Check on the existence of report directory

Reports will be written at rtcBot/test/reports/<report_name>

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26689004

git-svn-id: http://webrtc.googlecode.com/svn@7372 4adac7df-926f-26a2-2b94-8c16560cd09d
talk: removes empty directories base and sound.

git-svn-id: http://webrtc.googlecode.com/svn@7373 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixing build issue with L-sdk

Based on https://codereview.appspot.com/153000043/

BUG=https://code.google.com/p/chromium/issues/detail?id=420293
R=niklas.enbom@webrtc.org, serya@chromium.org, yfriedman@chromium.org

Review URL: https://webrtc-codereview.appspot.com/29659004

git-svn-id: http://webrtc.googlecode.com/svn@7374 4adac7df-926f-26a2-2b94-8c16560cd09d
Merge webrtc r7301 to M39 branch. 

TBR=juberti@webrtc.org 

BUG=414211

Review URL: https://webrtc-codereview.appspot.com/29679004

git-svn-id: http://webrtc.googlecode.com/svn@7375 4adac7df-926f-26a2-2b94-8c16560cd09d
Merge webrtc r7310 to M39.

TBR=pbos@webrtc.org
BUG=420753

Review URL: https://webrtc-codereview.appspot.com/29689004

git-svn-id: http://webrtc.googlecode.com/svn@7376 4adac7df-926f-26a2-2b94-8c16560cd09d
Mac: adds missing _DEBUG flag to mac debug builds.

BUG=3836
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25769004

git-svn-id: http://webrtc.googlecode.com/svn@7377 4adac7df-926f-26a2-2b94-8c16560cd09d
Only configure the SSL library in one place.

Build settings now use use_openssl in both Chromium and standalone builds. It
moves all the platform-specific SSL-related build checks to be conditioned on
this flag as appropriate.

This is to avoid colliding with Chromium's transition away from NSS.

BUG=chromium:413497
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29559004

git-svn-id: http://webrtc.googlecode.com/svn@7378 4adac7df-926f-26a2-2b94-8c16560cd09d
Prevent reading outside iSAC bitstream, if the stream is corrupted.

BUG=chrome_373312(#24)
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20089004

git-svn-id: http://webrtc.googlecode.com/svn@7379 4adac7df-926f-26a2-2b94-8c16560cd09d
Add an SSRC filter to neteq_rtpplay

This allows the user to set the desired SSRC if the input file
contains multiple streams.

BUG=2692
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30609004

git-svn-id: http://webrtc.googlecode.com/svn@7380 4adac7df-926f-26a2-2b94-8c16560cd09d
Set NetEq playout mode through the Config struct

This change opens up the possibility to set the playout mode when
creating the NetEq object. The old methods SetPlayoutMode and
PlayoutMode are still available, but are deprecated.

BUG=3520
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23869004

git-svn-id: http://webrtc.googlecode.com/svn@7381 4adac7df-926f-26a2-2b94-8c16560cd09d