# M-Audio Fast Track connection



## Dan Moroboshi (May 21, 2011)

My dear coleagues, please, consider that I'm very beginner in this area.:help:
I am investigating several options for room acoustic analysis.
Nowadays, my preference is at:
- Condenser Microphone: Behringer ECM8000;
- Pre audio interface: M-audio Fast Track, due to USB interface for windows 7 laptop and RCA outputs:sneeky:.
My questions is in relationship to connections of Fast Track:
1-) Is it possible to install the mic directly to port on Fast Track?
2-) If 1-) is possible, how to place it correctly?
3-) Is it preferably to use RCA connection instead of TRS for output to my receiver or Integrated amplifier? Longer cable is needed in this case.
Thanks for any help.:T


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## EarlK (Jan 1, 2010)

Dan Moroboshi said:


> My questions is in relationship to connections of Fast Track:
> 1-) Is it possible to install the mic directly to port on Fast Track?
> 
> *YES, the Behringer mic can be connected directly into the Fast Track mic preamp .#2, in the following picture. Make sure you turn on 48 volt phantom power for a Behringer mic.*
> ...



<> EarlK


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## Dan Moroboshi (May 21, 2011)

Thank you so much for your very complete and comprehensive answer! You're number one!

I believe that this component belongs to a cheaper set-up to measure the room effects in order to optimize it. In addition to that, fewer cables, connectors and other stuffs are needed. Is it true?


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## Cyber_Murphy (May 25, 2010)

Hmm.. interesting,, thanks..

I too am looking for a USB interface for my laptop and had stumbled across the 'Fast Track'..

How do these type of interfaces get 48v without an external powersupply? If it's DC-DC conversion surely that cant be good for noise??

Also, are these types of 'USB soundcards/Intefaces' "full duplex", Eg, can they send out a signal via the output while receiving a signal via the input, at the same time via USB ??

Cheers..
CM


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## AVoldMan (May 15, 2011)

Cyber_Murphy said:


> 1) How do these type of interfaces get 48v without an external powersupply? If it's DC-DC conversion surely that cant be good for noise??
> 
> 2) Are these types of 'USB soundcards/Intefaces' "full duplex", Eg, can they send out a signal via the output while receiving a signal via the input, at the same time via USB ??


I would have to second your questions. I am also thinking about the Avid M-Audio Fast Track for REW and recording purposes.

1) The Pro version has its own seperate Power Supply. Would it have less noise?

2) Does the "full duplex" function allow REW sound card calibration? Or am I not understanding something?

Does anyone have either of these models and be able to comment further?


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## jinjuku (Mar 23, 2007)

REW has full support for the Tascam US-144MKII. You may simply want to consider a product where it is a known quantity in regards to REW. I don't know where it stands SQ wise to the M-audio. I simply use it for room correction.


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## AVoldMan (May 15, 2011)

jinjuku said:


> REW has full support for the Tascam US-144MKII. You may simply want to consider a product where it is a known quantity in regards to REW.


I have seen that product. What I was interested in with the Avid M-Audio Fast Track was the included recording software "Pro Tools SE". Does anyone have a comment or experience on this particular recording software or other suggestions?


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## AVoldMan (May 15, 2011)

Dan Moroboshi said:


> I am investigating several options for room acoustic analysis.
> Nowadays, my preference is at:
> - Condenser Microphone: Behringer ECM8000;
> - Pre audio interface: M-audio Fast Track, due to USB interface for windows 7 laptop and RCA outputs.


I'm interested in whether you have purchased the above equipment or found something else. I seem to be following down the same track, but with additional home recording preferences. Any updates?


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## Dan Moroboshi (May 21, 2011)

For sure it is a very good option. However, it is also more expensive. Probably, the "quality" of the results from this Tascam model is very well, but alway it has its price.
"there is no free lunch".

There is no update. I'm sorry. I am stuck in the decision of what and how much cables I will need. You know, I am an audiophile and it is almost impossible to me to purchase a very simple cables. For me, they should be 'audiophile grade'.


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## anibal1966 (Sep 30, 2006)

I have taken this same M-Audio Fast Track mkII plus ECM8000 mic.
I am getting started just now, following the REWV5_help.pdf document.
As i have no SPLmeter, I have the following doubt regarding "SPL calibration"
- I have downloaded and installed the ECM calibration file
- I have followed instructions to set a reference SPL. At a "not to high, not too low -to my ears-" SPL of the sweep signal I have arrived to a reference SPL of 93dB (as read in the SPL calibration window).

Are this two actions compatible? Is this the right way to proceed?

May I proceed to measure then?


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## SAC (Dec 3, 2009)

anibal1966 said:


> ... Is this the right way to proceed?
> 
> May I proceed to measure then?




Maybe.



But at the risk of scaring a few folks, please stop and listen and explore a few additional suggestions that WILL result in an increased understanding and effectiveness in addressing whatever acoustical measurement issue with which you are concerned.

And we will start with "calibration" - as it is directly tied to what you are doing and what your goals are...

In answer to the above question, IF all you are doing is making frequency response measurements and you (Still!!!) think that they actually provide the degree of insight you need to analyze a situation, sure.

But if you are going to take advantage of the powerful time domain capabilities for the additional causal insight provided, the frequency and gain calibration are optional, while you MUST implement the easily accomplished hardware loopback correction whereby the output of one of the channels is fed back into the same channels input. (And configure this so that the software are aware of it). (And later you can discover how to do this with a software file.)

As all equipment has some degree of propagation delay - the time it takes for a signal to be processed, this time must be accounted for and subtracted from the measurement, other wise the measurement will be off by some amount - in some cases exceptionally large, in other cases small, but in all cases normally unaccounted.

I know many are preoccupied with frequency calibration and gain level calibrations, but I am curious as to why.

While I too ordinarily like to have all levels calibrated on principle, I wonder if folks really have any idea as to why they think they need this?

I now that this may seem to fly in the face of common sense, but have folks stopped and examined exactly what they are measuring?

Since when is an absolute gain level calibration in terms of SPL necessary? (if you can even really see SPL and not dBfs) Unless on requires a comparison to some 'external' reference there is little use for this capability. In fact, the myriad typical uses are simply for relative comparison, in which case we only care about the agreement or the difference between two relative levels. In which case we don't care what the meter reads, as long as it is consistent for both measures!

And as I have had the actual dubious honor of baby sitting a measurement rig overnight in a field dealing with the myriad spiders an dew (as such rigs have a habit of growing feet) and performing environmental Noise Level Analysis (NLA) that requires a FULLY calibrated and certified test set up -= as well as a software package certified for such use - I have never needed a calibrated gain level. And believe me, if you have been through this where the data is certified for admission as evidence in a court of law, the first thing a sharp lawyer will do is scrutinize the platform and software package employed - and if you are not using a compliant package, you will have wasted allot of time, and most likely your fee in doing so! Bottomline, there are uses for calibrated gain levels, but I suspect few here will ever encounter them.

Likewise frequency calibration. Although for those interested in this, for almost any measurement mic worth purchasing, individual calibration files are available or supplied by the manufacturer. In the event of less expensive mics, generic calibration files are generally available as well based upon an averaged sample of mics (as opposed to an unique individual calibration file made specifically for that individual mic).

But again, what are we comparing? Doe one expect that whatever the are measuring is properly referenced to a DC to gamma rays flat response? And as the frequency response of almost any source is so dramatically effected by boundary proximity and the superposition if direct and indirect signals within a bounded space, I would love to see the source that actually perform as most expect based upon some idealized free space mounting in an anechoic chamber! Thus more than a few should be questioning the adequacy of the usual supplied speaker documentation! Or at least asking what other measurements are more useful - and why...

Again, what are we comparing with what? If you assume an ideal behavior, well, what they say about assumptions is indeed true!


Thus, here are a few simple rules of thumb:

If you have a mic frequency calibration file, that is easy to load. 

If not, and IF the application requires one (FEW DO!), either pay the manufacturer or a qualified individual with a certified calibrator (available for about $300 and up) to generate it for you. Or buy a high quality calibrated measurement mic! While there are a few that seem to keep going into and out of production, about the cheapest you should be looking at for this category is an Earthworks M30.

I would suggest that if you need (as opposed to simply wanting) calibration, quit messing with a hobby mic and thinking you will hot rod it (you won't!) and invest in a quality professional measurement mic with is associated cost. Oh, and then worry about never dropping it!:yikes: Otherwise you will be fine.


And gain calibration...

Also, if calibrated gain levels are required, you WILL want to also seriously examine your computer and software package as well for its calibration process. None of these combination which provide for such capabilities are cheap! And if you are not intimate with that (or dedicated Noise Level Analysis (NLA) applications, you probably don't need it! Most do not. 

If you are dealing with voltage reference levels and digital saturation (dBfs) levels, great. This is another issue entirely.

But if you are dealing with with SPL gain levels, again, what are you doing and trying to achieve??? In typical measurements we only care about relative levels and the differences before and after or between two relative positional measurements. As long as we use the same setup for both or all measurements, the relative differences are consistent and we're done! 

And I love how folks assume that some $40 RS reference meter is itself calibrated! Generally at best, all you are doing is creating agreement between a standalone SPL meter and your computer based SPL meter. I wont ask why you need two of them when the computer based platform provides SPL, RTA and a beau coup assortment of other much more capable tools, but hey - I realize that many assume that they need an entire raft of frequency domain tools. That is, until they discover he implications of behavior in the time domain. And suddenly the assortment of SPL meters and RTAs are the first things left out of the bag - and for good reason.

Otherwise, if gain calibration is indeed required, go back and re-read the above and start assembling the budget necessary to invest in s certified NLA measurement rig and software and mic. Start saving now, as you will not like the price when you investigate such a platform. Oh, and if this is capability is appropriate, simply invest in the appropriate $300+ calibrator - generally capable of more than one frequency or level - which quickly bumps the entry price to ~$400+.

Thus, if one is doing time domain measurements (and lets inquire as to if not, why not?), you need to do the loopback hardware propagation delay correction. Its easy and simply requires a jumpier and a few software settings to make the programs aware of its presence. Not much more than setting up a mic in Yahoo Messenger or Skype. THIS IS NECESSARY. otherwise all of the precise time measurements are wonko.

So do as you will, but please become familiar with what exactly it is that you are doing, and please become at least as smart as your tools. As if you are not, its possible to get all sorts of utterly fascinating measurements that might cause you to do all sorts of fascinating things - and they will be meaningless, or worse, incorrect.

And the first of the assumptions that one might want to examine, is why one thinks that the frequency domain, which is derivative and not causal, is so important...and conversely, why the time domain perspective, which provides in depth insight in to the discrete causal elements is not being investigated - and thus actually acquire and use the tools that enable you to actually resolve those frequency domain anomalies some feel are so important!!! 


Epilogue:
I know at least a few are saying to themselves: "Gee, I just about got to where I had an idea about what was going on and here comes this wacko talking about all sorts of strange and confusing terms and turning my world upside down again!" And aside from the omitted colorful expletives, I am guessing that I got it about right... 

Right?????

Yes, the perspective is a bit different, and it involves unlearning a few assumptions, and embracing a few new ideas, but its NOT hard, and once you unlearn a few old habits, you will see that so many things that have been really complex problems will suddenly LITERALLY cease to be problems. In other words, if you address the fundamental relationships in the time domain, the environmentally based driver-driver and speaker-room anomalies that manifest themselves as problems in the frequency domain , simply disappear. Sounds crazy I know, but its true.

And in doing so you will discover the true advantage that tools with time domain capabilities, such as REW, offer. Not only can you resolve the issues in the frequency domain, but you will quickly discover that you can both understand and accomplish MUCH more! And that the increased understanding of what is happening, and of what you can literally fix, will be a substantial improvement over what you could do previously.


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## anibal1966 (Sep 30, 2006)

First of all, thank you very much for your kind & detailed answer.

My goals when starting with REW are two:
Learn about home acoustics (so your information is most interesting to me).
Optimise my hifi set, whose performance is highly compromised by suboptimal room &limited placement options.

So just a newbie with hobby intentions. At least I have no assumptions to unlearn.

I can understand that I don't need an absolute SPL measure, is the curve -relative values- that matters. I am concerned that the 93dB value departs so much from the reference in the REWhelp guide (80), just for clipping issues.

I have followed the guide step by step, soundcard calibration included. Is that the "harware loopback connection" test that is so important for time domain measurements? If not, I would be grateful for a link.


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## SAC (Dec 3, 2009)

The larger issue "to be unlearned" is the reliance upon the frequency response and the frequency domain in general as somehow being causal. This is reflected in the reliance on free field anechoic frequency response plots to frequency response plots of rooms and the myriad questions of "how flat" a rooms should be.

Regarding speakers, it is interesting that few demand accurate spatial polar dispersion information and driver alignment and power response data - as each driver typically does not radiate energy over the same spatial range. And seldom is the signal alignment of the various drivers revealed - issues that directly impact the degree of resultant polar lobing that increases with off-axis orientation. This (and other related issues such a spatial loading) directly impacts the behavior of a speaker in a bounded space. Instead we are generally 'satisfied' with small signal analysis and plots of an isolated, anything but real world, free-space anechoic measurements totally unrelated to a devices real world use. Thus, a knowledge of how a speaker will interact with a bounded space, being that such interaction is critical to determine the total actual response in a bounded space, is essentially left to chance.

Instead you address the _vast_ majority (all non-minimum phase relationships) of what appears to be an anomaly in the frequency domain in the time domain. Thus, understanding the concept of superposition and how this behaves in the time domain is critical. And understanding how this information is displayed in the time domain becomes fundamental.

Which rather begs the question as to why so many even bother with the frequency response rater than simply focusing upon the basic relationships as they are displayed with respect to time. Thus, for energy above the modal range, the ETC response is fundamental, not the frequency response. The extent to which this is not the case provides an indication of the degree to which old assumptions need to be 'unlearned' and new information embraced. 

It is to this issue that one must become familiar. This is where the 'quantum leap' during the last 40 years has occurred in acoustics, and as such it is quite new for most. And the implications are most profound. (And the degree to which this is not understood is illustrated in the recent discussion of whether Omnimic and XTZ are adequate measurement platforms - seeing as how they focus entirely on the frequency domain to the exclusion of any time domain behavior for use with regards to small acoustic spaces.) And awareness of the fundamental primacy of time based tools should render a discussion of such platforms compared to others such as REW (and others) that include quality time based tools as moot.



Regarding hardware propagation delay compensation - aka "loopback correction":

Every time based measurement platform, in order to be complete, must address hardware propagation delay. The process is the same for all platforms. Only the particular 'configuration screen' in the software will vary. Note, this is not "frequency calibration".

Perhaps the best illustration of the physical configuration is provided in the documentation for FuzzMeasure, explained and pictured here at this link.


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## gerchy (Aug 5, 2011)

Hi,

I too have the Fast Track and got the message that the highest level is too low.
I changed the channels - from right to left and the message was gone.










I uploaded the Behringer ECM-8000`s cal file, synchronize the SPL meter to 90 dB (the actual reading on the sweetspot was 75 dB) and I was able to make some quick measurements.
Any thoughts?
:innocent:


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## EarlK (Jan 1, 2010)

gerchy said:


> Any thoughts?


> Your soundcard loopback measurement isn't valid ( if this is what you used ) . This was in your .mdat file .

> ReRead the help files and then do another SoundCard Loopback calibration .


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## gerchy (Aug 5, 2011)

Ah yes, the loopback connection!
I guess 1/4” instrument input would do the job ...

There is still message that Level is low (when input channel is set to right) and that Level is high when input channel is set to left ... (gt-m2.mdat).

The measurements with loopback connection didn't give any error messages. (gt.m3.mdat).
Should loopback connection always be in place?


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