SpiceWare's Blog

A custom DPC+ would need to use a power-of-two sized sample space, so it could be 2k or 4k and you could just use what you need. Let me know what is best. You can still start the table at any offset of 32 bytes if that helps.

I think 2K will be the way to go then. Unless I'm mistaken, the index into the 32 byte sample just loops - ie: only resets to 0 after reading #31. If so, we'll need something to reset the index back to byte 0 so that the words don't start playing back "mid-word" when the sample buffer is filled in with the next word to play back.

 

I've decided to go with Zarvox for now so I can add custom phrases. It might be interesting to record samples off the AtariVox, though it'll be a while before I'm ready for that.

The lower sampling rates are introducing aliasing. If you first run the .wav through a low pass filter, ideally, one that is very steep and cuts off all frequencies above half the sampling rate, it may sound acceptable even at 1k, or perhaps less.

 

EDIT: And when I say the sampling rate, I mean the rate at which the 2600 plays it back.

good to know as the 2K and 1.5K samples were taking up too much room.

 

Hmm - looks like there's one included with OS X, just have to figure out where & how to use it! Will research it more after work.

http://testtone.com/developers/apple/au-low-pass/review

How often will you be updating AUDV ? NTSC 2600 line rate is 15.7kHz, so you would low pass filter below 7.8kHz and use a 15.7kHz sample rate.

 

Just for reference, telephones use 8kHz sampling.

I'm currently updating it it every scan line using one of the incremental data fetchers, which simulates what we'll eventual be doing using the DPC+ music routines. The 1K sample uses 24 as the increment, so it updates once every 256/24 = 10.7 scanlines. 1.5K uses 36 (7.1 scanlines), while 2K uses 48 (5.3 sl) and 4K uses 96 (2.7 sl)