GeorgeVW

Been using the version 2 of the SpaceStation (less power passive rather than active xover) and even that version sounds absolutely huge. It's the first stereo rig I've ever used where you can experience the stereo pretty much everywhere in the room, not just in the "sweet spot." Last couple of shows have been Helix 1/4" outs to the SpaceStation and XLRs feeding the PA.

If it's a third-party shop and they're not L6 certified, they probably can't get the parts, hence "irreparable," at least by them. Component-level repairs have become very much a thing of the past for most modern tech, as it's mostly board swaps these surface-mount, wave-soldered days.

Rupert Neve-design heritage preamps on the Clarett combine with Thunderbolt speed makes it a no-brainer. Aside from the fact that, unless you're only ever going to be using a single mic at a time, there's smply not enough I/O for it to be a serious audio interface. When you're recording, you need tools that simply get out of your way and let you focus on the creative process, not tools that you constantly have to figure out how to work around their limitations.

Point source vs. environment. And a transducer is a transducer is a sensor is a rose is your ears mounted on the side of your head. You're asking for a unicorn. The sound of an "amp in a room" is requisite on having a) an amp and b ) a room. And no matter how you measure it, what you're always going to end up with is the sound of _that_ amp in _that_ room because you can't stop that butterfly from flapping it's wings halfway around the world. This debate meets the original definition of "moot point" i.e. one which is subject to endless debate without every reaching resolution.

A little big, but you can fit all your acccessories and such in with it, too. I use one of the smaller ones of these to carry 9 mics for the P.A. system. As long as they've got refurbs in stock, the $100 price is pretty good for what it is. https://www.monoprice.com/product?p_id=24585

As a primarily improvisational band, we do a lot of live recording, and that starts to become a very complex setup with the routing on the Helix not quite flexible enough (that I've been able to figure), to do wet/dry/wet and regular stereo out at the same time. You'd need to get the signal taken off immediately post amp model to go both separately to a separate send and fold it back into your secondary stereo outs at the same time, while the post amp (stereo wet) signal needs different mix levels for the isolated wet mix and the regular stereo mix. Don't see a way to do that with existing routing options, so it's one or the other, not both.

Biggest problem is that it doesn't record well with traditional stereo miking techniques (and I'm not sure how a M/S recording would work, since it is, in fact, a point source). You can get a little idea by checking out the demo videos Aspen has up on YouTube (look for CenterPointStereo or CPS) under phones, but the videos come nowhere near capturing how enveloping the soundfield is live. I'd have to try a binaural recording, since that does a really good job of capturing a space, and listen back under headphones to see if that captures it (now where did I put that dummy head?).

Not so much "don't believe"(hey, I've got good friends who work in both places) as in believe that it's discardable in real-world applications. Your proposed experiment doesn't test what you think it's testing, though. Can a person detect 20ms of delay? Certainly. That's been well established. But here's the experiment that you need to do. Shot an IR of a hallway from 20 feet away from point A. Place two flat-response, full range speakers (B and C), one 20 feet down that hall, one 20 feet closer. Balance level between the two so that observer at point A gets the same perceived SPL from both. Apply the IR and a 20ms delay (or whatever is necessary so that DSP latency plus delay line are equal to 20ms) to the signal going to the nearest speaker. Switch between the sound sources randomly, rigged so that the person doing the switching doesn't know which position of the switch is which source. Do the test multiple times, shifting the L/R orientation of the speakers for randomization purposes between test sessions. Can the test subject consistently, above the level of pure chance, tell which speaker is actually 20' away and which one is virtually 20' away? That's the experiment that will determine the answer.

Not buying it when it comes to playing guitar onstage with a band. In an environment when your ear is hearing both the original string impulse (even if only very faintly) and an amplified source, yeah, makes sense, and I've experienced it in a studio setting. But in an environment where background noise is masking your string sound, nope — don't buy it.

Let's see that assumption stand up to a double-blind test before we get too attached to it.

Stopped by to visit Aspen Pittman on Friday to pick up one of his SpaceStation CenterPointStereo cabinets. Interesting concept — think of it as Mid/Side recording flipped around for a playback system. A 3-way front-facing system (coaxial low/mid driver and a super tweeter) reproduces the L+R signal (or what's in common for both left and right channels) and a full-range speaker placed underneath it fires the difference signal (L-R) sideways and 180°out of phase with the summed front facing output. If you've ever heard the Fender Acoustasonic SFX amp, that's Aspen's patented system at work. While it doesn't give you the hard left/hard right you would get from stereo speakers or a LCR (Left/Center/Right) setup, the big advantage of it is that the whole audience (and the band on stage) gets to hear a big stereo image. Huge spacious delays and reverbs and, man, oh, man, you can hear the Leslie swirl, all from a single source-point. With a 300° dispersion, the sound is very even in level almost everywhere. The surprising (and slightly freaky) thing is that the stereo passes through open doors and windows, since the "space" is created by the way your ears decode and combine the two sound waves. I can play something in the studio and, with the door open, someone standing down the hall can hear it with the same sound field I hear. No more having to sit in a small "sweet spot" to hear a large soundfield( which has always been the problem with using stereo in a live situation — most of the audience doesn't get to hear a fairly large part of the performance all that well). While by itself it's pretty darn cool, I figured that using it as the post-amp effects amp and feeding the signal from just before the post-processing chain to a separate "center" channel, I could get a pretty good approximation of a wet/dry/wet rig in much less space and with only the Helix and two power speaker cabs — in my case, the Spacestation SFX Mk2 (there is a newer version, but Aspen gave me a deal on the previous model) and an EV ZLX12P powered cabinet. So, inspired by a combination of the supplied Helix templates for FRFR Wet/Dry/Wet signal chain and Parallel Spans, I've started tweaking up some patches to take advantage of the expanded soundfield this system offers. As a quick review of the signal flow, the chain looks something like this: guitar input > mono stompbox-type effects (anything you'd normally stick in front of the amp) > amp+cab (doesn't matter if your using an amp+cab block or separate amp and cab/IR block) > Split (still experimenting with the variations I can get with Y vs A/B splits) with one side of the split going to Send 1 which feeds the EV cabinet and the other going to > stereo post-processing (delays, mods, verbs, whatever) > L/R outs > Space Station. Slowly coming up with some basic principles for designing patches to take advantage of this. It seems to be good to use 100% mix on the first stereo effect in the chain to maximize the "stereo-ness" of the final sound, since you've already got a speaker covering the mono "dry" part, and you want to maximize the "wet" to the stereo field. Sticking a stereo limiting amp on the tail end (the LA Studio comp in limiter mode) seems to make for a better result, especially if you're doing some crazy stereo filter stuff in the post section. As I get some full band practices and gigs under my belt with this system, I'll report back and add to this thread. So far, in the studio, it's pretty durn impressive, and I'm looking forward to pushing it forward. It seems to me like there might even be some way for Helix to do some variation of the routing and phase-flipping stuff in firmware so you could feed a summed signal to one speaker and a phase-reversed difference signal to another and get a similar effect. Hmmmmm … that's probably worth some further thought as well.

It's also valuable to bear in mind that 1ms in latency is the equivalent of standing a hair less than 1' further away from the sound source (using the average 1125 fps free air speed of sound), so even a 10-12ms total latency is only the equivalent of wandering towards the front of a stage by not even the length of your average guitar cable.