solarbird: (made her from parts)

I thought I was going to be posting about a new microphone I built today! I was pretty sure I could get the kit construction finished and even make a couple of recordings.

And I could’ve, but no, I had to have one of these moments:

…wherein on page 24 of the assembly instructions it talks about connection sealing material and special kinds of thread-locking fluid which aren’t strictly necessary but are definitely good to use, if, you know, you just happen to have them around, and if you don’t, well, you can add them later but EVERYTHING WILL EXPLODE.

Particularly the connection sealing material. Apparently.

None of this was in the components and tools list up front. Of course. So thanks, now I get to order all that stuff, which I already have, and it’ll get here Wednesday. (No, it’s not at the local hardware. I did check!)

Anyway, this is how far I got, and where it’ll be until Wednesday:

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solarbird: (molly-oooooh)

I wanted to post about the cool second-generation crystal microphone today BUT NO IT’S ALL STUPID AND NOISY AND I DON’T KNOW WHY but it sounds like either a really bad cold solder joint (please be that) or a bad transistor (@&*$&!!! special orders please don’t be that) and I don’t know which.

It’s too bad because I came up with a nice little jury-rig jig (say that five times fast) and so the backplate of the housing came out really well and I was looking forward to showing that off. Fingers crossed this is some sort of Surprise It’s Easy! fix – that would indeed be a surprise, to be honest about it, but a pleasant one.

In the meantime, enjoy this video of Overwatch players in custom game mode making some genuinely gorgeous Genji Beams. These are effects created by lining up opposing teams of Genji players opposite each other, in continuous-shot-deflection mode, and hitting them with various weapons. The shots bounce back and forth between the teams, and you get some really neat graphics interactions. It’s pretty cool and occasionally hilarious. Enjoy:

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solarbird: (pindar-most-unpleasant)

i was going to post the more than vaguely punchdrunk tweetstorm from my 13-hours-and-counting marathon run-in with Microsoft technical support about why windows hadn’t been letting me install security updates since AUGUST directly into the blog, because some of it is pretty funny

and i’m gonna do that anyway

but the seventh (7th!) tech sport did something i specifically said we cannot and must not do and destroyed my desktop machine, which drives all my audio software

(seriously, completely levelled it, i’ve got partition recovery running right now)

(broken boot table, no more linux system partition, no more swap partition, can’t even get to windows loader because grub2 is 100% made of “wot?”)

so i just storified the twitter rant instead of making a fake collection here (it’s pretty ragehappy) and then played the hilariously stupid current special brawl in overwatch, which is all pharah and mercy (phamercy brawl! <3) and double TriplQUADRUPLE kill Play of the Game until i felt better.

because it turns out i get pretty good at smashy brawls when i’m, like, really mad and have rocket launchers.

so, yeah. fun? enjoy my ranty goodness while I’m rebuilding my machine, again.

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solarbird: (music)

Oktava has some great microphone designs. But the quality of the components can be pretty random, particularly in the used market, since a nontrival number of those were made in the early post-Soviet era. My two 012s sounded pretty different – one in particular rather unpleasantly harsh – so I implemented Recording Magazine’s recommended component upgrade* on the harsh one, which we’ll call Nr. 1.

Nr. 1 may have been modded a bit before. It’s certainly been opened before; one of the three screws was stripped and useless, the other was jammed badly. I had to drill both out, so I’m hoping I can order replacements. The third was fine.

As soon as I had the microphone open, I saw what Recording meant by random components. The key transistor was a make so old that it had a metal shield ground cap, and separate lead to that cap, something I haven’t seen in gear made after about, I don’t even know, 1978? I also saw what they meant by “fragile circuit board,” because wow, you could lift these circuit board traces with an overly-aggressive hair dryer. Take care!


Comrade!

Still, it was mostly a matter of being methodical and not rushing things, and in good time, I had the key components upgraded, with no surprises other than the transistor’s extra lead.

These are three unmodified before/after snippets in one recording – recorded under identical conditions other than the internal microphone electronics – of Oktava mk-012/mc-012 nr. 1 in my studio. Even on laptop speakers, I can hear the harshness, particularly in the first sample. In all cases, it’s pre-modification first, then post-modification after:

Oktava MC-012 nr. 2 sounded very different to nr. 1, before; opening it, I could see that the components used were of a significantly more modern variety. It may well have been made later, which would be part of that. Now, the two microphones sound much more like each other, indicating that nr. 1 really was meaningfully different in component quality.

Here is a recorded comparison of nr. 2 (still factory) and nr. 1 (upgraded). These recordings were made simultaneously, with the two microphones right next to each other. The differences are much subtler, but I think the upgraded nr. 1 has a bit more presence – or maybe sense of stage – than the factory nr. 2. Despite being mono recordings, it’s almost like there’s a slightly better stereo image in the modified nr. 1… but give a listen and hear for yourself, see what you think.

You’ll definitely need headphones to have any chance of hearing anything interesting here. Factory nr. 2 comes first in all cases:

So, all in all, very glad I did this to nr. 1; pretty sure I’m going to go ahead and do it with nr. 2 as well, though I expect a much less dramatic change.

The only thing I’m thinking about now is – there’s a bank of capacitors in back. They’re good ones – Philips, not generic, which have a good durability and spec-compliance record. (I don’t know whether they’re original; some Oktava 012s shipped with quality caps already in place, and their track record has improved with time.) So I shouldn’t need to upgrade them – and the article at Recording Magazine says not to bother if you already have “improved” capacitors.

But I don’t know how old these are, and electrolytics have a lifespan. That’s measured both in calendar time (years), and in use – tho’ the latter is in tens of thousands of hours, and these mics are certainly nowhere near that.

The small downside is time spent, the large downside is the possibility of circuit board damage, which wow I don’t want. The upsides would be 1. possible sound improvement if they are aging already, and 2. Never having to think about it again, in practical terms.

So I dunno. Get it out of the way, or leave sleeping caps lie? Hm.
 
 


*: errata for the linked article: Capacitor “C6” in the parts list is actually capacitor C1; there is no “C6” in the build description or circuit diagramme; I assume this is a typo.

Also, some of the items in their parts list are no longer made, but they have exact replacements. R1/R2 exact replacement part number as per my October 2016 Digi-Key invoice: MOX200J-1000ME-ND. Capacitor C1 (listed as “C6” in parts list, see previous paragraph) current part number: 445-4737-ND. Capacitor C2 current part number: 399-1418-ND. Capacitors C3 and C4: 4073PHCT-ND. Capacitor C5: 4047PHCT-ND. Mostly, the substitutions are lead-free versions replacing earlier versions with lead.

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solarbird: (made her from parts)

I made a thing! It’s called a focus knob. It’s quite simple and normally you’d built it into an electric guitar as a guitar mod, but since I don’t have an electric guitar, I built it as a pluggable external box.

Basically a mild high-pass filter that serves to pull out ‘boominess’ from instruments, it puts a bit more of an edge on an instrument’s sound – the more you turn it up, the greater the change. As effects go – on my zouk, anyway – it’s pretty subtle. But it’s also the kind of shift that is multiplied by later effects added in, and changes how later-in-chain boxes like distortion pedals work.

As you can see from the instructions here, the wiring takes all of about 20 minutes’ time. But it’s good warmup for making a bunch of component upgrades to my Oktava 012s, and I already had all the parts from the Great Radio Shack Lootfest of 2014. Plus, hey, cool hard candy tin!

So I have a a HARD CANDY knob now. It goes from Hard to Candy. Yum. 😀

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solarbird: (pingsearch)

So the new iPhone is out, and as predicted, it does away with the standard, unencumbered, unrestricted-by-patent 3.5mm audio connector. You can read about the release on BuzzFeed’s pretty decent writeup if you like. And this matters, even if you have an older phone, or an Android phone, because Apple is the kind of 10,000-pound-gorilla that can shape markets in this area. Even if you’re not an Apple user, this throws expectations around for the future.

There is an adaptor – really, a mini-interface-card disguised as a cable adaptor – to let you use 3.5mm devices with the lightning port. It has to contain a D/A converter and a small amplifier. One will be included with the new phones, and it costs $9 and doesn’t make your cable weird – it’s not some big block like the previous 30-pin to Lighting interface, and it’s not $30.

I have concerns about how good a job a $9-retail D/A converter and amp unit is doing to do at rendering quality audio. It will be very tempting to make it deliver “meh” quality output, and push people to new gear. That’s short-sighted, but let’s not pretend that stops anyone.

Countering that concern is the fact that at least at one point, Apple required a specific D/A converter for the Lightning audio standard: this one. I have no idea whether that’s still a requirement. But if it is, I’m willing to assume a baseline of competence for it – anything else would’ve been suicidal for the spec right out the gate.

I’ve heard a lot of people talking about whether the new interface is built for digital rights management (DRM) as the long goal. I genuinely don’t think so, because it doesn’t really add much capability they don’t already have. Sooner or later, you have to go to analogue, and unless they want to remove the capability to connect to high-end audio equipment – and Bluetooth does not cut it for audiophiles, or necessarily even mid-philes – there has to be a way to hook up to standard, not-Apple gear.

You can’t get around that. Lest people forget, an Apple-provided solution for this already exists in the form of the dock – shown on the iPhone 7 front page, too. It’s not going away. And the reason it won’t go away is that while audiophiles are not a big market, they are exactly the kind of lifestyle market Apple wants and needs in order to support its brand, and more importantly, its markup. That’s not tech; that’s image management. Even without Steve, Apple knows its image.

Similarly, they can’t cut off concert musicians and DJs from plain old analogue output. There are too many audio pros out there using phones now, and while that market isn’t actually large, it’s a market Apple still invokes in image, and it’s too perceived as cool for Apple to throw overboard without throwing another serious wrench into its branding.

And frankly, with the recording industry betting what’s left of the farm on streaming, they don’t really don’t seem to care much about DRM on plain audio anymore. The RIAA destroyed the value of owning music, so from their point of view, who cares? Music is the billboard, not the product. I just really can’t see this as “HDMI for audio.”

So from a consumer standpoint, mostly I see “Apple has made your headphone cable annoying.” Even that’s assuming you’ve got your own headset and aren’t using the one Apple included, which most people do and will continue to do.

Now, this does get more complicated for musicians and DJs. Even if the included little cable adaptor is good – and let’s say it is straight up great – then you can’t trivially run the new devices on power and interface directly to performance gear anymore. That’s a headache. “Oh shit, I forgot to charge my phone” becomes a critical failure. Best case is you get a new device for that – and the dock is not suitable, you need something you can’t knock over or drop – which means one more damn thing to buy and carry around and/or lose.

Let’s also say you’re using some sort of audio software on the phone, and it doesn’t have a way to save files that you can transfer to other devices. (Even the software I have which does this doesn’t do it easily or well, it’s kind of a pain in the ass and I don’t do it. I use the headphone jack.) And a lot of software – like 8-bit emulator sequencers, and like Animoog, which I have actually used on multiple released tracks – just doesn’t do it. So that just got more annoying on newer hardware too. Another dock or another cable or another whatever. It’s one more step.

But, interestingly, not on the iPad. So far, I’ve heard no rumours that the iPad will drop the 3.5mm connector. And the iPad – particularly the iPad Pro – has very un-phonelike things like a keyboard case and special connector, and art stylus/pencil, and so on.

So what I’m thinking – particularly with the Pro – is that Apple is seeing a differentiation opportunity between “phone” and “pad,” and that they’re pushing “iPhone” to “purely consumption device,” paralleling their attempt to push “iPad” towards “creation device.” That’s not the actual usage out there – lots of people use the iPhone to make things – but it’s coherent market segmentation, and marketroids love their market segmentation.

Also, the iPad isn’t nearly as space-constrained as the iPhone. It’s just not comparable. On the iPhone, replacing that jack space with bigger battery and camera means vastly improved camera and about an hour extra battery life. On the iPad, it’s not a big enough percentage of space to care.

If the next generation of iPad keeps the 3.5mm analogue headphone jack – while adding support for the new Apple wireless headphone specs, of course – I’ll take that to be pretty solid supporting evidence. We’ll see.

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solarbird: (molly-oooooh)

In yesterday’s post, I posed a question: do USB chipsets matter in the 2.0 environment? I had reason to suspect they might.

The answer is holy crap yes they matter they matter so much it is unbelievable.

First, let me talk about what prompted this research, so you’ll know why this matters.

On my old sound interfaces I had live monitoring in hardware, so I didn’t have a lot of need to care about latency. Since that won’t mean much to most people, I’ll explain; when recording, it’s good if you can hear yourself, in headphones. If you’re multitracking, it’s critical.

My old audio interfaces did this with direct connections in the hardware. Whatever came in the microphones also went out the headset. There are advantages to this method, but also disadvantages, in that you aren’t actually hearing what’s being recorded, just what’s being sent in the microphone jack.

But now, I have this shiny new 1818vsl, which doesn’t do hardware monitoring under Linux. Higher-level kit generally doesn’t provide that; they’re assuming you have enough computer that your computer can send back what is actually being recorded, effects and all, and that you’ll do that instead.

This means I now have to care about latency in my system. Latency is basically delay, between mic and computer, and computer and headset. And if the computer is feeding my monitor headphones, that delay matters. You want to hear yourself live, or close to it, not with, oh, a quarter second of delay or something horrible like that.

Now, the good news was that straight out of the box on Ubuntu 16.04 (the latest long-term support version), I had better, lower latency numbers on my new 1818vsl than on my old hardware, when I was using that on 12.04. I could get down to a buffer size of 256 samples, and three frames, which gave me about 30ms basic latency – roughly half what I had with my old hardware and old install. I could use it as-was.

But I couldn’t go any lower on those buffers. One more setting down, and even playback would lag. It’d be okay until the system had to do anything else, then you’d get a playback pause, or a skip, or if recording – presumably, I didn’t bother trying – lost sound. That’s unacceptable, so 30ms was the lower limit, and I wasn’t sure it was a safe lower limit.

And that’s what got me doing all that chipset research I talked about yesterday, and I ordered a new USB card (plugs into PCI sockets) based on that research. I was hoping for a couple fewer milliseconds of latency, that I wouldn’t actually even use; I just wanted a safety margin.

So that new card arrived on Sunday, with its OHCI-compliant chipset made by NEC, and I popped it into the machine and started things up with normal settings.

At first, I was disappointed, because I only saw about half a millisecond less lag, instead of the 1-2ms drop I’d hoped to see. But across tests, it was more consistent – it was always at that same number, which meant I could rely on that 30ms latency in ways I wasn’t sure I could before.

They I decided to see what would happen moving the sample buffer setting one level lower, into what had been failure mode. And the result was 1) it actually worked just fine, where it hadn’t before, and 2) when running analysis, tests showed much lower latency at that setting than with the previous USB ports.

That was an ‘oh ho‘ moment, because it implied that the 256-sample run rate was basically the spot at which the on-motherboard USB could just keep up, and trying to run faster wouldn’t actually produce any actual processing improvement. It’d try, but fail, and time out.

So I did a couple of recordings on that, and they all worked. Then I dropped it another level, until finally, I just said hell with it, let’s just set it as far down as the software will allow and see how hilariously we explode.

I just successfully recorded test tracks four times with these settings, on the new card:

0.7 milliseconds isn’t even something you think about on USB 2.0. 2.8ms, maybe, okay. I’ve seen that managed a few times before, and that’s genuinely indistinguishable from realtime/hardware monitoring. But 0.7ms?

Seriously, this is well into “…is that actually possible?” territory. I’ve never even heard of someone running over USB 2.0 at latencies this low.

So, I guess it looks like the chipset matters a whole lot. Maybe not for most applications, and maybe not in the same way as in USB 3.0 or in FireWire, were there are serious compatibility issues. But in the 2.0 world, in realtime audio, it appears that the chipset makes all the difference in the world.

And yet, I can find this nowhere online. I’m beginning to think nobody bothered until now. Certainly when I’ve asked about it, the response has “why are you on USB get firewire” or “why are you on USB get PCI” because sure I want to throw out all this hardware and start over THANKS NO.

I think USB users have been trained just to accept it and deal. But surprise! You don’t have to! You can actually get a better USB card, if your system allows it, and it’s $30 instead of $1300!

So, HELLO, OTHER SMALL-STUDIO MUSICIANS! You want a chipset that uses OHCI on the USB 1.1 level even if it’s a USB 2.0 card or later because the 1.1 layer still matters, and still gets invoked by the higher-order drivers for card management. See previous post for why that’s important.

This means avoid Intel and VIA chipsets, and look for NEC or SiS – or anything else that loads OHCI drivers and not UHCI. If you’re on Linux, you want to:

cat /proc/interrupts | grep usb

If you see “uhci_hcd” in there, you have a UHCI chipset running your USB port and getting a new USB card with an OHCI-compatible chipset (and disabling whatever’s already installed) might help you with your latency issues.

Good luck!

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solarbird: (pingsearch)

I’ve been trying to find out whether there’s any sort of difference between USB 2.0 cards, specifically as addresses the needs of digital audio workstations on Linux.

Very few people in linux communities seem to have addressed this question at all, and none I can find on the audio side. (Firewire, oh my gods yes – huge lists. Just not USB.)

But I did a lot (a lot) of digging, and discovered via the Linux USB kernel driver dev mailing list(!) that while there’s not much difference on the USB 2.0 side, there are important differences on the 1.1 side. These difference manifest in two different driver models. That still matters at least a little bit in 2.0, because those 1.1 drivers still get loaded.

Anyway, that difference is that there are two very different driver interface models. One is UHCI, created by Intel and used by Intel and Via, mostly. The other is OHCI, which Compaq pushed when it was still around, and Microsoft preferred; it has less intellectual-property load, and NEC, SiS, and some other makers use it. If you see a “Mac compatible” card? It’s going to be OHCI.

The OHCI model puts a lot more of the business of doing USB into hardware on the card. UHCI has the processor do that work. And while that isn’t a heavy load, it is a nonzero load, and more importantly means that UHCI chipsets require more CPU attention than OHCI chipsets, on a recurring basis. And that is something we don’t need in a digital audio workstation; there are only so many board interrupt opportunities; I want them for moving data, not servicing USB mechanics.

Once I knew that, I did more searching and found people saying how switching to a NEC chipset card had (in one case in particular) ‘saved their bacon’ specifically on their digital audio workstation. They were using ProTools on Windows, not Linux, but it was still with a USB audio interface.

The chipset used by my on-motherboard USB ports is, of course, Intel, and therefore UHCI. (And UHCI drivers are actually loaded, I checked.) There’s also an on-motherboard hub between the outside world and the one true root device; that doesn’t help anything either. So there’s a nonzero chance I’ll see improvement both from changing from UHCI to OHCI, and from moving to a true root USB device instead of a hub device. It won’t be much, but I’m only looking for a few milliseconds of latency here. And even that’s more for… reliability buffer, I suppose? Yeah. Reliability buffer, rather than pure necessity.

I’m mostly posting this 1) so I remember it and 2) so other people looking for this data can find it. HI! I can’t be the only one!

I’ll update this post if I get interesting results.

eta: INTERESTING RESULTS AHOY: CHIPSETS MATTER SO MUCH OMG. I’ll write up a post with details, post it tomorrow.

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solarbird: (banzai institute)

I’ve been playing with that ‘added pressure adds bass response’ idea, for use with these piezo pickups. I made a little wooden chamber that would let me add light pressure, as with the bridge pickup design. It would be held down with a clamp for testing, but would isolate that pressure from the piezo itself.

Anyway, I made a bunch of recordings, two for control, and eight with a range of pressure in the chamber. The controls were made with the pickup taped to the front of my zouk with double-sided tape (standard attachment), and with the pickup directly clamped to the front (also a standard attachment) and come first and second in the recording. The other eight were with the pickup in the test chamber, with increasing amounts of pressure on the crystal, applied by inserting paper as seen here:


With thin cardboard and two sheets of paper

Note again that the clamp is not adding pressure to the disc in any way.

Audio samples in a single mp3, here. There is some extra noise in these recordings; I was trying the modular approach again and that’s the result. I think the TRS connectors are inherently noisy. But that’s a separate matter.

I also ran spectrographic analysis on each recording, and combined those into a single animated gif that cycles through the recordings in order. Here’s the key for both. The gif is repeating, so each frame is labelled in the upper left.

 1: taped to top
 2: clamped directly to top
 3: in chamber, no paper
 4: in chamber, thin cardboard (0.46mm)
 5: in chamber, cb+1 sheet  (+0.11mm)
 6: in chamber, cb+2 sheets (+0.21mm)
 7: in chamber, cb+3 sheets (+0.31mm)
 8: in chamber, cb+4 sheets (+0.42mm)
 9: in chamber, cb+5 sheets (+0.52mm)
10: in chamber, cb+6 sheets (+0.63mm)

You’ll note in both the graphs and the audio that bringing in the chamber at all, even with no additional crystal pressure, caused a big drop in high-end oversensitivity, and boosted the low-end. That was interesting; I have suspected for a while that the crystal side of the disc would actually be better as a source-facing element, but there are physical issues to doing that, since the wires have to attach on that side.

Adding pressure continued to boost low-end response through test 7, without inhibiting high-end response. After that, I think additional pressure began to overcome the benefits, and you see a return to a more midrange-heavy sound – though in all cases, I think it’s better than either traditional mount.

This is consistent with tests made in the bridge pickup from last week, and reminds me of a diagramme I saw of a period crystal microphone that implied the crystals themselves would be set up forward-facing.

Anyway, data! And lots of it, for lots of your crystal/piezo experimental needs.

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solarbird: (asumanga-yay)

A couple of months ago, I tried making a Cortado-based bridge pickup for the octave mandolin. It worked okay – better than the ad-hoc clamp arrangement I’d bodged together for last January’s Conflikt show, but not what I’d hoped. It was a lot more stable, but still needed lots of equalisation help.

I’ve built a new one, with a new design! It’s much better. Here’s an mp3 of the previous design alternated with the new design, on octave mandolin – no eq of any kind, no effects, just raw output from the old design vs. the new. (Old design is first.)

While working with the previous attempts, I’d figured out what really improved things was the right kind of pressure on the piezo disc itself. My thumb was pretty optimal, but you can’t exactly do that and play at the same time. The clamp wasn’t bad, but it was slippery and awkward and actually came off on me during rehearsal, so I didn’t trust it. Most piezo-style pickups live under the bridge of an instrument, but you can’t do that the usual way with this one, it’d be destroyed by the pressure.

So I went about trying to fix that.

First was to take the bridge plate and add a wide, flat channel – one wide enough specifically to contain the entirety of the Cortado piezo element. I made it by wrapping sandpaper around a flat piece of metal, and scrubbing back and forth to excavate out the wood I needed removed.


This is actually a new bridge plate.
But it’s made of the same material, so no real diff.

You need to sand away enough wood to make room for the piezo and all the tape wrappings, and some extra. But you do not need to sand away enough for the wires soldered to the disc – you want to avoid those entirely.

Keep sanding away wood until the bridge slides freely over both the new channel and the piezo, like so:

What this makes is basically a wide clamping chamber around the pickup element itself. It doesn’t do any clamping yet; it just creates a space for it. At this stage, in fact, if you hook it up and try it, there’ll no change in sound from the previous version.

(In fact, the “old design” recording I used in the sample is actually this version at this point in the process. I verified that it sounded exactly the same as the previous version, as predicted, which means I’d re-established the old baseline. Important for science!)

But now, of course, I have a clamping chamber! We just need something to apply pressure.

So what’s our clamp? Pieces of paper. Post-it notes, to be specific, just because they were handy. The right number of sheets in this exact case turned out to be four.

Five also worked, and did not feel like too much pressure inserting the papers under the bridge. But it did sound like a bit much compression, tonally, so I went back to four.

(Here’s that sample track again, alternating old design and new, old first.)

The beauty of this is that since it takes several thicknesses of paper, and since that paper be changed out without taking apart the pickup, you can use any number you like. You could even adjust the tone on the fly.

Interestingly, the pickup didn’t even get quieter with more paper. I’d worried about that, but didn’t need to. In fact, adding more sheets made it louder, meaning that the pressure is not so much “damping down treble” as it is pulling up bass. Which, in turn, makes me wonder if it’s not so much “resonating better” as moving the zero/no-vibration point of the crystals’ charge state from all-electrons-in or all-electrons out (doesn’t matter which) to a more middle-range position, which…

…hm. Actually, that’s interesting. No, that’s really interesting. That would explain why the pickup got louder with more clamping, rather than muffled or…

…huh. This is an hypothesis. If I’m right, I can make my next crystal mic substantially more modern sounding, by enclosing the piezo in a small clamping chamber, which is, like this, attached to the resonating disc of the microphone, and possibly…

…possibly I should take my SRMD meds now or I’m going to be up until 5am next Thursday playing with crystals and possibly taking over the moon again, aren’t I? Yeah. I am. Okay. BRB.

So. Yeah! I’m super-pleased with this result. I’m also thinking that maybe this could be used on other items that have flat surfaces which need pickups – like, a piano, maybe – and instead of the bridge, as here, you use a weighted flat bar of some sort across the pickup plate to create the clamping chamber. Then you’re off with tonal control via paper again. I have no need for this functionality at the moment, but it strikes me as legitimate nonetheless.

And most importantly for me, I now have a much more conventional DIY pickup for the octave mandolin. Here y’go, doc – just plug ‘er in, and we’re off.

Much better.

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solarbird: (banzai institute)

I have some mics that react badly to phantom power, so I made a phantom power blocking took. I had an old Smarties block handy, so I made it out of that.


It seemed appropriate. ^_^

I also went at improving an effects box I built a while ago called the Trash-o-Matic 68000. You would have heard it on Daleks Behaving Badly (Dalek Boy), a joke track that really needs a shorter edit, because it takes way too long to build up.

The best part of this box is the Berthold Ray effect circuit that I legit invented. That must have happened in a pretty heavy Science-Related Memetic Disorder attack (or spark hyperfocus, if you’re a Girl Genius reader), because I was trying to figure out how the hell it worked and I am here to tell you that this is some serious-business Oscillation Overthruster bullshit right here.

It’s basically a multi-store self-reducing sampler feedback effect with frequency shift that’s using the device’s amplifier as a delay loop and sending the amplified samples back to the input via a combination of the internal system ground and negative phase of a balanced signal lead. Both matter. I… don’t entirely know why.

Anyway, the whole thing is noisy as hell, and much of that is the platform I was building onto, and I was hoping to fix that. I was able to reduce noise levels somewhat – no, that’s unfair, meaningfully, it’ll be easier to gate out noise now – but it’s still buzzy as hell.

I’m kind of interested in seeing if I can re-implement the Berthold Rays in a less trashy environment. Sure, it’s fun in this mess of noise and grind and crunch, but it’d be nice to have in a cleaner box as well. Maybe I’d use it more then.

I’ve never used it for music before, but here’s a little bit of noodling I call “Broken Music Box, Found After a Fire,” played on Irish Bouzouki and run through Trash-O-Matic voice 6, with full Berthold Ray attack:

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solarbird: (Lecturing)

Building microphones is fun and seems to be of interest to readers, so here’s a collection post for posts about that! These posts discuss building both microphones, and, when applicable, their matching microphone driver circuits and/or pre-amplifiers.

Building a Carbon Microphone:

Related posts:

Building a Crystal Microphone:

Building a Ribbon Microphone:

Other microphone and preamp customisation/modification posts:

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solarbird: (korra-excited)

The last parts for the carbon microphone arrived yesterday! And once I got back to the Lair, I set about adding them to the circuit and building out the final version. Here’s a quick sample recording I’ll talk about more later.

I’m starting with the previously-discussed circuit, now taken out of the test harness and reassembled on one of the small breadboards. The new isolation transformer on the left – basically the outputs of the original circuit are attached to one side, and new outputs picked up off the other. This serves two purposes: first, it eliminates some kinds of hum noise if they start to crop up, and second, combined with 2K of resistance on the other side, brings the line-level(ish) output down to microphone level.

(These are standards which matter in a studio and … not many other places. XD )

The zig-zag in the resistors doesn’t serve any function purpose other than fitting into a smaller space – ideally, I guess I wouldn’t’ve been making needed values out of collections of other values? But I had what I had.

Signal flow is right to left in this photo. Underneath, at the top and bottom of the board, I’ve built wire rails to connect the components. That probably means I’m not really using the breadboard entirely as intended? I don’t even know. It holds everything in place and that’s definitely what I intended. XD

To house all this, I’m using an “experimenter’s case,” which is basically old-radio-speak for a metal box. XD It has soft metal on two sides, easily drillable and workable, and a hard case. I’m using one side for input, the power lamp, and the on/off switch; the other side is for output, or, as it turned out, outputs.

The carbon element (in the can) is connected to the driver/amplifier circuit via a 1/4″ TRS phone jack – like an old large headphone jack – with the two leads to the carbon element being on tip and ring, and the shielding ground being on the sleeve. (Tip, Ring, Sleeve: T R S.) That socket is on the left in the above photo; the middle component is a small LED, to indicate power on/off, and the right is a BIG CHUNKY POWER SWITCH. I love big chunky power switches. CHONK

For output, I quickly realised that I could have both balanced XLR output at microphone level, and line-level output on a phone plug, if I could find a way to isolate the chassis ground from the phone socket’s sleeve connector.

Normally, both being grounds on the same circuit, they’re connected automatically. Finding one that isn’t already connected is actively difficult! But careful use of electric tape did the job; I drilled the mounting hole larger than it needed to be, and basically lined anyplace the case and the socket would touch. Isolation achieved!

If you look for the blue and white wires, you can see where the TS (mono) phone plug is tapping the raw (line-level) amplified mic signal, just before it’s fed into the isolation transformer.

The transformer is really pretty optional – powered carbon circuit signals are pretty high as microphone signals go, and as I mentioned above, we’re actually reducing that signal to create the balanced XLR output on the other side of the transformer. But it’s nice to have the option of using line level, since it already exists. That’s what built-in sound inputs like on your laptop want, too, so there’s a point to it.

And here’s the whole driver/amplifier circuit, with a battery holder made of velcro.

Is that cheating? Holding the battery down with velcro, I mean. totally cheating I’m hoping it works out – I didn’t have a 9v battery case and it seemed excessive to try ordering one.

That LED power indicator? It’s warm white, left over from another project. I was planning on putting in your typical red LED, but realised that if they’d had a power indicator on one of these in 1932 or whenever, it most certainly would’ve been a little incandescent bulb, and it may and may not have had a colour lens. So I went with warm white, because period accuracy! Sort of.

The neat thing about the way this circuit works – and all carbon microphone driver circuits work – is how it points you right at vacuum tubes, and from there transistors, conceptually. It really, really does.

See, in tubes and transistors – which are both signal amplifiers – the input signal is used to create an amplified copy by controlling how much raw input power is let through, from another source. That’s why tubes were called “valves” originally; it’s because they are valves, electrically controlled, and regulating the flow of electricity from an input, just like the valve on your faucet controls the flow of water from the plumbing.

In this case, exactly the same thing happens yet again. But the input signal is sound pressure (how loud the sound is), which is controlling how much electricity is let through from the battery. And those changes in sound pressure – and therefore electrical flow – make the electrical copy of the sound waves.

Neat, huh?

Anyway, that’s the inside. Let’s look at the case!

I really like how chunky and primitive it looks. This is an old experimenter’s case; I’ve had a box of random cases in which I can build things for a while, and I don’t even know where I got this one, or when. If you saw it on the set of a 1950s television SF show, nobody would give it a second glance.

Always document your builds! You never know what might confuse people later. And by people, I mean yourself, after you’ve come down from the science-related memetic disorder high. I want at least the theoretical possibility of using this amp with other carbon elements, so writing down how the interface works is pretty important!

Except for the glare from the power light, I think this would be the Radio Shack Catalogue photo from, say, 1975:


Good, Better, or Best? Probably “Good.” It is just carbon, after all!


Or maybe this is the catalogue shot? Not sure.

Finally, here’s a test recording I made, using both outputs (phone/line level and XLR/balanced mic level) at once, hooked up to two different inputs on my board. I put both recordings in the same mp3; one’s on the left channel, the other’s on the right. The two tracks should be pretty much identical – being the same signal picked up at two different places on the board – and I wanted to see if that actually happened. Fortunately, it did!

Well, eventually it did. This is actually the second time I tried this, because the first time, I discovered that I’d managed to wire the two outputs up as electrical inverses of each other. Playing the two tracks back at the same time resulted in massive waveform cancellation. Which was hilarious, but also a good indicator; they wouldn’t’ve cancelled so well if they weren’t really similar. 😀


EXTREME WAVEFORM CLOSEUP

So that’s about it for this project! I’ll most likely do something to the ring to control the elastic better. And I’ll probably build a case for the whole kit, like I’ve done before – mics should have cases for protection! – but that’s a separate project.

This has been such a fun build, you have no idea. If you have any interest in this kind of DIY audio, I totally recommend this as a fun, easy project. Particularly if you don’t have studio gear, because you can look up the line level part of the output to damn near anything (including a PA system, I might mention) and it’ll work.

As always, more and bigger pictures on my Flickr account. And if you’re out of work, that’s a great time to listen to the new (NSFW lyrics) single! It’s awesome.

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solarbird: (Lecturing)

My carbon microphone is actually in a can now! I’m super happy with it, and yes, it works, even if most of the circuit is still in a test harness. The can itself does not contain anything important in the way of circuitry, though it does contain quite a bit of grounded shielding which is very important, and that shielding is carried forward out the cable, which is a TRS phone plug.

I made the decision to put the support circuitry in a separate box for a couple of reasons. First, the signal level is pretty high, so a decently-shielded cable should prevent most issues from being actual issues. Second, the support circuitry for these things tended to be external in the originals, so it’s period-accurate. Third, opening the can to replace a battery every 15 hours of use or so was going to be annoying; much easier to do it an an external experimenter’s cabinet.

And yeah, one of the downsides of this mic is the high power consumption. I thought about trying to tap phantom power, but from what I can tell reading up on it, the amount of power actually supplied – voltage aside – is dramatically lower than a battery’s supply and is not well defined, by which I mean is not defined, so… yeah. Batteries it is.

Anyway! I decided to make it look like a vintage carbon-element microphone of its era, only using elastic instead of springs. Springs look cool, but they’re also noisy as hell and create other issues, and it’s easy to tell that they would’ve used elastic if they’d had it just by how quickly they changed to using elastics as soon as they had elastics to use.

BUT SOLARBIRD, HOW DID YOU DO IT?

Well, first, let’s ask the eternal question: ah, junk shop, is there any build problem you can’t solve?


No. No, there is not.

If you’re wondering, that’s a vaguely-20s-looking kind of drink can holder in the middle, and I believe a hand towel holder designed to hang from the top of a door.

My only worry was whether the towel ring assembly was just screwed into the plate, or bolted to it; most excellently, it turned out to have been attached with a screw. So I just separated the ring from the existing base, sanded the attachment point down a bit to make it nice and flat, and attached it to a small metal L, using a rubber plumbing washer to make up the extra space.

Attaching to a standard microphone stand attachment is just as easy; unscrew one of those infinite number of cheap mic clips that are floating around every studio ever, toss the bit that holds the mic, drill out the hole in the L metal to be large enough, and pad with more plumbing washers.


Result!

Now the ring is ready to attach to any standard microphone stand.

Now, the can itself was a little more of a trick. This is all mechanical construction, not circuitry, since all that will be in an external box. But! There is some electrics, because given my particular… affinity… with RF (and radiating it, hi, it’s solarbird for a reason), I wanted a good heavily-shielded pickup enclosure.

Did you know you can buy adhesive tape made of copper? With conductive adhesive? It’s made exactly for this purpose. I love it. I started by lining the back, and peeling excess up the sides a bit intentionally. You want a good amount of overlap with this tape.

Lining the sides involves another ring of copper metal tape, with – again – overlapping tabs made of the excess height. Getting the height right is really simple – just put the tape in and cut inwards, using any common scissors – this may be metal tape, but it’s pretty thin, and no special tools are required.

After I took that photo, I realised I needed smaller tabs, so I went through and made another set of cuts, halfway between each existing cut.

Once you’ve fiddled around with it a bit, you’ll end up with something that looks like this:

Make sure the copper is well rubbed down against each other, so the conductive adhesive can really carry current without adding any resistance. Again, no special tools, a fingernail is fine – but make sure it’s well stuck down.

You might also notice in that photo, a small black line – I broke the tape, and fixed it with just a small piece to cover the gap. As long as you have well-connected metal throughout, you’ll be fine. We’re talking very low power with RF noise, in most circumstances, so you don’t have to worry about carrying power or anything like that.

Unless you have a tesla coil, maybe. That’s different.

Now, I also needed a grill for the microphone, and – importantly – it had to be a conductive grill, because I need that RF blocking all around the carbon element. I know, I know, some of you are going, “it’s a carbon element how are you doing anything to it?!” and all I can say is I have recordings and I have to ground myself with a wrist strap if I’m using AKG microphones, and again, supervillain.

Fortunately, a material that serves this purpose quite well is common and cheap: aluminium window screening! I’m kind of annoyed with myself, because I threw away a bunch two weeks ago – used but still clean and good – because I had no thoughts I’d need it. What was wrong with me? I can’t even tell you. Moods. So I had to stealbuy some. Fortunately, it’s pretty much dirt cheap.

I got a ring of heavy flexible rubberised foam to make a structural ring, and measured the right size just by pushing the screening into the bottom of the can until I had good edges. Since it is window screening, it has a lot of room to compress, and that helped. You want that excess screening material, for reasons which should be obvious momentarily.

In the above photo, I’ve sized the screening material, and am getting ready to make a ring of copper tape to surround both the inner and outer layers of the support ring. This is partly structural – you can see that my support ring is not a single loop, but a bent straight piece – and partly to help make sure of good, solid contact between the metal screening, through copper tape, to the interior copper shielding of the can. Make this part a little too large, if anything, and you end up with a solid pressure-contact connection.


Swaaaaank

Holding the carbon element in place is also a job for foam. In this case, I have some high-density impact-absorbing foam left over from my case making projects earlier, so I just used that. It can be a very rough cut, as long as it’s just a tad bigger than the ring it’s going into. It’ll compress, and that provides a little more outward pressure to make the grounding contact between the grille and the interior shielding better.

You’ll also note inside the can, against the back, I’ve placed a spacer ring. This keeps the grille’s support ring from going too far into the can; it’s just a physical element, since you want the carbon element nice and forward, and not sinking into the can where sound would get echoey.

What you can’t see is an important step I … didn’t remember to photograph. Sorry! And that’s drilling a hole for the cable. The cable is three-conductor; two signal leads (which connect to the two contact points of the carbon element) and one shield ground. The shield ground gets soldered directly to the copper tape, which is why you use copper metal tape instead of some other metal. This is a little tricker than you might expect, mechanically; I had to use higher temperature on the soldering iron. I think it mostly has to do with soldering wire (physically complex, wicks well) to a flat surface (physically simple, does not wick well). Just take that part as read.


[Win95TADA.wav]

Some of you might be looking at this picture and going, “…wait. The carbon element is right up against the grill. It’s touching, and metal, and isn’t that a problem?”

Well spotted, you! It would indeed be a problem! I solved it by cutting out a couple more layers of plastic window screening – also left from another project – to provide an insulation barrier between the metal screening and the actual pickup element. If you don’t have any of that, hosary would do fine – the same material used in pop filters. Acoustic transparency is important here, of course, but to be frank – less so than usual.

“That’s not like you.” Yes. But think about it, I mean, what’s a telephone use? Oh yeah, THICK PLASTIC WITH SOME HOLES IN. Does it hurt the sound? NOT VERY MUCH, because it’s a carbon element with a range of around 300hz to 3500hz, maybe. So you can afford some loss in the high end, because what you don’t lose, the mic will lose for you. And this screen material is plenty acoustically transparent for these circumstances. So would be hosary, or the right foam, or all sorts of other things. Just, you know, use good judgement.

Anyway, the build stuck on hold – ON HOLD! HA! – for a couple of days as I wait for the 600ohm isolation transformers to arrive. They aren’t essential, necessarily, but do reduce RF noise and hum in some circumstances, so I’m going to use one. I’ll post more when I’ve got more done. I’m really pleased with how it’s coming along; this is fun.

Bigger pictures on Flickr, as usual.

ps: People were asking for a sound sample. This is from the test harness, before I built the ‘can.’ New recording sample next time.

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solarbird: (made her from parts)

I have a waveform! I did a bunch of testing of various parts and such, discovering that the carbon transmitter (microphone) I had was indeed fully functional, and really, it came down to “not enough voltage.” 1.5v is pretty borderline for carbon microphone power, so that’s fine.

The waveform is kind of lopsided, but that’s because this test harness doesn’t have the balanced output yet; I’ve ordered the balanced transformer, and that should help. It’s also a bit noisy – here’s a sample – which is partly related to SHIELDING WHAT IS SHIELDING because it’s a test harness.

Talking of, here’s what version one looked like:

This is a direct implementation of the first half of the circuit described in this instructable, which runs off a 9V battery. Once the rest of the circuit is added, it’ll have balanced output, which is pretty snazzy.

This is a closer view, and also after I added an LED, because hey, LEDs!

After that, I tried a smaller capacitor, and that’s working fine – mostly, I’m just picking from what I’ve got, which goes along with what the circuit designer described.

I’m going to play more with the circuit a bit, pending getting the matching transformer. And some shielding.

And, of course, the can. 😀

ps: this harmonica “bullet mic” looks pretty cool too.

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solarbird: From moongazeponies on deviantart (pony-pinkie-hax)

Or more correctly, a few other projects. Or hilarious things I found out.

ONE! I need a bullhorn for the chorus of this song. YES A BULLHORN. Yes, I can fake it with plugins but searching around for advice online, the most common answer by far was Just Get A Bullhorn, They’re Cheap. And they are, you can get a decent one for $9.

TWO! I’mma gonna build a carbon button microphone – think antique telephone, like those things from the 1930s-1960s – and nobody can stop me. Turns out the parts are cheap and you don’t have to spend $250 for that one from Gold or whoever.

THREE! As I posted on Facebook over the weekend, turns out those old Square credit-card readers for your phone or tablet that let you take credit cards at shows? They’re purely analogue devices, which is why Square upgraded everyone for free last year. Meaning they output sound. Meaning I just recorded the audio of my Costco card and this is hilarious.

I need some quarter-inch magnetic tape, stat. XD

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solarbird: (molly-oooooh)

A Germany company has shipped a delay fx pedal that uses a floppy drive as magnetic media to run its delay. That’s… interesting… and strikes me as likely to be really noisy, but on stage, probably not enough to care.

What makes me think about it more is data rates. Are they floppy-native digital? Are they formatting mp3? If so, 320kbps is very high quality, and the faster floppies managed 500kbps, so we’re good there, and you could ignore FAT and just write a digital data stream at that speed, it’d work.

But what if you intentionally racked that down? I kind of like the idea of intentionally under-quantising your delay pedal. Crank it down to 48kbps or something, have your delay sound like a cranky land telephone line.

Or maybe they’re bypassing the digital part entirely – what does floppy drive sound like as an analogue magnetic media? What do dropouts sound like on a floppy disk?

That would also let you play with different rotation speeds, of course.

Oh wait, look, they have a video. (Scroll down at the link.) Apparently, it’s analogue. That’s fucked up! I kind of like it. But it does lose the possibility of digital data loss, which – depending upon what you’re going for – is kind of too bad. Low data rates combined with this environment could make some really awful/awesome noise.

eta: In comments, John posted a link to this awesomeness, go play that, you need to right now.

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solarbird: (banzai institute)

Leannan Sidhe and I were kicking around at an estate sale – she was looking at some PA kit that had been advertised, turned out really not interesting – and I noticed a little neglected microphone pre-amp sitting in the corner. I’ve never bothered with separate microphone pre-amps, much less tube-driven ones, but I was curious about it.

So I went to the manager running the show, and said, “I don’t need this, but it might be fun to play with, what’s your best price?” And so I strolled off with it, and today I set up a pair of side-by-side M-Audio NOVA large-cap condenser microphones to make some simultaneous recordings. Both mics ended up going through my TASCAM interface, with one going through the tube preamp first, then to the TASCAM with the TASCAM’s gain cranked down to zero. The control mic gain on the TASCAM was set to match final recorded levels. A few samples are linked below.

First thing I noticed: jfc this thing has gain. If I need something LOUDed at the pickup level, I now have that piece of kit. I kind of had that kit already, but that was the ribbon-mic preamp I built specifically for the ribbon mic I built, and that can’t provide phantom power like this does. (And it shouldn’t; phantom power destroys some ribbon mics, including mine.)

The second thing I noticed is that… the difference is pretty subtle. I mean, I expected that. And part of that might’ve been having both lines going through the TASCAM at the end – but it had to go through something for digital conversion, or I can’t record.

In studio, I can hear small but audible differences. The TASCAM’s preamp seems to like mid-bass more than the ART TUBE MP. I think there’s a little quicker response in low base in the ART, in a way that I recall from tube amplification equipment like EICO and Dynaco gear.

Outside the studio, though – on a good consumer headset on my laptop? I’m not hearing much of any difference in 320kbps mp3. I think I’m hearing a little in uncompressed WAV files, but not a lot. That may be the laptop’s D/A converter, I don’t know. On the laptop speakers, I don’t hear anything different – though really that has to be expected.

Worth it? For what I paid, sure! I have a serious business gain DI/pre-amp out of it. Sound-wise? I dunno. I really do think there is some subtle difference and if I’m in an environment where I’m having to rein in mid-bass and pop the low end a bit, maybe it’d be better to do it with this thing than in equalisation later. Probably would be, in fact. But it has a pleasant enough sound to it, regardless. I’ll probably play with it on bass guitar, later.

Anwyay, here are some recordings – they’re edited so that consecutive repeated musical phrases alternate between the ART tube amp and the TASCAM interface’s built-in mic preamp. What do you think – do you hear anything?

Irish bouzouki: WAV mp3
Octave mandolin, tuned to open E5: WAV mp3
Bodhran, two different strikers (traditional, bamboo): WAV mp3

eta: I make a point of not talking American politics here much, but I do elsewhere, and watching the GOP’s civil war start in earnest is kind of neat.

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solarbird: (pingsearch)

On my digital audio workstation, I run two monitors, and I do it in xinema mode. This is because Ardour – like the overwhelming majority of apps ever – wants all its windows to be on one desktop. xinema mode is how Xwindows does that.

It’s great for Ardour. I have mixer on one screen, editor on another, it’s lovely.

But a couple of system updates ago, a lot of apps started getting “better” ideas about where “centre of screen” is, by which I mean, instead of getting centre of monitor, they were getting centre of desktop.

Which means that many applications now start split in half across two monitors. Including, annoyingly, Ardour’s startup menu and open-file menu; it remembers opening positions for main windows but not startup dialogues or splash screens.

This is incredibly annoying.

So is there a way for it to get, idk, “monitor centre” when it asks for “screen centre,” or to set a default location for unspecified-position windows, or… stuff? Because yeah. Annoying. So annoying.


Looking for the Grammy Awards Long List nominee post? Thank you for listening, and for your consideration.

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solarbird: (made her from parts)

We’re heading up to Vancouver tomorrow for VCON! We’ll be there for the weekend, hitting Chapters and Siegel’s Bagels and picking up some desperately-overdue cider rations and kicking around town. Mmm, Growers, how I miss thee. If you’re around, yell!

Also, there’s an exciting special event coming up here next week; you’ll want to read about it. More on that below the fold.

Right now, let’s talk Digital Audio Workstations.

First, what are they? Simply put, Digital Audio Workstations are software implementations of the physical hardware you’d use in a large recording studio to record your music. They include virtual mixing board, virtual patchboard, virtual tape recorder, virtual cables, virtual effects plug-ins, virtual equalisation – and depending on the package, even more.

The goal is simple. If you can do it on one of these:


I’ll be in my bunk

…then you should be able to do it in your digital audio workstation (or DAW) software.

Of course, it’s not quite as simple as basic recording. Were it, you could get a little digital recorder and be done. What that giant hunk of hardware – or your software DAW – gives you is the ability to record several tracks of sound, separately or all at once.

A DAW lets you play those tracks mixed together in a synchronised fashion, move and edit your recorded sounds, adjust their levels (both relative to each other and in absolute terms), adjust equalisation, add effects such as reverb or distortion or overdrive or whatever you have plugins for, and so on.

Some DAWs include integrated MIDI support; some include sequencers as a core component. Some even support remote boards that give you all those sliders and knobs, so you don’t have to use the mouse or keyboard so much. Those are cool, and easier to use in some important ways, if less portable.

But at the most basic level, you have recording, editing, mixing, and playback. At the most basic level, you have GarageBand.


I will not be in my bunk.

Now, I’m not mocking GarageBand. GarageBand is a great introduction to concept, and surprisingly capable. It makes a whole bunch of tasks really easy, has integrated MIDI support, and includes a bunch of virtual MIDI instruments.

While from a features standpoint it’s pretty limited, and while it handles tracks in a way that implies they’re less generic than they are by naming them after instruments and making them sticky in weird ways which might confuse you later, it’s still a great first experience.

If you just want to get the idea with GarageBand before tackling something more complex? Go right ahead. Because I am not going to lie to you: the learning curve on the more advanced DAWs can be brutal. Particularly on the free/open source ones.

So, what’s out there? Well, if you have the money, and a Mac, I hear great things about Logic Pro. For both Mac and PC you have Pro Tools, which is called an industry standard because it is one. Pro Tools Express is free with some hardware purchases – but it’s also limited enough that I wouldn’t use it myself. Reaper, for Mac and Windows, has fans in the professional community. (And as Tom Smith noted last week, IK Software is having a big sale right now. This is relevant to your interests.)

But we’re about dirtball DIY. Let’s talk building your own kit, and doing it the cheapest way.

There are really two topics here: hardware and software. We’re already talking software, so let’s carry on.

The cheapest route, in dollar terms, is always open source. Linux is free software. You may have to be able to do a lot of internals work – no, that’s not fair; you’d better be ready to rip its guts out – but you can do it.


Afraid? You will be. You will. be.

Audacity is a relatively-simple open-source DAW. It runs on Windows, OS X, Linux, and some Unix OSes, not that you’re likely to run into those. It’s easier to set up and it works. I ran into its limitations in the first hour, but that’s because I already had aggressive goals; it’s the GarageBand of the open source world.

Ardour is my workhorse, and it is a monster. It runs atop specialised sound server software called JACK, and runs on OS X and Linux. If you run it on Linux, you’ll have to grab PulseAudio by the throat, slice off its head, and salt the ground on which it dies. This will not be easy in some Linux variants (Ubuntu, I’m glaring hatefully in your direction) but it must be done. Ardour is monstrously frustrating (at times), is possibly the most difficult to learn software I’ve ever used outside of 3D modelling…

…and it can do anything. But it will make you cry getting there.

MusE has a fair bit of traction in electronica, because it’s really a sequencer. But it also has DAW capabilities, and the stated intent is to expand into the DAW arena. It’s Linux-only. If you anticipate a lot of sequencer use, and have relatively light physical instrument requirements, give it a look.

Rosegarden started out as MIDI and composition software, and that’s still where its heart is. But, as with MusE, it’s headed into DAW territory and added at least some of the basics of the functionality. If you like sheet music composition and MIDI, you may want Rosegarden.

So, what about the hardware? I’ll approach this from the idea that you’re building a new box for this, or upgrading an old one substantially. If you’re not, well, skim this anyway.


Screw you, Best Buy

Here are things not to care about: what the case looks like. How cool anything on the motherboard sounds. (We already talked about external sound interfaces; if you skipped it, go read up.) The graphics card. You’re not doing video: you do not care.

What you do care about: fan noise. Bus throughput, on the hard drive side and on the USB chain side. (I’m assuming you’re on USB and not FireWire or Thunderbolt, mostly because that’s where we are in the technology curve right now.) Raw CPU power. Lots and lots of RAM. If you want to spend some money, throwing some dosh at an SSD drive is not misallocated funds.

Basically, you want to build a lean box dedicated to math – because math drives your virtual effects – and moving audio data around, and nothing else. Every other toy, every other frob, adds interrupts and takes CPU and bus time away from what you’re doing with audio. Rip that shit out.

One particular task you’ll want to figure out is probing your USB bus for onboard devices. A lot of motherboards will share device assignments between on-motherboard equipment and external USB ports. This is technically correct – the best kind of correct – but in high-demand applications results in more interrupts on the bus and slower throughput. This can and in my case did result in higher latency and buffer overruns. Find and use ports which are unshared for your external audio card.

Also, for Linux in particular, you may find that wireless internet will be a problem. It’ll work, but will interoperate badly with your realtime kernel, hammering you with interrupts and popping you out of realtime mode.

Some people ditch networking entirely. If that’s not okay, go wired. If you must go wireless, get an external wireless bridge and connect it via ethernet cable to your wired (and realtime-kernel-compliant) ethernet card. This will solve many weird network problems.

But I said we’d talk about hardware, dammit! So okay! Where do you get performance hardware for cheap?

Well, you shop around, of course. Check your local parts stores, but the cheapest route I’ve found is to get a copy of CPU magazine’s motherboard roundup issue – preferably the last couple of years’ worth – and to go the gaming kit-out sites.

Yes, I know, I just talked about case mods and all that: don’t care. You don’t go for the frills: you go there for the motherboard clearance sales, because last year’s gaming l33tness is this year’s dogshit, as far as they’re concerned, and they just want it gone.

As a result – the fire-breathing motherboard inside my DAW? 75% off retail. The CPU, 60% off. The RAM, sadly, not as much, but still: bargains are to be had, and I had them.

When browsing, though, choose wisely! Look over the supported hardware list for your operating system and DAW and follow them. The last thing you want to be doing is tracking down some obscure kernel bug and finding that it’s only fixed in a downstream revision your distribution doesn’t even support yet, so you end up installing a custom kernel configuration and doing haxx0r insanity, not that I know anything about that.


Fuck yeah, meme baby. Fuck yeah.

And that’s an overview! Believe it or not, that is an overview; there are an endless series of twisty passages you can run down on this topic, all alike. I’d browse a little, pick one, and dive in.

If you’ve already built a DAW, what do you use, and why? What problems did you hit that I haven’t covered? Is anybody out there using Thunderbolt yet? Share your experiences!

Finally, I teased an announcement up top. It’s super awesome. Get this:

NEXT WEEK, we have a special event! We’ll be kicking off a series of monthly guest DIY posts with one from JEFF BOHNHOFF.

You may know Jeff and Maya Bohnhoff from their YouTube hit, Midichlorian Rhapsody, or some of their many albums and awards. Jeff and Maya also built Mystic Fig Studios, and Jeff has engineered and recorded literally dozens of albums in his 30-year musical career.

And next week, Jeff will be stopping by here, to talk about DIY sound control in your home studio. We’re thrilled to have him, and YOU WILL WANT TO READ THIS, if you have any DIY recording interest at all.

Until then – see you in Vancouver!

 


This post is part of The DIY Studio Buildout Series, on building out a home recording studio.

Mirrored from Crime and the Blog of Evil. Come check out our music at:
Bandcamp (full album streaming) | Videos | iTunes | Amazon | CD Baby

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