Dean Roddey

More basic: Proper wiring--- not cheap stuff thrown together by hack electricians looking to save $1.50 on every house they complete in a subdivision. Are you saying because it puts out RF or because it causes hum in equipment? If the later, I'd think that pretty much every piece of equipment has a switching power supply these days and it wouldn't really much matter. Maybe amps still don't, I dunno.

In some cases it's also because they were DI'd in and never went through a mic or amp.

And the blessed day has arrived, and the people were glad, and they did feast upon the lamb, and the calf, and the shoat, for in that day the dream became a reality. I finally got the studio into its basically final configuration today, other than adding more bass traps when I can afford to. I've got ten traps in there, and three more on the way, which should do me pretty good on bass management. This is a purely in the box studio. I DI in bass and guitar (and use Amplitube/Ampeg plugs) and use BFD for drums. Only vocals use a mic. The new desk is bigger but, since it's really a corner type desk, it leaves more space out into the room than the smaller old table did since it's thinner in the front, and of course having the two layered desk makes way better use of the space available. Here you can see the Behringer DCX2496, which provides the crossover and EQ for the monitors and sub, and the little Presonus volume/headphone controller, and the padKontrol. And of course the new Mackie HR824's. There was just enough room to slip in the little speakers between them and the monitors. On this side I've got the Solo/610 and the M-Audio keyboard. The computer is just down there on behind the desk on this side. So there it is. Be it ever so humble, I'm glad to finally get it all worked out. Now I can concentrate on making music.

That's kind of a different thing. A digital recording medium won't add any color to the recording. So the thing with tape is that it's both the recording medium and it's a source of warmth and compression because of the way tape works, it affects what is recorded on it. In the digital world, it just captures whatever comes in with no (or very, very little) coloring of the sound. So in the digital world, you either use warm input sources (like a tube based guitar amp) or you add that warmth artificially after the fact with processing. You can argue that one digital medium or encoding technology is more accurate than another, but that's about as far as it would go in terms of how it affects the sound. None of them would be inherently more likely to emulate tape warmth, since they all just reproduce the incoming signal that's sampled, and as long as they are pretty close to each other in accuracy, they should sound about the same. I've read some arguments about very high rate, single bit samples that DSD uses, as apposed to high quality PCM type sampling. I never really got the feeling that there's any true consensus that one is significantly better than the other in practical terms.

That doesn't mean it still isn't even harmonic distortion and compression and low end boost, which it still is. If it's gotten hijacked by marketers, that's a separate issue.

Actually it's the very cleanliness of modern digital recordings that is the argument for using tube gear during the input phase of the chain, to keep it from sounding completely sterile and overly clean. That doesn't mean tubes sound 'more musical'. That's not a measurable quantity. It's just that overly clean, overly accurate recordings are not really even desirable for some types of music, though it is for others. And there's a difference between 'warm' and 'bad'. Warm does not imply that the recording is less than high quality, any more than a recorded guitar is not a good recording because the amp was distorted. You are still very cleanly capturing the source material. The same applies to other instruments when that is appropriate and desirable for the type of music. If people think it sounds pleasant, then they do. The goal of the listener is not to receive the cleanest possible recording, but music that they find moving and pleasant or stimulating to listen to. Many people find that that warm sound with even harmonic distortion is very pleasant.

It IS pretty straightforward Dean; differences of +/- 2db are audible. Anything beneath that isn't math at all (OP was asking about science/math). Various people claim to be able to hear very small differences in volume. I think that the 2dB minimum is not universally held, even by scientists, and it depends on the frequency of the sound. But somewhere between 1 and 2dB I agree is likely the limit absolute best case. But even if those who claim to do better were correct, the point is that the volume difference per sample step is hugely less than that, so it wouldn't matter. As to your other comments, you can believe whatever you want, but most folks would disagree with you. You obviously have some anti-tub bias for some reason, but I don't think that many people would argue that all those classic tunes that were recorded with all or mostly tube based equipment don't sound quite pleasant. And certainly most people would argue that tube based amps sound smoother and nicer than solid state ones when pressed hard.

Just for argument... At 24 bit sampling depth, there's not going to be much in the way of stair casing in digital recording either. It's there but on an infinitesimal level. That's 16 million possible steps from 0dB down to the floor. At some point a sufficiently fine grained digital system becomes effectively equivalent to an analog system in any practical terms. Since the 'measuing device' here is the human ear, that's far, far beyond the human ear's ability to distinguish one sample level from another. If you have a -120dB floor, that's 0.0000075dB per step. Most folks can probably barely hear a quarter dB change reliably, so that leaves more than a bit of leeway. Even if you aren't using the full drange (peaking at -6dB, say) that still leaves at a tiny fraction of a dB per sample stepping. Obviously the sample rate plays into that as well, but you are free to do 24/96K, which would just be so far beyond any human ability to distinguish that it's not worth worrying about. Particularly when, whichever way you record, you are going to put it out to 16/44.1 digital format anyway.

Actually, it's pretty straightforward: 1. High end rolloff from the tape 2. Some low end boost from the tape 3. Addition of even harmonic distortion by the tubes 4. A kind of very natural and smooth compression that occurs with tape and tubes as they are driven harder (which is why people like tube based guitar amps so much, they take very good advantage of this effect.) There's not any koolaid there, it's clearly a very warm sound and you can hear it on many a hit of the past which were done on analog gear, using lots of tube based equipment and analog tape. It's a sound that we find very pleasant and musical as a rule. It can be emulated in various ways, and it's not necessarily a sound you want if you are going for super-clean and pristine. But it's a sound very associated with rock, funk, soul, R&B, blues, etc... If you can get it the old fashioned way (when that's what you want), then I'd argue for that. I use a tube based mic pre/DI box so that I can get that sound naturally for guitar, bass and vocals, instead of adding it after the fact with plugs.

My dual core (2.4Ghz) has no problem with BFD and an Amplitube or Ampeg instance being unfrozen at once as 2.7ms, i.e. at good tracking latencies. Can't have a few convolution reverb busses at the same time though or it will go over the edge now and then. I don't know what DAW you use, but in SONAR I can just push the latency slider to say 64ms and it can handle a quite substantial amount of stuff at once for mixing purposes, where latency isn't an issue. So basically I track everything with minimal or no processing, with the latency down to 2.7ms. Then go into mixing mode, push up the latency high, and start throwing in the multiple reverb busses, all the drum processing, etc...

Not sure where you'd use 4G RAM. I guess some new big orchestral, multisampled plugins could benefit. Most I've ever used is about 670M with a couple of big soundfonts. Drum synths. A drum synth like BFD itself can use more than 4GB of RAM, leaving aside the OS and DAW software. I use the 40 velocity level version of it right now. If I flip the switch that tells it to load all the samples (for the small drum kit I use, not a big one) into memory, it will zoom past 2GB and start swapping like crazy. I've not let it continue since it would just kill the machine. But if that small kit at 40 layers eats that much memory, clearly a Neil Pert type kit, at the 128 layer level, would probably hit 4GB or more. So what you end up with otherwise is a scheme where it loads the first parts of the samples into memory, just enough to get the sound playing while it loads the rest of the sample from disk. So you really have to have a separate hard drive for its samples and it beats that drive hard and can get behind in really frenetic sections and glitch (not during the final mixdown which is done offline, but while tracking/mixing.) With a 64 bit system that had say 8GB, you could just let the whole kit load to memory and have a very much higher performing system. I'd love to go XP 64, but my machine has to double as a software development machine during the day (when I'm not wearing the cape and tights.) I'd really have to buy a second machine in order to do that. In the meantime, I think that just going up to a 32 bit quad core would be a better investment for me.