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confused About buying UAD

audiofreak

New Member
hello,
i am new to here , i have built a studio and i use DAW for all recording and mixing, and i want really to buy the UAD ultra pack, but i am getting so confused as i heard many many great reviews and opinions about it but in same time i heard those horrible stories about latency prob, cpu load and most of all not working with AMD 64 bit processors, now i will list what i have and what i want and please i need honest help and opinions as i am gonna spend money on this and please excuse my lack of knowledge :)
1-my specs are
- AMD xp X2 4800
- SLI mother board
- 4 gigabyte of ram
- MOTU Traveller sound card
- AKG and shure Mics
- yamaha O10V digital mixer
so will UAD work with my specs ?

2- my first prob is i need decent compressor , gate and limiter so i wanna know if i can use UAD for on recording and not only in mixing or mastering and if i can so then how ? i mean i tried to put software compressor on record input in cubase with my Motu card but it didnt work, and after reading i saw that most sound cards dont do this and they are not able to compress or add any effect on the recording , to be more specific can UAD act as hardware outboard compressor, right now i am using my yamaha onboard compressor which for me sound sucks,
and quality of UAD is superb as i heard so i wish to use it that way

3- i read that the UAD cards now are outdated and u should wait for UAD-2, now i dont much care as long as it really does the job but if it doesnt then this is big problem, i mean making problems like latency or havey cpu load

sorry for sounding stupid but i am really new to this and want to learn :)
thx in advanced for any help
 

robi

Member
To 2: No software can do this. You need a compressor before the ad-conversion to achieve what you want.

I don't use cubase but I think it has the abbility to print to tape the audio with effects while recording (I read that anywhere) but this is a different topic 'cause the audio is eg distorted at the a->d stage that comes before any software dynamics will work it.

To 3: If there will be a new product at aes is guesswork 'till 20th but if I were you I would wait the two days, maybe, maybe not...
 

TheHopiWay

Active Member
On occasion I've used plug-in compressors on the front end of a recording in DP.
The drawbacks are;
1-you can't monitor the results in real time due to latency,
2-the routing scheme is cumbersome.
3-You'll need to manually compensate for the latency by nudging the track earlier in time equal to the system latency after it's recorded.

Bring the source into your DAW to an Aux channel (rather than an audio track) and make it's output \"bus 1\".
create 2 more aux tracks both with inputs set to \"bus 1\".
Insert a compressor on \"Aux 2\", make it's output \"bus 2\" and set \"bus 2\" as the input source to an audio track.
To monitor the performance in real time you'll need set the out but of that 3rd aux you created to the headphones.
Note that you won't actually hear the compressed track in real time while recording, you'll instead be hearing the input side of that chain.

The above method is why I have about 5K invested in a few hardware compressors.


:wink:
 

robi

Member
...but how would that avoid eg digital clipping at the converters? I have to miss something here? :?
 

secretworld

Active Member
Hi on cpu load, I have virtualy none and can use my 2 uads int the 90% dsp. No problems here. Recording with Uadplugs is possible but monitoring won´t be latency free. I always record vocals straight and aply plugs afterward.
You don´t state the mobo you use, but I would check for possible incompatibillities there are quiet a few with recent (dualcore NF4?) mobo´s.
Look in the working setup stickie post!
 

audiofreak

New Member
guys i really so appreciate your replies,
so from what i read from you i will wait another week and the new products will defently come out ?

regradiong compression then i must buy hardware one,
i am thinking of buying FMR really nice compressor , i have read many great reviews about them what you think ?
 
G

Guest

Guest
in SX you can compress as you write to disk you just select an insert on the input channel you are coming in on. and you can listen in real time with low latency sound cards.
 
Look at your signal path.
If you are going mic->mixer->sound card->software , a compressor on the input of your software won't catch clips before they hit the A/D of your sound card.
Use an outboard compressor if you really need this.
The FMRAudio RNC is a nice cheap option for two channels if you don't like the compressor on the mixer.
You may just be recording too hot a signal if you are coming from the analogue world. I now I was until I realised that most sound card's A/D equate an analogue 0dB signal at -14/-18dBU.
IOW, you can set the meters in your software to hit at -14 dBU and still have a nice hot signal and headroom.
 

audiofreak

New Member
Phil Chamney
that is exactly my setup
but can u explain more to me please ?
should i turn the pad gain on my mixer to the minimum ? and how much the mixer should output ? in other words the mic channel in mixer how much should it hit as output ?
and same how much input on my sound card i should max recieve ?
about FMR compressor being nice cheap option ? does this mean it is not that good or not much better than my yamah onboard compressor ?
i only need it for vocal or maybe guitar that is all what i record
sorry for my noobness
 
You need to check the specs from the mixer and sound card manufacturer.
I'm just saying that an analogue device that has meters calibrated at 0dB can be recorded in some (check) sound card mixers at, say -18 dB (@ +4) to allow optimum head room.

The confusion comes in that, with your preamp hitting zero, one assumes that the sound card should do so...not necessarily true.

Here is a quote from Lynx audio cards:
\"The LynxTWO is calibrated for -16, +20dBu in the +4 position.\"

I believe Paul Woodlock has gone over this recently as well:
http://forum.cubase.net/phpbb2/viewtopi ... highlight=

I've found the trick is to adjust your incoming signals with your analogue meters (close to -6 to 0dB since they should have headroom), and ignore the soundcard meters, as most won't let you adjust the input.
If you incoming signal on your input channel in your software is -20 to -8 or so, you should be ok. Your mileage will vary with different equipment.

Sorry, I've probably just confused you more...read a bit of those threads I linked.

By the way, the RNC is a pretty nice sounding comp, and fairly well accepted as such.
and...the UAD1 plugs sound great for the audio programs I do. :D
 

audiofreak

New Member
thx,
if i got it right
at mixer record at -6 db then route to sound card and record at -18 db
all this to have enough headroom right ?
 
I don't know how the yamaha digital mixer handles it.
Two different analogue preamps I have clip at +16 and +28 :eek:

Get your input close to 0dB on the analogue meter, and see what your soundcard meter reads.
Probably somewhere between -14 and -18.

My take on this...I've found that if you engage your soundcard or software mixer, and tweak your preamp or mixer while watching the input, you will find a good medium for that equipment.
 

audiofreak

New Member
thx alot really man
i have read my yamah mixer got 100dbu headroom unless i am mistaken
after reading alot
i decided to get both FMR audio the compressor and preamp
-but still i am confused about UAD cards
the fairchild alone is killer
i kno i am in UAD forum but are tc powercore etter as i see they are new and have no probs iwht motherboards
 

robi

Member
ok, as you seem to be new to this, in plain words it's more simple as you might think.

You don't want distorted audio and distortion - as I guessed from the beginning and you proofed me right later on - is happening (at least) on a->d stage. So forget about any numbers, just patch a comp/limiter bevor your converter (as the first step) and get it \"as loud as you can\" bevor the audio clips indicated at your a->d converter.
The fmr rnc will do a good job on this.

Powercore? Get one..., too, with all their revebs, maybe the sony dynamics but... you'll want some ua, too, I promise. Problems they have both as well as any other hard- and software.
 

audiofreak

New Member
thx robi but i am really confused ,wait for new products or it will take months.
i wnat also to know one UAD card how many plugins can it run before it hit its pwoer cieling ?
 

Eric Dahlberg

Purveyor of musical dreams fullfilled.
audiofreak said:
project thx,
but the clipping is happening , i see my travller led hits red all time
Just turn down your inputs. With 24-bit recording, there really isn't any need to record with as high a level as possible. That was only important with 16-bit or analog tape recording.
 

secretworld

Active Member
most converters sound best at ~-12 to -20db (well not the converters but the analog part before them), most linear!
 
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