Improving error message from neteq_rtpplay

If a packet with unknown RTP payload type is inserted, this CL
will make sure that the error message is a little more detailed
and gives a better understadning of what to do.

BUG=2692
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7603 4adac7df-926f-26a2-2b94-8c16560cd09d
Enables AIMD control by default.

BUG=1788
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7604 4adac7df-926f-26a2-2b94-8c16560cd09d
Prepare for removal of PeerConnectionObserver::OnError.
Prepare for removal of constraints to PeerConnection::AddStream.

OnError has never been implemented and has been removed from the spec.
Also, constraints to PeerConnection::AddStream has also been removed from the spec and have never been implemented.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7605 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove the codec_type_ member from AudioDecoder

It isn't actually required, as evidenced by the comparative ease with
which it can be removed.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7606 4adac7df-926f-26a2-2b94-8c16560cd09d
Change the PeerConnection proxy templates to use blocking method calls instead of using Thread::Send.
The problem with Thread::Send is that it processes incoming pending messages and for the proxies,
this can mean that multiple incoming calls can concurrently run on the same thread, resulting in unexpected behavior.

See e.g. crbug.com/429740 (and more)

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7607 4adac7df-926f-26a2-2b94-8c16560cd09d
ReAdd PeerConnectionInterface::AddStream to fix Chrome build.
AddStream(MediaStreamInterface* stream, const MediaConstraintsInterface* constraints);
This will be removed once Chrome has been updated.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7608 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove the useless dummy state parameter to WebRtcG711_*

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7609 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7610 4adac7df-926f-26a2-2b94-8c16560cd09d