Sound Blaster Live Processor

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I've done some research on the subject and want to build a surround sound processor/preamp. I want to take a sound blaster live card & run a dd or dts signal to it then take the the analog outputs and matrix them together to get 12.1 surround sound. There have been some post on the sound blaster live cards before, but nothing quite like I'm talking about. If there is anybody that has done something like this, has an idea on how to, or is interested please let me hear from you.
 
audioaction said:
Somewhat. Iwould take the rear to channnels and matrix them together for a center rear channel. And do kinda of a delay for the other channels.

What you have here is typical of the 6.1 systems, such as the Dolby Digital EX system or the DTS Neo standard. They also specify sending the matrixed centre surround channel to 2 centre rear speakers to keep it out of the "dead" spot directly behind the listeners' heads.

Where are the other 6 speakers?

:)ensen.
 
Hi AA,

There were a couple of other recent threads that dealt with a similar application and you may want to look them up.

http://www.diyaudio.com/forums/showthread.php?s=&threadid=17907&highlight=soundcard

http://www.diyaudio.com/forums/showthread.php?s=&threadid=17737&highlight=soundcard

http://www.diyaudio.com/forums/showthread.php?s=&threadid=17557&highlight=soundcard

I know the on-board sound of my Gigabyte motherboard can decode AC3 and output 5.1 analog so I imagine that all these basic DSP chips will have similar features. There will likely be some programming, especially if you want to use the card without plugging it into a mobo bus and you will likely also need some kind of memory to store your program.

That said, most consumer sound cards don't have the analog quality to compete with a well done analog 2.0 matrix decoder done. Most of them barely have SNR of 80dB so you may be better off getting a cheap used Yamaha receiver or try this.

http://www.sound.westhost.com/project18.htm

As for trying to mix L and LS to get a left centre channel, you will only get success if two things happen. 1) The mastering engineer intended there to be such a signal and encodes that signal into both the L ans LS channels. 2) You know how that information is encoded, if it's there.

The first situation is highly doubtful as the mastering is (now) usually done under strict guidelines for AC3 or DTS decoding.

http://www.diyaudio.com/forums/showthread.php?s=&threadid=17864

In the second situation, you'll need an option for both summing or difference signal and then be able to boost or attenuate the signal by 3dB.

:)ensen
 
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