Bumblebee Pro Active DI Kit Pre-Order Offer October 24, 2017 14:52
Longtime DIYRE followers will know that we go way back with Latvian designer Artur Fisher. We helped Artur launch his first product, the RM-5 ribbon mic, back in 2011, and our customers have been continually blown away by his designs since then.
So I'm proud to team up with Artur again to launch his new active DI kit, the Bb-D2.
Bb-D2 Design and Sound
The Bb-D2 is a premium direct box that reflects Artur's no-compromise design attitude. The active circuitry is all discrete, featuring matched JFETs, and the output transformer is a custom designed to compliment the active stage.
All of this sweating the details has resulted in a world-class DI that captures instruments with natural clarity and just a tiny bit of analog finesse. The JFET input stage ensures a high input impedance, making the DI great for high-Z sources such as passive guitar/bass pickups, Rhodes, and piezos.
Beginner Friendly Kit
Part of what makes our partnership work so well is that, like DIYRE, Artur is as serious about the DIY aspect of his products as he is about the design. The Bb-D2 kit comes with every part needed to build the DI and is accompanied by a step-by-step instruction guide. Also, after having handled many support tickets for his ribbon mics over the years together, I can tell you that Artur is very responsive to questions.
Artur is currently funding his first production run of Bb-D2s, so now is the best time to get in on special pre-order pricing. Until Wednesday, Nov 8 the full kit will be only 119 EUR (down from 169 EUR street) with the code DISCOPAMPDI.
Check out the pre-order page on Bumblebee Pro's site for full offer details and to place an order.
Colour Lineup Update August 2017 August 31, 2017 09:26
Recently we made full sample videos of every Colour module we have available so you can hear each at a few settings on a variety of sources. These samples give a nice accurate picture of what you’re getting with each Colour. However when I’m looking to buy anything that makes sound, there’s no replacement for seamless A/B audio samples, and switching back and forth between videos just doesn’t cut it for those purposes.
So we've taken our audio samples and put them back-to-back so you can hear every Colour at the same settings on a single source. Not only does this let you hear the overall character of every module, but it gives a good sense of which Colours are subtle even when pushed, versus others that start gnarly at low settings and collapse even more as you push them.
For a few highlights, check out how the bass comes alive with the 15IPS, or how the CTX adds a subtle weight to acoustic guitar at medium settings. The Distortastudio is an easy favorite for blowing up drums and electric guitar, while the Colourphone really sits the same electric guitar in a nice, brash, fuzzy place. Or check out how Pulse makes a great vintage slapback on drums. Check out the samples below:
"Explain Like I'm 5": Audio Transformers May 25, 2017 12:28
What are transformers?
Transformers are those huge, heavy, primitive-looking parts you’ll see in both vintage and modern audio gear.
They look something like this:
What do audio transformers do?
A lot of things! I guess that’s why we keep them around, primitive and expensive as they are.
Transformers do a lot of different jobs in audio gear, including:
- Stepping voltages up or down: increasing output level of microphones, bringing instruments down to mic level, etc.
- Providing balanced inputs and outputs
- Impedance matching
- Eliminating ground loops
- Blocking DC while passing audio signal
How do transformers work?
A transformer is just two long wires wrapped around one magnetic core. Signal passes from one wire to the other, but the wires don’t touch. What sorcery is this?! It’s a funny feature of our universe called “electromagnetism,” where electric current creates magnetic fields and vice versa.
Modified and reused under the GNU Free Documentation License
The electrical current running through first coil of wire (the “primary”) creates a magnetic field in the core. This magnetic field then induces a corresponding voltage in the second (“secondary”) coil. Voila! We now have the same signal at both sides of the transformer without a single electron making the journey from one side to the other.
How do transformers step voltages up or down?
We only get that same exact signal on both sides when both coils have the exact same number of turns around the core. By changing the number of turns in each coil, we can directly change how much signal is transferred between them.
For example, if the primary coil has 200 turns and the secondary has 100 turns (we call this a “turns ratio” of 2:1), only half the signal will be transferred. This is called a “step down” transformer. However, we could turn that same transformer around and use it as a “step up” transformer to double the signal!
If that sounds too good to be true (free gain!), it’s because we’ve only considered half of the equation—if a transformer steps up voltage, it steps down current by that same amount, and vice versa. Another way of saying this is that transformers can’t create or destroy power (power being the product of voltage and current). You could think of transformers as trading voltage for current, while power stays the same.
For example, let’s say a 1:2 transformer sees an input of 1 Volt at 1 amp current. At the secondary, the voltage will be stepped up to 2V, but with only 0.5 amp of current available. So a 1:2 transformer for voltage is a 2:1 transformer for current.
How do transformers match impedance?
Sometimes it’s important to match the impedance of two devices that are being connected (see "ELI5: Impedance" for more on that). For example, your guitar won't sound very good if you plug it right into a mic preamp—the output impedance of the guitar is just too high to transfer all its signal to the preamp. So we use a DI box to step the guitar’s output impedance down to mic level.
And guess what's in a passive DI box—a transformer! Transformers can step impedance up or down in the same way they do with voltage and current. Except whereas they change voltage by the turns ratio and current by the inverse of the turns ratio, they change impedance by the square of the turns ratio.
So let’s look at a DI box as an example. A typical passive DI box transformer has a turns ratio of 12:1, which means it will step down the guitar’s output impedance by 144:1 (12 squared). A typical output impedance for a single coil pickup is around 20k Ohms, which our DI box will step down to 138.8 Ohms, which is typical of a microphone. Now we can run that guitar directly into our mic preamp with no impedance issues. Transformers win!
Why do transformers sound so good?
Of course, we don’t just keep transformers around to do technical jobs—they also sound really good. There are a couple reasons for this, mostly having to do with the unique ways in which they "fail" to be perfectly clean and linear.
Like all analog components, transformers clip when given too much signal. Transformer clipping happens when the core saturates and can't contain any more magnetic flux. This sets a hard limit on the amount of signal the transformer can pass and generates harmonic distortion.
What makes transformer saturation so lovely is that the distortion it creates is inversely proportional to frequency. Which is a fancy way of saying transformers create more warm, gooey, low-frequency distortion and less harsh, bright, high-frequency distortion.
Transformers also exhibit another distortion phenomenon called "hysteresis." This is where the core, after getting magnetized by a signal, stays magnetized for a short period of time after the signal is removed. Hysteresis creates low-frequency, harmonic distortion at all signal levels, not just when the core is saturated. This same effect is a large part of the desirable sound of analog tape.
Why is transformer inductance important?
Inductance is a measure of how well a component converts voltage into magnetic flux. We're concerned with inductance in audio transformers because higher inductance in the primary coil translates to better low-frequency response. Inductance can be increased with either more windings or a more permeable core material (see below).
How does core composition affect the sound?
Different materials have different abilities to contain magnetic flux—this is called “permeability.” Core materials with higher permeability create higher primary inductance, and therefore better low-end response. However, more permeable core materials will also saturate faster than less permeable ones. Ah, nature, where everything's a tradeoff!
The most common core materials for audio transformers are M6 steel (steel with a bit of silicon) and nickel/iron alloys. Cores with high nickel content are more permeable and more expensive, with less hysteresis than steel cores.
In general, steel will have higher distortion at normal signal levels due to hysteresis, while nickel will have higher distortion at higher levels, due to saturation. For this reason, you'll often find high-nickel cores in high-quality transformers designed for lower signal levels, and steel cores in cheaper transformers or those designed for high signal levels.
What’s a zobel network?
Transformers, like all other parts that exist in the real world, have unintended effects called “parasitics”; ie, extra resistance, inductance, and capacitance that an ideal transformer would not have.
Sometimes these parasitics will combine to cause the transformer to ring in the audio range. That is, if fed a certain frequency, the transformer will keep ringing at that frequency even after the signal is removed.
A Zobel network is a simple, passive filter consisting of a resistor and capacitor placed after the transformer to eliminate ringing.
What'd I miss?
Thanks for reading! Let me know in the comments below if you have any questions about transformers I didn't answer.
Designing a 500-Series Optical Compressor Pt. 1 March 30, 2017 13:33
It’s time to let folks in on the next collaboration between JC-diy and DIYRE. Our first effort resulted in the EQP5 passive equalizer. This time around we’re doing a classic-style optical limiter.
I have always been a fan of the classic optical leveling amplifiers, and for good reason — their musical processing and simple functionality (just two knobs, one for reduction and the other for gain make-up) make it hard to produce bad results. The most famous leveling amps are the LA-2A and LA-3A from Teletronix (later Urei) which are lovely not only in the character of their dynamic processing but in the inherent tone of their signal paths. It has long been my desire to create a DIY-friendly project whose behavior evoked the spirit of these classics while being accessible in both cost and build, so now we’re making it happen!
We are tentatively calling it the OLA5 (for Optical Leveling Amplifier 500 series ).
So What is a “Leveling Amplifier” Anyway?
“Leveling Amplifier” is a name given to early optical compressor/limiters, as their intention was to level (average) a signal’s dynamic range to make it easier to balance in a mix. The gain reduction element in these designs used a voltage divider featuring a light-dependent resistor (LDR) driven by a light source (an electroluminescent panel or an LED) to drop the level of the signal being processed. As the intensity of the light source (driven by its sidechain amplifier) increases, the resistance of the LDR drops, causing a reduction in level. Because the reduction is produced by a passive device (a resistor) there tends to be less obvious distortion in optical compressors compared to other designs with active gain components.
Additionally, LDRs possess an inherently non-linear release characteristic which changes based on how hard they’re driven. Subtler processing yields soft, forgiving time constants, but wind them up a bit, and you’ll have plenty of grab followed by a fast initial release that slows as it gets closer to full release. This non-linear response gives these processors a unique and musical ‘leveling’ characteristic (as opposed to a more linear, VCA-based compression most of us are already quite familiar with). This characteristic is highly prized for its ability to flatter vocals, bass, and guitars in particular.
The LA-2A was a tube unit, and the LA-3A was all discrete solid state, and both had input and output transformers. By contrast, the later LA-4 was all IC-based with an electronically balanced input and a transformer balanced output. There are already existing DIY projects based on the LA-2A and LA-3A which are wonderful. If you build one of these you won’t be disappointed. However, my goal for this project was to create a simpler and more cost-effective build that still retained the classic leveling characteristic that we love in those old designs. So I chose to start with the LA-4 circuit, and then tweak from there.
Where’s the Love for the LA-4?
Despite the almost universal love for its forebears, Urei’s LA-4 hasn’t enjoyed quite the same enthusiasm from the pro audio community. In fact, I have more than once heard it referred to as the “red headed stepchild” of the LA-2A and/or LA-3A.
Why the disrespect?
Well, the overwhelming gripe people have with the LA-4 seems to be with the signal path of a stock LA-4, not so much its dynamic behavior. Audio passing through a stock LA-4 has a tendency to sound a bit dark or ‘veiled’ while being a bit abrasive in the midrange, particularly when working with stronger signals. Both of these characteristics are a result not so much of poor design, but of the proliferation of the less-than-stellar RC4136 quad opamps that make up almost the entire signal path.
The lackluster 4136 lowered production cost and simplified the build, but it also limited the speed of the audio (called ‘slew rate limiting’) making signals dull. The remedy for this is a switch to better opamps, which is a challenge given the unique pinout of the 4136. But we’re talking an entirely new, ground up DIY project here, so opamp selection isn’t a problem for us... we can use whatever IC’s we want!
Upgrading the opamps reveals the gain reduction circuit's refined, vintage dynamic character for which classic optos are coveted. My first prototype coupled my own electronically balanced input stage with the LA-4’s original output circuit (with modern substitutes for the actives), and it sounded great. I have since swapped the output stage for the proven output arrangement I used in the EQP5 Passive Equalizer. So this unit’s signal path offers the same, exceptional performance as that in the EQP5 including the option for the vintage discrete opamp/transformer output bundle to give it more old-school love!
Expanded Feature Set and Functionality
We've added or improved features compared to the LA-4 design, including sidechain HPF, better stereo linking, a legit, discrete opto cell (instead of the off-the-shelf vactrols that everyone else is using). I will cover all of that and the path we took to get here in a future post, but our immediate need is for feedback from you regarding the possible inclusion of one specific feature.
Question for you: Wet/Dry Mix?
I must admit this feature, included on the prototypes, has become a personal favorite of mine, but one that some may feel doesn’t really belong on a compressor of this heritage—the ability to do parallel processing. It seems more and more hardware and software dynamics processors these days offer on-board parallel functionality, but is this because people find it really useful, or just because it gives marketing something to boast about?
The beauty of the LA-4 sidechain is that it excels at leveling audio signals in a musical, but not overly heavy-handed way. Yet I find that this musical sidechain, when slammed hard, does, indeed, have an attitude of its own that blends well with many unprocessed signals to affect both the perception of size as well as really sticking a signal dynamically “in your face." Under mild-to-moderate processing (which is not uncommon for such a true leveler such as this) the usefulness of parallel processing is less obvious, but when you really wind it up the process can be quite magical.
So this, dear readers, is the primary question we are struggling with regarding the design of the OLA5: should we include parallel functionality, or should we simply respect its classic heritage and stick to purely serial processing?
What do you folks think? We've included some samples and clarifying info below. And please feel free to comment or make suggestions on any other aspect of its design too. It’s not too late to change things, and we wanna know your thoughts!
Rock Mix Samples
- Dry - Unprocessed
- Series - Uses substantive, but gentle series leveling
- Parallel - Uses a slammed signal underneath the dry signal to fill in the gaps and make the overall blend bigger and fuller
- Just the crushed part of the parallel compression
Guitar and Vocal Samples
Why Two Knobs for Wet/Dry? In the original prototype I used a single ‘blend’ knob that adjusted the balance between the unprocessed and processed signals, but I found that I couldn’t quite tweak its ballistics to my liking — everything from about nine o’clock to three o’clock on the pot sounded remarkably similar with only the outer throws of the pot yielding much perceived control over the balance. This really bugged me in use, so I switched gears and chose a separate fader for the “dry” input signal just like the processed signal has. While this is a different approach to a blend knob, I find it quite simple to make the desired adjustments, and both knobs provide a proper feel. Plus, those who aren’t interested in parallel processing can simply turn the dry signal all the way off.
"Explain Like I'm 5": Printed Circuit Boards December 01, 2016 13:23
What is a printed circuit board (PCB)?
Before I answer that, let's talk about how circuits were made without PCBs. Before the advent of PCBs, people soldered components and wires right to each other, usually with the help of some sort of rigid board. For example, the technique shown below is called "turret board," where wires and parts are soldered to each other and to turrets to complete a circuit.
As you can see, assembling electronics used to take a lot of time, skill, and focus! Then PCBs came a long and made everything a easier by building all those wires right into the board itself.
PCBs are rigid boards with pre-wired circuitry, plus some extra perks like soldermask and silk-screening (more on those to come). So these days, instead of painstakingly cutting, bending, and soldering each connection, the assembler just has to solder each component in the right place and the PCB does the rest.
What are PCBs made of?
PCBs are made of several layers, like an onion parfait. These layers are: substrate, copper, soldermask, and silk screening. PCBs can have several of each of these layers, but most audio PCBs are "two-layer" boards (top and bottom) with this makeup:
Let's take a look at a very simple PCB, our L2A Passive Re-amplifier, layer-by-layer. To keep things simple, we'll only show the top copper, soldermask, and silk-screen layers.
The core of a PCB is fiberglass. Its job is to be rigid (to hold the parts in place) and non-conductive (so electricity won't pass from one side of the PCB to the other). The holes you see in the substrate are for mounting components to the PCB.
This is where the circuit happens. The copper circles you see are called "pads"; this is where the parts get soldered to the board. The lines are called "traces"; this is what carries the electricity from one part to another. There are only a couple of traces visible here because most of them are on the bottom side of this particular PCB.
Although the pads and traces are made of the same thing, we only want to solder to the pads. So we cover the traces in a layer of polymer called "soldermask" that keeps us from getting solder on the traces.
Finally, we print some silk-screened labeling so that the humans can know where to put the parts.
Can I make my own PCBs?
Yes! Start by learning your way around an ECAD (electrical CAD) program. The most popular free programs are EAGLE, DipTrace, Upverter, and KiCad. My favorite of the bunch is Diptrace, though EAGLE is somewhat of an industry standard among DIYers. Here's what the L2A board looks like in DipTrace:
The best beginners' guide I've found to PCB layout is Dave Jone's PCB Design Tutorial (PDF).
Once you've got a layout, you can either order your PCB from a manufacturer, or etch your own at home.
OSHPark is an amazing, game-changing service that charges only $5 per square inch for three copies of your PCB. Unless you plan to make lots of PCBs at home, you won't beat that price rolling your own, and you certainly won't approach the same quality.
If you just love the idea of making your own PCBs or can't wait for OSHPark to deliver, Make Magazine has a great rundown of all the ways you can roll your own.
Any other questions about PCBs you like me to answer?
Let me know in the comments!
Troubleshooting Audio Electronics in 3 Steps (Without Any Test Equipment) October 31, 2016 13:48
You finished your DIY project, but it doesn't work. You don't have any test gear, and even if you did, you wouldn't have any idea how to use it. You're screwed, right?
Wrong! There's a lot of troubleshooting you can do without any test gear or electronics background.
Below, I walk through the 3-step troubleshooting process that we guide our customer's through when their assembled kit isn't working.
These steps don't require any special tools, just some attention to detail. And in the vast majority of cases, they're all that's needed to find and fix the issue.
1. Eliminate Varibles
Before you go digging through your non-working unit, make sure the problem is actually in the unit!
Some of these checks may seem very stupid and obvious. But that's the point of this step—to uncover anything you might have forgotten because you were distracted by your DIY project.
- If the unit needs power, is the power on?
- If you’re testing with a condenser mic, is phantom power on?
- If the unit is a 500-series module:
- Is the module screwed into the rack completely?
- Have you tried the module in other slots in your rack?
- Have you tried other modules in the same rack?
- Are you patching to/from the correct slot in the rack?
- Is the routing to/from your DAW correct?
- Are you using the correct cables for input/output? (Balanced, unblanced, etc.)
- Have you tested your signal path with a known working device?
Once you're confident the problem lies with tour DIY unit, move onto the next step.
2. Visual Checks
This is the step where you're most likely to find your error. In fact, about 90% of the support requests we receive are resolved by simply going over pictures of a customer's build with a fine-tooth comb.
Is everything in the right place?
Double check the placement of every part, especially resistors, transistors, and integrated circuits. Cross reference against everything available to you: assembly guide, photo of finished unit, bill of materials, etc.
Note: you can’t reliably check resistor values with a multi-meter after you’ve put them in a circuit. Use your eyes and the resistor color code.
Are all polarized parts oriented correctly?
Make sure that every polarized part (capacitors, diodes, transistors, and ICs) is inserted in the right direction. Refer to our post on the subject for tips on identifying polarity.
Is every solder joint good?
Make sure that every solder joint in your build is a shiny, uniform, “tent” around the lead, and that excess leads are clipped right to the top of the solder joint.
If you see joints that could use some love, follow this re-flowing process:
- Get your soldering good and hot, make sure the tip is clean (no black oxidation) and tinned (plenty of solder covering the tip)
- Heat the solder joint, touching the iron to where the pad and the lead meet
- Heat for 2-5 seconds until the solder melts completely. Sometimes the solder will seems to suddenly “snap” onto the pad. That’s what we want!
- Add a tiny amount of solder to the molten joint. Leave the iron for another 2 seconds after this
- Remove the iron
3. Replace ICs and/or Transistors
In my experience, if you've made it this far without finding the problem, the most likely culprit is a damaged chip (IC) or transistor. These parts are sensitive to small electric shocks, which can happen during shipping, building, or even just sitting on your desk.
Without a dedicated test jig, the best way to determine if an IC or transistor is broken is just to replace it. And if you don't have the electronics knowledge to identify which chip is broken, the best path is brute force: replace them all!
It may be counter-intuitive, but it's usually better to replace every IC/transistor at the same time, rather than one-by-one. This is because, if you replace one of the parts that isn't defective, the defective one(s) could damage it and send you back to square one.
When All Else Fails... Seek Help
If the steps above don't reveal the problem with your build, it's time to seek expert advice. If your project is a DIY kit, hopefully you can get support from the company that sold it. If you bought the kit from us, just drop us a line via the support page.
But if you project isn't a kit, or doesn't come with support, you may be able to get some friendly advice at the GroupDIY forum.
Let us know below if you have any troubleshooting tips we missed.
Designing a 500-Series Pultec-style EQ Pt. 2 September 14, 2016 15:35
Today's post is part 2 of a series by Joel Cameron of Rascal Audio about designing a new EQ kit.
Okay, so back in March I wrote about the early development of a 500-series DIY Pultec EQP-type equalizer which we have subsequently dubbed the ‘EQP5’. At that time, we had the initial working prototype that I had built into a small cookie-type tin, and we received a lot of encouraging interest and feedback from you guys (thanks so much!). Since then Peterson laid out proto pcbs in 500 format, built a set up and sent the unit to me for testing and troubleshooting. Well, we’ve tested, tweaked, cut traces, soldered jumpers, etc, have generally played with this thing for a while now, and have settled on the final circuit design.
The Final Circuit Design
The final circuit, barring any last minute changes, is the ‘four-band’ EQP type filter with two frequencies per band: 30Hz and 60Hz for both LF+ and LF-, 10kHz and 16kHz for the HF+ and 10kHz and 20kHz for the HF-. Adventuresome builders will be free, of course, to tweak frequencies to heart’s content, and alternate component values to assist them in doing so will be included. The input and output of the basic kit include electronically-balanced I/O with an optional discrete opamp/output transformer output option as well (for a bit more girth, dynamic imaging and overall vintage vibe).
I gotta say, this thing sounds wonderful! I am really pleased with it. I’m confident this little guy will find a welcome home in 500 racks across the globe, from the small, high quality home studio setup to heavyweight, well-appointed facilities alike (after all, who couldn’t use a few more Pultec-style channels for tracking and mixing? And one could load a 10-slot rack full of these for less than $3000, including the rack!). It’s great on everything.
So you wanna hear it?? I have made some audio samples for y’all to check out:
HF- CCW (off)
HF+ 10k @ 12:30
LF- 30Hz @ 11:30
LF+ 30Hz @ 11:00
HF- 20k @ 11:30
HF+ 10k @ 1:30
LF- 30Hz @ 12:00
LF+ 30Hz @ 8:30
HF- 20k @ 11:00
HF+ 10k @ 2:00
LF- 60Hz @ 11:00
LF+ 30Hz @ 10:30
Custom Transformer/DOA Output Stage
I’m also pleased to report that the transformer used in this prototype is a new trifilar design that we developed specifically for this project (and others forthcoming...). It is essentially a trifilar version of the more typical quadfilar API 2503-type output used in similar circuits. This steel-core output transformer provides the same 1:2 connectivity as is common for such discrete op amp driven outputs without the waste and expense of the fourth, generally unused winding. If you prefer to use a quadfiler design (in case you already have an API 2503 or Cinemag CMOQ-2, etc. that you wish to put to good use.) the pcb is designed to readily accept those units in the same 1:2 configuration, so you’re good either way.
On a similar note, this new output transformer will be right at home in any API 312-type build, providing the same 1:2 output, again, without the expense of the fourth, unnecessary winding. Just wire the colored leads the same way you would a 2503-type, and you’re good to go.
This transformer, when used with a discrete opamp, provides greater dimensionality compared to the stock output. Transient signals, such as drums, percussion/loops, strummed acoustic guitars and thumping bass, particularly benefit from this treatment, appearing to lean forward from the speakers with less actual level needed. And, as the audio samples demonstrate, the stock, electronically-balanced output is no slouch either, showing remarkable musicality with a touch more clarity while preserving the original dynamic content of the material. Both outputs yield a wonderfully organic result.
What Makes This Design So Musical?
So why, exactly, is this filter so musical on such a broad range of sources? What is it about this design that has placed it among the most desired and coveted of all EQ’s in the history of recorded audio? Well, two things certainly contribute: for one thing the filter itself is entirely passive, and secondly, it’s curves are broad and musical.
With regard to the passive nature of the Pultec EQP-type filters it is useful to note that, in theory, two equalizers that share the same transfer function will sound the same regardless of whether one is active and the other is passive. (Google ‘transfer function’ if you want to know more about that). In practice, however, it certainly appears the passive designs I have had the pleasure of using (such as passive models from Spectra Sonics consoles of the early 1970’s, Langevin’s EQ251, and yes, the beloved Pultecs, among others) do yield a smoother, more natural characteristic (particularly when boosting hi-mids and highs) than their modern/active cousins. I mention this, because simply saying “passive eq’s are smoother than active eq’s,” isn’t really a true statement, though in practice this often seems to be the case, and smoothness of this filter, in particular, is quite lovely.
In addition to the natural quality common to many passive filters, the EQP filter in particular (originally designed by Pulse Techniques co-founder, Eugene “Gene” Shenk back in the 1950s), provides broad, gentle curves that are great for overall sweetening duties. The low frequencies may be labeled “30” and “60” Hz, but their impact reaches well into the midrange. The high frequency curves are similarly broad, making them very powerful for balancing the frequency spectrum. In fact this filter design was intended for balancing overall audio spectrum on program material (i.e. buss outputs or whole mixes) rather than honing in on problematic frequencies within individual sources the way more current EQ designs are intended to do. As such the EQP-type circuit is uniquely capable of enhancing your audio in a way no other can.
The result is euphonic – you’ll want to put it on everything!
So check out the samples, feel free to ask questions or make comments, and we’ll keep you posted on progress!
Thanks for reading.
"Explain Like I'm 5": Common Parts Markings and Polarity August 18, 2016 11:22
In the five years we've been selling kits, we've received a couple thousand support tickets. And I've found that ~90% of issues come down to two common errors:
- Parts in the wrong place
- Parts in the wrong orientation
While these errors are easily avoidable, they're not "dumb" errors. Identifying parts can be really confusing, since every kind of part follows a different convention.
In this edition of "Explain Like I'm 5" we'll cover how to identify the value and orientation of the most common parts.
Resistors feature colored bands which indicate their value (resistance) and tolerance (resistance range). To identify a resistor, check the colors of the bands against a color code chart or look them up on a color code calculator.
However, if you have a multi-meter there's an even better option! Set your meter to read resistance (Ω symbol) and probe either side of the resistor. Just keep in mind that the actual resistance will vary based on the tolerance of the resistor. For example, the resistor in the photo below is rated at 910Ω with a 1% tolerance and measures 904Ω.
Resistors are not polarized, so there's no correct orientation for them.
Electrolytic Cap Value
Electrolytics are one of the least forgiving components: they are polarized and they tend to explode spectacularly when they're inserted backwards. On the bright side, they're perhaps the most completely and clearly marked parts there are.
Electrolytics' value (capacitance) and voltage rating are marked right on the body, with units specified and everything!
Electrolytic Cap Orientation
And manufacturers are so serious about making sure you don't reverse electrolytics that they marked their polarity twice. How nice of them! An electrolytic's positive lead is longer and the negative lead is marked on the body with a stripe and minus signs.
If only all caps were as clearly marked as electrolytics. Most caps are marked with a three-digit code for value and one letter for tolerance.
The three-digit code indicates the cap's value in picofarads. The first two digits are the first digits of the value, and the third digit is the number of zeroes. So, in the photo below, the cap on the left is 100pF (10 + one zero) and the cap on the right is 100,000pF (10 + four zeroes).
If, like me, you can never remember the metric units, you can use an online converter to convert that 100,000pF to something more readable like 100nF.
Unlike electrolytics, most other caps are not polarized. The very few exceptions, such as tantalums, have their polarity marked on their bodies.
Diodes' names are marked right on the body (though you may need a magnifying glass to read them).
A gray or black stripe marks the cathode (negative) lead. Just align the stripe on the diode with the stripe on the PCB and you're set.
It doesn't get easier than identifying an LED. Searching for the "red LED" from the BOM? It'll be the red one.
And just like electrolytic caps, an LED's positive lead is longer than the negative lead.
Transistors are very straightforward to identify because, instead of a value, they have a model number which is marked on the body.
Since different transistors have different names for their leads, the most reliable way to identify orientation is by shape. Simply match the shape of the transistor's body to the shape marked on the PCB.
Integrated Circuits (ICs)
Like transistors, ICs have a model number which is marked on the body. There's often a batch number, too, which can be ignored.
IC manufacturers indicate orientation in a couple different ways. First is with a notch on one side the body (between pins 1 and 8). This notch is usually shown on the PCB as well. Second is with a dot next to pin 1.
Recording with the CP5 Colour Mic Preamp (Video + Stems) July 21, 2016 16:53
The DIYRE team spent a couple days in the studio recently to record new backing music for our how-to videos. We recorded everything with our CP5 Mic Preamps and various Colours so you could hear them at work.
You can download the tracks as hi-res stems via the link below. The only processing on the tracks is the Colours used during tracking. See the session notes below for mic and Colour specifics.
The stems are released under a Creative Commons license, so feel free to use them in your own tracks or remixes.
Watch the Recording Process
Check out the video below to see exactly how we set up and used the CP5s + Colours during tracking.
- BPM: 144
- Bit Depth: 24 bit
- Sample Rate: 48khz
- Converters: Lynx Aurora 16
- Mic Preamps: CP5 Colour Mic Preamp
|Kick Drum||ElectroVoice RE-20||15IPS Tape Saturation Colour|
|Snare Top||Shure SM57||Rogue-Etc Air Passive EQ Colour|
|Snare Bottom||AKG C414 B-ULS||Distortastudio Cassette 4-Track Colour|
|Drum Room||Flea 47||Toneloc Compressor Colour|
|Guitars (Fender Classic Player Jaguar > Vox AC15)||Shure SM57||Pentode Tube Saturation Colour|
|Bass (Fender American Standard Precision)||Direct Input||DOA Colour + GAR1731|
|Synth (Korg MS-20 Mini)||Direct Input||15IPS Tape Saturation Colour|
|Tambourine and Shaker||Neumann KM84||15IPS Tape Saturation Colour|
Our 10 Favorite Places to Buy Parts (that aren't Mouser or Digi-Key) June 03, 2016 14:33
A huge part of what we do here at DIYRE is "sourcing"—finding the best vendor for each part in our kits. What that means in practice is countless hours of scouring the internet and navigating clunky, decade-old parametric search forms for deals on parts.
Some people actually enjoy this (those people are broken inside), but for those of you who don't we've compiled this list of our favorite sources.
- Redco Audio: One-stop shop for jacks and cabling. Simply the best prices anywhere for Neutrik jacks.
- OSHPark: Get your own PCBs made for cheap in the US! OSHPark has an awesome online ordering system, charges a very fair price of $5/square inch, and sends you 3 copies of your PCB within 2-3 weeks. We use them monthly for our prototypes.
- Apex Jr.: Specializes in buying vintage overstock. Great for old tubes, transformers, and weird stuff!
- Edcor Electronics: Transformers made-to-order in the USA for incredible prices. We use their PC10:10k in our L2A Re-amplifier.
- McMaster-Carr: The hardware superstore with the coolest website in the world. Get your nuts, screws, standoffs, etc. here in every size and finish imaginable.
- MonoPrice: Known for cheap earbuds and USB cables—but have you seen their audio adapters section? The perfect place to stock up on those magically disappearing TRS adapters.
- Bitches Love My Switches: Cheap guitar pedal parts with an attitude. Great prices on 1/4” jacks, switches, knobs, and cases.
- Tayda Electronics: Tayda was introduced to me as, “the site that brazenly undercuts everyone else.” I’d say that’s about right. Tayda stocks generic versions of the essentials for prices that beggar belief. And here’s a pro tip: always check Tayda’s Facebook page for their monthly 15% discount code before your order.
- All Electronics: You might be aware of the huge distributors like Mouser and Digi-Key. But sometimes their gigantic catalogs are more confusing than convenient. That’s why I like All Electronics as a general parts store—they have almost everything you’d want, but not in every variety, brand, etc.
- AliExpress: Need 1-100 of a unique part of questionable quality and mysterious origin? Look no further than AliExpress. How about 100 mic capsules for $0.07 each? Or 100 Neve-style knobs for $0.50 each?
I hope this list saves you a few hours hunched over a computer (that you could have spent hunched over a soldering iron!) and leads you to some good deals. If you know of any gems we missed, please let us know in the comments.
Designing a 500-Series Pultec-style EQ Pt. 1 March 24, 2016 12:21
Today's post is by Joel Cameron of Rascal Audio, who's collaborating with us on a new EQ kit.
I love DIY! I got my start building gear almost two decades back by scouring the internet (a much smaller internet back then) for schematics of classic gear in hopes of building the stuff I couldn’t afford to buy. This was long before sites like DIYRE came along, of course, and I had to figure out how to do things pretty much from scratch. Along the way I made a lot of mistakes, of course: bad grounding (“hummmmm.....”), popped caps, burned up power supplies, toasted transistors and opamps, etc. But each lesson learned was invaluable, and after a while I figured out not only how to make great gear, but I began learning how circuits worked and what it was made these old designs so great. I eventually began to come up with circuit ideas of my own...
Well, I’m pleased to say that my latest idea is one specifically aimed at the DIY community—a 500-series Pultec EQP-style equalizer!
Drawing Inspiration from a Classic
For more than a decade now the filter topology used by Pulse Techniques (aka “Pultec”) in their EQP variants (EQP-1/1R/1A/1A3/1S3 and EQH-2) has been my absolute favorite EQ circuit. Appropriately referred to as a ‘program EQ’ these units paint with broad, deeply enhancing strokes that make them a proper choice for both tracking and mixing. No surgical maneuvers here... this is all about tone! Of particular interest (especially for those working in the DAW environment) is the inductor-based HF boost band which can add clarity and sheen while remaining entirely sweet without any hint of harshness (try that with most EQ plugins!!!). And the LF controls can add immense, unflappable fullness to your low end as well as tame the unwanted mud and weight from bottom heavy sources. And because the Pultec filters are passive, they do all of this while sounding totally natural. In fact, its effect feels so natural you need to be careful not to overuse it.
Recently I put together a mix room in my house, and I’ve been craving a few more Pultec-style channels for processing stems (I mix out of the box). Wanting to save rack space and eyeing the empty slots in my 500 rack, it hit me: I need to build some Pultec-style EQs to fill those slots.... and (lightbulb!!) what a perfect project for the DIY community!
The basic EQP-type filter circuit is actually quite simple, requiring surprisingly few components, so I just needed to add high-quality gain makeup and I/O and we’d have a powerfully musical device that anyone can build.
I contacted Peterson at DIYRE to see if he had an interest in a project like this, and he was game, so I made some drawings and sent them on to him. I also breadboarded the initial concept and sent the contraption on to Peterson and the gang to get their stamp of approval.
The first prototype. It sounds much prettier than it looks.
As of now we have a tentative, proven design that sounds amazing, though before we commit to a final product we wanted to run the overall idea past DIYRE’s loyal readers to see if anyone had some thoughts they wanted to toss in to make this truly killer.
What We Have So Far
The EQP5 (its working title... “EQP” for obvious reasons and “5” for 500 series) will feature an enhanced version of the Pultec EQP-type filter (‘enhanced’ in that it has four independent bands, not three as original EQP’s do). The original design uses a single control to select the frequencies for both the LF+ and LF- sections simultaneously. But these two sections really are separate in the circuit, and the single control of the original is a 2-pole switch, so... we’re separating these into separate switches, so you can boost at one frequency and cut at another, dramatically increasing the usefulness of the thing.
A four-band Pultec!
Each of the four bands has a pushbutton switch for selecting one of two available frequencies per band. This circuit is a broad brush, and the frequency selections are broadly musical over a variety of source material including individual tracks and complete mixes. The use of pushbutton switches keeps the project affordable and the build simple (and also keeps the front panel from becoming too cluttered for big fingers!
The stock design features an IC-based gain make-up (the passive filter has about -16dB loss for which we need compensation) and electronically-balanced I/O. There will be an option to have the gain provided by a discrete opamp driving an output transformer. Any discrete opamp compatible with API’s 2520 footprint can be used (including the RED-25, ML2520 and others available from DIYRE).
Spot the 2520-style opamp and output transformer in the prototype.
Little known fact: the last Pultecs made were had solid state gain makeup provided by an API 2520 discrete opamp driving an output transformer, so this approach is definitely the way to go for a more vintage vibe. It adds a more three-dimensional fullness that seems to reach beyond the speakers, directly engaging the listener. A jumper is included with the optional output to allow the selection of either output topology, so you won’t lose the option of a cleaner signal path if that’s what you want for certain applications.
Questions for You, Dear Reader
Okay, all of this has been tested and sounds fabulous. Here is where we really would like some input: I originally intended the pcb to provide four frequency options per band with any two of them user-assignable (via jumpers) to the front panel pushbuttons. The only concern for doing this is that it might offer additional confusion to newer DIYers, plus it would preclude any ability to silkscreen the front panel with chosen frequencies (which can be disconcerting to some users). To keep things simpler we could simply choose the stock frequencies ourselves, two per band, and have them screened on the panel like normal. And then for those who are more adventurous we could make faceplate available that has no frequency labels along with a chart of alternate capacitor values, so users could experiment to their hearts’ content.
Frankly, those of you who have used Pultecs know how odd the stated frequencies are—how often do you see 20Hz or 30Hz on any other equalizer design? The truth is that these given frequencies affect content well into the midrange, so the labels can be a bit misleading; you really have to trust your ears more than a frequency printed on a faceplate. As such, I think that a faceplate without screened frequencies along with giving the user the ability to program their own choices via jumpers is a useful idea, but what do you think?
Should frequencies be marked on the front panel?
Should there be multiple frequency options?
Any other thoughts you’d like to share?
Thanks for reading. We’ll keep you updated on our progress!
"Explain Like I'm 5": Resistors March 17, 2016 16:45
What are resistors?
Resistors are one of the core building blocks of electronic circuits. Even the shortest signal path in the studio can contain dozens or hundreds of resistors.
Resistors are incredibly simple components: they’re basically wires that don’t conduct as well as regular copper. The degree to which they're bad at conducting (or good at resisting) is their resistance.
But despite being simple, they’re also incredibly powerful and versatile. The circuit inside our SB2 Passive Summing kit, for example, contains only resistors.
A through-hole, metal-film resistor.
What do resistors do?
They resist the flow of electrons. In other words, they limit the amount of current that will flow in a circuit. In the trusty electricity/plumbing analogy, resistors are different widths of pipe.
What can resistors do in a circuit?
Countless things. Resistors are required for creating filters, setting the brightness of LEDs, setting power supply voltages, controlling the response of a microphone capsule, etc. This is why you’ll find resistors in practically every electronic circuit.
Two resistors configured as a voltage divider
By Velociostrich (Own work) [CC BY-SA 3.0], via Wikimedia Commons
What do resistors’ specs mean?
Despite being the simplest of components, resistors have a lot of specs. For simplicity’s sake, I’ll just go through the most important here.
- Resistance: The degree to which it resists the flow of electrons, expressed in Ohms. A 1 ohm resistor is one where 1 volt will create 1 amp of current, or 1 watt of power. Nice and tidy!
- Tolerance: The precision of a resistor’s value, expressed as a percentage. For example, a 100R 2% tolerance resistor could have an actual value 98R and 102R. Most often in audio we use 1% tolerance resistors.
- Wattage: How much power a resistor can handle. Resistors dissipate power by turning it into heat. If a resistor gets hotter than it can handle, it’s value will or change or (more fun!) it will combust. The most common wattage in small-signal audio is 1/4W.
Do resisitors have a sound?
No. For all intents and purposes, resistors do not have a “tone” of their own. They don’t saturate like transformers or have phase effects like capacitors. They may have different self-noise levels and tolerances which can affect the performance of the circuit as a whole, but in general resistors by themselves do not have a "sound."
How do I identify different resistors?
Through-hole resistors (as opposed to surface-mount, which we’re usually not using for DIY) are wrapped in a number of colored bands which tell us the resistor’s value and tolerance.
The metal-film resistors that are most common in DIY projects use a five-band code, where the first three bands represent the first three numbers of the resistance value, the fourth band is the multiplier (ie., how many zeros come after the first numbers), and the fifth band is the tolerance as a percentage.
Of course if you don’t want to bother learning or looking up color codes, you can always identify resistors with a multi-meter.
Why do resistors common resistors have such weird values (4.7, 6.8, etc.)?
Back in the day when resistors had very wide tolerances of 20%, it made sense to manufacture only values that were about 40% from each other with a bit of overlap. Thus 1.5, 2.2, 3.3, 4.7, 6.8, and 10 became the standard values for each decade (10x, 100x, etc).
Nowadays tolerances are much better and you can buy a resistor in practically any value. However, the old common values are still made in greater quantity, so they’re cheaper and more reliably stocked. This is why you still see many more 47k than 50k resistors, for example.
How’d I do?
Did that make sense?
Did I miss anything?
Please let me know in the comments below!
"Explain Like I'm 5": Why do I need a reamp box? February 17, 2016 18:05
When we first launched the L2A Re-amplifier kit five years ago, I got a lot of emails asking simply, "what is reamping?" A lot's changed since then. By now, it seems like most people are familiar with the process of patching their recording gear into their guitar gear and then re-recording that "reamped" signal.
However, we do get a lot of questions along these lines:
Do I really need a dedicated device to reamp? Haven't people been reamping since before there were reamps?
Fair questions! The short answer is no, you don't need a dedicated reamp box to start reamping. But for ideal performance in a wide range of situations, you're better off with one.
Can't I just connect a cable right from my interface to my amp?
Technically, yes. But you may get a lot of noise.
Pro-audio gear uses balanced connections, while guitar gear is unbalanced. Connecting the two systems directly creates a path for noisy ground currents to flow into the audio paths.
A reamp box like the L2A solves this problem by isolating the grounds with a transformer. Through the magic of electro-magnetism, the transformer allows signal to pass from the input to the output without a direct connection between their grounds.
But don't take my word for it, here's what the ground lift on the L2A can do:
Additionally, patching right from pro-audio to guitar gear can cause an impedance mismatch. Most of the time this has no audible effect. But sometimes it can register as the reamped signal just sounding "not right." A reamp box can prevent this by recreating the typical output impedance of a guitar pickup.
Can't I just use a passive DI in reverse?
Very clever! But not ideal.
A passive DI is a step-down transformer (usually 12:1) that steps an instrument's volume and impedance down to microphone level. Using it in reverse flips the transformer's ratio, so the DI will step up your signal by 12x. So if your line-level signal for reamping is a standard +4dBu, it will leave the reverse DI at a whopping +25.5dBu! This will clip most guitar pedal and amp inputs.
So, while the reverse DI trick does provide ground isolation, the high ratio of the transformer makes it less than ideal for level and impedance matching.
Can I get started reamping without a dedicated re-amplifier?
Absolutely. I'd never advocate putting your music making on hold while you wait for a piece of gear. If you've got a track that needs reamping today, go ahead and try one of the options above—it may work just fine.
However, if you do get too much hum, or your guitar gear just doesn't sound "right," we do happen to stock the most affordable re-amplifier on the market.
DIY Recording Equipment FAQ January 20, 2016 12:56
Why should I build my own gear?
- To save money on gear. For example, our L2A Reamplifier kit is about half the price of an assembled equivalent.
- To obtain vintage gear you'd never get your hands on otherwise. Just because Neumann doesn't make the U-47 anymore doesn't mean you can't!
- Build cool stuff that doesn't exist on the commercial market.
- Deepen your understanding of the tools of your craft.
- Build something amazing from scratch.
Is DIY gear really as good as the commercial stuff?
Yes, sometimes better. If the design is good, the components are good, and it's built relatively well, the gear you build yourself will be every bit as good as a similar commercial unit. While every manufacturer would like us to believe their gear is magic, at the end of the day they use the same basic components and are subject to the same laws of physics as the rest of us.
I say "sometimes better" because commercial manufacturers often make design or component-quality compromises to meet a certain price point. As DIYers we have the luxury of setting our own price points, and so can choose to use an over-rated power supply, boutique components, a heftier chassis, etc.
Ok, so if it's really just as good as the commercial stuff, why is DIY so much cheaper?
- Commercial gear makers do lot more than just put components together. The brilliant people who design great pieces of gear and invest their time and money to bring that gear to the market deserve to be well compensated. Check out our podcast on "Why is Pro Audio Gear So Expensive"?
- If you are counting your DIY time in terms of dollars, it's often not any cheaper than buying retail. The real savings tend to happen at the extremes of the spectrum, with cheaper stuff like mic cables, which are really cheap to DIY, and the really pricey stuff, such as the Drip 670 which costs roughly $45,000 less than the original.
Do I need to understand electronics to build gear?
Nope! Building gear from a kit is more like putting together a puzzle than troubleshooting your home wiring. It requires patience and care, but no knowledge of electronics theory.
Is building electronics dangerous?
If you stay away from high voltages (tubes!) and wall power, then no. This is why all of our beginner kits are completely passive, and why the 500-series is so popular in the DIY community.
But isn't soldering difficult/dangerous?
True, soldering involves melting metal with a hot pointy thing. But with a little care and practice, it really is so safe that kids and do it. Making perfect, shiny solder joints 100% of the time does take a bit of practice, but it's nothing you can't handle and basic audio projects don't require 100% perfect soldering anyway. Think of it like learning an instrument: anybody can learn C-G-D on a guitar on guitar in a couple hours, and that's all they need to know to play a good number of songs.
What if I break something or get stuck on a project? I don't want to end up with a pile of broken parts.
Five years ago, I would have said this was a very valid concern. However, today there are several companies offering full kits with step-by-step instructions and support, so there's very little danger of completely botching your first project.
How much do I need to spend on tools to get started?
About $50. Better yet, borrow tools for your first project!
Is building my own gear one of the steps on the way to studio ninja-hood?
How do I get started?
What does a resistor do? What's a BOM? Etc.
My question wasn't answered here—what should I do?
Ask us! And and help us improve the FAQ by posting your question in the comments below. We'll get to it as soon as we can.
Introducing Our New 3630 Mod Kit (FAQ) January 13, 2016 13:33
We're excited to introduce our first mod kit: the 3630 Parts Upgrade Kit. The Alesis 3630 is a (in)famous, budget compressor/limiter that can be found in either the racks or storage closets of most studios. Our new mod kit cleans up some of the sub-par components that hinder the 3630's sound.
What does this mod do (in plain english)?
Basically, the Alesis 3630 features a solid circuit that's compromised by crappy parts. This mod replaces those parts with truly pro-quality components for better sound and performance. It swaps out ICs and capacitors in the signal path, and also strengthens the power section with fresh diodes and bigger caps. The end result is a more solid low end response, lower noise floor, and a more transparent sound.
How much does this mod improve the sound?
We'll let you decide for yourself from the samples below. Listen especially for the improved bass response in the kick and the transient detail in the acoustic guitar.
For better audio quality, click through to Soundcloud to download .wav files.
Does the mod make the 3630 sound as good as the soft diminishments of our psyches as we fade away into nothingness?.. or like an LA2A or something?
No probably not. But it will delight your heart and validate your diy spirit, that spirit being what most likely caused you to purchase such a diamond-in-the-rough in the first place.
How much does the 3630 Parts Upgrade Kit cost?
$50 American dollars.
What's in the kit?
Good stuff! Highlights include silver mica capacitors, our all-time favorite Panasonic FR-series electrolytic caps, and THAT Corp. 2180 VCA.
Is this a difficult mod to do?
Nope, not really. I wouldn’t recommend it be the first kit you ever take on, but anyone with intermediate soldering ability should have minimal trouble. The hardest part is desoldering the pre-exising components. But since they’re crap, you don’t need to worry about damaging them!
Check out the step-by-step instructions to see exactly what's involved.
Will doing this mod void my warranty with Alesis?
Is this mod worth the investment?
If you already own an un-modded 3630, I think it's a no-brainer. For the price of a couple cables, you can give your 3630 a second life as a very usable, high-fidelity compressor.
If you don't own a 3630 yet, you can also get in on this modding action by getting a used 3630. You should be able to find one on eBay for under $100. In which case, you'd be looking at $150 total for a respectable, stereo, hardware compressor. Check out.
Designing a Tin Can Piezo Microphone December 01, 2015 18:08
Today's blog post is written by Glen van Alkemade of Zeppelin Design Labs.
Most DIY audio enthusiasts are familiar with the standard “tin can mic” (or variations thereof), in which a piezo disc is taped onto the bottom of a tin can and then plugged into a high impedance voltage amplifier, like a guitar amp. These types of microphones are nice because they are so easy to make, but they are quite limited in most other areas, including bandwidth, microphonics, signal-to-noise ratio, impedance matching, and cable driving. These issues usually result in a very noisy and brittle sounding microphone. To overcome these limitations, we at Zeppelin Design Labs developed the “New and Improved Tin Can Microphone” (Figure 1), which is described in detail in this Instructable. Using common household items along with a simple circuit, this mic has a tonality similar to the classic tin can mic (for all you Tom Waits fans), but improves upon nearly every other feature.
This article will explain the essential design considerations we put into our Tin Can Mic. You should be able to adapt and apply these principles to your own unique project.
Figure 2 (Click to enlarge)
Figure 2 illustrates the essential components of our Tin Can Mic. These are the parts that we think are key to the performance of the microphone. They include the Resonator (A), the Shielding Container (B), the Balancing Circuit (E) and its Grounding Wire (F), and the Resonator Suspension system (I). Following are some comments on each of these features with suggestions for variations and custom mods.
Figure 3 (Click to enlarge)
The heart of the mic is a tiny phantom-powered circuit (originally designed by Alex Rice) which creates a balanced output signal from the piezo disc. A balanced signal is extremely quiet, providing high signal-to-noise ratio. Further, this circuit provides a high input impedance to the piezo disc, and also a low output impedance to match a mixing console input. By properly matching the input and output impedances we achieve a much wider bandwidth than the original tin can mic, and also the circuit can drive the signal along a very long run of mic cable. Figure 3 shows the schematic of our circuit, which we call the Cortado. Our Instructable includes a complete bill of materials from which you can build your own circuit, or you can get a kit from Zeppelin Design Labs.
Speak To Me, or, That Really Resonates With Me
The one piece that most affects the tonality of the microphone is the Resonator, the thing the piezo is stuck to. A piezo disc does not respond to sound in the air; it is only sensitive to vibrations in a surface to which it is stuck, or “coupled”. For the resonator, you want an object that is stiff enough to propagate sound vibrations, but light enough to respond to your voice in the first place. For example, a thin piece of foam padding is very lightweight, and your voice is loud enough to cause it to vibrate. But it is very soft and flexible, and the energy of your voice is quickly absorbed by the foam. A plate of glass will transmit sounds instantly and with great fidelity, but it takes a lot of energy to get a plate of glass to vibrate. You could use a piezo to record the sound of marbles pattering on a glass pane, but even if you shout at that glass, your mic will likely hear nothing.
For these reasons, we settled on a Styrofoam cup for our resonator. There are many other common objects you could use for the resonator: paper cups -- small or large, waxy or plain; plastic cups; plastic food containers of various materials, shapes and sizes; aluminum cans; and of course the original, a steel soup can. Each one will sound significantly different. Even trimming the curled lip off the end of a paper cup will affect the tone. Experiment!
Stick it to me!
For maximum bandwidth, the piezo needs to be tightly coupled to the bottom of the resonator, Figure 2 (C). We think the best thing to use for permanent installation is a good quality double-sided tape (carpet tape), but various glues could work too. Glues will likely filter the sound in different ways. If you want to try the piezo on lots of different resonators, you can just use masking tape or painters tape temporarily.
For the piezo to respond to the vibrations in the Resonator, it is important to use very fine, flexible wire between the piezo and the circuit (D). We use 30 gauge wire in the Cortado. We have found that the wires inside a computer monitor cable make an excellent choice.
EMF Defense Shield
To eliminate noise from the circuit, every element must be carefully shielded, from the piezo all the way to the recorder input. This is why the Resonator, piezo and balancing circuit must all be contained within a conductive Shielding Container, Figure 2 (B). Additionally, the Shielding Container must be grounded to the circuit (F), and you must use a decent-quality shielded mic cable coming from the circuit output (G). You may want to wrap up your circuit board in electrical tape, and secure it to the inside of the Shield Container. In our design, we use a standoff and a couple of tiny screws to mount the circuit board to the bottom of a soup can. Just make sure the circuit can’t short out to the shield, or bump into the resonator.
The shield can take many forms, but must be made of a conductive substance, and must close around the mouth of the resonator close enough to intercept electromagnetic fields that may be approaching. Thus a trash can would not make a good shield, but a length of pipe would be excellent.
I Can’t Stand the Suspense
To eliminate microphonics (the transmission of unwanted signals produced by mechanical vibration), we suspend the Resonator in the Shield with rubber bands. See Figure 2 (I) and our Instructable. The point is to string something elastic back and forth across the shield to form a square into which you can push the resonator. Rubber bands work great; so does sewing elastic, or even lightweight bungie cords. Place the suspension so as to nicely balance the resonator. Be careful to leave clearance for the circuit below the resonator. If the lip of the resonator insists on projecting outside the shield, just trim it off.
We go an extra mile with our Tin Can Mic: we suspend the Shield itself in a Hoop (Figure 2 (H, J)). We use rubber bands for this too, but boy would it look great with little springs! To the hoop we glue an adaptor nut so we can mount it to a mic stand. You will devise any number of clever ways to mount or suspend your mic.
So those are the basic design elements: a balancing, impedance-matching circuit; stiff, lightweight resonator; well-coupled piezo; careful shielding and grounding; and a springy suspension system. Within these general guidelines, you can make a million different microphones, each with a unique tonality.
The $50 Quick & Dirty Soldering Setup October 16, 2015 17:00
So, you want to start saving money and learning electronics by building your own gear. Great! But you don't have any of the tools.
No tools? No problem. You can get up and running with a very respectable setup for under $50.
The $50 Beginners' Shopping List
You only really need three basic tools to get started: a soldering iron, solder, and wire cutters. I made the shopping list below based on what I think are the best deals on Amazon right now (10/2015). The images and links are affiliate links, so we'll get a little commission if you buy something on Amazon after clicking them.
1. (A Good) Soldering Iron - $39
A soldering iron is a pen-shaped hand tool with a hot tip. You use the tip to melt metal to other metal to create electrical connections.
Your soldering iron is the most important part of your DIY toolbox. The better your iron is, the better your solder joints will be and the better your DIY gear will perform. So invest in a decent iron like the $40 Weller WLC100.
Don't even think of getting a $25 pencil iron at the hardware store—that can only lead to buying another, more expensive iron later, or giving up the hobby out of frustration at your crappy iron.
2. Solder - $6
There are two main types of solder: "tin" and lead free. I highly recommend using old school tin solder; it's easier to work with, it makes better, longer-lasting joints, and it actually makes less toxic fumes than lead free.
This spool of 60/40 (tin/lead) solder is a good diameter for circuit board soldering and will last you for several projects.
3. Wire Clippers - $5
Once you've soldered those parts together, you'll need to trim the left-over component leads. This small wire clipper will do the job neatly for $5.
Off to a Good Start
21 Ways of Stating Ohm's Law August 11, 2015 16:31
Ohm's law is the key to understanding basic electronics. It describes how the three elements of electricity—current, voltage, and resistance—relate to each other. Ohm's law can be expressed as an equation three ways:
- I (current) = V/R
- V = IR
- R = V/I
Which is crystal clear if you've studied electronics for years. Most of us need to hear it rephrased in plain language dozens of different ways before it clicks.
All of the statements below are simply ways of restating the equations above. Each one is pretty dense and many will be counter-intuitive. Take your time to unpack them, and leave your questions in the comments if anything doesn't make sense.
I hope one of them makes Ohm's Law click for you.
- Voltage is how much current will flow through a conductor of a certain resistance.
- Voltage is the resistance of a conductor given a certain current.
- Resistance is how much current will flow given a certain voltage.
- Resistance is how much voltage will be generated by a certain current.
- Current is how much voltage will be generated by a certain resistance.
- Resistance is the ratio between voltage and current.
- Current is the resistance of a conductor given a certain voltage.
- Resistance is how easily voltage can increase current.
- Voltage makes current flow through a conductor.
- If voltage is fixed, increasing resistance will decrease current.
- If current is fixed, increasing resistance will increase voltage.
- If resistance is fixed, increasing voltage will increase current.
- If resistance is fixed, increasing current will increase voltage.
- Current is proportional to voltage; resistance is the constant of proportionality.
- Without any of the three, there’s no electricity. (Try putting zero in any of the equations.)
- Current moving through a conductor creates voltage.
- You can’t have voltage without current, current without resistance, etc.
- “Danger: High Voltage!” could also be correctly written “Danger: High Current and Resistance!”
- If resistance is very low, you can get a ton of current with a very low voltage. (eg. 1 Volt / 0.001 Ohms = 1,000 Amps).
- If resistance is very high, you get very little current even with very high voltage (eg. 1 Volt / 1M Ohms = 1 micro Amp).
- If you know two, you can always figure out the third.
"Explain Like I'm 5": Opamps July 21, 2015 14:09
What are opamps?
Opamps (or op-amps, or operational amplifiers) are small, inexpensive integrated circuits that can be used to do a ton of different things in audio electronics. They can apply gain or attenuate a signal, create filters, present desired input/output impedances, oscillate at specific frequencies, etc.
They’re usually manufactured on a black, spider-like component with at least five terminals:
What do opamps want?
Instead of talking about how opamps work from an electronics perspective, it’s easier to think about what they want. Opamps want their inputs to be equal all the time. So if the voltage at both inputs is 0, the opamp is happy–it doesn’t need to do anything at all. But the second we change the voltage at one of its inputs, the opamp will jump into action immediately to give the other input that same voltage. Absolute equality–that’s all it wants.
Opamps are so committed to this single desire that they will burn themselves out in a plume of noxious smoke before admitting defeat.
How do they get what they want?
Opamps use their outputs to make their inputs equal. But they can’t do this on their own–they need us to provide a feedback path. Let’s say we kindly oblige and solder a wire between the output and - input pins. (This is called negative feedback.) Now whatever voltage the opamp sends to it’s output gets immediately sent to the - input as well. In other words, if the + input is fed a certain voltage, the opamp can immediately make the - input the same by adjusting its output voltage. In other words, it can get what it wants!
Now let’s do that with numbers so you can see what it looks like. Say we connect a microphone with a 1V output to the opamp’s + input. The opamp wants to make - input 1V as well so it sets its output to 1V. And since we’ve attached a negative feedback wire, that 1V is immediately sent to the - input. Now both inputs are sitting at 1V and the opamp is happy.
How we make them do what we want?
That’s all well and good, but if we stopped there the opamp would only be good for passing unity gain. The real fun comes in replacing that feedback wire with some more interesting components. Let’s say we replace it with two resistors configured as a voltage divider.
A voltage divider in the negative feedback loop
These resistors will take the voltage from the output and cut it in half before it reaches the - input. Let’s go back to our example from the previous section. If the output were still set to 1V, the - input would be at only 0.5V. And that’s not what the opamp wants! So now it adjusts its output to 2V to get 1V back to the - input. The opamp is happy again and our microphone signal is 2x (6dB) louder at the output of the opamp.
And gain is just the beginning. We can put all sorts of stuff in the feedback loop: capacitors and inductors for filters, diodes for clipping, transistors for variable gain–you can even put other opamps in there! The point is that the opamp just wants one simple thing–to make the inputs equal–and we can make it do all sorts of stuff just by making it work harder to make that happen.
How do opamps work?
I honestly have no idea. I’ve never designed one or even bothered to look at one’s schematic diagram. But that’s the beauty of integrated circuits: you can treat them like a black box. If you understand their specs and theory of operation, you can use them without knowing what’s going on inside.
Why are there so many kinds of opamps?
All of design is basically managing tradeoffs: more gain vs. more noise, greater bandwidth vs. less stability, greater precision vs. higher cost, etc. There are hundreds of different opamps and they all do the same thing but make different tradeoffs. So one opamp may be ultra-low noise, but have stability issues at high gains. Another may be very low cost but have lots of noise. And so on.
What are discrete opamps?
99% of opamps you’ll come across are monolithic, as opposed to discrete. They look like the spider thing below and are manufactured in massive quantities on silicon chips.
A standard monolithic opamp IC
But some folks in the audio community are not content to pick from the variety of monolithic opamps available. They prefer to roll their own by placing individual components on a printed circuit board. They are big and expensive and usually sport “worse” specs than even low-end monolithics, but they have certain benefits like being proprietary and high-margin entirely customizable to suit the designer’s ears.
Do different opamps sound different?
There's much debate over how significant the audible differences are between different opamps. Some people (often musicians) claim to hear a "night and day" difference when they swap the opamps in an old piece of gear. Some people (often engineers) claim there's no difference at all.
I've spent a good bit of time listening to different opamps in a controlled environment and my take is that audible differences between opamps are vanishingly small when they're operated in their linear range. When you start to ask them to do stuff beyond what on their spec sheets, all bets are off.
So, swapping an opamp that's just a buffer between two stages in a compressor? Probably not gonna hear much of a difference. Swapping opamps in a circuit with a hot input signal and excessive amounts of gain? You will hear major differences, akin to using two different mic preamps to apply a lot of gain.
Did I miss anything?
I hope that helped clarify things a bit! Is anything unclear? Let me know in the comments if you have any other questions about opamps I can answer.
How Good Are Your Ears? Identify This Mystery Colour July 15, 2015 18:01
We're wrapping up production of a new Colour with a mystery collaborator. The Colour is based on a vintage piece of gear. Can you guess what it's based on from one audio sample?
Comment with your guess below. The first person to guess correctly will get one for free when it launches. (But we won't reveal who got it right until the launch.)
"Distortastudio" Colour Design Pt. 3 July 09, 2015 16:39
Back in April, we began designing a new Colour module to emulate the sound of an old Tascam 4-track tape machine.
When we left off in the second blog post, we had sketched out the basic circuit for the bright, crunchy mixer-section distortion and were all set to begin work on the dark, gooshy tape section.
Well, the universe had other plans. That week I heard from Jens at Eisen Audio, who designed the TM79 Colour. After reading my blog post, Jens revealed that he'd been working on a new cassette-inspired Colour that was "obvious and shitty and musically relevant and reminiscent—at least in some dimensions—of an Aiwa Walkman I had in the 90s."
Now there's a lot about this business I don't know. But I do know the world probably doesn't need two cassette Colours coming out around the same time. So Jens and I decided to join forces to make two separate-but-complementary Colours: a 4-track mixer distortion (Distortastudio) and a cassette tape and electronics emulator (currently named "LOWSPEED"). This way users can have the mixer and tape distortions on their own, or combine them for the ultimate 4-track Colour.
Our Colour, the Distortastudio, is dangerously close to being done. Parts and PCBs are on their way—we just need to put them into kits and write the documentation.
The final Distortastudio prototype
Jens has some more tweaking to do on his Colour, but initial listening tests have been really promising.
The most recent "LOWSPEED" prototype
Thanks for following along, and please stay tuned for the upcoming launch of the Distortastudio kit!
Introducing the CP5 Mic Preamp Kit June 18, 2015 16:50
I'm really excited to share the news with you of our upcoming preamp kit. But let me get to your question first:
"Why release a new mic preamp in 2015? Isn't there enough water in the ocean?"
Fair question, and well asked!
I've wanted DIYRE to make a mic preamp since day one. And I'm not the only one—"diy preamp" is #1 search query that brings people to our site. It's also the #1 thing people type into our store's search bar.
But despite all the interest, we've been steadily disappointing preamp seekers since 2010. To be honest, I've just never felt we could bring something special enough to the table.
We don't do vintage clones; our friends at ClassicAPI and Hairball Audio do a great job of that. And we're not the kind of evil geniuses who cook up radical new designs; Eisen Audio and Louder Than Liftoff do that better than we could hope to.
What we do is make DIY kits that are radically affordable, exhaustively documented, and constantly improved via the feedback of a community of engaged DIYers.
I've never seen how we could use those qualities to produce a preamp capable of distinguishing itself... until now.
Introducing the CP5
The CP5 will be an ultra-transparent, single-channel preamp for the 500-series in beginner-friendly DIY kit form. And it will have three big features:
- The most affordable 500-series mic preamp on the market, bar none
- Stepped controls for quick recall and stereo operation
- Colour stage with separate gain control
High-End Parts, Low-End Cost
We're all about making great gear radically affordable, and the CP5 will be our biggest coup yet. We've managed to come in well below the bottom of the market on cost while matching the high end on quality. I anticipate that the price will cause people to underestimate the CP5's quality. But don't be fooled—there will be nothing "low-end" about the CP5. We're using exclusively best-in-class parts: WIMA film capacitors, THAT Corp. ICs, custom stepped controls, US-made metalwork, gold-plated circuit boards.
We'll reveal the exact pricing on 7/1, but the kit price will be about half that of the cheapest 500-series preamp available from retailers.
The most recent CP5 prototype
Transparent or Coloured
But what really makes the CP5 special is the addition of a Colour stage. With the ability to choose your tone from among the ever-growing variety of Colours, the Colour stage makes the CP5 the most versatile preamp ever. Add some transformer goosh with the CTX, tape saturation with the TM79, tube harmonics with the Pentode, etc.
The Colour control allows you to dial in the amount of coloration without affecting the overall gain of the preamp.
The Design Team
For the design we teamed up with GKL, a team of brothers Eric and Peter Gaskell who have quite literally dedicated their lives to the nitty gritty of audio electronics. They teach and research audio at McGill University. Eric was involved with the design of the AEA R84 ribbon mic and published an AES paper on the psycho-acoustics of op-amp distortion.
They've studied with George Massenberg (once they took me to a party where George had cooked a Mexican feast—yes, his golden touch extends to nachos). In short, mic preamps are in their DNA and I'm confident the electronic design couldn't be in better hands.
Pre-Order Beginning 7/1
Just like we did with the Colour platform, we'll be launching the CP5 with a pre-order next month. Final pricing and production photos shall be revealed then. Stay tuned here or join our newsletter to be notified the moment the pre-order is open.
Questions? Comments? Please let us know below!
Pre-Order the New Active DI kit from Bumblebee Audio May 04, 2015 15:07
I first met Latvian designer Artur Fisher in 2010, when I launched DIYRE as a project directory. Since then, I've followed along as Artur has painstakingly built a small line of world-class recording equipment: first the RE-254 ribbon transducer, then the RM-5 ribbon mic, then the Bp-P1 inline mic preamp. The common thread between Artur's designs is custom parts and circuitry in service of performance. Where most designers are content to choose from among pre-rolled solutions, Artur rolls his own. Everything in the RM-5 mic is custom—down to the XLR jack—and the Bp-P1 uses fully discrete circuitry. You can understand why he only releases a new product every couple years.
So I'm really excited to announce that Artur's newest kit, The Bb-D1 Spark DI, is now available for pre-order. The Spark is an active, phantom-powered, transformer-balanced, direct input box. And in keeping with Artur's tradition of no-compromise design, every component has been chosen or customized over several years of development. The circuitry is completely discrete, the transformer and chassis are custom, and he's even built in power supply conditioning to ensure optimal performance, no matter the quality of your phantom power source.
For a limited time, Artur is offering DIY kits for the Spark at a deeply discounted price. I think this is a phenomenal deal: a) you get a truly world-class, boutique DI box for €99 and b) you get to support a talented designer who's dedicated to the DIY community. Grab the pre-order here before May 15th:
Here are the pre-order details:
- €99 for full kit (~$110 USD)
- Price after pre-order will be €129
- The last day to pre-order is May 15, 2015
- Use the code "LOVEFET" in the shopping cart (before checkout) to claim the discount
- Kit includes everything needed to build the DI, including screws and wire
- Complete, step-by-step assembly guide
If you have any questions about the Bb-D1 or the pre-order, please leave them in the comments below and Artur or I will do our best to answer them.
-Peterson Goodwyn, DIYRE
"Distortastudio" Colour Design Pt. 2 April 17, 2015 17:58
In last week's post we identified three distinct distortion sounds of the Portastudio: mixer-only, tape NORMAL speed, and tape HIGH speed. We asked you which sound you preferred and the consensus was that... there was no consensus and we should try to incorporate options for all three sounds.
So we're going to try our best to make a Colour that incorporates all three tones of the Portastudio. Hopefully we'll be able to fit all three modes onto the board along with jumpers to set your preference. Since all three modes include the mixer-only circuitry, we're starting our circuit R&D there.
The Portastudio Channel Strip
After confirming that it did indeed sound awesome, we began our investigation into the mixer section by opening up the 464. It's actually a pretty complex piece of gear; there's a forest of film and electrolytic capacitors, a few big noise-reduction chips, the motorized parts for the cassette deck, etc.
The mixer itself however, is fairly simple. The green board you see here is the underside of the mixer. It consists almost entirely of opamps and resistors (including potentiometers) with a few caps in the filter section. On top of that is the a sub-board for the balanced input circuitry, but since that comes before the stages we're distorting we can ignore it.
It's simple enough, in fact, to identify the major circuit blocks without doing any reverse engineering. An adjustable gain stage (Trim) is followed by a couple adjustable filters (EQ section), followed by a passive fader (channel fader) followed by a buffer, followed by a passive pan control.
Just in case there was something fancy going on, we ordered the service manual from eBay so we could look at the schematic on paper. As suspected, besides the discrete balanced input circuitry (which we're ignoring), the 464's channels strip is basically a series of basic opamp circuits. And from what we can tell from other schematics available online, every model and revision of the Portastudio uses roughly the same approach for the channel strip.
So, nothing fancy about the circuitry we're distorting here. Just some textbook opamp designs being driven really hard.
Operational amplifiers (opamps) are relatively small, cheap integrated circuits that provide different amounts of amplification depending on the circuitry you place around them. They're kind of like cheat codes for electronics: with just an opamp and a couple resistors you can get a very stable, low-noise gain stage that would take 10+ parts and a lot more design time to make with discrete components.
So how do we make an opamp distort? Opamps have a hard limit to the amplitude of signal they can pass. This limit is set by the voltage of the power supply voltages (also called "rails") and the opamp's own ability to "swing" within those rails. For example, the power supply of the Colour 500 Palette is the 500-series standard of +/-16VDC, or 32 volts total. Most opamps can swing within roughly 2 volts of each power rail for a maximum peak-to-peak voltage of 28V, or +22dBu (for a point of reference, that's 18dB above professional line level). That's our hard limit; nothing will be amplified past 28V.
Let's see what happens if we feed an opamp a signal that's 3V peak-to-peak and set the opamp to a gain of 10x. The opamp will try to drive those 3V peaks up to 30V. But it can't! Anything that should have been above 28V gets "rounded down" to 28V, while everything below 28V gets amplified normally. In other words: clipping! In other words: distortion!
Like microphones or preamps, opamps come in a bewildering variety: low-noise, low-power, high-current, wide-bandwidth, etc. And also like microphones or preamps, there's much debate over how significant the audible differences are between them. Some people (often musicians) claim to hear a "night and day" difference when they swap the opamps in an old piece of gear. Some people (often engineers) claim there's no difference at all.
I've spent a good bit of time listening to different opamps in a controlled environment and my take is that audible differences between opamps are vanishingly small when they're operated in their linear range. When you start to ask them to do stuff beyond what on their spec sheets, all bets are off.
So, swapping an opamp that's just a buffer between two stages in a compressor? Probably not gonna hear much of a difference. Swapping opamps in a circuit with a hot input signal and excessive amounts of gain? Check out the samples below.
The first chip we tried in our mixer-section prototype was the trusty TL071. At "normal" signal levels, it sounded very similar to the actual TASCAM channel strip. But at even moderate distortion levels there was a gross crackling on the transient hits (this was partially due to our using only one opamp for this prototype, but it was nonetheless more pronounced with this chip).
Then we swapped the crackly TL071 for a UA741, truly one of the crappiest, most obsolete opamps (on paper) still manufactured. Voila! The crackle was gone. But so was a lot of the high end and, wow, that hiss!
Finally, we tried the obvious: the NJM4565, the same opamp used in our origninal 464. Now our prototypes started sounding almost identical to the original channel strip. (The crackling was still there--better than the TL071 but worse than the UA741. But we figured out how to eliminate the crackling completely by using two opamps and described above.)
So next time someone tries to tell you that all opamps sound alike, tell them they're just not driving them hard enough!
The Circuit So Far
To get maximum clipping and character out of our opamp, we're both a) applying copious gain to get the signal into clipping territory and b) regulating the power supply voltages down to lower the clipping point of the opamps. Here's our circuit so far:
You'll notice that there are two opamp stages. We did this because we found that, while we could drive a single opamp into clipping, the gain required came with too much noise and other nasty, non-musical artifacts. By spreading the same amount of gain across two opamps, we get the same amount of distortion with far less noise.
The ICs you see in series with the power supply rails are fixed regulators, which simply knock the supply voltage down from +/-16V to +/-9V. This gets us in the clipping range of our 464 Portastudio, which is powered with +/-10V.
Roll the Tape
We feel we're pretty dang close with the mixer section. The current circuit with the NJM4565 opamp sounds very close to our 464 channel strip. All that's left is to nail down the exact internal gain settings so that users will have a wide range of tones with the Colour Palette's Saturation knob. Next week, we're onto the "tape" half of the circuit. We're aiming to incorporate the both dark, grainy tone of the normal-speed mode, and the brighter, but more rounded (than the mixer) sound of the high-speed mode. We just brainstorming right now, so if you have any circuit ideas we're all ears!
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