Me guesses not everyone has access to it. It's probably a "devs-only" page or something. Of course, if you don't have access, YOU should considering you're one of those devs.
Maybe one of the mods can add you to the list. In the meantime, lemme make a brief explanation here, and the code is in the arcs, of course.
CVSD normally compresses the samples to just a sign bit. That tells the decompressor whether to add or subtract the delta from the derived sample. If delta was a constant, clearly you would only ever have slow saw-tooth representations of the original wave. The way CVSD works is to change delta after every sample depending on some criteria. The "normal" algorithm changes delta if you have three or four 1s or 0s in a row. In my code, I changed delta larger if the current bit matched the previous bit, and smaller if they differed. That was the original code. One bit per sample means it was equivalent to 8:1 compression compared to what you would store in the Genesis to playback. Of course, it really isn't suitable for music.
So my latest experiment decided to bump the samples to two bits instead of one. We still have the sign bit, but now an extra bit is used to tell the decompressor which way to shift delta. Instead of implicitly shifting based on matching previous bits, we now shift according to which will give less noise. That's what the last two arcs have used, and at two bits per sample, you get four to one compression. It can be suitable for music - some songs seem to encode very well at 4:1, while others show a little more noise than you'd like in your music. So this may or may not be suitable for people to use, depending on specific circumstances.
Again, remember that this is compressing 16 bit audio. While the code decompresses to 16 bits, only the top 8 bits are used with the YM2612 DAC. I may make another example of using 10 to 12 bits with the 32X audio.