Your killer Digi 001 rig

[mechanic]

I'd imagine that it's got a better clock just because it's newer technology and the power supply is internal instead of running off of the damn PCI buss. If you think about it, clock stability relies greatly on a good power supply, and whatever the computer is sending out through that PCI is crap. Just a better power supply should improve a lot of things with the clock and converters, theoretically. Havne't heard one yet, so I couldn't tell you how much. That's what I'm hoping to find out.

...if this is the case, it would hardly make sense to purchase a $1300 word clock device, when you could just get an 002 rack, right? Any comments???

[mattmlpdx]

I'm looking to get a secondary drive for the PT system I am trying to get up and running. This will be for thee music and PT session files themselves. Any suggestions? I was leaning towards a FireWire drive...

[djgout]

i've never had a 001 b/c i've been laptop bound. but i've owned an mbox and now switched to the 002.
yeah the clock on the 002 is most likely way better than the 001, simply because it's newer. with the rate (no pun intended) that all of this technology is increasing the newer the converters the better they are is a good philosophy right now. think about it. HD was the first to get up to 192kHz. the 002 is a more powerful converter than a mix system from 2 years ago.
i've compared mixes on my 002 with mixes done on a tdm system and it's obvious that the 002 does sound better.
the only complaint i have with it is the lack of AES and a wordclock input. it's a real drag being limited to 48kHz through lightpipe, I'd like to be able to get 16 channels in and be able to work at 88.2.......but i can't afford HD yet, not until my first 7 or 8 grammys.......

but yeah, there's work arounds for what ever system you're using.

[trodden]

I figured i'd resurrect this instead of posting something new that kinda deals with what we've been discussing.

Latency, When ever a plug in is used, there is a degree of latency. What about, say you like what the plug in is doing, so you decide to commit it with the audio suite function, then lose the plug in. Will there still be an amount of latency after the track has been audio suited rather than just using the plug in?

thanks,
brandon

[JGriffin]

re: Input latency: to Electricide's question--"wouldn't most DAW have latency around 3ms? With some you can tweak down to 1.5 I've heard. The TDM systems avoid this, but for $10000. Does any DAW at or under $1000 avoid this really?"

ProTools 002 has a "zero latency" mode for recording that routes processing power to get the latency as low as possible. I don't know if it's less than 3ms, and you do sacrifice using plugins on the input channel, but it's cooler than the 001 or mBox.

re: "Latency, When ever a plug in is used, there is a degree of latency. What about, say you like what the plug in is doing, so you decide to commit it with the
audio suite function, then lose the plug in. Will there still be an amount of latency after the track has been audio suited rather than just using the plug in?"

Audiosuite-ing the track will fix the latency problem, since you're now just playing back a rendered audio file rather than sending that file through what's essentially a small program running on top of a larger program. If you choose to keep your plug-ins running so you don't have to commit to a setting permanently, you can have ProTools tell you what the plug-in delay is, then shift your track backward in time by that amount. It's important to do this on a duplicate playlist so that if you decide to change or remove the plug your tracks can go back in sync easily by selecting the original playlist.

[Electricide]

If you choose to keep your plug-ins running so you don't have to commit to a setting permanently, you can have ProTools tell you what the plug-in delay is, then shift your track backward in time by that amount. It's important to do this on a duplicate playlist so that if you decide to change or remove the plug your tracks can go back in sync easily by selecting the original playlist.

wait..hmm. backward in time means earlier than it should be, right? If it's a reverb, or other effect in which you have a mix of dry sound and effect return, are you suggesting sliding the original track ahead in time so its effect return is where it should be? Because then the original signal is ahead of everything else.

The solution I see is to duplicate the audio track on another track, move it ahead in time the appropriate amount, send to the plugin, and only use the return, and blend the return in with the original unshifted audio file.

That's a lot of work. DWLB, please explain more detailed if I'm missing your point.

[leigh]

ProTools 002 has a "zero latency" mode for recording that routes processing power to get the latency as low as possible. I don't know if it's less than 3ms, and you do sacrifice using plugins on the input channel, but it's cooler than the 001 or mBox.

Well, if they're calling it zero latency mode, it better be less than 3 ms, it better be 0 ms!

And the only way to do that is to monitor from the input signal itself, not from the signal after A/D and then D/A conversion steps. This requires a little analog mixing of the input signal with the output of the DAW. Not all that tricky, although the 001 doesn't offer this. However, I thought that the Mbox did.

Leigh

[JGriffin]

wait..hmm. backward in time means earlier than it should be, right? If it's a reverb, or other effect in which you have a mix of dry sound and effect return, are you suggesting sliding the original track ahead in time so its effect return is where it should be? Because then the original signal is ahead of everything else.

The solution I see is to duplicate the audio track on another track, move it ahead in time the appropriate amount, send to the plugin, and only use the return, and blend the return in with the original unshifted audio file.

That's a lot of work. DWLB, please explain more detailed if I'm missing your point.

If you're using an aux to send multiple sound sources to, say, a reverb, that's a different story than what I'm talking about (in fact, if you're using pre-delay on your reverb plug, it won't really be an issue at all since the reverbed signal is delayed anyway). I'm talking about a plug or set of plugs on a particular audio track.

Example:

Say you've got a bass guitar track and on that track, for some insane reason, you've instantiated a Metaflanger, a Renaissance Compressor, a Focusrite d2 6-band EQ and a Waves Q10 EQ. Command-click (control-click if you're in Windows) on the level indicator, and the display will toggle between level, peak, and plug-in delay. The delay is represented in samples, and for the above combination of ill-advised effects ProTools tells me the track will be delayed 78 samples, which means the entire bass guitar performance will be 78 samples behind the rest of the tracks, assuming for this example that there are no other plug-ins on any other track. This may not be really good for the feel of the tune. If you were to create a duplicate playlist for the bass track and shift it 78 samples earlier in time, the bass track will hit the plug-ins 78 samples earlier, get processed for 78 samples' worth of plug-in delay, and come out exactly in time with the rest of the band.

Now put the problem in the larger world of plug-ins on every channel: a gate and EQ on the bottom snare mic, a compressor and EQ on the top, compression and EQ on the kick drum, limiter on the overheads, Eqs on all the guitars...prety much your whole mix is being pushed apart at the sample level. This can in extreme cases seriously de-groovify your mix, and in non-extreme cases at the very least your top and bottom snare mics are now a little out of phase. So, to correct this, we do the same thing we did with the bass in example A, by finding out the plug-in delay of each track, and shifting the audio, everything lines back up. Just make sure you do it on duplicates of the original playlist so that if (and of course, when) you need to do a mix with different plugs or no plugs you ca put your tracks back where they belong.

I hope I explained that okay. Does it make sense?

Hopefully the new PT's auto-plug-in delay compensation feature will render this whole exercise meaningless.

[chuck anderson]

I use Digi 001 PT. I added Panasonic WZ-96 converters for 8 more channels of A/D and D/A connected by lightpipe. I mix through a Neotek Series II. I sync up my MCI JH-110 8 track one inch recorder with an Adams-Smith Zeta III. It took some messing around to get it running but its pretty stable and reliable. The tape machine chases a timecode track recorded in ProTools. I have eight Dolby 361 units with SR cards that I have yet to hook up to the 8 track (the dolby units are pin 3 hot and I'm switching them, but its tedious).

[JGriffin]

Well, if they're calling it zero latency mode, it better be less than 3 ms, it better be 0 ms!

Agreed.

And the only way to do that is to monitor from the input signal itself, not from the signal after A/D and then D/A conversion steps. This requires a little analog mixing of the input signal with the output of the DAW. Not all that tricky, although the 001 doesn't offer this. However, I thought that the Mbox did.

Leigh

I believe it's an input-only monitor setup on the 002, yes. The mBox has a little knob that allows you to mix the input and output signals. I've never used the mbox for overdubbing so I don't know how well that works.

[Electricide]

yeah, DWLB, you're referring to inserts, where 100% of the signal goes through plugin before monitoring.

I guess I was thinking in an aux send, aux track plugin scenario.

Anything is better than my current setup, in which there is a recording latency. It monitors in perfect sync, but when the audio file is written to disk, it's like 40 ms behind. Slide city, ick.

[JGriffin]

yeah, DWLB, you're referring to inserts, where 100% of the signal goes through plugin before monitoring.

I guess I was thinking in an aux send, aux track plugin scenario.

Yeah, I had not considered what to do in an aux send/plug on aux return situation prior to this thread. There was a thread about the trick I was talking about over on the computer board a few months back. I'll have to think about the send issue and see what seems to make sense.

Anything is better than my current setup, in which there is a recording latency. It monitors in perfect sync, but when the audio file is written to disk, it's like 40 ms behind. Slide city, ick.

Wow! What are you using that does that? Strange.

[Electricide]

Nero soundtrax.

Programs with intentional misspellings send red flags in the first place.

Oh well, it was free, already installed. Just debating my next (first) move

[twitchmonitor]

I'm going to ignore the latency side issue and just respond to the initial post.

I'm running PT 5.2.1 on a G4. Outboard gear includes a Soundcraft M12 board (which is use mostly for preampls, but I'm going to try to start mixing out of the box with it), Studio Projects VTB-1 pre, RNC, Audio Buddy 2 chan pre, BBE sonic maximizer, DOD rackmount compressor, TEAC M3 board (for pres), Yamaha spx 90 effects unit and some other odds and ends, patchbays and a bunch of mics. I'd like to upgrade my pres but I had to invest in a bunc of pres since I like to be able to record full bands.

I'm pretty stoked with my setup and I get good results.