TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH.

BUG=3570
R=juberti@webrtc.org, mallinath@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7070

Review URL: https://webrtc-codereview.appspot.com/20999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7120 4adac7df-926f-26a2-2b94-8c16560cd09d
ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that.
BUG=3789
TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate*
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7121 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix the RTC+Chromium GN build.

LOGGING_INSIDE_WEBRTC was being set in the inherited config, whereas in the GYP build this define is not inherited. This caused duplicate logging macros to be defined in Chrome files dependening on WebRTC targets.

Move LOGGING_INSIDE_WEBRTC to the common config (non-inherited).

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7122 4adac7df-926f-26a2-2b94-8c16560cd09d
Change explicit static cast from int to uint16_t to implicit cast of 0u.
BUG=3663
TESTED=local windows build with VS2013.
R=harryjin@google.com, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7123 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixes two issues in how we handle OfferToReceiveX for CreateOffer:
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.

2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.

BUG=2108
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7068

Review URL: https://webrtc-codereview.appspot.com/16309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7124 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix up configs applying to GN build.

The audio_processing target didn't have the build configs applying to it which led to some logging errors.

TBR=kjellander

Review URL: https://webrtc-codereview.appspot.com/22339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7125 4adac7df-926f-26a2-2b94-8c16560cd09d
Add a new class InterfaceAddress inherited from IPAddress to keep track of IPv6 Address flags.

Skeleton put in place in Network::GetFilterIPs() which will be used to
filter addresses

BUG=3773
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7126 4adac7df-926f-26a2-2b94-8c16560cd09d
include cstdlib for free() and abort()

This previous CL added uses of free() and abort() without including cstdlib:
https://webrtc-codereview.appspot.com/22449004

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23559004

Patch from Mostyn Bramley-Moore <mostynb@opera.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7127 4adac7df-926f-26a2-2b94-8c16560cd09d