rlsg7229@uxa.cso.uiuc.edu (Robert Lewis Spence) (05/31/91)
I am currently working on an application that will take audio information directly into the computer and process it in real time. What this means is that instead of running a program that records a length of sound and stores it in memory, I need to pull in a single value and do something with it, then get the next value, etc. I'm therefore trying to use the Macrecorder as a simple A/D converter a sample at a time. I'm not the most advanced programmer, so I have some questions: 1) I know Farrallon released some public domain source code for the Macrecorder (Available from SUMEX) but I have been having problems getting it to work. Is that because I have the latest version of the recorder? The code only pulls in about half a second of sound no matter how long I wait for it the record. The recorder works fine with the software Farrallon sent with it..] 2) Are there tech notes available on snd formats, interfacing with the modem port, or anything to do with Macrecorder? How about books or magazine articles (Such as MacUser or MacWorld)? 3) What system tools are available that will enable me to directly access timing circuitry with a higher resolution then the standard 1 sec. Mac time routines? I need these so I can sample at standard intervals, so I need a clock with resolution to the jiffy or better. Thanks in advance, Rob
paulr@syma.sussex.ac.uk (Paul Russell) (06/07/91)
From article <1991May31.151435.10888@ux1.cso.uiuc.edu>, by rlsg7229@uxa.cso.uiuc.edu (Robert Lewis Spence): > I am currently working on an application that will take audio information > directly into the computer and process it in real time. What this means is > that instead of running a program that records a length of sound and stores > it in memory, I need to pull in a single value and do something with it, > then get the next value, etc. I'm therefore trying to use the Macrecorder > as a simple A/D converter a sample at a time. Not wanting to dampen your fireworks, but you won't be able to do much processing on your samples. At 22 kHz you get around 45 microseconds maximum per sample to do any real-time processing. Even hand-coded assembler isn't going to get very much done. You could probably do a few very simple calculations (eg: simple filtering) and stuff the data into a buffer, but there just isn't enough time to do anything more complex. In order to do serious real-time processing of sound you might want to consider using a DSP coprocessor (eg: the Motorola 56001, as used on DigiDesign's AudioMedia board). Of course, if you're only interested in doing something very simple with the sound then please feel free to ignore any or all of the above. :-) //Paul -- Paul Russell, Department of Experimental Psychology University of Sussex, Falmer, Brighton BN1 9QG, England Janet: paulr@uk.ac.sussex.syma Nsfnet: paulr@syma.sussex.ac.uk Bitnet: paulr%sussex.syma@ukacrl.bitnet Usenet: ...ukc!syma!paulr