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If you had $1100 to spend on a nice preamp or A/D Converter what would you choose?


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That's not completely true. Any time you use a one-way, non-packetized, type of connection, then you have to have two separately running clocks. So the clock in the interface inside the PC has to derive a clock from the incoming samples, which would inevitably introduce jitter.


Though it would introduce a little more latency, using a packetized, asynchronous connection like firewire would avoid that. Then the PC is just receiving data packets and it's clock doesn't have to sync to the converter in any way. It just has to clock out the samples for monitoring during the tracking process.


I'm not sure if other such non-packetized connection types, like AES or MADI, get around these issues.

 

 

So Dean you're saying that if you had something like the API, the best connection to avoid two clocks would be an audio inteface like firewire to accept the API's SPDIF signal? Or what is the best way to connect the API to the PC to avoid jitter (and minimize latency)?

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So you have been happy with the API? Do you use the Mbox for your monitoring and D/A?

Thanks



I am definitely happy with the API. I am still messing around with the different tones you can have with the pre's, and my two condenser mics (Rode NTK, and Oktava MK-319 Floating dome mod. I do run my MBOX to for monitoring D/A. But in your case, you might want to focus on getting an amazing preamp. I think most would agree on here that a great pre can make a so-so mic sound incredible but a great mic probably won't sound nearly as good without a nice preamp. In that case, you can't go wrong with anything API has made in my opinion. :thu:

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So Dean you're saying that if you had something like the API, the best connection to avoid two clocks would be an audio inteface like firewire to accept the API's SPDIF signal? Or what is the best way to connect the API to the PC to avoid jitter (and minimize latency)?

 

 

It won't minimize latency. The latency will be slightly worse, because the data is gathered up into small bundles and sent over to the DAW. It should minimze jitter, since the converter's own clock is the one that's time stamping the samples. So I guess it's whether you feel that a ms of extra latency is worth the lowered jitter. I think that a firewire interface adds about 1ms of latency in general.

 

The reason I mentioned it is that that box doesn't have such an option, I don't think. I think it's S/PDIF or AES only, but you should verify that yourself.

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"Wax rhapsodic"... Nicely done!

I guess back to the original poster's question. I'm in the same boat and price range actually and I'm just looking to spend say $1500 and get the best combo of preamp/interface for that money. Build me a system!
:)
OR is this price range too low to truly gain that extra benefit in sound over say a $500 setup?

I see the original poster bought the API. But starting from scratch I would still need an interface to my PC since that does not use firewire.



If quality is your goal, for about $1600 you can get a Lachapelle 583s preamp and an Apogee Duet, but the Duet only records 2 tracks at a time.

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Don't forget the room treatment. It's SO important, it can't be overstressed. I've put a stupid amount of time and exploration into it, but it's well worth it. And it's going to suck up a fair amount of bucks.......



Don't overlook this. Some of the best advice I've read so far.:cool:

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If you can get the Trak2 for $1100, I'd say go for it...if you're looking for clean preamps. I'd put its preamp on par with about anything out there, but they are very clean, along the lines of GML or Earthworks preamps. They are very accurate, which isn't surprising considering they come from a converter manufacturer.

If you're looking for preamps to add mojo, though, then you'd be better of with the Great River...but if you get the Apogee now you can always add your mojo preamps later, and you'll still find uses for the Apogee preamps.

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Dean, with the RME 9632 card, for the output does that need to go into a mixer first before going to your monitors?

 

 

No, they are line outs designed to go straight to your monitors, or through other stuff long the way, such as a monitor management systems. The RME allows you to select the standard balanced -4dBU levels, or the consumer -10dB and they have an even lower low gain setting. Same with the inputs.

 

 

BTW, the latency issue with something like the API A2D? Just run the analog outputs to your headphone system and mix with the track. Zero latency.

 

 

But you couldn't monitor then, right? If they are only A/D, and not being used as D/A, then you could only hear what's going in, you couldn't hear anything you are tracking against.

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Here's my latest tune. It was all done with the LA-610 and RME card:

http://www.charmedquark.com/Web2/TmpAudio/TheHiddenHand-Preview05.wma

It's obviously not the ultimate in uber-crisp clarity, since it's very much intended to be kind of vintage sounding, which is what the LA-610 really does well, and I think the kind of sound most folks in the rock, pop, or alt type worlds would want. The electric guitar and bass just through the LA-610 to the 9632, and the vocals were Pearlman TM-1 -> LA-610 -> 9632. The only things that didn't go through it are the drums and the organ.

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Dean, it took me a minute to figure out your question. The API has analog outs along with the A/D. I route those analog outs into my headphone amp along with playback from my DAW. Instead of monitoring what I'm recording routed through the DAW (latency), I'm hearing the API pres. Mixed in with the backing. I've done this with a small mixer in the past but now do the blend in the headphone amp. It's a pretty common method for overcoming latency.

 

When I track a whole band's basics I can't do this but the ambient level in the room seems to mask the latency.

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Oh, OK. As long as you don't need to hear effects applied the DAW, that would be a good scheme. For me though, I never am in that situation. I never track with reverb or other effects. Those are always applied in the DAW. Well, for that matter, since I use amp sims, the whole guitar/bass tone is applied in the DAW, so I always have to monitor through it. But even if not, I'd still have the reverb and effect issue. Not tracking with effects is so flexible and the ability to comp is hugely improved without reverb or other time based effects on the raw incoming signal.

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Dean, it took me a minute to figure out your question. The API has analog outs along with the A/D. I route those analog outs into my headphone amp
along
with playback from my DAW. Instead of monitoring what I'm recording routed through the DAW (latency), I'm hearing the API pres. Mixed in with the backing. I've done this with a small mixer in the past but now do the blend in the headphone amp. It's a pretty common method for overcoming latency.


When I track a whole band's basics I can't do this but the ambient level in the room seems to mask the latency.

 

Lee, what is your setup for mixing through your monitors then?

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I use a Digi 002. When playing back I just use the internal converters out to the monitors. 002 converters are subpar I know, but I have what I have and do quite well so far with it. Getting good conversion going in is my priority. D/A is important but not priority over A/D.

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If you don't use Pro Tools and have a firewire port, this is a killer rig. It sounds light years cleaner and clearer than my Digi 002 rack. Preamps are also good. Don't skimp on the a/d d/a convertors. As long as you are using decent equiptment and mics, this is the most important link. The unit sounds incredible and is super fun to play around with.

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This is a long thread. I read most of the first page. With my Digi001 convertors I could hear a big difference with the Great River Pre. When I upgraded to Pro Tools HD, I thought the HD sounded slightly better. Then I upgraded to Aurora Lynx convertors. The difference between the Digidesign HD Convertors and the Lynx sounded negligible and I was wondering what the hype was.

 

I'm trying out the Focal Solo6 Be monitors. I've been using the Mackie HR 824 and NS 10s. The Focals sounded better and revealed much more. Then I tried switching between the Digidesign convertors and the Lynx and the difference was substantial. I just didn't have the monitors to hear the difference.

 

So my conclusion is that you need three things. Good monitors, good pre and good convertors.

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I've used and sold MAudio for sometime....can't recall ANYONE ...EVER...liking their gear over anyone elses!!! On a budget...think MOTU and Focusrite...have had nothing but great success with both, in my studio (RockDawg Studios) as well as my former place of employment...rhymes with sitar center!

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The aforementioned pre's are very nice but I thought we were talking about budget, if I could interject an off the wall but highly relevant thought....I would spend a lot more money on sound control in my listening environment first, then improve my monitors, long before I would worry about mic pre's and converters...just based on my personal experience as a studio troubleshooter.

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mBox isn't made my mAudio - it's made by Digidesign. It's the cheaper hardware interface for ProTools LE.

Improving your listening environment is vital, no doubt, and so is having good monitors - IF you're going to be mixing things at home, at least. That said, all good monitors and acoustic treatment will do if you have bad preamps and converters is highlight the fact that you have bad preamps and converters. You need every piece of the setup (including room and instrument sounds, players, microphones, etc...) to be at least a certain minimum quality level. That doesn't mean you need to spend thousands of dollars or anything crazy like that, but it does mean that blowing all your money on one part of your signal chain will give you only limited results if you cheap out on another part.

Also, why did you split your reply up over three posts? :) Just curious.

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