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1) What is the reason for two sets of surround speaker outputs on the 3803? I'm not talking about the back surrond/zone2 outputs, but the fact that there are A and B outputs for the surround speakers. Is this for people with an extremely long room who want two sets of side surrounds, like in a movie theater and have 9 speakers set up in a 7.1 configuration?


2) Not a big deal, but are the digital inputs on the 3803 compatible with 88.2? I have a RP-82 that "re-masters" CD's from 44.1 to 88.2 and my current Denon AVR-1700 cannot recognize the 88.2 signal. This is not a big deal I was actually just curious to hear the difference, if any, and to know if modern receivers are compatible with this and if it is just the fact that my receiver is a few years old as to the reason why it doesn't support this frequency?


Thanks
 

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How does one go about wiring the 'B' surrounds if, say I wanted them to be the REARS in a DD/DTS 7.1 mix...AND the Rears in 5 channel music??


Can this even be done?
 

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flynbw01: To my knowledge, I am not aware of any consumer grade CD or DVD player that outputs an 88.1 kHz signal. The standard digital output frequencies are 44.1 kHz, 48 kHz, and 96 kHz. They can be PCM, DD or DTS, depending what type of disc you are playing...
 

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Jeff: I'm not totally sure what you mean but the 3803 has A surrounds, B surrounds and can have a pair of Rear surrounds. These can be configured in several different ways to give you the effect you're looking for. Checkout the Denon manual link above for the best info...
 

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The panasonic rp82 can resample a regular cd to 88.2 khz and output it through it's digital outs. It plays fine in my HK avr525 even though the receiver shows 96khz. Guess they only have 96khz and 192khz icons and 96 is the closer one.
 

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i'm confused why anyone would want to resample existing 44.1 material to 88.2? As I understand it a higher sample rate is only useful for the original sampling or digitizing process. Once the file is in the digital realm and the sample rate has been set, doubling the sample rate does not change the available frequency spectrum of the sound (like when the sound was originally sampled - nyquist theorem etc.). Increasing the sampling rate at playback just takes the existing wave and doubles the information. It seems like this just makes more work for the processor at zero sonic benefit. Am I wrong?
 
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