I'd like to axe a question about the Fender Super Champ XD

[ggm1960]

 

You dont have the latency issues you would have recording with an amp sim either.

 

 

I really like my Digitech GSP1101 along with the control 2 pedal for recording but I also like Amplitube 3 and the Line 6 POD Farm software stuff. If you have a second computer/interface to run them on, you don't need to worry about latency on the recording computer either.

[stevieb]

I've had a "SCXD" for about two years, now. It's been my go-to amp for most of the rehearsing I do, both here at home and with my buds, because it's light and easy to carry, saves me from having to bring pedals, and does the job. Works just fine for playing out (we don't get real loud) but I prefer to bring my SF Deluxe Reverb, as it sounds better and looks more "serious." I was able to get the SCXD to sound VERY much like a vintage (tube) Champ, in a careful A/B comparison.

 

I have not found it to be noisy. In fact, reason I bought the Super Champ over the Vibro champ is 'cause the SC had less noise, not for the extra power.

 

12AX7 tube is NOT use for pre-amp gain, it does something else that I can't ever seem to remember. Pre amp is solid state.

 

Only real complaint I have with the amp is I can't get tremolo and reverb from it at the same time. So, if I want that, I gotta bring a pedal.

 

I upgraded the speaker to an Eminence Ragin Cajun speaker, cause the amp lacks bass. Noticed a modest change, if I had to do it again, I might not. That speaker gives the amp a more "taking care of business" look from the rear, FWIW.

 

Mine's for sale, btw, because I now have an all-tube, Ampeg Jet J12T- I rarely use any effects other than tremolo (I am FRUIT for tremolo!) and reverb, and only use one or two of the SCXD's amp models. I am liking the Jet both for it's all-tube tone, 12" speaker (that brings the bass) and it's look- the blue tolex is just so cool. If I had never scored the Jet, the SCXD would have continued to serve me well, probably for years and years. PM me if you are interested in buying it.

[blue2blue]

 

You dont have the latency issues you would have recording with an amp sim either.

If it's digital, it's got latency, but, of course, not on the same scale as routing through a computer, with all the extra buffering needed to cushion the various aspects of a computer based rig. It's probably more like the relatively short delay of any digital multi-FX or dedicated rig-sim (like a POD, etc). That's not going to bug many folks, in all likelihood, but there are some sensitive souls it will make uncomfortable, at best. I know.

 

As anyone who's had to sit through one of my tirades about manufacturer's calling 5 or 10 ms near zero latency but calling 2 ms zero latency knows my thinking on fuzzy use of the word, latency.

[WRGKMC]

 

If it's digital, it's got latency, but, of course, not on the same scale as routing through a computer, with all the extra buffering needed to cushion the various aspects of a computer based rig. It's probably more like the relatively short delay of any digital multi-FX or dedicated rig-sim (like a POD, etc). That's not going to bug many folks, in all likelihood, but there are some sensitive souls it will make uncomfortable, at best. I know. As anyone who's had to sit through one of my tirades about manufacturer's calling 5 or 10 ms near zero latency but calling 2 ms zero latency knows my thinking on fuzzy use of the word, latency .

 

 

If you consider analog delay, the time it takes for sound to travel through an analog unit or pass throug a mixer, then yes. If you're considering a ping test using direct monitoring, no. My PCI cards have no appreciable "audiable" latency. I can, and do set my buffers close to max so I dont have to deal with resource issues mixing. I do have a delay punching in because of that latency being set high and the gui is slow to respond. Meter movement is slow and selecting buttond like record or stop take time to react, but that doesnt bother me. I'm used to that coming from an analog background and having mechanical delays with tape recorders. If I'm punching in a part I simply capture the note sooner so it overlays properly or record to another track and edit. no big deal.

 

I do stress though, I dont monitor my signal through the system, I monitor it coming directly back out of the sound cards so I'm hearing an analog feed. The tracks are synced to record and self adjust the data so the tracks align. It may be different with Firewire or USB because you're dealing with a communication bus. With PCI you can route the signal for real time recording when set for direct monitoring.

[blue2blue]

Again, if an AD and a DA process is involved, there is latency. (And, of course, there is latency in analog circuits as well, but that is properly measured in microseconds, not milliseconds.)

 

That's all I'm saying. The latency from a digital converter's 'direct' onboard monitoring may not bother some folks. It will bother others.

 

Yes, I'm well aware that some people sit across the room from their amp and that the speed of sound is roughly 11 feet per 10 ms. But I certainly don't sit across the room from my amp -- and I've done enough direct-in recording of electric guitars in the past that I've come to expect a certain, shall we say, immediacy [pun acknowledged] in the sound.

 

And the sophistication of the human auditory systems is such, at least for some folks, that the delay implicit in a guitar sound emanating from an amp 10 feet away is already expected and processed by the brain.

 

I noticed this phenomenon when I first started going to see stage plays from the second row of a large theater with a digital sound system. It was like watching a movie with the picture and sound out of sync. That puzzled me, since I was already 25 or 30 feet from the actors. But it wasn't that there was a delay -- it was that it was the wrong delay for that distance.

 

Now, there are apparently some older PCI (and maybe even outboard) devices that do have a direct analog input monitoring capability, but if you've got a multichannel unit (ie, more than stereo analog in), it's almost certainly using a DSP mixer in its direct onboard monitoring, and, of course, that means at least a tiny amount of latency.

 

The end run on that is to use a strictly analog chain in front of the converter for cue monitoring (which is what I do). Obviously, though, that means tracking guitar while forgoing any digital FX on your guitar signal (unless perhaps you can fold some of that into your cue mix without too much timing mayhem).

[WRGKMC]

 

Again, if an AD and a DA process is involved, there is latency. (And, of course, there is latency in analog circuits as well, but that is properly measured in microseconds, not milliseconds.) That's all I'm saying. The latency from a digital converter's 'direct' onboard monitoring may not bother some folks. It will bother others.

 

 

My PCI cards use 36-bit Embedded DSP

They are Delta cards that contain a 36-bit embedded DSP enabling a software-driven patchbay/router for all analog and digital I/O

[blue2blue]

From http://www.m-audio.com/products/en_us/Delta1010LT.html...

 

 

software controlled 36-bit internal DSP digital mixing/routing

If you're suggesting the $250 M-Audio Delta cards have a multi-channel, VCA-controlled analog mixer embedded, I'm afraid I find that somewhat difficult to believe.

 

I note that the ~$750 rack version uses the phrase zero latency monitoring -- however it clearly says it uses "software-controlled 36-bit internal DSP digital mixing/routing" -- so, like many other manufacturers (including MOTU* who made my 828mkII), it certainly appears that they use the fast and loose definition of zero.

 

It's gotten to the point that a number of prosumer recording manufacturers are selling multichannel preamps that have integrated AD/DA -- with no analog pathway whatsoever between the preamp's analog input and the analog output. (Behringer and Alesis are among those companies.)

 

 

*MOTU called the 828mkII near zero latency in the marketing materials for it; it was one of the first multichannel FW devices and when other manufacturers put out products in that class they used the rather dishonest term zero latency for their products -- so when MOTU made the relatively small revisions for the 828mkIII they made one big paper upgrade, changing their marketing language to zero latency -- even though they very clearly indicate that it's still a DSP process.

[WRGKMC]

 

 

 

The 1010 rack card is basically the same as the LT version. The major difference is the 1010 has has a rack breakout box. The card connectors are wired differently to prevent people from using the LT cards with the 1010 rack units. The cost is low on the LT cards because thay dont have mic preamps or phantom, They are bare bones line level inputs only with two channels allow low impediance mic inputs but they only have board jumpers that can be set for line level or the signal up 10 or 20db with a couple of $12 op amps. So the cost is low because you have to provide your own mic preamps.

 

The DSP is used on the "processed signal" to provide low latency with the software. Thats why I highlighted that in my post. In direct mode the inputs are connected to the outputs

and attenuated between the inputs and outputs by the driver. This is simple stuff here, its not rocket science and its cheap as hell to do circuit wise. I think you may be overthinking how complex this routing is. Its bare bones attenuation.

 

The drivers do not attenuate the gain of the input signal. Since the boards are all line level there no need. You can overdrive the preamps on the boards and use the drivers to attenuste the feed to the output and it wont remove the distortion. If you read the manual can be misread to make you believe it does. I actually confirmed how the cards work in direct mode with their engineers when I got them and they worked the way I suspected. The Driver has attenuators between the inputs and outputs but the signal itself doesnt get digitized. Its simple steped digital attenuation that acts like a variable resistor between the ins and outs. The preamps themselves do not vary gain sensitivity. I had an ISIS card years before and they did it the same way. As I said I dont know if this is uniquer to all PCI cards but is is in the ones I've owned.

 

Motu is a completely different setup in comparison. It has an interface card that acts as a multichannel communication device between the computer and for the rack unit.

The signal is digitized inside the rack unit and the card acts as a multichannel bus to the main board. Thats a whole different thing. You're dealing with communication latency there simular to Firewire.

 

The M-Audio 1010 uses a multichannel analog cable between the board and the breakout box. All the box contains is analog components so the length of that cable is is important in signal loss, thats why they keep it 6' long.

They charge you $450 luxary. I simply substituted a patch bay and patch cable and had the exact same thing for around $50.

I did get the schematics from the engineers on the site when theyre dongals were on backorder for my cards.

I actually made my own cables which was a bitch by the way soldering up 40 analog connections. It wasnt the quietest cable butit worked.

 

I will contact their engineers again and see if I can get them to post a description of their direct mode routing.

[blue2blue]

Yeah... I get what you're saying, I think. The fact it's step attenuation rather than fully continuous is certainly interesting. But... I'm presuming the 'direct out' signal comes out on one of the 8 analog connectors, right? Does that connector also output signal from the computer? I'm pretty sure it does. That would require summing of the computer's PB after DA with the direct monitoring of the input if that direct monitoring is analog. Also seems to imply signal routing microrelays. Even replacing DVCA's with DVC stepped attenuation (perhaps a combined function microrelay?) -- I'm thinking that's a lot of extra analog circuitry per channel -- when it's clear that the current 'affordable' gear paradigm is heavily dependent in the 'chips-with-i/o' paradigm.

 

I certainly could be wrong. I know that unit is a pretty old design -- and there certainly was a time when the digital side was the 'expensive' side (in the sense that we didn't have the plethora of modular digital solutions available to gear designers now).

 

Still, my gut and my brain keep whispering digital pathway from input to cue -- primarily because of the summing and routing issues -- and let's not forget, the M-A marketing materials specifically say, "software controlled 36-bit internal DSP digital mixing/routing," as we've both now pointed out a couple times.

 

BTW, sounds like you were describing the original MOTU 828 -- which, IIRC, did use a card-based computer interface (although I remembered it as having the AD/DA on an outboard 2U chassis). The 828mkII was strictly outboard, connecting to the host via FW. The mkIII Hybrid, in addition to the amazing paper latency upgrade ( ) between the mkII and the original mkIII, has a hybrid USB2-or-FW interface.

[WRGKMC]

Heres the responce from M audio. He basically describes it the same as gating instead of attenuation but its basically the same thing.

 

Direct monitoring

I've been having a forum debate on how direct monitoring works with your Delta 1010LT cards I'm hoping you can clarify.

 

My understanding is in the direct monitoring mode, the analog input signals are toggled to connect to the outputs and the signal remains analog and does not get digitized. This accounts for the zero latency. The controll panel simply attenuates or pans by attenuating the signal between the inputs and outputs.

I realize the cards use DSP on the processed signal to lower processed signals latency, but thats a different thing all together and was not the topic were wer debating. You would expect latency monitoring a processed signal because its converted A/D processed then converted D/A before its heard. That round trip takes time especially if softeware effects are involved.

 

If you could give some clarification it would be greatly appreciated.

We often discuss these kinds of topics over at HC and one moderators doesnt seem to understand how direct monitoring with zero latency can be achieved. I dont feel comfortable correcting others if I have doubts on a subject so I figured I'd ask the experts. Thanks

 

His reply was

 

If you configure the Delta control panel to run the corresponding inputs through their outputs, then you plug in an input, and you go to monitor it with headphones for example, it will be 0 latency monitoring for sure, as it will just play through the hardware. Yes, it will technically pass through the card, however I think one way of describing it is a software "gate" which controls the physical signal flow. If it's enabled to pass audio through, it will, at 0 latency.

 

If you set up your DAW software in order to monitor, that software monitoring is not, and most likely will never be 0 latency. I know my two Delta 1010's, using the latest Beta driver are achieving a 1.9 I/O latency using Reaper x64 for Windows 7 Home Premium, and that's the best I've ever seen it.

 

It basically goes the same with most of the Delta line, the 44, 66, 1010, and 1010LT. The AP192 is a different beast altogether though.

 

Right now there is no input level control over the inputs in the latest drivers, however there is pan controls. In older drivers, yes, there are input level controls.

 

For sake of argument, if I were to configure the control panel to send inputs 1/2 directly to outputs 1/2, and the same for the other inputs and outputs (3/4 to output 3/4 and so on), the signal does not get processed digitally before being sent to the output. It only gets digitally processed if you're using the Software Return output, and you have it configured to run through DAW or other processing software before being sent to the output.

 

 

This is what I was trying to describe and he just reconfirmed it. You can have virtually zero latency with direct monitoring and only have to deal with the analog latency which is nill. The drivers used also show no latency multitracking when set for direct monitoring doing a ping test or otherwise.

 

I do know hat you mean by some of the manufactures not having any direct monitoring though, and the issues involved monitoring a processed signal especially through amp emulators. I cant tolerate that kind of latency. Its like listening to yourself play through an echoplex with the mix knob fully tweaked. landing directly on a beat takes guesswork anticipating where the note should be.

 

Heres the path the best of my knowlege. I havent seen the flow diagram to this setup but itys pretty easy to figure out with simple deductive reasoning and how things function.

 

 

[ATTACH]335202[/ATTACH]

[blue2blue]

OK... sorry to drag this out but I still wonder if there's a potential communication breakdown.

 

If we really want to get to the bottom of this, could you please ask them if, with the input routed 'directly' to the output if a) indeed, there is no AD/DA of that input cue signal through the unit's converters and b) there is the capability of blending playback from the computer software with the directly routed input cue through that same output?

 

IOW, is the 'directly monitored' input cue strictly analog and -- if so -- can you also route playback from the computer to the same output simultaneously, mixing that analog input cue with playback from the computer -- which would mean that the computer playback would go through DA and then be analog summed with the still-analog input cue?

 

(If so that's a nice feature -- and quite rare on a contemporary device of any price, from what I've seen.)

 

 

Also, could have been a 'slip of the keyboard' -- but the fact that the M-Audio rep wrote...

 

If you set up your DAW software in order to monitor, that software monitoring is not, and most likely will never be 0 latency.

... gives me pause. That 'most likely' could safely be said to be never.

 

And that makes me wonder if he might not be trapped in the common marketing speak mis-definition of zero latency (as being only a couple ms or so). Of course, he could have just been erring on the side of being extra-cautious. I suspect it's a habit one gets in dealing with many members of the public day in and day out.

 

 

BTW, I'm only a mod of the songwriting forum -- not a tech/recording oriented forum -- and we pride ourselves over there on not having to know anything about technology.

 

(But, yeah, I have sort of been around and I do sort of know my way around digital audio issues. I do get lost in some of the sampling theory math, I'll admit. But in practical terms, in the 14 years I've been running a DAW, I've gained a pretty good grip on the general process. )

[Bookumdano2]

For recording direct, I been using one of these lately. http://cgi.ebay.com/ART-SGX-2000-Multi-Effects-Processor-Pitch-Delay-Reverb-/310321010014?_trksid=p4340.m263&_trkparms=algo%3DSIC%26its%3DI%252BC%26itu%3DUCI%252BIA%252BUA%252BFICS%252BUFI%26otn%3D10%26pmod%3D230601452233%26ps%3D63%26clkid%3D176603834017312492

 

hahaha... I completely forgot that I have one of those too. Bought it new in 1989 or something based on a review and to this day, have not powered it up. The Fender XD seems cool at only a couple hundred bucks but maybe I'll take a closer listen to what I already have and then pop over to a GC or BestBuy to try the Fender out for myself to see how it compares.

[WRGKMC]

Also, could have been a 'slip of the keyboard' -- but the fact that the M-Audio rep wrote... ... gives me pause. That 'most likely' could safely be said to be never . And that makes me wonder if he might not be trapped in the common marketing speak mis-definition of zero latency (as being only a couple ms or so). Of course, he could have just been erring on the side of being extra-cautious. I suspect it's a habit one gets in dealing with many members of the public day in and day out. BTW, I'm only a mod of the songwriting forum -- not a tech/recording oriented forum -- and we pride ourselves over there on not having to know anything about technology. (But, yeah, I have sort of been around and I do sort of know my way around digital audio issues. I do get lost in some of the sampling theory math, I'll admit. But in practical terms, in the 14 years I've been running a DAW, I've gained a pretty good grip on the general process. )

 

He was speaking of the signal when its set for processed. You cannot have zero latency with a processed signal. Low yes, some computers are extremely fast, but never zero. Processing takesd time.

 

How low that latency of course is dependant on what you set the latency for, how fast your computer can process the signal, etc.

 

I use a 3G single processor and can get that down to around 10us for a single track. Its better at 100us to be safe. I have "no" reason to run it that way though.

I leave the latency set stupidly high so I can run 16 tracks with the maximum number of plugins without dropouts or digital static.

I think I have it set for 2048 or 3072 or something like that. If I toggel to the processed mode thats like a second or two delay.

Thats only a problem if I want to punch in or use the meters accurately because I use direct mode recording. Mixing its just an anoying time delay making audio changes.

 

One thing to note, the control panel is partially shared by sonar. Sonar uses the M-Audio ASIO drivers so when you togel the switch in the mixer view, that selects between processed and direct its actually accessing the control panel toggel.

 

Also when you open up audio options (or properties) in Sonar it opens the M Audio Controll panel. I'm not sure other interfaces intergrats this way when programs install but this is how its work with both the cards I have now and the previous Isis 8 track card I used to use and still have. Both worked the same way as I've described so its not unique to theis manufacturer.

[blue2blue]

Ah, OK, then, I think we've got that ironed out. Like I said, a great feature that would seem to preclude having to monitor through an outboard analog mixer to avoid cue latency on active inputs! (Note my own probably over-caution with that seem. )

 

Too bad that feature isn't widespread among other affordable interfaces. I'd love to be able to forgo monitoring through my analog board -- especially when we're going through a hot spell.

 

BTW, my MOTU doesn't have continuously variable input attenuation but it does have single attenuation switch to toggle between -10dBV and +4dBu and another software switch to give a 6 dB boost at the input. But, of course, no direct analog monitoring, unfortunately.

[WRGKMC]

Ah, OK, then, I think we've got that ironed out. Like I said, a great feature that would seem to preclude having to monitor through an outboard analog mixer to avoid cue latency on active inputs ! (Note my own probably over-caution with that seem . ) Too bad that feature isn't widespread among other affordable interfaces. I'd love to be able to forgo monitoring through my analog board -- especially when we're going through a hot spell. BTW, my MOTU doesn't have continuously variable input attenuation but it does have single attenuation switch to toggle between -10dBV and +4dBu and another software switch to give a 6 dB boost at the input. But, of course, no direct analog monitoring, unfortunately.

 

Yes it is a nice feature. I guess I'm spoiled by it. I can of course use an analog board too if I wanted. I could run the 16 inputs and outputs through a mixer but theres no benifit in doing so.

 

I do run the drums through a mixer to act as a preamp because the cards inputs are line level. For the vocals I use separate preamps or feeds off the PA mixer recording live. The guitars I either record from line outs on the amps or I can use a Low to High impediance transformer and connect them directly into the interface line inputs. The Line level preamps in the cards are strong enough to record direct like that with loud guitar amps. The control panel has 6db boost. It boosts the digital signal being recorded, not the preamp gain though.

I alsi have the guitar preamp effects units to record withand they're all line level so I'm good with those. I also have separate mic preamps and can record from those adding vocals too.

I eventually want to get one high quality preamp for that, but I dont know how much my voice will benifit from it.

 

I do have an 8 channel preamp coming in. May be in today as a matter of fact. I plan on using that for the drums instead of the mixer. I can use the mixer for the amp mics and DI feeds from tha bass if I want. I may try it, but my main theory still remains. The more unnessasary crap you put before the interface, the more analog degration occurs recording. I have no problem walking over to the bass amp for example and adjusting its line level send pot so it has an optimum meter reading on the daw for recording. Having a mixer at my fingertips would be handy, but you have to balance quality vs convenience.