@colinf: I'm just running Ubuntu, because the repository stuff is decent and I like to be able to use my laptop for everyday use and production.
Thanks for the lsmod tip, i'll try that out
edit:I sort of got it working by setting the periods to 3, but now it just throws out hundreds and hundreds of xruns, I don't know what to do :S
What are your settings for jack if you don't mind me asking?
I cannot see snd_ua101 in lsmod :S
I'm running Ubuntu 10.04 myself, but with a self-compiled 2.6.35 kernel: the 2.6.32 kernel that's in the repository doesn't have the snd_ua101 driver. You said you'd already tried a 2.6.35 kernel yourself: where did that come from?
I assume you're using QJackCtl: have you got 'Realtime' ticked in your settings? I've found 'Frames/Period' of 256 and 'Periods/Buffer' of 2 to be about as low as I can go with the UA-1000, and bumping up the 'Priority' to 89 seems to help.
Update to the RME Babyface: I talked to one of the developers. There were several parties involved in creating the BabyFace, and not all seem to agree in releasing the driver sources to create an open source driver. So unless somebody generates a protocol description by looking at the data stream, there will be no driver for Linux.
What a waste.
I find 700 bucks for 1 ADAT port and 2 analogue ports a bit steep anyway. You can almost get a proper sound interface for that money. And the end of proprietary interfaces is near!
I've compiled my own 2.6.35 kernel now and it's kind of working. I can get jack running but it xruns quite a bit and in the connections it only shows 4 inputs and 4 outputs. It also doesn't allow me to choose the sample rate, not matter what I set it to it runs it at 192000 even though the UA1000 only goes up to 92000. I have Realtime checked. snd_ua101 is now showing up in lsmod, so thats good I suppose.
edit: got it working! Thankyou so much for your help! It turned out you have to turn the device off when you want to change the sample rate, so I just turned it off, switched sample rate to 48000 then turned it on and it works! thank you so much!
That sounds like progress...
You have to switch the UA-1000 off to change the sample rate: the rate is set from the sample rate switch when the device powers up.
According to the owner's manual, the UA-1000 supports 192 kHz 4 channel playback only. The 'USER SET' position of the switch defaults to this sample rate. The only way to change the 'USER SET' sample rate is from the Windows control panel: as far as I know, there's no way to change this under Linux.
I've never tried the 192 kHz sample rate myself, so I don't know whether the xruns will go away once you get it running at a lower rate.
Looks like our messages crossed! Great to hear you've got it working: have fun!
I compared the m-audio fast track pro to the built-in Intel HDA interface in my laptop and found no significant difference. I set the alsa-mixer gains to zero and used an external mixer with preamps. I gave more details in a post on another audio interface thread.
If you only want two input channels, you might consider this option. I would be interested in knowing other peoples' experience.
The M-Audio(Or any external interface) is the preferred solution for at least two reasons, one the AD conversion is likely to be slightly better(Though I have not measured this yet, this will be a small difference but noticable on good equipment with good ears) but the larger reason is simply because it will move the AD/DA outside of the box and be FAR less susceptible from interference from within the computer, a common source of problems.
And I am fairly certain I am forgetting one or two reasons, of course the OP was also looking for preamps with 48v, and this thread was started nearly a year ago:)
I use a Cakewalk (Edirol/Roland) UA-25 EX and I really love it. As I bought it for mobile use I decided to buy an USB interface. Otherwise I think I would have bought a PCI device like the M-Audio Audiophile 2496. The UA-25 EX works just out of the box with e.g. Ubuntu or AVLinux. It has got two XLR/TRS combo jacks with phantom power, 24bit/96khz quality (simplex) or 24bit/48khz (duplex).On my new system I use AVLinux and I can go as low as 5 ms latency without issues. This experience is based on projects with about 6 stereo channels and some simple plugins (like reverb). If you've got a project that is more complex and you want to make sure that the UA-25 EX is good enough for that project, you may send me your project and I will do some testing with it.
The decision may depend on what you plan to do. If you want to record more than two channels at once, the UA-25 EX will not be your friend (it's a two channel device).
To answer your question about mics: This is a very complex topic that can't be answered with a few sentences (and the answer depends on what you are going to do). You just can't compare a Rode NT1A against a Shure SM58 (or Beta 58A). These ones uses completely different techniques and they aren't made for the same goal. The Shure is a good mic for "on stage use". E.g. it produces low feedback and it doesn't break if you accidentally drop it. The Rode NT1A can only be used in a (home) studio. It is a large diaphragm mic and it's very sensitive. On a stage you would hear a foot tap that's 10 meters away. That kind of mics are the best solution for studio purposes if you plan to record voice or some acoustic instruments like a guitar. It makes you sound "big". But: This doesn't mean that you can't use a Beta 58A in a studio. As I said, it depends...
I thought about the fact that this post is old. But I went through these old posts on audio interfaces when I was buying one, and I suppose others do too. So I thought new information might be relevant.
I know it is heresy to suggest using a built-in sound chip. I actually bought the Fast Track Pro, and spent a good bit of time comparing it, trying to find a reason not to return it and I couldn't. I wrote a bit more about it in the thread "Budget USB Interface for Ardour?" In particular, I recorded silence with both of them and listened to it greatly amplified and looked at the rms level in audacity. I recorded acoustic guitar with a big diaphram mic from several centimeters, with the guitar at a moderate volume. At a reasonably strong playback gain, I could hear no noise with either interface, only guitar. If I then increased the playback gain greatly in the silent parts of this acoustic recording, the preamp noise completely swamped anything else. I also tried to cause some change in electrical noise, when recording silence (ie, with no input, or preamp gains at zero, so I had some chance of hearing it) I waved the input cables around, clicked the mouse, keyboard, moved gui windows, etc. I then turned the replay gain up so the noise was loud, but heard absolutely no change from any of those things (I wish I had written down the numbers, but he noise level with the preamps off was much smaller, maybe a couple of orders of magnitude in amplitude, compared to having the preamps at a level sufficient to close mic a guitar with a large diaphram condenser mic) I also couldn't hear any difference toggling between two guitar tracks recorded with the two interfaces.
I think it might be interesting to do other audio tests, like record sine waves and so forth to check distortion from the DSP. I wouldn't be surprised to find the HDA chip is worse. I'd actually like to get my hands on another interface to try some more things. Maybe even just using both interfaces normally for a while, something might show up. But the m-audio also has drawbacks. It's difficult to get the drivers working, and sometimes goes into a crackly mode and needs to be restarted (in MSWin too), etc. I couldn't justify keeping it if I couldn't identify a single advantage.
I know the OP wanted preamps and 48V power. But the behringer 802 mixer with preamps and phantom power that I used is 58 USD on amazon, while the FTP is 165 USD.
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