Move thread_annotations.h to webrtc/base/.

R=andresp@webrtc.org, mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27579004

git-svn-id: http://webrtc.googlecode.com/svn@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
Thread annotation of rtc::CriticalSection.

Effectively re-lands r5516 which was reverted because talk/-only
checkouts existed. This now resides in webrtc/base/, so no talk/-only
checkouts should be possible.

This change also enables -Wthread-safety for talk/ and fixes a bug in
talk/media/webrtc/webrtcvideoengine2.cc where a guarded variable was
read without taking the corresponding lock.

R=andresp@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27569004

git-svn-id: http://webrtc.googlecode.com/svn@7284 4adac7df-926f-26a2-2b94-8c16560cd09d
WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t

We have to fix both at once, since there's a macro that calls one of
them or the other.

BUG=909
R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7266

Review URL: https://webrtc-codereview.appspot.com/19229004

git-svn-id: http://webrtc.googlecode.com/svn@7285 4adac7df-926f-26a2-2b94-8c16560cd09d
audioproc: Now also writes to output file in simulation mode

After changing to use wav as default file format no output was written in simulation mode.

BUG=3359
TESTED=locally
R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25639004

git-svn-id: http://webrtc.googlecode.com/svn@7286 4adac7df-926f-26a2-2b94-8c16560cd09d
Moved the filter calculation from analyze to process in ns_core

It makes sense to have it there if the analyze and process methods are called in different stages.
Tested over the entire QA set for bit exactness.

BUG=webrtc:3811
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29549004

git-svn-id: http://webrtc.googlecode.com/svn@7287 4adac7df-926f-26a2-2b94-8c16560cd09d
Reduce jitter delay for low fps streams.
Enabled by finch flag.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31389005

git-svn-id: http://webrtc.googlecode.com/svn@7288 4adac7df-926f-26a2-2b94-8c16560cd09d
Call NS AnalyzeCaptureAudio before AEC

This attenuates the noise pumping generated from the NS adapting to the AEC comfort noise.

When there is echo present the AEC suppresses it and adds comfort noise. This is underestimated on purpose to avoid adding more than the original background noise. The NS has to be called after the AEC, because every non-linear processing before it can ruin its performance. Therefore the noise estimation can adapt to this comfort noise, making it less aggressive and generating noise pumping.

By putting the noise estimation analysis stage from the NS before the AEC, this effect can be avoided. This has been tested manually on recordings where noise pumping was present: Two long recordings done in our team by bjornv and kwiberg plus the most noisy (5) recordings in the QA set.

On the other hand, one risk of doing this is to not adapt to the comfort noise and therefore suppress too much. As verified in the tested files, this is not a problem in practice.

BUG=webrtc:3763
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24679004

git-svn-id: http://webrtc.googlecode.com/svn@7289 4adac7df-926f-26a2-2b94-8c16560cd09d