Major Audio Regression

Hi Adobe,
First of all, I am seeing some great progress from you with your p2p H264 video calling which is set to change the world.
You have fixed the repeated frame issues in the 11.2 Beta 2 which is great.
Unfortunately, you have broken the audio which sounds slow and bubbly.
You can easily test a Flash H264 p2p call with our test page
http://pim-uat-06.requestec.com/rtmfp/p2p/fp11_2/
Press Connect on each side. Then have one side enter the peer id of the other and press Place Call.
I have put the source here for you all
http://pim-uat-06.requestec.com/rtmfp/p2p/fp11_2/src.zip
Please let me know if I can be of any further assistance.
I believe you send key frames every second to avoid the need for key frame requests between endpoints.
This does not sound like the most efficient way of doing things and will result in lower frame rate for a fixed bandwidth.
I would also love to know what you are doing for rate control. It seems you do have some rate control taking place but setting the camera bit rate too high one some connections will lead to massive quality issues.
Therefore I am considering analysing the various stats (packet loss and frame latency) in an attempt to tell the encoding endpoint to SLOW DOWN.
Ben Weekes
Technical Director, Requestec

'Slow and bubbly' is a great way to describe precisely what we're hearing too with 11.2beta. We're only focusing on voice & audio at the moment, but this version is virtually unusable on our hardware.
Tested on netbook, 1.6GHz Atom, 2GB RAM.

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