External USB Sound Card / Audio Interface

I am going to be using my new notebook to produce music using Reason and Record, and so I am going to need to purchase a suitable sound card or audio interface. Does anybody have any positive experiences with an external USB sound card or audio interface that they could recommend for me? I am not trying to spend more than maybe $150, unless the unit is truly amazing. Also, it would help if it had a built in headphone amp and came with suitable ASIO drivers. Any suggestions would truly be appreciated. Thanks!

 

I have the FiiO E7 and it's spectacular for $99.

 

What do you use it for if I may ask? Does it have a headphone amp, and do you by any chance know what it can replay at (24bit/192khz, etc)? Thanks again!

 

It's a USB DAC/Headphone AMP, but I don't think it does 24bit.

I use it at work to listen to music and whatever else out of my laptop because the onboard soundcard is wretched.

 

Great, +1, thanks for your help.

Unfortunately, I will definitely need 24bit/96khz, and I would prefer 192khz, although that part isn't mandatory. I saw the Creative SB1240 that seems to fit the bill pretty well. Actually, it seems to be almost perfect, except that Creative seems to have a less than stellar reputation in the audio production world. Unfortunately, I am too new at this to know why exactly. Any other suggestions that might fit the bill?

 

I believe the NuForce uDAC-2 does 96/192. It is a USB DAC and headphone amp, also. Check head-fi.org for some other suggestions if you're interested in higher quality stuff than Creative offers.

 

I'm looking into something similar and so far my list of top choices are:

Leckerton Audio UHA-4
iBasso D4 Mamba
Headroom Total BitHead

Check those out and see if they meet your criteria.

 

I really recommend you use firewire otherwise you can have bandwidth and latency issues with higher quality uncompressed sound files.

I have a presonus inspire 1394, it was 50$ used. Much better than that nuforce honestly, they talk about flexibility when it doesn't even have mic inputs?

 

Are you looking for Mic/Line IN?

The Nuforce uDAC2 is good if your only after output. Its got a headamp and supports 96khz/24bit playback. Works well with WASAPI or ASIO.

iBasso amps is very similar, except its portable and its got headphone out for monitoring and a line out/aux in.

 

OP, please give the SIIG USB SoundCard a try...i'm not positive it will deliver what you require (you can try digging up it's specs via google) but i'd say for the price it just might be worth trying out; I was truly amazed by the loudness and clarity it provided...it's the best $18 i've spent, ever!

 

Thanks for all the suggestions. I would like to have some sort of input, as I will be sampling sounds at some point.

 

Yes, you need something with XLR inputs for decent entry level professional mics, as well as phantom power. My suggestion was the only one which has these.
The annoying thing about firewire (1394) is that it needs an external power adapter unless you get a 6 pin port somehow. I was using mine with a PC card firewire adapter which had 6 pin and a jack for some kind of power, I don't know what it was supposed to be but the whole thing seemed to run fine on USB 5v power.

I guess it depends on how portable you really want this to be, if you are more into field recording stuff then I'd strongly suggest you get a zoom h2.

 

Honestly, it doesn't need to be portable at all. I am going to be doing all my music stuff in one place, and I don't plan on taking my audio equipment anywhere.

 

OK, so this next question should completely confirm my newb status .

Amazon, Newegg, etc has a model by Creative called the SB1240 that has a headphone amp and all the other stuff discussed in this thread. The positives seem to be the 114 SNR and the headphone amp, and the negatives seem to be the lack of an EQ (and I don't remember if it had any inputs or not). Is this model even worth consideration, or should I stick with the Nuforce or one of the other models suggested? I only ask because the SNR seems to be a good 15-20 db better than some of the other models, but from the input I've received it doesn't seem like I will be turning my headphones up loud enough for that to matter. Again, all input is appreciated. Thanks.

 

Hello fellow music producer.

I have never experienced any bandwidth problems with USB2.0 soundcards. I currently use a TC StudioKonnekt 48 (firewire) however which does give me problems but they are WDM driver related.

I reckon the USB2.0 sweet spot around 150 to 200 bucks is M-Audio Mobile Pre and M-Audio FastTrack Pro. FastTrack Pro has a headphone amp and M-Audio has traditionally been very good with their Windows drivers. I can personally vouch for the quality of their Delta drivers since I owned a 1010 for years and the thing was an absolute rock. Never once had a driver related issue with it.

It also packs MIDI i/o. You'll need the MIDI input for keyboard and remote control within Reason. It has a couple of mic pres with auto sensing inputs so you can plug mics, guitars, whatever into it and record right away. Doesn't matter what the output impedance is. It'll also run Pro Tools LE if you ever decide to swing that way.

Its a pretty sweet package and I can't think of anything else that gives you so much and the nice driver support for under 200 bucks. As a general rule, M-Audio owns the budget end of the market. The EMU stuff like 1616M and 1820M were great for a bit more cash and had tonnes of i/o but sadly they are long discontinued. You might be able to pick one up second hand but anything that doesn't have ongoing software support is something I'd tend to shy away from. Right now, the absolute biggest bummer I can experience is having something not work due to flunky drivers and inflexible mixer software. As long as I have the i/o I need, everything else is something I can live or work around.

I have also formerly owned an RME Fireface 400 and had it side by side with the Delta 1010. I also had the 1010 side by side with the StudioKonnekt. I also had lend of a Rosetta 200 for a month a number of years ago. The specs sheets throw alot of jargon at you like anti jitter technologies and SNR figures and frequency response curves but frankly, the differences in sound fidelity are so minute that I never considered the difference to be anything but negligible. The real reason to buy a soundcard is connectivity, driver support and the software front end (the means by which you route physical hardware i/o to virtual i/o). TC Near gives me some WDM problems (though ASIO is fine) but the software mixer is pretty awesome. You can route pretty much anything anywhere.

I know you said you need 24/96 but I cant imagine why.

The biggest difference in sound quality, by a thousand miles is going to come down to your skill at recording and mixing. I would also like to point out that 192khz is actually kind of bad. You gotta understand that the audible range of human hearing is *at best* up to 20khz. There used to be an issue where the highest audio frequency signal exceeds the sampling rate of the converter resulting in aliasing and this was a real technical problem to overcome in like, 1995. You had to design an anti aliasing filter with a steep enough cutoff to band limit the usable audio signal without a) introducing too much ripple into the pass band and b) didn't pass enough signal beyond cutoff to let audible aliasing through. It was expensive and difficult to design an analogue filter like that but it was largely fixed by oversampling.

However you only need to oversample enough to relax the requirements on the anti aliasing filter and band limit to 20khz. 44.1khz already does that (nyquist = 22.05khz) but the only engineering consideration here is to design an anti aliasing filter that can roll off sharply after 20khz. Increase the sampling rate to 96khz and nyquist = 48khz. The anti aliasing filter does not need to roll off sharply anymore and its easy to design. At 192khz, nyquist is so far beyond the upper limit of human hearing that you aren't going to get better results. In a number of ways it actually creates more problems because 192khz wavs are huge so you waste storage disk space. It is a huge burden on RAM. It is also a huge burden on your cpu because it needs to handle realtime signal processing on so many samples per second. Dan Lavry, a well respected pioneer in the field of digital/analog conversion and noise shaping explains more of the reasons why this trend towards higher sampling rates in multi bit converters is a bad idea. You can check out his posts over at the prosoundweb forums where he used to be a guest moderator. Actually, if it wasn't for established industry standards, he suggests that the optimal sampling rate for multi bit converters is around 60khz.

I record and mix at 24/48 because the cpu and memory load at 96khz is just too much. I can literally double the amount of signal generators and processers I can run in realtime if I switch down to 48khz. My .wav dump folder is also half the size.

 

It really depends on what you're doing. I have neglected my music and dismantled my interface with hopes of fitting it into my ODD slot :/
but the sole reason I got it was to get low latency. as low as possible. I only run ableton live, live work is the only thing I actually care about ever. I would like to build my computer into an instrument.
I suppose as long as you're not like me, firewire isn't a big deal and drivers are definitely more important. I take back what I said about ableton because I have just as much love for kluppe + jack on linux with faado drivers. others are obviously more focused on pure production i see.

 

You can get lower latency by doubling up on sampling rate. Actually this is the only real reason I can see for using sampling rates beyond 48khz. At 96khz you sample twice as fast, fill up a DMA buffer and trigger an interrupt in half the time. Therefore you get half the latency. However, I find my system perfectly stable down to a 256 sample buffer, which at 48khz is something like 5ms latency. This is at pretty high cpu loads too.

Delta 1010 was great for that. I used to rock that down to 64 samples and less than 1ms but these are just numbers. I find that I can play bang on to a metronome at 10 to 15ms even. I've played at much higher latencies than that too.

 

Yea I know.
And thats were it depends on the type of music you're playing, and if you're playing with other people. I could not possibly bear 10ms latency.

I'm not a professional, just surrounded by them (jazz and classical performance majors) so forgive me if my knowledge is skewed. but for example DJs and other highly synchronized live electronics musicians are extremely sensitive to latency.
I have heard people talk about having to adjust themselves when going from 3 to 5ms. No kidding.

The music I do also has a lot of live sampling, so the latency is doubled. AND sometimes I'm trying to sync up with these people at 300 bpm. yes, its essentially free jazz lol but that doesn't mean completely free of rythm.

 

Ehh. I'm an electronic music producer, but I've also been playing solo acoustic guitar for a long time. About 14 years now. 10 milliseconds isn't all that noticeable. Lower is nicer but I'm cool with 10. Its not like a session will suddenly just go to peices or anything because I can't have 1 millisecond latency...

And 300bpm...you for real?

 

I can't thank you all enough for all the valuable info that you're giving me. This is awesome!

Hayte, M-Audio is actually the other company that I was looking at. I am actually looking at getting their 49 key Oxygen keyboard in the near future, and I have had nothing but positive experiences with them so far. I will definitely look into the models that you suggested. And you are right on another front, as 24/48 is pretty much what I will be working with, I just thought that 96 might be a nice option, although I admit I am pretty uneducated on this subject so far.

Again, thanks for all the help, and any other advise you have will be greatly appreciated.

On a side note...300bpm???

 

On a side note, that M-Audio FastTrack Pro is now my top choice. The price increase over the uDAC2 is definitely worth it, and something I am willing to spring for.

EDIT: It says that it is only class compliant at 16 bit??? And does it support USB2.0? I saw a review that said it doesn't...

SECOND EDIT: It looks like that unit isn't even USB2.0. Honestly, it looks like that unit is pretty dated. The Ultra version, however, looks totally sick. Of course, the price skyrockets from $160 to $300.

 

fiio e7 + e9 combo is really nice

 

I have looked at those, and I agree that they aren't bad for the price. That Fast Track Ultra is freakin awesome though, and I am really going to have to use every ounce of self control that I have not to drop 3 bills on it. If I could find one that had exactly those specs, with only one or two inputs for a hundred or so less, I would be sold. It sucks that the Fast Track Pro is so outdated.

 

I'm not sure where you are getting your information from but FastTrack Pro definitely does 24/96 because I've used one. You can't do 4in/4out at 24/96 though. I also never had problems via USB.

I have an Access Virus TI and I've used it as a USB1.1 audio interface. You don't ever get problems unless you do things like attempt to run multiple USB devices off the same port together (i.e. via a hub).

 

If you are willing to spend more I hear Echo is good. I learned about Echo because on another forum, they topped the after sales satisfaction poll with RME in second place and those two were out front *by a mile*. I can vouch for RME under Windows 7 for sure and I think I should have stuck with it but Fabrik C/R and the remote coaxed me away.

AudioFire 2 is the only Echo product that seems to clock in at under 200 bucks but doesn't give you alot for the money. The AudioFire 4 is comparable in terms of features to FastTrack Pro and I've heard its name ring out with its bigger brother, the 8 but on Amazon.com its nearly twice the price. I think you might want to take a step back and make a list of the things you really need as well as the upper limit of your budget.

 

Hayte, thanks for clarifying. I was looking at M Audio's website, but I was getting confused.

So far, my requirements (I think) are:

1. 24/96
2. USB 2.0
3. Decent headphone amp to power my HDJ-1000s (the new, special edition model).
4. At least one input.

I have a question about this subject as well. When you get a better sound card / audio interface, it will have a better chip / processor in it, which should take come of the audio workload off of the CPU, correct? I'm not worried about my CPU, mind you, I'm just trying to get a feel for how all this works. If I think of any other requirements, I will come back and add them. Again, thanks for answering all my questions!

Also, I have heard of RME, but I know nothing about them, starting with what model I should be taking a look at.

 

Nah it doesn't work like that.

A soundcard is just an array of inputs, an analogue to digital converter (ADC), a bus interface, a digital to analogue converter (DAC) and an array of outputs.

There is some cpu overhead at the driver level which takes some explaining...where to start?

When you record an analogue source, the electrical signal gets quantized. Its a realtime process and is synchronized to a clock source so the bitstream that is produced is constant. As you are recording, the bitstream gets fed directly into system memory (RAM) via a process called Direct Memory Access (DMA) from the USB or Firewire bus. Wherever your soundcard is connected.

With streaming comes the concept of buffering which is where latency comes from. The bitstream gets fed into RAM in small blocks or buffers. I have my ASIO buffer set to 256 samples, so what happens is that every 256 samples, your soundcard generates an interrupt. The CPU queues up the request for a new buffer and then generates another buffer. This will happen every 256 samples. The time it takes for the buffer to fill up, generate and resolve and interrupt thus readying the next buffer is your latency. It will be on the order of milliseconds.

If for whatever reason, the CPU fails to generate a new buffer in time, you get an audio drop out or popping/crackling noises. We call this a buffer underrun. Theres lots of reasons for this but most can be mitigated by not loading the cpu down to full capacity and by using larger buffer sizes (more time for a CPU to execute an interrupt request).

There is some cpu overhead associated with the interrupt handler and the generation of interrupts but it should be more or less the same for every soundcard, assuming the drivers aren't badly written. But in the scale of things, this overhead is very small compared the cpu load from running signal generators and signal processors in your DAW. These are things like software synthesizers, samplers, effects like compressors, EQ, filters, reverb. These are all realtime mathematical processes and some are worse resource hogs than others. SIR reverb + long impulse response + high sampling rate (say 96khz) is going to slay your CPU for instance. Thats realtime convolution on 96,000 samples per second with 32 bit precision.

CPU load will eventually become an insurmountable problem but you can mix efficiently to mitigate it to some extent. For instance, you don't want to be running 4x the same reverb instance as channel inserts. Instead you want to run 1x reverb on a send bus and route however many inserts you want through the send bus. Thats 4 times less cpu load to achieve the same thing.

You'll pick all that stuff up as you learn. About RME. I'd forget about them unless you want to spend $700+

 

You rock. Thanks for all the clarification.