Path: utzoo!utgpu!jarvis.csri.toronto.edu!mailrus!cornell!uw-beaver!zephyr.ens.tek.com!tektronix!reed!chaffee
From: chaffee@reed.UUCP (Alex Chaffee)
Newsgroups: comp.sys.mac.programmer
Subject: Re: System 7.0 sound
Message-ID: <14075@reed.UUCP>
Date: 6 Feb 90 05:05:44 GMT
References: <1629@ndmath.UUCP> <2850@draken.nada.kth.se>
Reply-To: chaffee@reed.UUCP (Alex Chaffee)
Organization: Reed College, Portland OR
Lines: 28

In article <2850@draken.nada.kth.se> d88-jwa@nada.kth.se (Jon W{tte) writes:
|There are two ways of doing twin sampled sound channels.
|Both use several short buffers, like 0.1s each.
|
|Way 1: Quick & dirty
...
|Way 2: Scientific & Cycle-stealing
|
|Sample the sounds at 11kHz, play them zero-padded at 22kHz:
...

|would become
|
|2000 3000 4000 5000 A000 B000 C000 8000
|
|which you would then digitally cut-filter at 11KHz and regain the
|previous signal (note: real-time filtering !) This is the method
|used by over-sampling CD players, by the way.

What a great idea.  But what do you mean by "digitally cut-filter?"  Would
you just divide each sample by 2?  And since it should be "real-time," do
you think this could be implemented as a modifier?

-- 
Alex Chaffee
chaffee@reed.UUCP
Reed College, Portland OR 97202
____________________