That being said I really do like their design. The spring reverb in my guitar amp sounds great but does not have tone control and get overly bright with a single coil guitar.
You nailed it. An old friend of mine wanted to get me to start a boutique stompbox business with him since I was a bit more technically inclined (ie, I could read a schematic and solder).
I just couldn't do it. It's too much. I love messing around with that stuff, but I could not be bothered to wade through all the cruft and politics of that. I just like music.
BTW, you should be able to cut the highs well enough with your guitar tone knob or EQ on the amp. You can get something like: https://www.youtube.com/watch?v=ZIU0RMV_II8 (Fender Stratocaster -> Fender Showman w/ spring reverb. I was saddened to find out just now that he passed away this spring)
He constitutionally can't - he is the definition of uncompromising. I don't play guitar, but I want to buy one of his amps because of how it is made. Take a look at this:
And every one is exactly the same because he writes and then follows explicit directions for making every connection.
Doesn't even look like he has any dealers up here—but I was able to find a Premier Guitar demo video from last year's NAMM:
Didn't realize he was from Tone King previously.
I haven't opened my Orange Tremlord. I expect it to look nice, but not nice like this. The Orange Tremlord has a very nice reverb with a real reverb tank. And is imho a very nice amp:
- cool, but they don’t work
Even in the crossover analog/digital world (chase bliss being one of the biggest examples) there’s some cool and fairly new stuff happening, like fully analog circuits that are completely controllable over midi or other lfo type control signals.
In the fully analog realm, yeah, there is a bit of what you could call cork sniffing, dressing up the same circuits that have been used for the past 30 years in a boutique package and selling it for twice as much.
Many of the "revered" circuits perform very differently due to component variations (e.g hFE or Vgs), and also have substantial drift in performance with temperature. So even different units from same manufacturer will sound different.
Senses are a funny thing, including hearing. When playing, I can easily hear subtleties like whether the Jazz III pick I'm using is made of Ultex or nylon. But I can listen to recordings I made, and not even be sure which guitar I played. (I've been playing for 35 years and recorded many albums.)
So these subtleties and details are real, they're audible (in certain circumstances), and the mockers generally don't know what they're talking about. But they're also easy to overfocus on.
I wonder how much of that is component tolerances. If you have two identical circuits with +-1% in resistors and +-5% in caps, how different would these sound with the components in each circuit at opposite ends of the tolerance range?
Which is why Behringer is good enough for most people.
Reasons to avoid
- poor build quality
- lawsuit-happy (suing forum posters, forums)
- flagrant low-quality ripoffs of other brands gear
- buys up good brands and makes them cheap
Want to help rookies? Steer them away from the $500 tube-screamer or dual-blues-breaker clones that you need to be on a waiting list for 2 years before you can even buy one! :)
However, the BBD sound can be almost easily emulated in digital. On the "wet" return line of a digital delay put a N dB low Q high pass filter before say 200 Hz, a N dB low Q low pass filter above 6-8 KHz, add some gentle clipping and some noise, compress overall wet dynamics and feed the return back to the delay input. Then adjust delays, feedback and filters amount to make it sound like a real one. The trick is simulating the progressive way in which the sound when going through a BBD device will lose in fidelity while getting more distorted and noisy, so we filter and add those artifacts inside the feedback path.
> The finished music track is recorded simultaneously to two matching tape machines, then replayed with both decks in sync. The playback-head output from the two recorders is mixed to a third recorder. The engineer slows down one recorder by lightly pressing a finger on the flange (rim) of one of the playout reels. The "drainpipe" or subtle "swoosh" 'flange flango' effect "sweeps" in one direction, and the playback of that recorder remains slightly behind the other when the finger is removed. By pressing a finger on the flange of the other deck, the effect sweeps back in the other direction as the decks progress towards being in sync.
The story goes John Lennon coined the name. Ken Townsend at Abbey Road had worked out a way to mimic vocal double tracking without actually recording the singer twice using this technique. They called it Artificial Double Tracking, or ADT. But Lennon would just call it "Ken's Flanger" and the term stuck.
If you don't clock it at all, you'll just be sending out the constant voltage equal to whatever your first capacitor has charged, which results in no audible sound. You can adjust your clock speed and have it play back at speeds slower or faster than the incoming rate, with the pitch changing too, just like slowing a record or tape machine.
The output isn't really discrete though, only the internals, and even then like you say, even logic transitions aren't actually instant (but at a rate so much faster than audio we ignore it). The BBDs use filters on both the input and output that remove the very high-end of the audio range (a lowpass filter).
These filters will be made to match or be lower than the current clock rate . One before going into the BBD removes any frequencies too high to be stored accurately (an anti-aliasing filter), and then one after the BBD removes any high frequencies which were created by the near-instant shapes of the capacitors discharging (a reconstruction filter). These means your output audio is "continuous", but also has zero frequency content above your filter cutoff (okay there's a teeny amount of information, nothing in reality is perfect).
To create the delay effect you take a long line of them (usually 512-4096) and the length of your delay is equal to the number of "buckets" times your clock rate. If you had a near infinite amount of buckets and ran them so fast there was no loss in fidelity from the sampling rate, you'd basically have a tape delay. The fun thing about BBDs though is that they don't always pass on their charge to the next bucket perfectly. So longer delay lines mean introducing more errors accumulating as your audio passes through the delay line. A near-infinite BBD would turn into audio mush. Though you can still get a very high number of buckets and a very high clock rate for near digital fidelity, but at that point it's getting more expensive than just using a digital effect, and you're ditching the unique charm of a nice effect.
Here are some more detailed explanations of a bbd   The posts by Richard Crowley in  are particularly cool to the HN audience I'd hope.
> Curiously enough, BBD were a pre-cursor to EEPROM, DRAM, and FLASH RAM. Microscopic capacitors are used to store the data. In the case of DRAM, the capacitors aren't terribly good, and they must be "reminded" (refreshed) on a regular basis (many times per second). Very similar to BBD.
 okay technically 1/2 the clock rate because of nyquist-shannon, I just mean it will be linked to the sampling rate.
There are a lot of digital effects out there. They digitize the input signal, do some calculations and again convert to analog for the output. A lot of reverb units are like that. Some amplifiers have digital reverb builtin.
1) Split analog signal
2) Pass through one side to output (probably with a signal boost)
3) Take the other side & digitize
4) Apply effect to digital signal
5) Convert output digital signal to analgo
6) Mix into the output of step 2
I find this to be a very audible difference.
Elk Echo Machine v3 (EM-3) from 1970
If the only "digital" part of the circuit is the clock then couldn't you replace that with a quartz oscillator that you can wire an analog modulator for in order to get the switch timing for the capacitors banking the sound? Am I oversimplifying this?
If I'm at all correct than I would also say I don't know of any product that does so since it sort of sounds like it would be going full analog for the sake of going full analog and wouldn't affect the tone AFAIU but sounds like there's no technical reason it couldn't be done.
Lots of heads and carefully tuned feedback (possibly also across different heads adding phase changes) could help in that direction. It would sound interesting, though the resulting contraption would also be big, clunky and expensive.
Not exactly portable, but definitely worth the build.