Wednesday, January 07, 2009

Ardour on Mac OS X

Ardour is an open source, free to download digital audio workstation that runs on Mac OS X and Linux. According to Ardour's website: "Above all, Ardour strives to meet the needs of professional users. This means implementing all the "hard stuff" that other DAWs ( even some leading commercial apps ) handle incorrectly or not at all. Ardour has a completely flexible "anything to anywhere" routing system, and will allow as many physical I/O ports as your system allows. Ardour supports a wide range of audio-for-video features such as video-synced playback and pullup/pulldown sample rates. You will also find powerful features such as "persistent undo", multi-language support, and destructive track punching modes that aren't available on other platforms. "

I've used a variety of DAW's over the years (primarily Digital Performer, Pro Tools, and Audio Desk), as well as tape machines. I've long been a fan of Mark of the Unicorn audio hardware (I currently own 3 of their units: 896, 896HD, and the Traveler) because they employ very clean converters and preamps, while being relatively cost effective. However, unlike when I first locked into MOTU's gear some 10 years ago (the original 2408), there are alot of cost effective audio units out there these days. More to the point, eventually I would like to move all my systems over to open source platforms or AT THE VERY LEAST have the option to walk away from Microsoft and Apple if I decide to. Microsoft's future is looking more and more dismal all the time, though I've never been a big fan of trying to do audio on windows, and Apple is becoming more of a fasion company than a tech company. Their new laptops are a perfect example.

But, before I get too far off on a tangent, this is what led me to look for another option. Ardour can be installed on Mac or Linux and for Mac, it uses the native aqua ui, and *not* X11 (thankfully). I wanted to see if I could test out Ardour with my existing rig before starting to explore options of new interfaces and I was very pleased to see that it installed and worked flawlessly. Thanks to Jack (which must be installed as well) using the Mac OS's core audio interfaces, and MOTU allowing their interfaces to use the same, I was able to fire up Ardour, select my interface, and get working with very little tinkering. I downloaded and installed the applications, setup a project, and recorded a 4-track drum part in less than an hour, and was pleased to see also that I had access to all my AU plugins. Everything sounded great straight up and it was fairly easy to use, with knowlege of any other DAW.

So, for now, lets say this looks like its going to be a fun project and hopefully in the long run, will give me an option when I decide to buy new hardware, to buy hardware with open source drivers so that I'm not tied to an operating system. I'll be updating as I continue to work on this project with tutorials of anything I can't find on the internet, as well as my thoughts and experiences along the way, but for now if anyone else is feeling a little depressed about the direction of apple... give ardour a try with your existing hardware... its easy to use alongside whatever you're using now, and will help you evaluate your options down the road.

Thursday, November 06, 2008

MOTU Fast Lane USB

Just when I'd finally forgotten about Apple taking away serial ports from their machines, I decide to add a small midi setup to my project studio only to realize: My old MOTU MIDI Timepiece is now a 1U place holder. Well, these days I don't have use for a full blown MIDI interface, and since I've used MOTU products for many years, I decided to look into their low end offerings.

The Fast Lane streets for about $70. Its a simple 2 in 2 out interface that connects and is powered via USB with an optional thru switch in case you want to use it without a computer connection as a dumb forwarder for device control. Straightforward enough, right? MIDI's been around since the early 80's and remains a robust standard for all kinds of sequencing applications. It hasn't really changed alot. So why does this new fangled USB MIDI interface slow my whole DAW down when I try to use it? Using the Fast Lane in Digital Performer, I've experienced serious performance issues and an increased amount of random crashes (always an occaisional problem in DP).

Not a whole lot else to say, it is after all, a fairly simple device. Still... if you use MIDI much, its likely you'll find the need for more ports sooner or later... either way I'd recommend shelling out a little more cash for a higher end interface. For now, I'll use my Traveler when I need MIDI while grumbling about my worthless serial MIDI Timepiece.

Tuesday, October 21, 2008

MOTU Traveler Firewire Audio/MIDI interface.

This firewire interface from Mark of the Unicorn really *IS* a traveller, in so many ways. I've been using it for several years now, with very few problems. Its a feature complete workhorse for anyone doing mobile recording, and an asset in the studio as well. Let me run down the prominent features for you:

  • Firwire bus powered
  • 4 onboard preamps w/ phantom
  • Full bussable mixer
  • 8 audio ins/outs at a max of 192k/24bit.
  • 96k optical in/out
  • 96k AES/EBU in/out
  • MIDI in/out

It also has ADAT and wordclock sync. The preamps on this guy are real clean and smooth... nothing spectacular, but pretty transparent. The biggest advantage of this unit is it doubles as a decent studio capture device (the converters are real clean) while being the most fully-featured mobile I/O you can find. At 1U, it has the same footprint otherwise of a 15" laptop.

Drawbacks: the mixer, while extremely versatile, is very difficult to use without MOTU's cuemix software component. In the rare event that you are using it without the software component (which it will work as a full fledged mixer standing alone), be prepared to be frustrated. Also, this isn't really a complaint about the traveler, but something to consider... firewire bus power is great, but in most cases where it will be most useful (no outlet in sight), it won't get you far as its going to help drain your laptop battery much faster (and if you have an Apple, that's already too fast). This unit will link up with any of MOTU's other interfaces to give you more inputs in various configurations. They're very expandable, but be careful because monitoring gets a little weird using more than one MOTU box at a time.

MOTU bundles the traveler with AudioDesk, their neutered version of their flagship sequencer, Digital Performer. While AudioDesk was once a very decent sequencer for simple audio tracking, it seems the company has begun to ignore it. It hasn't seen an update in quite some time and the last time I tried to run it, it wouldn't stay open without crashing. One last gripe: for a long time I've debated getting a working linux studio running, but Mark of the Unicorn has made it very difficult for any sort of linux drivers to be made for their products. Though I've been a longtime customer of theirs, this one stance is making me seriously reconsider when the time comes to upgrade again.

All in all, this is a prett killer interface if you record on the road at all. With just a laptop, the traveler, and a few decent mics, you can do a very high quality recording with no external power, no additional hardware... AND when you bring it back to the studio you can plug it in to the rest of your MOTU gear and expand your input section with 8 more high quality, balanced, 192k/24bit inputs. All in all a great value if you have need for recording on the go.

Sunday, March 09, 2008

The boogeyman of mixing and his worst enemy (understanding phase problems).

Now, recently I've been trying to ignore technical issues in favor of KISS (keep it simple, stupid), because creating a great recording is only 10% gear, maybe less, BUT when you're recording and mixing in tiny, poorly designed rooms, its nice to have lots of things in your mental toolbox.

The biggest boogy monsters of most home studios are phase problems. Phase problems can result from all kinds of different scenarios... cancellation from multiple sound sources (as when you're listening on your studio monitors), cancellation from multiple capture sources (most often from attemping to use multiple mics to capture the same sound source, or capturing different sounds in close enough proximity that you're getting alot of bleed), and they can even be introduced as a result of the distance at which you mic a source.

Which seems to make this a good time to explain what phase cancellation is. We even have a neat little visual thanks to wikipedia. Basically, in theory, if a particular soundwave is 180 degrees out of phase with another wavelength of the same frequency, those two completely cancel each other out, resulting in no sound.

Now of course, this isn't exactly the way it happens in real life (but this is the principle that all these new fangled "sound-cancelling" headphones are based on) because very few sound waves come completely isolated, but think about the implications: The closer two wavelengths get to being 180 degrees out of phase with each other, the more they're going to start disappearing. This means, not only do you have to worry about the scenarios I described in the last paragraph, you also have to start thinking about different sound sources that have similar frequency information (for instance, the low end of an electric guitar might share alot of frequency range with the mids of a bass guitar)... this is generally thought of as "masking".

If you're thinking about this alot right now, it might seem like a big scary mess, but dealing with it in the studio isn't actually that difficult. There's some simple techniques that will bring you "phase enlightenment". First, in your monitor setup, you *need* a mono button. The easiest way to know if there are stereo phase issues in your mix is to switch it to mono and listen for what disappears. Even if you can't nail it down to something specific, but it sounds weird in mono... you have phase problems... go back to anything thats panned and work with it till the mono mix is as solid as the stereo mix. If you're thinking to yourself, "who cares though, if it sounds kick ass in stereo, no one's going to actually be listening to it in mono"... remember that there are thousands of different stereo speaker setups in millions of different rooms out there, in which you have no control over how those sound waves are going to be interacting... thats part of the job of someone mixing or mastering a record, to make it as solid as possible for all of those combinations (I'll address some other ideas on that specific topic in a later post).

One more note on stereo phase cancellation. Anytime you're trying to get a stereo mix from a pair of microphones, remember the 3-1 rule. Any microphones that are micing the same source should be at least 3 times as far from each other as they are individually from the source. Which means for drum overheads, if they're 4 feet above the drums, they need to be 12 feet from each other, which will not be practical for most situations. In some cases, you might be better off using an xy pair, which is a very specific method of stereo micing. Take two mics of the same type and line the capsules up, facing each other at exactly a 90 degree angle (or more). This way the overlap of the two pickup patterns is minimized, resulting in a lower chance of phase cancellation.

As we slowly move to less complicated sound fields, the solutions become slightly more complicated. Anytime you start to feel like two instruments are "fighting" each other in the mix, they're trying to share the same frequency ranges and there's just not enough space for both of them to be dominant across the whole spectrum. This may or may not be phase-cancellation related, but if you use this technique, it will help eliminate phase-cancellation issues between different instruments in your mixes. Think of your mixes like a sandwich. What needs to be at the base, and what needs to be at the top? What needs to be the dominant flavor, and what should merely be an accent? If you apply that theory to mixing, keeping in mind what you can compromise on, your mixes will clean up right fast. So your electric guitar sounds beautiful and full with this huge bottom you love, but the bass has no definition; what's the important part... can you sacrifice some of that low end on the guitar for the bass? Each instrument needs its place in the frequency spectrum. Not to say that there won't be overlap, there will be alot, but if you map out the frequency spectrum based on where your different instruments really shine, they stop fighting with each other, and where they do, you know where you can start cutting frequencies out to make room.

Worst case scenario, when two instruments are fighting each other, you can always solo them out to try and make them play nice, isolate where their overlap is, and mold them together. However, it's always a good idea to not spend too much time with anything solo'd. You start to have the tendency to create a great sound for the solo'd instrument and forget about the mix, then when you switch back, you've changed the whole landscape and you have to start over again.

Alright, so what if the problem is just one instrument fitting into the mix? The first thing to do is to invert the phase of that instrument in the mix and see what happens. It might just be a sonic revelation! I hate using exclamation points, but when you hear a phase problem pop out at you, it really can be eye-opening (or ear-opening I suppose). Most audio sequencers will have an invert-phase plugin you can put on a channel, otherwise, if you're still in the recording phase, you can try to invert phase by keeping a few mic cables around that are wired backwards. Make it a weekend project, so you have them around when you need them in a pinch. Just swap the hot and cold wires on one end of the cable... easy to do with some cable ends and a soldering iron (what?!? you don't have a soldering iron? how long have you been running a home studio?).

One last scenario in this long winded post... and really this is what motivated me to write this in the first place this morning. When you're recording an instrument and the sound you get on the recording just doesn't line up with the way you hear the instrument. In this case, it's time to start thinking about mic placement (which you've already thought of right? that should be priority #1, but that's another article too). So, it is possible to have phase cancellation using one microphone recording one sound source. This is particularly true with instruments that produce purer notes and waveforms (say, violin) than more complex ones (distorted electric guitar), but its another thing to add to your toolkit, nonetheless. If you try to imagine a sound as the physical waveform that it really is you can see where overlapping waves from the same sound source can produce phase cancellation at the right distances. There's a funny thread over at the sweetwater forums that talks about a specific scenario with a violin like this, combining reflections and direct sound at particular wavelengths. The nice thing about that thread is the note at the end from a proffessional engineer reminding us to not get too caught up in technical details, which is full circle for this post. The bottom line is: move the microphone. Don't try to fix it in the mix. Keep moving the microphone until you have the sound you want. Microphone selection, but really microphone placement is the holy grail first important step of a good recording. No amount of great eq'ing or fancy preamps and effects will do you better than moving the microphone around and getting used to really listening to things and how sounds change.

So, basically if you're letting all this information turn into soup in your head like I did, you might be curious about the physical wavelengths of different frequencies... I stumbled across this wavelength calculator. I want to reiterate not to think about this stuff too much when you're actually working, trust your ears, but its nice to have some idea of the theory, and kind of cool to think about the lengths of different sounds. Just remember, there's no secret sauce.

What is the boogeyman's worst enemy? Mic placement. Know how to position your mics and experiment with moving them around in relation to each other and to your sound source and you can solve almost any problem. Some of the other tricks I've mentioned will still be super useful, but the fundamental is always where you put the mic and how the thing sounds in the room. Concentrate on what you hear and where you put things in the room... don't pull out the big complicated words if you can fix things by moving a mic.

Friday, February 01, 2008

AKG C3000

The AKG C3000 is one of AKG's low-end large diaphragm condenser mics. When you can still find it, it streets for $300 or less. This mic is no longer produced by AKG, but is widely available at affordable prices used. It was replaced in AKG's product line by the C3000B (notes on the differences between the two will come later. The C3000 is a large diaphragm condenser with a switchable pickup pattern, behind the main large diaphragm capsule is a small-diaphragm capsule, the output of which is combined with the main capsule to produce either a cardioid or hypercardioid response. It also has a low cut (6-dB/octave slope below 500 Hz) and a standard-ish 10 db pad.

Ok, enough on the specs. I was very excited about this mic when I first picked it up, and after numerous attempts at giving it a fair shot, I was a little disappointed. While it does have the "big" large diaphragm condenser sound of its bigger brothers (the C414 or the AT40XX series), it sports a higher noise floor than those mics and some strange emphasis in the high mids, that, while useful in some cases, I found to be overall an annoyance. During my tenure with this microphone (I used it in sessions for almost 2 years) I was able to apply it in a few cases, mostly on instruments, where it sounded really great, provided that the room I was recording in was exceptionally silent and the instrument lent itself to a particular wanky brightness (certain acoustic guitars and brass, also had luck with a saxophone with this guy). In general, I found myself favoring other large diaphragms in my collection at the time over this one in side by side auditions (the AT4050, and sometimes even the CAD E100, and of course, when I had access to it, the Nuemann TLM103, no surprise there).

Now, presumably AKG discontinued the original C3000 because of some of the very annoyances that I've mentioned. The C3000B is a significantly different design, sporting a single capsule design and a fixed pickup pattern. The C3000B has its pad before the transformer that steps up the signal, which means that it technically has the ability to handle 10db higher SPL than the original C3000 with the pad engaged (though I can't imagine many applications that you'd want to use this mic for where you'd be pushing the 140 or so db SPL limit). It's also been said by other reviewers that the electronics in the C3000B are of higher quality than the original, and that the capsule is closer in design and materials to the C414 than the original. I haven't had the chance to test out the C3000B so I can't speak to it personally, but it seems to be a considerable improvement over the original.

Though this is a reasonably decent mic for the price, I would definitely say there are other mics out there with similar specs in the same price range that will give you better results on a wider range of sources. Outside of the C3000B, AKG's new perception series looks like a promising alternative in the same price range, not to mention the numerous other low-cost manufacturers releasing mics in this price range... in general, unless you're looking to amass a collection of low-cost mics with different characters, I'd say steer clear of the original C3000 in favor of other large diaphragms in this price range.

(normally I try to include a scan of the frequency response for a mic, but I was unable to find one for this particular model. If you have a resource for this, please drop me a line).

Monday, January 28, 2008

Build your own ribbon mic.

Just stumbled across this tutorial to build your own ribbon mic (for about $150 - $200 in parts) in the reviews section of the most recent issue of Tape Op magazine (which just happens to be the coolest thing since sliced bread... free subscriptions even). I haven't gotten the tutorial yet, but when I have the money and the time, you can bet I will. In the meantime, if anyone reading this has done it and wants to share their experiences, drop me an email and I'd love to hear about it (and/or post it for other curious folk).

The Tape Op review says, "I'm very confident it will utterly smoke any of the bargain ribbons currently on the market." My mouth is watering already.

Thursday, January 10, 2008

Photos from the Ten Speed session.

Here are a few photos of our makeshift rig we put together in a friend's house for the recent recording I did with the band Ten Speed. The record has yet to be finished, as we're still working on vocal tracks, but should be some cool sounds once it's all done:My little control room area was upstairs from the room that the band was in, and though we had the amp sectioned off and band members wearing headphones, we recorded drums and guitar together in the same room so they could make eye contact and have that personal interplay. There were even several songs we recorded vocals and kept the live takes. We recorded to a Tascam 388, an old 1/8" 8 track reel to reel. We mostly avoided using the built in mixer, as these old tascam boards are very blocky, both in the pre's and the eq's/ We did use the built in pre's for one of two kick drum mics (beater side AKG D11) and for a snare mic (57) which sound pretty decent. The kick drum signal was mixed together live to tape with an AKG D112 on the resonant head run through a dbx 376 tube pre with some light compression and eq'ing.




















For overheads we used the wonderful Rode NT5's, which I've already given a bit of treatment. We ran these guys through two Studio Projects VTB1 tube pres. The VTB1 has both tube and transistor circuits with a knob to control how much you're driving the tubes. They tend to overdrive pretty easily when you're moving into the yellow and redzones on the tube side, so we used a little of both and kept them backed off on the input stage to stay "in the green" as it were.

The guitar was recorded using the Nady RSM2 Ribbon Mic. We auditioned several mics on the guitar amp, including the Sennheiser 421, Shure PG 81, and the classic 57 on the grille technique, but ultimately the ribbon mic a few feet off the cab yielded the richest, most balanced sound of all the ones we tried. We ran the ribbon through a dbx 576 tube pre with some light eq and compression from the 576 as well. In general we slammed the guitar to tape pretty hard for that smooth tape compression.

The guitar amp was tucked back in a closet as you can see, and we got a nice thick low end from it thanks in part to the closet capturing the sound coming from the back of the cab.

Vocals and harmonica were recorded live on several tracks and a few of them were keepers. We used the Sennheiser 421 large diaphragm dynamic for a live vocal mic. We originally tried the Rode NTK, but there was far too much bleed for it's delicate sensibilities. The 421 did an adequate job though I missed some of the wide openness of a condenser. All in all the 421 catches the rich low tones so well that it makes up for some of what you'd miss in the delicacy of a condenser, and it's ability to be focused in a sometimes loud environment makes it an ideal choice for an application like this.
All in all the session was a hell of alot of fun and we got some great sounds. Unfortunately things got cut short when the neighbors complained, but we're set up in a new location to finish tracking vocals and bass and hopefully shortly thereafter we'll be going into Pragma Studios to dump to digital and mix.