes no sense and would cause hard-to-debug errors. Therefore
I'm adding an explicit check so this isn't allowed.
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6621 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 11:39:19 +00:00
2bb1bdab8d
Preserve RTP states for restarted VideoSendStreams.
...
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.
Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 13:06:48 +00:00
73823cafa4
Add initial gn build files for video_coding and video_processing.
...
BUG=
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
03c817e405
Fix pacer to accept duplicate sequence numbers on different SSRCs.
...
BUG=3550
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6610 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 10:20:35 +00:00
7832648824
Add missing break introduced in r6603.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6607 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 17:04:00 +00:00
bee164a214
Fix test issues and a win compile error introduced with r6605.
...
Also changes the name of a variable which has been hijacked by windef.h (included by windows.h), which forces #define near and #define far upon us. This issue was introduced via the following inclusion chain:
bwe_test_framework_unittest.cc includes
paced_sender.h
tick_util.h
windows.h
windef.h
And causes EXPECT_NEAR(foo, bar, near); to expand to EXPECT_NEAR(foo, bar,); generating a very confusing compile error.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6606 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 17:00:06 +00:00
875ad49dee
Revert conversion from TickTime to int64_t in paced sender.
...
Introduced with r6600, causing flakes in SuspendBelowMinBitrate. The reason for this flake is currently unknown.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6605 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 15:27:55 +00:00
b9f5453e29
Add boilerplate code for H.264.
...
R=mflodman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17849005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 12:42:07 +00:00
20c1f56992
Configure RTX send status on new modules.
...
Fixes bug where newly-allocated modules wouldn't send payload-based
padding (or probably not send over RTX at all).
As the newly-added test exposed lock-inversions shown on tsan in
VideoReceiver, VideoReceiver was thread-annotated and locks taken less.
BUG=chromium:391085
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6601 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 10:58:12 +00:00
88e0dda475
Introduces PacedVideoSender to test framework and moves the Pacer to use Clock.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6600 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 09:20:42 +00:00
720964faac
Fix memcheck error in r6594.
...
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6596 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-03 21:14:07 +00:00
c8364539d3
Fix for FEC decoding with sequence number wrap-around.
...
BUG=3507
R=stefan@webrtc.org
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6594 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-03 16:49:30 +00:00
69ef9911e4
delay_estimator: Allows dynamically used history sizes
...
Gives the user a possibility to dynamically change the history size. The main advantage is, for example, that you now can start with a wide delay range and over time decrease the search window to lower complexity.
Adds
- two new APIs.
- and updates unit tests.
- a history_size member variable to BinaryDelayEstimator.
- two help function re-allocating buffer memory.
One thing that makes this a little complicated is that you are allowed to have multiple delay estimators with the same reference, so changing the buffer sizes at one place will automatically give you a mismatch at other places.
BUG=3532, 3504
TESTED=trybots and manually
R=aluebs@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6593 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-03 14:59:03 +00:00
224a140339
Make experimental NS API not purely virtual
...
Because not all subclasses will want to bother overriding these methods.
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6592 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-03 13:41:39 +00:00
c0ba4392f1
common_audio: Removes macro WEBRTC_SPL_SHIFT_W16
...
We should avoid macros in general (see style guide). This shift macro is not a severe one, since there is a check for negativity.
BUG=3348,3353
TESTED=trybots and manually
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6591 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-03 13:38:53 +00:00
38214d53db
EchoCancellationImpl::ProcessRenderAudio: Use float samples directly
...
This patch lets EchoCancellationImpl::ProcessRenderAudio ask the given
AudioBuffer for float sample data directly, instead of asking for
int16 samples and then converting manually.
Since EchoCancellationImpl::ProcessRenderAudio takes a const
AudioBuffer*, it was necessary to add some const accessors for float
data to AudioBuffer.
R=aluebs@webrtc.org , andrew@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6590 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-03 09:47:33 +00:00
93426cd2ff
Implement BUILD.gn for desktop_capture.
...
BUG=3441
R=brettw@chromium.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6584 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-02 15:47:12 +00:00
74bf7a6523
Add tkchin@ to OWNERS.
...
Adding myself to OWNERS of subdirectories containing iOS bits. Added niklas.enbom@ for audio_device and wu@ for everything else.
R=niklas.enbom@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6578 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01 18:38:28 +00:00
9138eb649b
Fix compile error introduced with r6571.
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TBR=mflodman@webrtc.org
BUG=3527
Review URL: https://webrtc-codereview.appspot.com/20799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6572 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01 12:44:05 +00:00
5779ca478d
Fixes a potential BWE clock mismatch bug.
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Since libjingle provides a packet arrival timestamp to webrtc, and the clock in remote bitrate estimator and the clock used for packet arrival timestamp can be different. This can cause the bandwidth estimator to malfunction.
This CL changes the remote bitrate estimator so that packet arrival timestamps never are compared to the time taken from the internal clock.
BUG=3527
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6571 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01 12:14:49 +00:00
6d21ddca5f
audio_processing/aec: Refactors NonLinearProcessing to prepare for NEON optimizations
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Puts functionality necessary to calculate sub-band coherences into a function.
BUG=3131
TESTED=trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6570 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01 10:03:42 +00:00
