MACKIE ONYX 400F (audio interface)

[Brittanylips]

Originally posted by DemoKing Hey Brittanylips- I think that in some cases, things can definately be better for less. Now with coverters, and this coming new age, I think that with the grade of converters in the 400F, if the surrounding circuitry is decent, they could be better than a new apogee. Its not unheard of. On another note, considering the cost of components in the apogee, do you think they have twice as much money in a 2 channel converter box as Mackie does in a 20 converter box with firewire, drivers, a mixer and 4 preamps? Hell no they dont. They are probably using the same converters in fact. Wouldnt surprise me at all. Are they worth 20 times the price? Nope. Could they sell them cheaper and broaden thier audience and sales? Yep. You said they couldnt, I disagree. They sell for that much money cause people buy them. Pull the market out from under thier feet and then you have $600 Apogee boxes, brand new. I'm gonna buy me a 400F this week and compare it to my Delta 1010 and see how it fares. Peace Paul

 

 

Hey, whatever works for you!

 

I've been away for the past few days and am in a bit of a rush, so I'll just add a few thoughts and then catch up on all this down the road.

 

Two quick points:

 

One, it costs more to produce high performance converters than low performance ones, just like it costs more to produce high performance cars than low performance ones. It would be great if that weren't true, but it is. The profit margins for Apogee are not as great as you might think, and they really can't slash their prices down to Mackie's.

 

Second, the same component in two different pieces of equipment sounds different. Again, you'd hope not, and the tech revolution constantly raises performance as it lowers price, but all is not equal. Old-tech analog components (from wiring to power supply) are still part of the equation, and they don't obey Moore's Law.

 

When the MOTU HD192 came out, I recall that MOTU publicized the fact that it used the same converters as the Digidesign 192 I/O. However, as much as MOTU's HD192 was a bargain and sounded great, it didn't sound as good as the 192 I/O.

 

-Peace, love and Blips

[blue2blue]

Originally posted by composer I have read about the brittle ness as several other forums including Mackie....however these posters at Mackie are new users, so this leaves me suspect. However, at gearslutz, these users are long time members. ...

I like GS, but there are some deep cultures of bias, there. It has to be remarked that there are a lot of people talking, there, but not all that many knowing -- depending on the forum.

But there are some real heavyweights, too. Including some legendary figures. (Hell, Angelo's over there all the time. ) Overall, it's a great place, but think of it as a wiki without even social rules... what's the phrase? Caveat lector? Reader beware.

[MikeRivers]

Originally posted by Batters If all the expensive units are using the same chips for the converters; then isn’t the accuracy of the digital clock the most important factor to consider next in this debate ?

That's important, too. But there are many other design factors, and that's where the "craft" comes in. These days it's possible to make a pretty darn good A/D and D/A converter set by buying some decent quality chips and powering them up, remember, half an A/D or D/A converter is an analog design. The things that make great analog designs great apply just as much to converters as to preamps or power amps or mixers.

Assuming that small budget studios won’t be able to afford a dedicated Master Digital clock unit, and use the 400F as the master clock reference ? If I’m not mistaken, no one has mentioned the 400F clock accuracy in this thread yet ?

The purpose of a master clock is to synchronize several units that have to work together in a system. If a single device sounds better with an external word clock than with its internal clock, that's a sign of a poor design (and, admittedly, some older and funkier products do indeed sound better with an external clock than with their internal one).

 

I have used the Wavetek 110 function generator (that's gotta be at least 35 years old) from my shop as a variable frequency word clock for my HDR24/96 when I've needed to vari-speed a track and I don't find that it degrades the sound - a real credit to the converters in the Mackie AIO-8 analog I/O cards. If that ain't a crummy word clock, I don't know what is.

 

A master clock can make a system work, but (claims of the Big Ben aside) there's no reason why it should make well designed digital equipment sound better.

[MikeRivers]

Originally posted by mandoman Some people are reporting on the Mackie board of low gain on line level inputs 5-8. Mackie specs those to 40db of gain, I would think that's enough to cover most prosumer -10db devices and even some consumer devices. Please include a low down on those inputs when you get a chance.

Funny, we're reading the same posts on "that other" forum. (and the one about the headphone extension cable, too) The way the input sensitivity is specificed for the 400F is kind of ambiguous. Knowing how much gain is available at the line inputs (or mic inputs for that matter) is of no value if you don't know the basic sensitivity of the A/D converter.

 

Craig certainly has the tools necessary to measure how many volts, dBu or dBV it takes to reach full scale at the maximum and minimum setting of the input gain controls. That would be useful information.

 

My guess is that at full gain, -10 dBV into a line input will give a record level in the ballpark of -10 dBFS. That should be enough for real "prosumer" studio gear, but, as I pointed out on the other forum, real consumer gear (he was using an iPod as a sounce) these days typically has a nominal line output level closer to -20 dBV, and doesn't have as much headroom as the -10 dBV studio gear of days of old (geez, has it really been 30 years?).

 

The reason why the -10 dBV operating level was established was so that semi-pro studio gear could operate with 20 dB of headroom using cheap ICs and cheap power supplies. But with today's low power portable devices (and the installed things with which they have to be level-compatible) and popular music being compressed so that very little headroom is required, designs are less adaptable than they used to be.

[freen-bean]

Very good of Dan to address our questions.

Originally posted by Dan Steinberg The MCU would probably not be available to the DAW at the same time, you'd most likely have to choose one or the other. While your banking suggestions is a good one, keep in mind that for that to be possible, it would mean that each software company would have to go back and re-do their MCU implementation to allow for this. I don't see them doing all this extra work to help us sell a certain piece of Mackie hardware better. !

I could be completely wrong, but it seems to me that this could be achieved without requiring DAW developers to rewrite their MCU implementations. Could there not be an MCU firmware update that would add an additional mode of operation, such that hitting a button or combination of buttons would toggle whether the MCU were controling the DAW or the interface mixer function? Does MCU DAW control use the entire range of possible MIDI channels/controllers? Is there any data that could be transmitted by an MCU that is not used by it's DAW control implementation? If not, what about some kind of middleware app between the MCU and the DAW that would allow one to control whether the MCU talks to the DAW or to the DSP mixer control software, through which the MCU could control the interface's mixer function? Technically, I suppose this wouldn't be simultaneously controlling both--but easy mode toggling is just as good for what I'm talking about.

Again, thank you for addressing these questions. And you've got a good point about 64 sample buffer settings providing low-latency already. I can't always run that low, though.

[Anderton]

>

Actually, I'm sorry to say I don't have the tools...haven't had a functioning lab for about 10 years, although I still have a Tektronix scope. I hope to re-assemble something decent, I need to get a function generator and a few other things...

[Anderton]

Well, I've been using the 400F as my primary interface since this thread started. I must say that it hasn't misbehaved, but there are few things you've asked me to check that I haven't done yet. One is to try it with my laptop and see if I can get the famous "whine," which so far has thankfully stayed far away. The other is the long headphone cable, and the line levels.

SI plan to wrap up my portion of the review in the next couple days with some final conclusions and a summary of what I've found. However, that doesn't mean the end of the thread -- it will stay around as long as people have something to discuss.

I will be starting another Pro Review soon, though, and I think y'all will find it interesting as well. As always -- thanks for your participation! This has been a gas.

[MikeRivers]

Originally posted by Anderton Actually, I'm sorry to say I don't have the tools...haven't had a functioning lab for about 10 years, although I still have a Tektronix scope. I hope to re-assemble something decent, I need to get a function generator and a few other things...

Just do what I did - get Neutrik to send you a Minirator and Minilizer for review, then take a looooooooong time to write the review.

[PianoMania]

Got my 400f yesterday. Upgrading from an Mbox with imac G5, 2gig. everythings working great, no whines or problems. Just pluged in and started using. I record primarily acoustic grand piano and the upper end is definately clearer. This thread assited in my choice and is much appreciated.

[blue2blue]

Okay... I'm actually still back on page 4 or 5 on my thorough reading of this thread but I've just scanned the back two pages or so...

I haven't read all the "whine" issue stuff but it suddenly occurred to me that I ought to comment on my own MOTU 828mkII "whine issue"...

I find that, with my MOTU 828mkII connected by FW to my laptop, the laptop's audio plugged into an input on my everyday listening setup (a Yamaha 'prosumer' receiver and some NS10m's) and any of the MOTU's analog outs also plugged into (another input of) the Yamaha receiver that I get an intermittent whine/electronic 'whirring' -- basically whenever there's any mouse action on my computer.

All I have to do is disco the audio output of one or the other to the Yamaha (which also has a TV with its questionnable audio wiring and grounding plugged into it, just to make the soup a little spicier) and the whine goes away. (Strangely, disconnecting the FW between the computer and the MOTU does NOT cut the noise.)


Anyhow, I figure it's some kind of ground loop issue, complicated by the panoply of digital and analog signals in a laptop (not to mention the inclusion of a TV in the overall picture)...

FWIW...

[Anderton]

That makes a whole lot of sense, given that it seems to be intermittent, product independent, and difficult to pin down...almost the textbook definition of a ground loop problem!

[blue2blue]

Originally posted by Anderton That makes a whole lot of sense, given that it seems to be intermittent, product independent, and difficult to pin down...almost the textbook definition of a ground loop problem!

You forgot the other classic symptom of ground loop problems... the afflicted engineer's head spinning around and around and split pea soup flying out his mouth...

[amplayer]

This may be a silly question:

Why is there such a huge difference between the 87 dB S/N and the 129 dB EIN in the specs?

I realize these are 2 completely different specs, but with an EIN of a whopping -129 dB, I don't get how ordinary S/N should only be 87 dB. Where is all the noise coming from?

[MikeRivers]

Originally posted by amplayer Why is there such a huge difference between the 87 dB S/N and the 129 dB EIN in the specs? I realize these are 2 completely different specs, but with an EIN of a whopping -129 dB, I don't get how ordinary S/N should only be 87 dB. Where is all the noise coming from?

You pretty much answered your own question - they're two different measurements. EIN is actually a computed number and represents the ideal noise level contributed by the input before you add any gain. S/N is the difference between the maximum possible output level and the noise present under (presumably) known conditions.

EIN is the measured noise output minus the gain, It's really only significant for the first input stage, but, for marketing reasons, seems to be getting attached to mic preamps (the whole box) because it's a nice low number. If the EIN is -129 dBu and you have 60 dB of gain (as measured by conventional means), you have 129-60 or -69 dBu noise just sitting there at the output with no input. That doesn't look very impressive, so they have to come up with a different way of saying it that doesn't look so dreadful.

Add 20 dB of headroom to that and you have: a noise floor of -69 dBu and a maximum output of +20 dBu, giving a S/N ratio of 89 dB. Close enough?

It really isn't all that simple because you're really comparing apples and oranges, but that's sort of the way it works. You can make the S/N ratio look better in most cases by simply reducing the gain because you're amplifiying the input noise less and the noise output isn't swamped by input noise.

The really important specification is the one that manufacturers don't publish - how much noise comes out under realistic operating conditions. Actually, this is test data and not a specification - a spec is something that you start with and then design and build to meet it, or figure out a way to measure it so that it looks like you're meeting it.

[UstadKhanAli]

Hello Mike!!!

[Anderton]

Yeah, he's handy to have around, isn't he?

[harryjames]

First, thank you Mr. Anderton and Mr. Steinberg for your insightful technical review on the new Mackie 400F. I have no recording gear whatsoever and have been researching and haggling over various equipment and ways to accomplish what I am after. What surprises me in ALL these posts is nobody discusses Tracktion2 at all??!!! I have tried the demo and it appears to be as good a contender as any DAW. They all suffer the same latency issues regardless of hardware. Let's hear about the Tracktion2 tests using the 400F, afterall, it DOES come bundled with the 400F. FYI, I am considering using a 400F, Tracktion2 with a pair of Yamaha HS50M NFM's for reference and mixing. I'm still kicking around which large diaphram mic. $300 doesn't sound like $2000 grand. Also, with all this latency, including using Tracktion2 on my P4 3.2HT 800FSB with XP stripped downt to a flying gas can; latency really turns me off; to the point I am strongly considering buying a Yamaha AW2400 and just using my laptop to "trigger samples" into the AW2400. As well the AW2400 has most DSP users would want on every channel strip and there is no latency or any of the PC/Mac maladies mentioned here in this post. Please understand; I am not bashing. I got real turned on when I first saw the Mackie 400F. But I would appreciate some feedback from ANYone who could provide truly useful information to me. I understand that software recording and using RTAS, VST adds a new dimension to "creativity" and I do not discount this at all. Last year I came close to buying a Digi002 and for the same reason I halted that purchase. So in my thinking the AW2400 provides a rock solid recorder with mix automation, great AD/DA's and decent pre's (no matter what pre's you have you need a tube pre-amp in most cases anyhow) and in a 12-track or 24-track mix, I don't see "triggering samples" via MIDI from the lappy using [Tracktion2 and Sampletank2 for example] being a problem outputting those into the AW2400. So all I need for my lappy would be a $40 USB/MIDI from M-Audio or elsewhere. Any and all feedback is appreciated. I have a $3000 budget max. Thank You.

[harryjames]

Thanks to everyone who is sharing their 2 cents in this forum. I didn't mention in my other post that I got very close to purchasing Mackie's 1640 Onyx mixer; but again; I was leary about latency monitoring in the grand scheme of things. I was dissapointed to learn that using the Mackie 400F and turning on the built-in 10 channel mixer basically defeats your DAW. I am finding it hard to believe that as sophisticated as the DAWs and other software tools [including RTAS, VTS, etc] have become, they cannot integrate with external hardware "in real time." This is beyond my comprehension as a songwriter. And I agree with an earlier post that when the 400F mixer is "on" you should have DSP control as well. At least the basics; compression, 4-band EQ, gating, de-essing, etc. Wow that AW2400 is sound better already. I do own and still use my Mackie 1202; only one of the phantom pre's still works ;;heh;; I think the power supply is on its way out. I got my money's worth ten years ago. But back on point, why put a "dry" 10 channel mixer in the 400F when you have to bypass the HOST DAW??! To be fair to Mackie; being able to connect several keyboards to the 400F and feed them to "where ever" is a plus in the I/O department. But it doesnt appear to be an I/O solution for my recording needs after all I have written.

[DemoKing]

I think you have it all a little skewed. You dont NEED that mixer, really, other than to route signals through it to where you want them to go and how. You dont need effects, as that is what the DAW is for. Everyihtg will still be done in the daw, mixed, and shot back to the soundcard, which will simply act as a DA and route it to whatever outputs you decide. The built in mixer is no big deal, and why does anyone really need to pay extra for reverbs and stuff like that in an interface when I have top notch reverbs available in Sonar 5, or any other program for that matter. Sonar for instance has a new 64 bit mixing engine. Mix there, where you should be mixing in the 1st place. UNless you have a big money console. I dont believe you do.

Latency isnt an issue either, these days, with a fast daw with good drivers. I can do software monitoring with my Delta 1010 through Sonar on a 3 year old computer at 2.9 ms latency. If yo u have a problem with latency, just monitor off the board or something, rather that with the DAW. Its so simple really, I think you just dont understand it all and need some hands on to figure it out for yourself. I think the 400F even has latency free monitoring on input. I dont see the problem here.

[MikeRivers]

Originally posted by UstadKhanAli Hello Mike!!!

Hey, stranger!!!!

[Anderton]

Let me clear up a couple misconceptions here.

First, latency depends heavily on the computer itself, and the complexity of the project. For example, while tracking, I usually have a bunch of "frozen" tracks and record with under 5 ms of latency (for perspective, the same delay as moving five feet further away from a speaker compared to sitting right next to it). But on mixdown, when I'll be using a ton of plug-ins and complex automation, I might bump the latency up to 10-15 ms so the computer can handle the load without "sweating."

Using the Onyx 400F DSP mixer does NOT disable the DAW in any way. What it allows selected channels to do is bypass the DAW, so latency doesn't come into play. In other words, if you're singing into a track, rather than listen to the output of the track, use the mixer to monitor the track input - presto, no latency. You won't hear the vocals with any effects you've added in the DAW, but I'm not sure that's such a big problem, as I tend to record vocals dry and do any processing during mixdown.

As to why I haven't covered Tracktion 2 much, it's quite a deep program that would deserve the same kind of treatment as we did here on Sonar and Ableton Live. Trying to cover it here would be like doing two pro reviews in parallel. And don't make too much of the fact that it's bundled with the Onyx 400F; it sweetens the deal of course, but there's no special synergistic engineering technique that lets it have zero latency or whatever. It's performance is pretty much the same as any well-written ASIO application with the 400F -- very little latency with a decent computer.

Now, having said all that, there is a reason why devices like the AW2400 and Roland/Korg/etc all in one recorders exist: For one thing, they don't involve a computer (at least not overtly), and some people are more comfortable with a dedicated machine than pressing a computer into service as a recording device. It's not a decision to be made lightly, as each has pros and cons. And of course, some people use both: All-in-one for live recording, then they bounce the tracks into a computer for editing.

[MikeRivers]

Originally posted by harryjames I am finding it hard to believe that as sophisticated as the DAWs and other software tools [including RTAS, VTS, etc] have become, they cannot integrate with external hardware "in real time."

Don't blame the software manufacturers, blame the hardware manufacturers - or to be more real world about it, blame the customers. They want "world class" specifications and low cost. The ASIO specification does indeed provide the hooks to control monitor switching in the hardware but nobody makes an audio interface that takes advantage of it.

 

In order to do that, they'd need to do what real tape decks did and put relays (or the solid state equivalent) in the audio path so that when recording the audio input is switched directly to the audio output, and when playing back, the audio output is fed from the D/A converter. This requires more hardware (more cost), some mixing capability in the audio interface, and the switching potentially reduces the S/N ratio by a couple of dB. That's enough to send the customers over to another brand.

But back on point, why put a "dry" 10 channel mixer in the 400F when you have to bypass the HOST DAW??!

So that you can do what you can't do with the DAW alone - monitor your inputs, whatever it is, without the delay of a round trip to the computer and back. You can still mix in the DAW, in fact if you have more than 10 tracks, you'll have to do some mixing in the DAW. But if you have eight tracks of drums, you can send those to one pair of returns to the mixer on the interface and have direct hands-on control of that in your monitor mix.

 

Consoles are a darn good idea. A console isn't just a funnel for all your inputs, it's a set of controls and indicators at the heart of your control room. If you're going to try to do without one, it's nice to have at least some of those functions still available even though the box they come in now isn't nearly as impressive looking to the visitors.

[wolfledge]

Due to some sort of incompatability with my old sound card - which has works with most of my software - I started researching what new sound cards I should be looking at.

I have been over a number and stumbled on this thread.
What a FANTASTIC thread!

I have been reading Craigs work since 1986 -
"Midi for Musicians" sits on the shelf behind me.

My current sound card has 8x8 In x Out at 20bit 44Khz.
For my home projects it has been fine - until
some software updates for a well known VST
started to hang - and the drivers for the sound card are no
longer supported..

So the discussion of ASIO and the WDM on this is of interest.
I would not want to fork out money and repeat the current problem.

Has anyone tested the ONYX 400F using WinXP SP2 and
Native Instruments KONTAKT 2 -- or REAKTOR 5?

I have read every post here tonight - impressive expertise has been contributing here.

One other concern --
@192Khz - how much disk is consumed per minute of audio?

[harryjames]

Thanking all three (3) of you for your replies. I am a hands on kind of guy and I am sure because I haven't got to touch these things "in the DOing process" adds to my confusion or misunderstandings as to how these things work. I understand software DAWs and how ReWire, RTAS, VST works and when I lay down my initial song I like to record with effects and require 4 inputs for stereo keys and phantom mic with stereo effects returns. For me effects are part of the performance. Going back after the fact and editing/adding effect to say a dry vocal is not how I wish to record. I do understand the benefits to recording dry tracks, omping, mixing and post effects. In my little experience meddling with USB and firewire devices and a couple DAWs, I just didn't care for the latency I experienced using headphones and made it difficult for me to enjoy the process. When I first saw the Mackie 400F, I got the distinct impression that it would be a great I/O to use with Tracktion2 (which I was already pleased with its easy of use including ReWire) and that I could monitor the DAW mix, with effects, while recording, through the 400F with "no latency." As to the comment inferring I am some consumer wanting $100 grand equipment for $1000 bucks bucks; (not an exact quote) this is not the case. I am very appreciative that one now can purchase a quality 24 track recorder for $2grand. I just have found it very challenging [if not impossible in my area] to physically get a hands on demo of any of these devices. Which makes it harder to grasp how these things work. I appreciate everyone's input here. Many of you obviously already are using DAWs such as Sonar and hardware I/Os including the 400F, and for what it's worth, latency doesn't appear to bother you. This is why it was my thinking to use my laptop to trigger samples and record them to a dedicated recorder. My laptop is a P4 3.0HT 800FSB 2gRAM 4-pin firewire, and of course I would buy an external 7200rpm 16mb cache 8ms HDD for dedicated recording whereas my lappy has a slow-but-steady 5400rpm 80gb. I understand making the choice to go computer based versus dedicated recorder is not an easy one. And I agree they each have their own nemesis'. There's a lot of bells and whistles RTAS and VSTs that would be useful and certainly spawn creativity and allow for creating genres of music I don't even currently write or play. This aspect of computer recording interests me. And with exception to MIC recording, computer generated noise [such as fans] is not even an issue then. This is why I have waited to research more and to try my best to make a quality decision as to what gear purchase. It's not about the best or perfect; but what works for me. Sooner or later I gotta go with something and get recording too and I understand those dynamics. I have a couple DSP RUs so buying the 400F I definately need to get a rack cube and get everything wired up; which in that case I would also get a line conditioner and probably a good mic preamp. Anyhow, thanks again. My hat's off to Mackie for their Onyx products including the 400F and even my 1202 mixer that is still plugging along. And thank you for this forum. This will help me to make a decision. ;;cheers;;

[sekim]

Has anyone done a multi-generation loopback recording test with this box that can post their findings? Granted this test reveals total A/D/A system performance, but given that nobody is complaining about A/D - but only the D/A - I'd think this test could certainly reveal whether the D/A is mis-representing or not. If it is, it's going to be very obvious after a few generations.