Audio Interfaces

So, I'm getting a bit fed up with my M-Audio Delta. Let me be clear, it's in no way dissapointed me. Right now, it's doing 24bit/96khz @ 5.3ms latency and is going rock solid.

The problem is that it only has 4 inputs, and I have 3 synths plus microphone, so you can see where this goes. Right now I just have everything going into a shit Behringer mixer then into the soundcard on two channels, which works, but means that applying effects is an all or nothing deal. So I'm on my quest for a soundcard with lots of inputs, don't care about multi outputs. I would prefer to stay at 24bit/96khz if possible.

Why can't there be a card on the market that just has a shit-ton of inputs, and only two outputs? Do people really need 10in/10out cards that much?

Ideally, I'm looking for something around $300. I had noticed the Phonic series of firewire mixers, offering 12 channels for $269, at 24bit/96khz. I even read a good review in SoS! But since SoS has not given a bad review to any piece of gear sent to them, ever, I am rather dubious about this. Still, it's the frontrunner unless someone can disuade me. Anyone know of any latency issues with firewire audio such as this?

The only bit of advice I can offer is don't get an M-Box. They've been known to be quirky with anything except Pro Tools.

Good to know, although they seem to be of the same "2in/186out" weird mentality.

You might want to check out the Echo Audiofire 8. It's a bit more than $300, but I've always found Echo products to have pretty good sound quality, and their drivers are always stable. I wouldn't recommend getting anything too much cheaper.

What's frustrating is, why can't they offer something like that that's 8in 2out and make it cheaper? Who the fuck needs 8 outputs?

i have had enough of my m-audio firewire audiophile... the drivers for this bastard have to be reinstalled just about every time i turn on my computer. it'll record from it's RCA input for about a minute before it buzzes like mad. fuck the thing, i'm poor and can't efford anything else though.

Hmm, that might be a PC issue rather than an interface issue, actually.

One big problem Im having with this Phonic Firewire Mixer deal is that they've obviously sent sales reps out to web forums to try and sell the product. I mean seriously, who signs up for an account on homerecordingconnection.com to post just this:

  Quote

I've just ordered the 24 channel version of this from my local high street store. However they were good enough to let me take away and try out this model first, the Phonic Helix 12 firewire, to see wheteher it was for me or not.

I have to say I am impressed by the performance of this desk. The audio signal from 4 microphones around my bands drumkit was streamed simultaneously onto 4 tracks on cubase just as described. The drummer could play along to Cubases click track and hear himself back, without any noticeable delay. The Firewire really does it's job. To try and push it's limitations we got our singer, bassist, and two guitarists to simultaneously play along to some drums and other instruments we'd laid. This meant 12 tracks of audio for Cubase to send back along the Firewire (which it does as a Stereo mixdown), to the desk for our musicians to monitor, while processing their 4 inputs and feeding it back to them again so's they can hear themselves. The desk had no problems. I believe the latency was as low as 6ms. (anything less than 10ms is undetectable).

So in terms of processing audio and getting it onto Cubase, without the need of a fancy soundcard this desk is great.

 

The one down point I found was that after setting all the levels nicely according to the desk meter reading, the corresponding input level on cubase was just a little bit lower than desired, resulting in audio waveforms that looked a bit underfed.

Unfortunately I could find no way of turning up the firewire master output on the desk. In fact, it's worth noting that the signal sent to Cubase is not affected by any of the faders or EQ controls. And also the desks effects are redundant in firewire mode. This isn't a huge issue, because you really want to be using cubase to add all the bells and whistles to your audio tracks afterwards. It does mean however, that the only control you have over the level of each channel on the desk is the gain control. A fader value of 0db seems to be assigned by default, regardless of how much you push the fader up and down. Moving the fader, will however make the volume of the track sound louder in your headphones or control room speakers, but it won't change the signal level in Cubase. Only the gain control will do that and at it's loudest it's just a little bit low.

Good news is that a simple solution was found. Opening the mixer in Cubase SX (press F3) and choosing to view the whole mixer + input channels, you can simply adjust the input gain for each channel being sent from the mixer. A little tweak for each channel here, completely resolved the issue, giving nice big healthy audio waves.

 

In case anyone else has one of these, here's a few things that seem obvious but took me a bit of mucking around to get.

1)In order to find the Phonic Helix inputs in the Input drop down box next to each track in Cubase, you first need to create them.

Go to DEVICES > VST CONNECTIONS.

Then Right click in the columns and select ADD BUS. Add several buses and give assign each one a relevant input channel from the phonic helix. Hey presto! They now appear in the INPUT drop down box next to each track.

2)Another thing which had my mate stumped was how you simultaneously record onto multiple tracks on Cubase. This was easy. Simply create a few tracks and assign each one an input channel from the INPUT drop down box. Select the very top most track then hold down SHIFT and select the bottom track. A multiple selection has been made. Now just hit record.

 

My overall opinion of this desk is that it's great value for money and not a thing wrong with it. Don't let anyone tell you Phonic are crap. It might be a little less robust than professional models, but look after it and you've nothing to worry about. The thing that has always got me is that these things call themselves 12 channel desks when theres only really 4 inputs worth using. The rest don't have gain controls and the number 12 is got from including stereo returns and auxilliary ins and stuff like that. But then, all desk manufacturers are at that game

Wow! Tell me more about the click track!!!!

if you get another m-audio soundcard, with the amount of inputs required, your pc will detect them as one single audio card, and you can use them together.

been using m-audio sice they were called midiman, and never had a single complaint.

currently running a usb audiophile and a pci delta 1010.

superb audio hardware.

incidentally, why doesn't your desk have auxiliary sends and returns?

why would effects be an all or nothing deal on a multi channel desk?

get a b etter desk... even in the behringer range, you can get a 1604 eurorack.. 4-mono, 4-stereo, 2 aux, stereo tape channel 3 buss out... all for about a hundred quid...

Interesting, so even a USB audiophile will link right up in that way to my Delta-66?

Well because I don't have any outboard effects, just VST's. So if I mix down to 2 stereo tracks and put that into the PC, all I got is two channels of audio that is 3 synths mixed together. Technically, I could use the other pair of inputs on the Delta to go through the mixer as an insert effect, I think, but that still wouldn't really give me the level of control I'm after.

Plus fuck an analog deck in front of the interface is my new way of thinking. I'd rather just go direct from gear to 24bit/96khz rather than adding to the noise floor going through a mixer.

answer to your first question... 'yes'

i see your point about a crappy analog desk.

word.

a decent outboard hardware effects unit is a good thing to own though, for many purposes.

incidentally, m-audio's tech support is quite good, although it sometimes takes a while.

edit: also, why bother recording at 96 if you're going to bounce (with heavy aliasing) to 44.1.?.. you'd be better recording at 24/44.1... less maths involved in the master bounce.

I'll add it onto the list of things:

More synthesizers.com modules

New audio interface

Proper monitors

etc...

It's really not important to me right now, as I can do tons with VST's. Just need a good way to apply them all.

Gordan Reid explains this better than I can, but recording at 24/96 even when eventually mixing down to 16/44 will produce better quality results. I'd have to look up the article again and I'll post it here if you like when I find it. Interesting stuff.

i've read loads of stuff on the subject....

my opinion is, it's better to stick to the sample rate you're going top end up with, because there's less maths in the bounce... keep the bitrate as high as you want... makes mixing easier if you have everything you need right there until you bounce.

opinions differ.

i was just reading this:

http://www.digitalprosound.com/Htm/SoapBox/soap2_Apogee.htm

48 vs. 96 kHz

i can't tell what's just made up to sell stuff, like those $300 a foot audio cables, and what's real

they start by saying that originally the need for sampling above 44.1khz was because of how shitty the filters were, then say the filters are much better now. so then they move to saying you should work above 44.1khz to capture the interaction of ultra-sonics and their beat frequencies that happen when multiple instruments interact.

then they say stuff like this

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On the face of it this is quite absurd. Do we need to capture “audio” signals at up to 96 kHz? Obviously not – such signals don ’t exist. However, some recent research suggests that the human brain can discern a difference in a sound's arrival time between the two ears of better than 15 microseconds – around the time between samples at 96 kHz sampling – and some people can even discern a 5µS difference! So while super-high sample rates are probably unnecessary for frequency response, they may be justified for stereo and surround imaging accuracy. However, it should be noted that many authorities dispute this conclusion.

all sounds like voodoo to me

do you guys know any sites where i can read more about this stuff?

But, the "maths" are there anyway, the only difference is when you're starting from 96khz source you have much more information from which to interpolate.

I have the 16 channel Phonic firewire mixer and I love it. It's got 2 firewire outs for some reason, I haven't figured out what I could use the second one for really yet. I guess I could process some channels on one computer, and others on another, If I had some cpu intensive effects I wanted to use. You can return the Main L/R (the firewire bus actually carries 18 channels), So you could have all your equipment going in, run separate VST effects on each channel, then send it back to the mixer for monitoring, which would be nice for jamming/playing live. I haven't been using an VSTs for a while though, I just use it to record.

Does it feel cheap at all? Are you running it at 24/96? What sort of latency are you getting?

I don't think it feels cheap at all, which is why i bought one. My friend got one, and I was impressed by how solid it felt considering the price. I am running it at 24/96, but I'm not sure of a good way to test the latency? It doesn't seem to be reported in the control panel for the interface

  loganfive said:

i've read loads of stuff on the subject....

 

my opinion is, it's better to stick to the sample rate you're going top end up with, because there's less maths in the bounce... keep the bitrate as high as you want... makes mixing easier if you have everything you need right there until you bounce.

 

opinions differ.

Its true that bitrate makes more of a difference than anything but I wouldn't go so far as to say the quality is better if you stick to 44.1 the whole time. I mean, this isn't about opinion, theres mathematical reason to believe that this is not the case. Especially when recording and doing processing with individual tracks afterwards. Besides granulation noise (which dithering fixes) theres nothing to worry about when downsampling).

Although its pointless to upsample anything that is previously lower unless you are a applying an effect(such as reverb) at a higher quality rate. I that case upsampling is okay, it actually helps you keep information.

caveat: have not read thread

answer:

i have a delta 1010 and i've had a long and wonderful history with it, sans a falling-out or two. after i'd had it for a bit under two years, it died on me. it may have been a power brownout that nuked two of my other boxes, but that's besides the point. i opened it up and found a capacitor had been blown, which was not hard to fix, thankfully. it worked fine again for a while, at which point it died again. later, it mysteriously fixed itself, and has been good since.

other than that ramble, anyways, it's great. latency is quite good, 8 channels of audio essentially, set to in or out in stereo pairs, so you can get 8in, 8out or 6in, 2out - etc etc. midi in/out/ w0rd muthafucka clox i never use

solid as a rock whenever it hasn't been busted. get the rack version, you boor, not the LT.

i've also had a tascam us-428. it's very plasticy and things have gotten broken on it. there is no way to custom-program the interface. since my audio box does not have usb2 (don't ask), it never did very will with audio anyways. on the other hand, i do get a lot of use out of it as a transport controls i can attach to a 15ft usb cable and have sitting somewhere convenient. but the delta was $100 more and oh so much worth it

before either of those i had a delta 2496 - also great, but a simple stereo card (+midi, spdif), and that's not what you're looking for is it??

edit: re-upsampling

sony or shrap or whatever was advertising some tv that had a 120hz framerate (that was the "innovation," of course it was diluted in marketspeak), double the signal source rate of 60hz. this struck me as absurd. all they could do, unless hdtv signals are some really funky calculus shit, is essentially show the same picture twice - which, if you ask me, would introduce MORE flicker than showing it once, for twice as long.

stupid.

upsampling helps when filtering out freqs above nyquist (aliasing).

adjective: yep its true that a lot of people have <15us auditory acuity so theoretically a >44.1khz sampling rate would benefit those ppl. if you want to read more about that, search the AES magazine's archives there's ben lots and lots of discussion going on about sampling rates VS audio acuity.

more ontopic, HAVE A LOOK AT THIS http://www.thomann.de/de/alesis_multimix_16_firewire.htm

now that's value for money! forget the phonic mixers..

now, about that click-track ...

iep, I'm not sure how that's a better deal than the Phonic stuff.. You can actually get the 16 channel Phonic for like $100 less than that.