As you probably know, the GT-100 is a pretty powerful bit of kit. But Roland, as is typical with them, does not provide an editor.

If you are unaware, there is a Boss/Roland enthusiast from New Zealand who goes by Gumtown on various forums who has written a bunch of editors for the entire GT-x line and the GR-55. They can be found here:

http://fxfloorboard.sourceforge.net/

I use the GT-10B, GT-100, and GR-55 editors and they really make programming those units as simple as possible.


Regarding mono-only on the GT: Here's a trick.

You can use the Return as an additional input and put it anywhere in the signal chain you like. Regular input can go to Channel A, Return can go to Channel B, put whatever effects you want, sum them before the output and Bob's your uncle.

Actually, Roland/BOSS has improved quite a bit in terms of software support in the last year or so... They developed something called "BOSS Tone Studio" (http://bosstonecentral.com/), which basically allows you to edit the chains on your computer (they support both PCs and Mac).

I did use FxFloorBoard for a while too, and it's an impressive piece of software considering that Gumtown probably had to reverse engineer some of the MIDI commands on the USB interface.

Agree that having any kind of software interface makes programming those things way easier, helping you understand where in the chain you are, etc.

I was aware of the "return" trick on the GT-100, but it's my understanding (from somebody that had tried it and posted back results on a forum...) that the problem was latency on the return chain (apparently, this latency is built into the circuitry)... there was certain amount of latency that made the configuration a bit hard to play... Did you try it? Did you perceive any latency at all on the return chain? The other problem would be, at least for the Railboard, the lack of phantom power (but this would be just a minor issue considering the upside of using the GT-100)

I haven't tried it with the Stick, but I have a bass with a GK-3B that I run like this:

Bass -> Input on GT-10B

Bass -> GK-3B -> GR55 -> Return on GT-10B


Then I blend both signals inside the GT-10B and send to one amp.

Never noticed any latency.
If you already have the GT-100, you lose nothing by trying it.

I'll try it out as soon as I get my stick and report back!

Sent from my Nokia 1100 using Tapatalk

Rob,



I've been doing a bit more research on this topic. On the "scarlett" family of products, I see that only the 18i8 has more than two XLR inputs. Are you able to plug a microphone on your "saffire pro 24"? I see that, as with the scarlett 2i2 or the 6i6, the saffire pro 24 also only has two XLR inputs (unless I'm missing something...)

Also, do modern DAWs (like PreSonus Studio One) support inputs from multiple USB sources? Since the USB interfaces should be treated as high end external sound cards, I'm wondering if DAWs can actually take signals from multiple sound cards or interfaces, perform some processing using the main CPU in your computer, and then route back the resulting signal to one (or multiple?) of the USB interfaces. Are there any limitations (either of the interfaces or DAWs, and leaving aside the fact that your computer needs to be powerful enough to perform the required processing on all of the interfaces at the same time, USB I/O limitations, etc.) to consider?

Also (related to FHF and online classes...), I wonder, what prevents a DAW from also using a computer's built-in microphone (and speakers...) as another source? I feel the answer is related to the low-latency audio drivers (e.g., ASIO, JACK, Core Audio), but I'll let you educate me here ... maybe you can share what's your audio routing setup for FHF?



Do you perceive any latency issues with Wine? I'm assuming you are using WineASIO (since JACK support seems to be broken), in which case latency should be non-existent.

My current audio interface (the Saffire Pro 24) does not reflect my knowledge of equipment: it reflects the financial constraints I had when I had to replace my previous audio interface quickly. I can use my mic and my Railboard together if I put the Railboard in mono mode, but that only applies to ASIO programs (like my DAW). The Focusrite Saffire driver does not support WDM remapping, so WDM programs (i.e. everything which does not support ASIO) only see the first XLR connector. From reading the manual, it looks like the Focusrite Scarlett driver has the same shortcoming. In that case, I do not recommend getting a Focusrite audio interface if you use Windows.

That depends on the DAW used, the operating system used, the audio interfaces used, and their drivers. Some drivers will aggregate multiple audio interfaces, though of course the interfaces would have to be compatible with the same driver (e.g. identical models or models within the same product line).

Depending on the DAW, your operating system's standard audio subsystem may be supported. If it is supported, then you can use your computer's built-in microphone and speakers: there may be issues with latency and synchronization.

For FHF, I run my mic into the first XLR connector. That is all there is to it, other than using the headphone output for my headphones. I use Saffire MixControl to mix the headphone output with my mic at unity and my operating system's stereo output at -22.5dB. This allows me to run the headphone volume high enough to monitor my voice in real time while keeping my operating system's stereo output at a sane level.

I only installed Linux last week (a very long story, indeed), after 667 days without it, and my current interface will not work. I could work around the issue by downgrading my firmware version, this too is a long story, but I decided it would be better to replace my interface.

Of course... mono mode! Most of the software I use supports both ASIO and Core Audio (in the Mac, which I'm currently using), as well as WDM (although I had serious latency issues in the past when trying to use it... not an option for me)



I assume that if all the drivers are ASIO-based, then they would be considered identical, i.e., they are all hosted on the same driver framework. Are you saying that, even if two interfaces are ASIO-based they would still have to be of the same model or product line?



Yes, given the previous discussion these two subsystems (USB interface and internal audio) may be driven by two distinct driver frameworks (e.g., ASIO and WDM), in which case latency/sync will be an issue (if it is even supported at all by the OS)



Got it.



Yes, I understand that FireWire audio support in Linux via FFADO is not without glitches and vendors do not typically offer support. That Midas interface is drool-inducing, congratulations Rob! But it's way more than I'm willing to pay (currently) for an audio interface, and I honestly don't have the experience required to be able to make the most out of it anyways. I'll do some more research on the driver support for the Focusrite USB 2i2 (with regards to Windows drivers and DAWs, even if I'm using Mac right now...) and report back.

The answer to your question is "yes," since the commonality of ASIO does not mitigate the various differences among interface hardware.

Is this because interface vendors deviate from standards within ASIO? I don't know enough about ASIO to tell how much the standard covers, and how much it leaves to interface vendors to screw up (ahem... "differentiate"). One would expect that if a DAW vendor understands the nuances of each interface vendor, then this wouldn't matter at all (that is, interface vendors don't have to understand each other's nuances since the DAW software would act as a bridge between them, converting and normalizing the differences, etc.)

You are misunderstanding the situation: the driver would aggregate the devices, not the DAW (i.e. ASIO is between the driver the DAW: device aggregation is between the driver and the devices). Audio interfaces are not all built the same way, even when they use similar chips, thus drivers tend to be specific to the device they control.