Some Honest Advice


Active Member
Sep 20, 2014
Gear owned
Tascam DM-3200
Hi All,

I'll try to keep this short - but am hoping to get some advice on where to go from here.

Intel Mac | 10.9.5
Dm-3200 currently with 2 ADAT cards (tried to move to the FW card to simplify the setup but just can't get it to work ... you can read the thread here :
MOTU 2408 MK3 PCIe 424 (connected via ADAT)
ProTools 9
Misc Mic Pre/Comps etc

I've been recording for awhile mostly in the analog realm (had a tascam M2400 years ago) then later with ADATS. I'm pretty comfortable on the software side but where I get hammered is when you get into the multiple layers of routing and clock.

I'm a pretty technical guy (work in Software / Hardware development) so i hate having to do this.
The reality is i don't spend anywhere near as much time in my studio as I want and I'm concerned my gear may be a little too much for me.

I tried to simplify the setup and get to recording with the FW card but as you can see from the thread above doesn't seem like that is going to happen.

I've read the manual (it's not great) , bought the DM3200 DVD and I get how to the use the console. I have no issue with that. getting the rig setup is another matter.

90% of what I do is project recording (tracking). Is this setup just not the right setup for me?
I feel like if I can get it setup - i'm good to go. but that has not been simple.

The path I'm on now is to use the MOTU 2408 with ADAT - everything seems to be setup ok in the MAC - i'm getting stereo (ADAT 1&2) into the console but that's it.

I am assuming that I should use the master clock from the board (i read you should do that with Pro Tools) and slave the MOTU to that.

Is there any walkthroughs of setups (routing on the board, on the mac, in protools)? I think I'm at the point where I'm psyching myself out now.

I would love to get a view of a tracking session setup on the board. How do you "flip" your channel to listen back from the DAW

I'm considering moving to a Presonus StudioLive board it just seems simplier and may be better for me at this point. The routing seems pretty straight forward. When you want to listen back from the daw (for overdubb'ing or whatever) you hit the digital in button on the channel :)

Any guidance/opinions would be welcome.
These are many questions in one topic, but if I were you, I'd get the IFFW repaired or replaced and forget about ADAT through MOTU and other complex schemes. My guess is that 80% of users have the DM with the IFFW card - it's simply the way to go.
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I don't understand. You said the FW card is only a few months old from B&H... why aren't you on the phone demanding a replacement???
I have my DM set up so there is no button pushing for playback... I route inputs to the appropriate recording buss during tracking, and simply play the DAW sequence to hear it. I never change the routing or "flip" the channels. Part of keeping the console set up this way is that I don't ever monitor tracking through the DAW. I could grab some screen shots....
I read your other post with your FW issues. It seems that you may have some delays to get past, but, like everyone said, the FW card is definitely the way to go. That being said, while you are waiting for the FW card issues to be sorted out, you do have a pretty good (but not as good as FW) alternative in your MOTU 2408/PCIe 424. There is no reason why you can't be recording 24 channels over ADAT, through your 2408 and into your PCIe 424, and into your DAW, and then returning 24 tracks of audio from your DAW back into the board via the same routing, if that is what you want to do.

From the PCIe 424 Mac manual...
"The heart of the PCIe-424 card is a custom-programmed VLSI chip
capable of simultaneously processing all 96 inputs and
outputs (192 channels total) at samples rates up to 96 KHz.
At 176.4 or 192 kHz, this chip can process 48 simultaneous
channels of input and output."

That's pretty impressive, but the ADAT cards you have in your 3200 can't do that. So, for 24 channels, you will need to set the 3200 at 44.1Khz or 48Khz, where the 3200 is using it's internal clock.

Do you need a step by step to get your routing down? I was about to do it, but thought I would ask first.
I can't give any advice about the Mac, its drivers or setup - but there are several experienced users here who can. After looking over your original message, it seems there are three potential problem areas - two of which might involve Mac/drivers/operating system/PT9 setup - and one pointing to a bad (or improperly seated) fire wire interface or card. These are frustrating issues, I know, but nowhere near a deal breaker just yet.

I'm considering moving to a Presonus StudioLive board it just seems simplier and may be better for me at this point

Yeah, you could do that; the S/Live board is simpler, indeed, but it's also less powerful in certain important areas, not the least of which, mix automation which it doesn't have. You'd be restricted to DAW automation which, in my opinion, isn't particularly wonderful in PT9 or 10. Secondly, the Presonus isn't anywhere in the same universe with regard to MIDI power either. I also suspect the S/Live isn't as robust in sonics as well, but that's immaterial at this point.

Bottom line: you're not the only person who's considered throwing in the towel and going in another direction. But I think you're much closer to a solution than you realize. Abandoning the DM in favor of another unit or solution might cheat you out of a potentially wonderful studio experience.

I can't tell you what to do, but you might consider swapping the IFFWire interface, reinstalling drivers, and making sure everything is set up correctly in PT9. We can help with a lot of that if you're up for it.


There's little doubt that there's a learning hump to get over before you can get the best out of your DM.... but based on your profession it's not beyond you. The DM feature set is worth the effort imo.

If you isolate your 2 problems it might make it easier - push to get your f/w card replaced that will be one issue solved - if the seller won't help then go to Tascam and tell them you're not getting support from one of their resellers.

Next is the routing - just view your DM as a smart switch and add that understanding to your computer skillset as networking knowledge. After you play with the routing tables for a while it will become second nature - basically "to's" and "from's". Once you have the basic setup done it can be a matter of powering the DM on and pushing up the stereo fader and perhaps a few mute buttons to get recording / mixing / monitoring.

One other tip is try an alternative firewire cable - it may be that simple.
Hello All,

Just wanted to clarify something in my post that may not have been clear.

I bought the original card from B&H returned it to Tascam (per B&H) because it wasn't working. Tascam sent me a second card, which I am having the same issues with. SO I am waiting for an RMA # from Tascam to return the second card and get a 3rd.

@Gravity Jim - I think I understand what you are saying. Do you do a lot of overdubbing in that setup? Any screenshots would be greatly appreciated.

@TascMan A step by step guide would be awesome. I'm going to try and get the MOTU 2408 setup and in use while I work to get another FW card.

@captdan I'm going to stick with the DM and hope that the issue is the FW card. My biggest concern is that there is so much in play i'm not 100% sure it's the actual card. BUT Ive isolated to that - just seems unlikely to have received 2 bad cards.
I have not even introduced PT into the setup yet. I'm trying to keep it isolated to as few variables as possible. Just looking to get up and recording ... I've been away from it too long (years) and really want to get back to it and not spending months on the setup. I do really like the flexibility and the power of the board. I'm just struggling a bit with the basic core setup. I get it all conceptually and understand digital routing and matrix routing etc ... just applying it to the DM is kicking my butt :) Add in the hardware issues and things get a little nuts.

@Drumstruck I'm on my 3rd cable. I thought that could be it as well. I'm going from 400 (DM) to 800 (Mac) i haven't read anything that would cause an issue. I was thinking about getting a FW 400 card for the MAC and bypassing the native 800 ports. What do you think about that? Do you have a certain cable that you would recommend? I just bought 2 from Amazon that had the best / most reviews.

Really appreciate all your help... this forum/community is really great.
As they say... "90% of network issues are cable related..." - good work on checking that first.

fwiw my setup is from DM4800 into a 500GB OWC hard drive (audio files drive) via a 3 metre firewire 400 cable and then from that drive into a MacBookPro (i7 / 8GB / Lion) over a .3metre Thunderbolt cable. I'm running Reaper for DAW with no issues.
I also have PT9CPTK but gave up on it recently because of random "can't read disk" errors. I'm sure this is a PT "special feature" as Reaper never misses a beat.
I've been meaning to chime in here for a while so here is my take. I own a chain of music schools that have full recording studios incorporated into each studio. In our first school we installed a DM 4800 and it was a long road to get it working consistently and to go through its painful learning curve. Without the help of some the amazing members of this forum I would have returned it for sure. Still to this day we have to unplug and replug the firewire cable regularly to avoid sample rate errors. But now we know how to use it for our needs and its a great board for our purposes. We primarily track bands, often with 6-8 musicians playing at once, so having the ability to send 6 custom headphone mixes, having 24 M/L inputs, 32 channels of FW I/O and many other features makes it a great product. From a marketing perspective, the mixer looks fantastic in our studio, its great to have a largish format board for the price. So we are happy now.

But in our new school we opted to install the Presonus 32.4.2 AI mixer. You can really tell it is a generation or two newer than the dm 4800. It was so easy to set up, immediately connected and never had one problem or glitch, and the learning curve was 90% faster, in no small part due to a much better written manual and because of my Tascam learning curve I understood computer recording in general much better. It has a whole bunch of features I haven't tested yet like the ability to remote control the mixer and custom headphone mixes from ios devices.

Both boards are great boards that have excellent sound quality for the price, lots of M/L inputs and some flexible routing. The tascam has significantly more power when it comes to accepting other I/O types, general routing abilities and the motorized faders are really missed on the Presonus. It can record up to 96kHz although I rarely record above 48. I also love the remote layer on the tascam because we mix primarily ITB so its a real luxury. The tascam is also much better if you plan to integrate some external tape/hard disc recorder because of its machine control functions and automation. Since I mix ITB I rarely use the 4800's effects, but when I do I find applying effects complicated and have to read the manual and watch videos to get it right. The presonus is much easier to use, its easier and faster to apply and rout effects (both for printing and monitoring), and has a more stable connection to the Mac/PT in my experience. The sound quality of the preamps seem comparable, but I haven't done a careful side by side test. However, it can only record up to 48 kHz. If you are working by yourself, don't need the flexible routing, connections to lots of other gear or if you ever plan to use the board as a live performance mixer/recorder, the presonus could be a better choice (also less expensive and has a built in meter bridge). You lose some routing flexibility for sure because some routing decisions are hard coded and made for you, but so far with one exception these have worked as I would have routed them anyway.

As for support, Its easy to get presonus on the phone and they are helpful, but I've only had usage questions, no actual problems. On the other hand, while Tascam has been easy to get on the phone and Redbus here on this forum has been an incredible resource, his help and advice has been a lifesaver, I find it simply unacceptable that I have owned the dm4800 for less than 2 years and I'm on my 3rd LCD screen. There is some major flaw in their design and they are not taking enough responsibility to fix it.

So in summary 2 great products, each with strengths and weaknesses. But based on your current situation, I might lean towards the presonus just because it works every time with no troubleshooting when connected to a Mac running PT.
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Well stated comparison; fair and balanced, IMO.

It's my twisted, subjective opinion that the DMixers operate best at 88.1/96kz, and a true audio comparison to other units would have to be done with that in mind.

My own conclusions comparing a bunch of tracks I've done at 48kz to those with 96kz is that there's a noticeable uptick in overall fidelity, dynamic range, and DSP performance at the higher S/R. I can't say why this is the case, other than to propose a theory - that the ADCs present audio more accurately at 88.1/96kz, which may lead an engineer to hear subtleties more easily and make adjustments accordingly. I can't say anything for sure, but I know what I hear in my room and it's not imaginary.

So, if sonics are the Number #1 desired feature set, the DM would likely have the edge here. But in the other areas you mention, perhaps not so much.

etc. :)

I will point out that if you have a DM connected to a Mac and have to continually unplug it to correct sample rate, you're not doing it right.
Thanks CaptDan. I need to try recording at 96 kHz, I've only avoided it because I was afraid running PT 10 I might have issues as I'm often tracking 15-22 tracks at once and I don't use HDX. But I do need to try it so I can compare to some of my other work, the world is so split on whether it makes a real world difference, but my gut says it will, if for no other reason than as you say letting us as engineers make better mix decisions by hearing details better in the confines of the studio.

I'm curious, I believe you hear a difference in your control room when listening to the hi res files but once the song is finally mastered to CD quality or worse and listened to on the typical consumer players (ios devices, youtube, car stereos, Beats headphones) in typical listening environments do you think you can tell a difference whether the original source was tracked at 44.1 or 96?
I will point out that if you have a DM connected to a Mac and have to continually unplug it to correct sample rate, you're not doing it right.

Believe me Gravity Jim, I am with you. Its crazy and somewhere somehow something must be wrong. But I have read every post in this forum, read the manual 20-30 times, and spent more time than I should have on various pro tools 10 forums. The reality is that occasionally and for no reason whatsoever, I will open a PT session that was tracked at 24/44.1 with the mixer correctly set and will get an error saying something like wrong sample rate. The only two ways we can fix it are to:

1. try to jog the FW card into recogizing the sample rate by switching it from 44.1 to 48 and back to 44.1, then reopen protools. this works 70% of the time, occasionally requiring 2-4 cycles of these steps.

2. reboot mixer then follow steps above. works another 20% of the time.

3. The other 10% I have to do all of the above, but also reboot the mac and once rebooted unplug and replug the FW cable.

If you or anyone knows a better solution or can give me some troubleshooting tips to check I'll try anything, but so far no one has been able to give any advice other than the two processes above.
It's no secret that ASIO-PTools flavors are VERY touchy about drivers and sample rates. With fire wire in particular, your maneuver range is narrow - meaning - if you even think to change your buffer rate when a PT session is open, you'll crash your session. If you pick a buffer rate that PT doesn't recognize even before launching a session, you'll receive a message, which I paraphrase: "Hey! Avid doesn't like that buffer rate! Choose another we like better!" :) ).

There's a sequence of steps that need to be done whenever you change a PT session from 44/48 to 88.2/96k. This involves changing the Windows audio drivers in the background, rebooting, then launching PT. If this isn't done right, you'll receive PT 'contentions' - half of which are irrelevant to the problem. It's as though Avid wrote a bunch of warnings, threw 'em in a bucket, and what you get is the -un-luck of the draw. :) In any event, the instructions for this process are in the IFFWDM 1.30x upgrade pack.

All that said, I'm a real fan of PT and how it works with the DM. I realize many users have problems, give up and 'reap' the rewards of other DAWs. I, however, have yet to find a deal breaking issue; far as I'm concerned, ANY computer - ANY DAW is fraught with issues that need attention. The only plug-and-play machines I've experienced are porta studios. :)

To your question about 96kz sonics after dithering to 44.1, CD and MP3. Absolutely, the sonic integrity carries through to the final product, including a lowly 128kbs MP3 file. I think it's the same thing as using a high end mic; what it captures more or less remains indelible to the recording. And the reverse is true too; if it sounds like garbage in the beginning, it can't be massaged into gold thereafter. But you already knew that. ;)

Run your own tests and derive your own conclusions. I was told - no - WARNED to avoid 96kz at all costs; it wastes HD space, causes 'artifacts' in down sampling, and warts to grow on your a**. And most of these came from 'experts' - 'industry leaders' self appointed to eradicate bs and half-truths from the world of audio. Perhaps at one time (circa 1999) some of what they said was true - well - maybe not the warts part. But today? Uh........methinx not. :)

And the reverse is true too; if it sounds like garbage in the beginning, it can't be massaged into gold thereafter. But you already knew that. ;)

Truer words have never been spoken. I learned it the hard way in the beginning of my learning curve. And I'm sure you know how hard it is to tell an anxious band paying by the hour for studio time that its worth spending another 30 minutes tweaking drum mics to ensure we track a great drum sound up front rather than investing hours editing, replacing and repairing poorly recorded drum tracks in the mix. I can see the look of annoyance when I'm bouncing back and forth between the kit and the control room listening to a snare hit for 15 minutes while making what seems like small tweaks but is actually getting the mics in phase and the kit sounding "on tape" like it does in the room.
I'm going to try recording at 96khz soon. I may even post some examples comparing the same selection at 2 or 3 different sample rates. lets see how much time I have...
Sounds good - or hopefully - will. :)

One thing to consider: source material is important to the tests. That means a drum machine and piccolo may not be the best choice. Perhaps heavy metal or ultra-compressed sources may not reveal much of a difference either. A rich acoustic rhythm gtr track, or maybe an acoustic jazz, folk, or scat vox sample would be better choices. The goal is to choose harmonically rich sources; overtones, solid bass, high end, mid range represented, etc. I think this is where the differences are perceived.

But I'll stop here; don't want to prejudice the outcome. :)


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