MACKIE ONYX 400F (audio interface)

[DemoKing]

I just saw one at Walmart with the TI chipset for $29...

[mige0]

Hello,

Had some nice time now to play around with this great mackie gear.
I took some curves which look really flawless, the squarewave response for example looks close to perfect.


1.)
Frequency range with 400F output wired to mic-in and mic-insert:
http://www.kinotechnik.at/pages/onyx/060101frequency.gif
"http://www.kinotechnik.at/pages/onyx/060101frequency.gif"

2.)
Simple THD measurement (sadly only the 16Bit setting works here)
http://www.kinotechnik.at/pages/onyx/060101thd.gif
"http://www.kinotechnik.at/pages/onyx/060101thd.gif"

3.)
Impulsresponse shown with 1kHz sqare on line-in dsp-routed to out
http://www.kinotechnik.at/pages/onyx/060101impulse.jpg
"http://www.kinotechnik.at/pages/onyx/060101impulse.jpg"



Some questions also came along:

4.)
Why is it, that only the 44kHz setting of the 400F seems to work "distortion free" when using the 400F as an extrnal D/A for the cd player connected by SPDIF?

5.)
It also appeares to me as if there are sort of timing problems resulting in audible distortion when the "internal clock" setting is chosen in such a combination?

6.)
Any suggestion on how to remain "SPDIF consumer" and also "SPDIF clock source" as the preferred settings when firewire is disconected and the 400F is switched off and back on?

For those who do accoustic measurement (like me from time to time) a more extended frequency range for the mic-in's would be nice.
In fact my trust into mackie engineering was the reason to buy this one, thinking that for sure it will match my mic's frequency response that is flat to 40kHz within 1dB.
7.)
Any improvements/ mod's possible there?



Thans for any explanations.
Michael

[harryjames]

Originally posted by DemoKing I just saw one at Walmart with the TI chipset for $29...

 

 

THANKS!! good to know.

 

i have a feeling my lappy is gonna "squeal" using the 4-pin firewire.

 

i like $29 bucks

 

Hey i am close to buying a MOTIF ES-6 but what do you people think of the Korg OASYS ?

 

seems pretty awesome although it kinda appears to be more aimed at live, repetitious songs; at least from the demos of Stephen Kay that i have watched. but i know with that kind of power, recording straight tracks within the sampler/sequencer should be child's play and it IS very intuitive AND the Korg uses same joystick controller that my i-3 has.

 

re-acclimating to mod wheels isnt my idea of fun.

 

;;cheers;;

 

Happy New Year

[MikeRivers]

Originally posted by mige0 Why is it, that only the 44kHz setting of the 400F seems to work "distortion free" when using the 400F as an extrnal D/A for the cd player connected by SPDIF?

The simple answer is that CD is only 44.1 kHz (and 16-bit). What's your test setup? What's connected to what, and what does the CD player have to do with it?

 

It also appeares to me as if there are sort of timing problems resulting in audible distortion when the "internal clock" setting is chosen in such a combination?

If you're feeding S/PDIF to the 400F and using the 400F's internal clock setting, the 400F isn't synchronized to the input data stream. You have to tell the 400F to use the S/PDIF input as the clock source. You probably also have to set it to the correct sample rate.

 

Any suggestion on how to remain "SPDIF consumer" and also "SPDIF clock source" as the preferred settings when firewire is disconected and the 400F is switched off and back on?

Not if those settings don't stick. There may be a "save as defaults" option somewhere on the Firewire control panel, or they may just expect you to set them correctly each time.

 

For those who do accoustic measurement (like me from time to time) a more extended frequency range for the mic-in's would be nice.

It's hard to see the detail up near the high end, but it looks to me like you're just starting to get a bit of roll-off, maybe 1 dB or so a few kHz shy of 1/2 the sample rate. This is as it should be. If you try to put anything higher than FS/2 into an A/D converter, you're violating the sampling law and you'll go to jail. I would think that as long as the distortion is sufficiently low, 96 kHz would be just fine for your mics with claimed response up to 40 kHz.

 

Interesting that on your noise spectrum test, there seems to be a peak at 6 kHz (6x the test frequency). It's 90 dB down, (about 0,003%) but it would be interesting to see if that's really a 6x peak or if it's a 6 kHz peak that's always there, with or without an input.

 

Can your test program measure IM (intermodulation) distortion?

[mige0]

hello Mike

First I'd like to thank you for your reply and Anderton for this special thread and in general for the idea of making pro reviews in this way as well as all the people contributing here.
Actually I was looking for a substitute for the disappointing Edirol FA-101 which I returned immediately few weeks ago. Ok it's not the same price as the Mackie but also not really cheap and the technical figures look good. But uuhh – it didn't even survive the first hour of listening.

On my search for alternatives I remembered that my Mackie VLZ mixers always work fine and so I came to the link to this review on the Mackie Homepage. I was glad to find here an in depth review covering almost any aspect I was interested in. After some hours of reading this made me decide to go for the 400F.

Now back to the issue.
As all my questions were a little bit mixed and packed in my first posting here, I'd like to break it into different parts and refer to the frequency response in this posting only.

The measurement software used, is pretty inexpensive and available under
http://www.audiotester.de/
"http://www.audiotester.de/"
(sorry Mike, not capable of taking IM measurement) and though it should support up to 32Bit I was not able to switch to 24Bit. This means all graphs are taken in 16Bit mode!
My guess is that the integration of the Mackie drivers into WIN XP is not quite perfect, as for example, I also was not able to find the WIN-standard mixers for the 400F. But this is something I would more like to address to the people of Mackie together with the non permanant console settings.

On the graph
http://www.kinotechnik.at/pages/onyx/60101frequency.gif
"http://www.kinotechnik.at/pages/onyx/60101frequency.gif"
you can see 4 curves.
The upper three curves are made by outputing the signal from the measurement software by the 400F and then connect a cable from this output to feed the signal to the insert point of the MIC-IN.
Basically this means, that the MIC-IN is measured without (!) its adjustable gain stage. I am not sure - because I do not have any schematics - but its most likely that from this point on its more or less the same circuitry than for the LINE-IN's (except for the balancing stages).
Every curve represents a different setting: 192kHz, 96kHz and 44kHz. It easily can be seen that there is only one fixed anti aliasing filter that meets the 192kHz requirement. The other curves just stop where they should have a steep roll off. On how this works for the settings less then 192kHz I didn't find the answer.

The lowest curve shows the behaviour of the complete MIC-IN. There the signal is wired not to the insert point but right to the port you normally would plug your MIC. (the gain is manipulated the way I could shown it on the same diagram). The setting here is 192kHz, as this is the most interesting one.
Here you can see that the –3dB point is around 6Hz at the lower end and 40kHz at the upper end of the frequency range. This is really not bad. only for acoustic measurements I would like to have this –3dB point somewhere at 60-80kHz. Maybe to change the appropriate capacitor is a modifying possibility. Maybe someone from Mackie can put some further light for me on this?


As you was asking for the MIC-IN gain in an earlier posting and I had my Minirator / Minilyser ready:
I measured about 80mV / -20dBu with full gain for turnig the red LED on.
the gain stage is around 60dB adjustable (if Anderton wouldn't be outside Europe I could make him an offer for Minirator / Minilyser / NTI Minstruments, used or new, but they run a webshop anyway
http://www.nt-instruments.com/X0-ASP-pLngCateId_88-pIntLevel_3-X1-default.htm
"http://www.nt-instruments.com/X0-ASP-pLngCateId_88-pIntLevel_3-X1-default.htm")
('')

greetings
Michael

next time about cd-player / spdif and distortion

[mige0]

hello Mike

to have a closer look to the THD behaviour of the 400F I made a clean setup and increased averaging to 50 measurements.

Again the the curves were made by outputing the signal from the measurement software by the 400F and then a connection was made from this output to feed the signal to the the MIC-IN.
In the two screenshots below (showing 1kHz and 2kHz respectively) you can see that 3rd harmonics are dominating and 2nd and some very high order harmonics are there as well but hardly can be detected.

http://www.kinotechnik.at/pages/onyx/060102_1kHz_thd_0db_FS.gif
"http://www.kinotechnik.at/pages/onyx/060102_1kHz_thd_0db_FS.gif"
http://www.kinotechnik.at/pages/onyx/060102_2kHz_thd_0db_FS.gif
"http://www.kinotechnik.at/pages/onyx/060102_2kHz_thd_0db_FS.gif"

Sorry Mike, for the missleading graph taken earlier.

To that point, I became interested in some further investigation to get clear whether this very small amounts of distortion come more from the MIC-IN or more from the OUT circuit.
On the two screenshots below you can see that harmonics get smaller when you decrease the signal of the D/A and they finally got lost in the noise floor of the MIC-IN when you further lower the OUT signal to about –20dB. Notice the DAW sliders positions. Giving less output right from the software has exactly the same effect.

http://www.kinotechnik.at/pages/onyx/060102_1kHz_thd_-10db_FS.gif
"http://www.kinotechnik.at/pages/onyx/060102_1kHz_thd_-10db_FS.gif"
http://www.kinotechnik.at/pages/onyx/060102_1kHz_thd_-20db_FS.gif
"http://www.kinotechnik.at/pages/onyx/060102_1kHz_thd_-20db_FS.gif"


This makes clear to me that the harmonics are coming from the D/A circuit.
I have to add, that the level was kept constant by the MIC gain manually. The yellow LED is turning on at a very exact –10dB which was very helpful in this as it also will be for setting the correct levels for mic recording.


So far what I could measure by ease, though its pretty time consuming. ('')
You mentioned that you are interested in IM distortion - the measuring software does not have this as an standard option but it can output dual tones at every level. If you tell me what frequencies at what ever levels you want to see intermodulated, I could perform it for you and post it next time.

+++++++++++++
added later:
Sorry, somtimes it would be better having read the entire manual first. ''
The measurement software is easily capable of performing IM and there is quite somthing on the help as well. Besides that, a veeeeeery quick response from the author helped me to perform best.
There seems to be no absolute fix rules (frequencie) but quite some standard guidelines for this kind of measuremant.


http://www.kinotechnik.at/pages/onyx/060102_IM_-10db_FS.gif
"http://www.kinotechnik.at/pages/onyx/060102_IM_-10db_FS.gif"

By the way, I also checked with the author about the quality of the sinus used.
I was told that it is calculatid on a 64bit basis which means this sinus produced by the software is virtually perfect and therefor has absolutely no impact on the graphs shown.
++++++++++++++


greetings
Michael:confused:

[DemoKing]

Michael- Its cool that you are doing all that even though it means nothing to me trying to decipher it
But... you tried another interface prior to it (Edirol I think?) What others have you compared, and what are your impressions from the 400F just listening back, and overall? Used the AD section yet?
Paul

[harryjames]

Originally posted by mige0 Actually I was looking for a substitute for the disappointing Edirol FA-101 which I returned immediately few weeks ago.

 

 

mige0; i came close to purchasing a Edirol FA-101 as well but after careful study of the "pictures" i realized there was a "toggle switch" on the inputs so you can only record using 2 of the 4 mic pres which became an "instant NO for me."

 

 

On my search for alternatives I remembered that my Mackie VLZ mixers always work fine...

 

 

mige0, since your Mackie mixer has worked well for you [as mine has as well] perhaps you should consider one of the new Mackie Onyx mixers with a Firewire card. These mixers allow you to send the entire computer mix back to the mixer ala 2 channel stereo with zero latency!

 

Some of us like outboard gear and are more productive than using the typical PC DAW setup. Sounds like it would behove you to take this into consideration in your upcoming music gear purchases.

 

;;cheers;;

[MikeRivers]

Originally posted by mige0 Actually I was looking for a substitute for the disappointing Edirol FA-101 which I returned immediately few weeks ago. Ok it's not the same price as the Mackie but also not really cheap and the technical figures look good. But uuhh – it didn't even survive the first hour of listening.

Funny about that. The Edirol interfaces have had good reviews in print and I haven't read of any sonic problems the few times these interfaces have been discussed on line. But when I asked my dealer about their new multi-channel Firewire interface several months ago, he said that the previous models didn't sound good so he wasn't going to rush to evaluate the new models.

The measurement software used, is pretty inexpensive and available under http://www.audiotester.de/ "http://www.audiotester.de/" (sorry Mike, not capable of taking IM measurement) and though it should support up to 32Bit I was not able to switch to 24Bit. This means all graphs are taken in 16Bit mode!

That will limit the lowest numbers of THD and noise that you can see, but other than that it won't hurt anything. It could be that the program doesn't work with the ASIO drivers and the WDM drivers are just providing basic support.

You might want to take a look at the Rightmark Analyzer to see if it works better with the Mackie drivers. It's really popular among those who test this stuff. Of course in either case you'll be limited by the accuracy of the output of the device you're testing since that's used as the input for the test, but at least it's a closed system.

The upper three [Frequency Response] curves are made by outputing the signal from the measurement software by the 400F and then connect a cable from this output to feed the signal to the insert point of the MIC-IN. Basically this means, that the MIC-IN is measured without (!) its adjustable gain stage.

This means you're sending the test signal in after the mic preamp. That's where the Insert Return is. You're not actually testing the preamp, you're testing the A/D converter, probalby essentially the same as the line input..

It easily can be seen that there is only one fixed anti aliasing filter that meets the 192kHz requirement. The other curves just stop where they should have a steep roll off. On how this works for the settings less then 192kHz I didn't find the answer.

It could be that it just never tries to send a signal that's out of the expected band. Back in the early days of DAT recorders, we could sometimes tell a difference in sound between 44.1 and 48 kHz, but with modern converters with oversampling and digital filtering, the difference is whether you're making the recording for audio (44.1) or video (48) purposes. And 88.1/96 kHz sample rates provide response well beyond that of concern by any but the most fanatic audiophiles and researchers. So knowing that the response is as wide and flat as it can be is sufficient. Often the rolloff for 192 kHz sampling is more gentle than the brick wall filters used with lower rates, so it could be educational to look at the shape of the filters,

I don't think there's a capacitor you can change to extend the bandwidth at 192 kHz. Most of that stuff is all done internally at the chip level today and the choice is made by the chip manufacturer. These chips are optimized for audio listening and not for measurement - that's why the Audio Precision computer interface box costs about ten times the price of a 400F, just for two channels. It's not built using dollar parts.

I measured about 80mV / -20dBu with full gain for turnig the red LED on. the gain stage is around 60dB adjustable

That's pretty reasonable. I found on the Onyx mixer with the Firewire card, that the clip light on the mixer came on less than 1 dB below digital full scale, and that you can increase the input level a bit more than 1 dB over "red light goes on" before the analog distortion starts to rise rapidly. I guess the 400F is probably about the same, but it's worth a check.

[MikeRivers]

Originally posted by mige0 I became interested in some further investigation to get clear whether this very small amounts of distortion come more from the MIC-IN or more from the OUT circuit. On the two screenshots below you can see that harmonics get smaller when you decrease the signal of the D/A and they finally got lost in the noise floor of the MIC-IN when you further lower the OUT signal to about –20dB. Notice the DAW sliders positions. Giving less output right from the software has exactly the same effect.

It's probably fairer to say that rather than the distortion coming from the D/A, that it comes from the analog output. Effectively it's the same thing since you can't separate them, but it rasises the question of wheter to point the finger at the analog or the digital designer. Perhaps where the problem liies is not in the choice of the chip or digital design, but what's between the D/A output and the line out jacks. But it really doesn't matter since there's nothing you can do about it but (as one user did) bypass the analog output chain and use an external D/A converter. From the THD graphs, it doesn't look there would be anything that would make the sound unlistenable (as at least one person has described it). This is why I'm curious as to what else is going on.

the measuring software does not have this as an standard option but it can output dual tones at every level. If you tell me what frequencies at what ever levels you want to see intermodulated,

The classic test that used to be standard for hi-fi equipment was someting like 60 Hz and 6 kHz mixed in a 4:1 ratio. That seemed to be pretty good for quantifying "toob" distortion but doesn't mean much for digital systems. I'll see if I can come up with something better for you, but you might try 5 and 6 kHz, and 15 and 16 kHz in the meantime just to see what you can see.

 

[Later] - Here's what I was thinking about, something that Paul Stamler wrote about the sort of testing that he does when looking for reasons behind stuff he hears that he doesn't like: (from Paul Stamler in a rec.audio.pro posting )

 

 

Two tones at around 19kHz, for example (I use 19 and 19.5) can reveal all kinds of misbehavior on the part of equipment, particularly solid state and more particularly digital stuff. So can three tones (I add a 9.6kHz signal to the previous two). These tests are useful for sleuthing out problems at the top end, because the test signals and at least some of the distortion products lie well within the passband. You find all sorts of stuff -- if the system just produces a 500 Hz difference signal, that's one thing; if it also produces 1kHz and 1.5kHz that means a more complex transfer function. And if there are IMD products at 5.1, 5.6, and 6.1kHz, then there's intermodulation going on with a 44.1kHz sampling frequency. Useful tool.

 

 

By the way, I also checked with the author about the quality of the sinus used. I was told that it is calculatid on a 64bit basis which means this sinus produced by the software is virtually perfect and therefor has absolutely no impact on the graphs shown.

But still, the actual test is only as good as the purity of the analog output that you're feeding into the path under test, and this is what's in question, at least by some. I assume that you see an essentially unreadable THD on the Minilyzer when you connect that to the analog output and look at the essentially perfect since wave generated by the computer?

[composer]

Originally posted by DemoKing Regarding the needing-more-midi ports problem I have... my drum module has a midi in, out, and thru... can I plug the out from the module into the 400F, and plug my keyboard into the thru and not have to screw around with changing plus every time? If so, that rocks. I'm assuming thru just passes signal on, I'm not very knowledgable on midi implementations....

Well I have the Blue SKy desk and the way it works is that you connect the other three speakers via the SUB, so somehow you can send the signal via SX to the correct speakers as opposed to using each output on the 400F. I'm not exactly sure how this concept works yet.

With NAMM coming up, it will be interesting to see what will be coming OR NOT, of course anything at NAMM will be 4-12 months before delivery.

[mige0]

Originally posted by DemoKing .... and what are your impressions from the 400F just listening back, and overall? Used the AD section yet?

paul – well, beside the very good specs, "just listening" is amazing and something like learning new what you already know: your equipment and your records. Sorry AD section not really used yet – judging from the AD section graphs, I am looking forward to the nice things left to discover...

Originally posted by harryjames mige0, since your Mackie mixer has worked well for you [as mine has as well] perhaps you should consider one of the new Mackie Onyx mixers with a Firewire card. .... Sounds like it would behove you to take this into consideration in your upcoming music gear purchases.

harryjames – yes, I was considering that seriously. But mobility is still something I do not want to miss, and compared by size and mass the onyx mixers lost by far.

Originally posted by MikeRivers I assume that you see an essentially unreadable THD on the Minilyzer when you connect that to the analog output and look at the essentially perfect since wave generated by the computer?

Mike – right, such small amounts of distortion are about ten times beyond the limits for THD measurement with the Minilyzer.
By the way, the sensitivity of the MIC-IN (with the use of the XLR connectors) is around 8mV with full gain turning the red LED on (80mV as stated above, only in "LINE-IN" mode on the TRS phone jack)

When checking the different options of feeding input to the channels 1 to 4 something interesting can be observed. Depending on being used as High-Z, Line or MIC Input, the 2nd order harmonics changes significantly. 3rd and higher order harmonics stay basically the same which are assumed to be from the output anyway.

MIC:
http://www.kinotechnik.at/pages/onyx/060103_XLR_1kHz_thd_-10db_FS.gif
"http://www.kinotechnik.at/pages/onyx/060103_XLR_1kHz_thd_-10db_FS.gif"

LINE:
http://www.kinotechnik.at/pages/onyx/060103_TRS_1kHz_thd_-10db_FS.gif
"http://www.kinotechnik.at/pages/onyx/060103_TRS_1kHz_thd_-10db_FS.gif"

High-Z
http://www.kinotechnik.at/pages/onyx/060103_high-z_1kHz_thd_-10db_FS.gif
"http://www.kinotechnik.at/pages/onyx/060103_high-z_1kHz_thd_-10db_FS.gif"

As can be seen the MIC-IN as well as the High-Z-IN add a pleasing amount of 2nd order harmonics.



greetings
Michael

[cgoobes]

How can this unit possibly exist without having the ability to switch between -10 and +4? This is completely rediculous. My M-audio Delta card that cost me 150 bucks has an amazingly eazy to use software control panel that switches these levels at any time with the click of a mouse.

LISTEN UP MACKIE>>>> GO LOOK AT M-AUDIO DELTA SERIES AND SEE HOW SOFTWARE CONTROLS SHOULD BE
Pair that with your converters and your interface and I wont have to spend 600 dollars more for a stupid fireface just so i can switch between pro and consumer line level inputs.javascript:smilie('')
mad

[the stranger]

Originally posted by Anderton But not so fast. Carbon-zinc and alkaline batteries have very different internal impedances, and when used with older effects that had shoddy power supply rejection, the battery type COULD make a difference in the sound.

I can here a difference between carbon-zinc and alkaline batteries. I can also hear d/a converters chirping. High pitched noise from crts. I hear all kind of strange stuff that most mortals don't hear.

And I have to admit that I'm very interested in the sidetrack concerning the d/a conversion. As noted, I'm very sensitive to high freq information.
I tend to get ear fatigue very easily.
And one of the most annoying experiences I have had with audio dealt with some crappy d/a conversion.
---

As for the review...this is the most comprehensive look at a piece of gear I've ever read. Great concept and application.

And I'm still only on page 8 or something.


One thing I would like to see is either another thread or maybe some exxpansion of this one to include the 1220, 1620, and 1640. To me, these appear to be exactly what I would want. The combination of analog mixer with firewire output is an idea that is way past due. It's amazing that digital mixers didn't start this way (maybe they did?).

I'm seriously considering one of the above. I have been doing audio on the computer for some time, but have yet to venture into the interface world.
Which is a good thing because the options and choices in this area have grown exponentially in the last couple of years.

Sorry for the long winded post that doesn't add much.

Thanks to the pros for the education!

[DemoKing]

Ok, got my 400F today.
Downloaded the drivers from the site this morning. Went to install the drivers... "you must have service pack1 or better to install this software"
Fine. Damnit, fine. Ok.
Drive all the way back home (no modem in DAW) to download service pack 1 or 2, and it turns out, due to someone that needs to be tortured and killed TONIGHT, that I cannot download those things. It wants to do it for you and install them on the machine. No option whatsoever of saving it to the desktop and making a copy and running off.
It does, however, oh thank the Lord above, give you the option of downloading the netoworking version of it, that is over 130 MB!!! In other words, in dial-up terms, about 8 hours for me. Thats if I dont lose connection in that time, or need to use the phone. So now what? Mass homicide spree? Smash everything I own in the street? Kick the dog? Beat the kids?
I'm waiting for someone to call me back that MAY be able to do the download for me in the next 48 hours. In the meantime, no listening test, no nothing.

[Anderton]

First of all, if you use anything related to USB, and particularly USB over MIDI, you'll kick yourself for not having installed SP1 earlier. There's a reason why so many programs require it for proper operation.

The dial up thing KILLS for this update. But if it installs on your computer it takes almost as long. Go to anyplace with broadband and bring a 256MB USB memory stick. Download to the stick. You should be able to do this at Kinko's or a hotel's business center (watch for viruses, though).

You didn't hear this from me, by the way, but there have been rumors of people installing the 400F with standard Windows XP and surviving

[DemoKing]

I got lucky. Borrowed someones Dell backup disk, that ususally wont work with anything but a Dell. It has Sp1a on it. I did the "upgrade" option, and it asked for a serial, I put in my current XP serial number, and it worked and didnt make me call for validation and {censored} again. The drivers installed in like 1 second. Works great so far.

So, 1st impressions... just a quick listening test THROUGH my board because I need a pair of 1/4 trs to MALE xlr to go straight to the monitors, the top end sounds GREAT. The space and clarity is much nicer, and things seem to sit outside the speakers more. I will post something more comprehehesive after a few hours with Sonar and recording some.
My last 3 years have been on a Delta 1010, so thats my comparison.

[cgoobes]

hey Anderton

Can you please tell someone important at mackie that ive got 700 dollars that will have to be spent on something else unless they can add some provision for consumer grade equiptment on the line inputs?

Ive got a 70's era studiomaster console that sounds sweeter than sweet... but the outputs are designed to feed consumer level devices.

I've talked to and emailed a couple of the tech guys at mackie and they say that its an issue they arent working on.

The m-audio delta series does it through software. I can't imagine it would be difficult.

PLEASE??

[DemoKing]

Slap an RNC between it and crank up the gain.

[MikeRivers]

Originally posted by cgoobes Ive got a 70's era studiomaster console that sounds sweeter than sweet... but the outputs are designed to feed consumer level devices. The m-audio delta series does it through software. I can't imagine it would be difficult.

This is the sort of thing that you run into when trying to use a 1990 console with a 2005 interface. The world of audio interfacing doesn't stand still. Think of all the money you've saved by not buying a new console in all this time.

If you only need two channels of level conversion, much as I'm hesitant to recommend something this cheap that's in your prescious montior path, the Samson C-Convert will fill the bill. If you need to convert 8 channels from -10 to +4 (one way only - the knobs on your console probably go down far enough so that you can feed it with the +4 outputs) Aphex makes a nice unit.

You can't continue to live in a cave for the rest of your life. Join the "professional" revolution.

[DemoKing]

So far, not much to say as I haven't had a chance to give it a run and put it under pressure.

So far though, no "annoying whine" that a few people have mentioned. I wonder if this isn't a problem amongst other equipment or certain combinations thereof.... grounding issues maybe?

Listening to about 4-5 songs from different bands/projects, I must say that I can hear the difference between the 400F and the Delta 1010. Its a pretty obvious difference. Not a huge difference in my untreated room, but enough to say "ahhhh" The high end is very, ummm, clear and present, if that makes sense to you. Not present in a boosted 12k kind of way, but present in that things are more clear, open, "clean". The music seems to be more outside of the speakers now. More 3D-ish.

The drivers loaded up in just a few seconds flat and within 1 minute I was passing audio. No problems or hangups (aside from having to install windows SP1 to get the drivers to install, but I guess I should have done that 3 years ago.)

The box feels sturdy enough, looks nice, and is surprising shallow. Its probably like 4 inches back into the rack. Come to think of it, I guess the Delta is about the same.

Turn off your monitors before you turn off the 400F though unless you want to buy new tweets.

More to come when I really get huffing with it in the next day or two. Anyone else had thier hands on one yet? Craig Anderton is doing a very good review over at his forum too.
__________________

[DemoKing]

-Much later post than the last one, even though at the same time right now, these 2 are a day apart-

Allright, I've been installing Sonar 5 P and DFHS all day, and thats kept me busy.
The WDM drivers would not work for me with Sonar 5. I tried and tried. THe ASIO drivers worked immediatly. But, with ASIO, the latency slider doesnt seem to do anything. Normal?
Things seem "ok" so far. I have a couple of issues to work through. FOr instance, the little music it plays when you shut it down. It has some digital static in it. Not terrible, but there.
Also, I'm not sure if Sonar is smart enough to get around this, but I record enabled 10 tracks, set all 10 to the same input and played guitar though the bos. With all 10 enabled (only one being onitored, the rest with the input monitoring turned off) there was static. Slowly turning off all the record buttons down to about only 2-3 of them on got rid of the static. Any idea why it did this?
MOre to come, I have a LOT to learn. In the same 2 days I've gotten the 400F, a Yamaha RM50, Sonar 5, DFHS, and a headache.
The DI on the 400F seems to sound very nice and clean.
More to come.

[cgoobes]

Originally posted by MikeRivers This is the sort of thing that you run into when trying to use a 1990 console with a 2005 interface. The world of audio interfacing doesn't stand still.

 

 

Hey Mike...

its a 70's board, not a 90's, and its really not a matter of saving money... this board was not cheap. It was designed to feed a certain catagory of tape machine with its 8 busses.

 

The issue here is not that my board is crappy... its that this is the only device that i have seen in this catagory of product that doesnt have this specific feature. Every competing product below and above its price range seems to have this feature... In fact, currently i have a MOTU 896HD (which i am very dissatisfied with for lots of reasons), but it has a +4 / -10 switch on every channel.

[cgoobes]

If you need to convert 8 channels from -10 to +4 (one way only - the knobs on your console probably go down far enough so that you can feed it with the +4 outputs) Aphex makes a nice unit. You can't continue to live in a cave for the rest of your life. Join the "professional" revolution.

Thanks for the suggestion on the aphex unit...

Unfortunatly, it has RCA ins, and XLR outputs... neither of which help in this case.

I really don't see how this is a matter of professional vs non professional. Really this is just a matter of a BIG company forgettting a small detail and not wanting to spend a small amount of effort to fix it. If you read back through this forum, there are plenty of other people who have expressed concern that the line level inputs do not have enough sensitivity. This could probably be dealt with by mackie in the software control panel. But it probably won't happen. Its just unfortunate that when I get rid of the 896HD, I will be having to spend more to do the same thing rather than less.

[MikeRivers]

Originally posted by cgoobes its a 70's board, not a 90's, and its really not a matter of saving money... this board was not cheap. It was designed to feed a certain catagory of tape machine with its 8 busses.

Ohhhh! Vintage! But you hit the nail on the head - it was designed to feed a certain category of tape machine, and that's not what you're looking to use it for. And I never meant to imply that it was crappy, it's just not plug-in compatible with the 400F, but there are ways that you can deal with that if you want to continue to use the board.

By the way, some boards of that vintage had switches to select -10/+4 levels. Why pick on Mackie? Why not complain to Studiomaster?

this is the only device that i have seen in this catagory of product that doesnt have this specific feature. Every competing product below and above its price range seems to have this feature... In fact, currently i have a MOTU 896HD (which i am very dissatisfied with for lots of reasons), but it has a +4 / -10 switch on every channel.

If you look hard enough, you can find something wrong with everything you look at.

Thanks for the suggestion on the aphex unit... Unfortunatly, it has RCA ins, and XLR outputs... neither of which help in this case.

So what does your board have, RCA in/out? TRS? XLR? Is your problem with the Aphex that you'll have to buy or make new cables too? The idea of the Aphex is that it boosts output levels that are too low. If your board has -10 dBV bus outputs, the Aphex box will boost those to ample level so that you can slam the meters on the 400F, but, yes, you'll need cables that go from the XLR outputs of the Aphex to the 1/4" TRS inputs of the 400F.

As far as sending the 400F outputs back to -10 dBV inputs on the console, that's no problem. Just turn down the input trims on the board or the output level of the 400F. Worst case, if there's not enough range on the control, you can put a 10 dB pad in line with the outputs. You can build this into a 1/4" plug with just two resistors. Again, connectors may not match the cables you have, but that's no big deal.

I really don't see how this is a matter of professional vs non professional. Really this is just a matter of a BIG company forgettting a small detail and not wanting to spend a small amount of effort to fix it. If you read back through this forum, there are plenty of other people who have expressed concern that the line level inputs do not have enough sensitivity. This could probably be dealt with by mackie in the software control panel.

I never liked the "professional" and "non professional" or "semi-pro" labels, but that's the way the industry has evolved. Back in the '70s, one of the ways that manufacturers could make gear as affordable as it was back then was to use a lower operating level. Today that's not necessary. And while we hear from the few coming here with concerns, there is a LOT more equipment in use today than uses the nominal +4 dBu operating level thnan the -10 dBV operating level. A company can't go on supporting old technology forever. Those who do are either charging more for it or are giving you less of other things that you want. You don't get something for nothing.

As far as a simple change in the software control panel, that's not the way to do it. The only way to boost a low input level in software is to multiply it, along with all the noise that comes along with the signal. What there needs to be is greater input sensitivity at the input of the A/D converter, and that's a hardware thing. They could put gain (hardware) at the input, but that would be extra cost and extra noise for those who don't need it.

Here's an example. I have a Creative Labs Nomad Jukebox 3 to which they "added" mic input capability with a firmware update. It puts 30 dB of gain after the input stage (which is what Mackie could do in software) but in the mic-in mode, the input is noisier than a cassette. So I use a preamp (a Mackie mixer, most often) when I want to use it with my mics.

This is a straightforward decision process - you choose what you can best work with. If it's too much trouble for you to deal with the level differences and/or connector types, pick another interface (but you'll probably pay more). If you acquired a vintage LA2 limiter (it has nominal +4 dBu in and out, and on barrier terminals, to boot) would you complain to UREI that it didn't accommodate the -10 levels on your console and you couldn't set the threshold low enough? Or would you adapt it and use it?

Keep your board if you like it, but bring it up to date so that you can use it with equipment that you can buy today.