Adding SSIM and PSNR videoFrame based functions

Review URL: https://webrtc-codereview.appspot.com/867005

git-svn-id: http://webrtc.googlecode.com/svn/trunk/src@2871 4adac7df-926f-26a2-2b94-8c16560cd09d
Moved calling points of spl_int routines from module init to module create routines.
Review URL: https://webrtc-codereview.appspot.com/789006

git-svn-id: http://webrtc.googlecode.com/svn/trunk/src@2872 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixed a bug in an Android macro definition in SPL that caused issue 833.
Review URL: https://webrtc-codereview.appspot.com/863007

git-svn-id: http://webrtc.googlecode.com/svn/trunk/src@2873 4adac7df-926f-26a2-2b94-8c16560cd09d
Return bw_estimate / num-streams from the multi-stream estimator for now until updating ViE APIs.

BUG=

Review URL: https://webrtc-codereview.appspot.com/842006

git-svn-id: http://webrtc.googlecode.com/svn/trunk/src@2874 4adac7df-926f-26a2-2b94-8c16560cd09d
Added test for NetEQ stereo with no input

This test verifies that NetEQ behaves as it should when using
stereo and not inserting any data before calling recout. The
main purpose of the test is to find memory issues.

BUG=http://code.google.com/p/webrtc/issues/detail?id=270
TEST=trybots, neteq_unittests

Review URL: https://webrtc-codereview.appspot.com/863010

git-svn-id: http://webrtc.googlecode.com/svn/trunk/src@2875 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix dummy RTP playout in NetEqRTPplay

Make sure that playing dummy RTP files works, i.e., RTP dumps with only
headers, no actual payloads. It turs out that some RTP dumps did not
state how long the actual packet was (including the discarded payload),
which resulted in packets with zero length payload being inserted into
NetEQ. Now, the dummy packet will contain at least one byte of payload
(set to zero).

Also added capability to filter out a specific SSRC from the RTP
dump file.

Finally, added SSRC printout from RTPanalyze, and corresponding
change in RTPchange.

TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/857007

git-svn-id: http://webrtc.googlecode.com/svn/trunk/src@2876 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixing LatestEstimate().

TBR=henrik.lundin@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/865005

git-svn-id: http://webrtc.googlecode.com/svn/trunk/src@2877 4adac7df-926f-26a2-2b94-8c16560cd09d
Continuing to rewrite custom calls.

Added new helper for getting regular input. Rewrote remaining input handling for custom calls. Added a --choose_defaults flag which makes it possible to default on everything (e.g. with the flag, choosing custom call will accept all defaults and go directly to the call).

The next patch will add support for overriding arbitrary choices using flags. That is the point I want to arrive at and this patch paves the way for that. Fortunately it gets rid of some repetitive and bug-prone code on the way.

BUG=

Review URL: https://webrtc-codereview.appspot.com/858005

git-svn-id: http://webrtc.googlecode.com/svn/trunk/src@2878 4adac7df-926f-26a2-2b94-8c16560cd09d
Makes it such that calling ThreadWrapper::Start(..) only returns false if a thread was never started. I.e. it will not return false if it was unable to configure the thread (however it does log such failures).

Review URL: https://webrtc-codereview.appspot.com/869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/src@2879 4adac7df-926f-26a2-2b94-8c16560cd09d
Hooking up EncoderStateFeedback to handle intra requests instead of trigger
ViEEncoder directly. This is one step towards adding send- and receive only
channels and getting rid of the default module.

Patch set 1 contains the reverted CL occasionally dead-locking:
http://review.webrtc.org/824004

BUG=769

Review URL: https://webrtc-codereview.appspot.com/859007

git-svn-id: http://webrtc.googlecode.com/svn/trunk/src@2880 4adac7df-926f-26a2-2b94-8c16560cd09d
Updating webrtc to r2880