Interviews

Behind the Gear with Tony Agnello of Eventide

On Temperance, a musical Reverb

Formed in 1971, Eventide soon became one of the first companies to build digital delays and other future-shaping audio processers. Their H910 Harmonizer may have changed the sound of records forever. Their SP 2016 was the world’s first digital audio processor, introduced the concept of the plug-in, and its reverb algorithms remain popular to this day. I heard about Eventide's Temperance plug-in, an interesting new concept in reverbs, so Tony and I got on a call so I could learn more.

I interviewed [Eventide co-founder] Richard Factor [Tape Op #130] in 2018, but when we first met, ages ago, you were doing a rack mounted plug-in with Joe Waltz.

That’s Manifold Labs and Plugzilla. But “ages ago” fits me. I started with Eventide in ‘73.

With Eventide being one of the early makers of digital audio hardware, did you ever imagine that it would all just be in a computer?

Sure. Yeah.

Once the H910 Harmonizer came out, you were basically writing code.

To my mind, all of this is obvious. So, when computers came out, it was clear that everything someday could be in the computer. In fact, one of the first products we made was a real time one-third octave spectrum analyzer for the [Commodore] PET computer, IBM PCs, and for [Apple] Macs. You could plug a card into the computer and the computer became a test instrument. No one had done that before. And then when DSP [Digital Signal Processing] chips happened, I founded a company called Ariel and we were putting DSP cards into PCs so people could write software for them. But all of this was foretold. When I was in graduate school at CUNY, I was reading Bell Labs' journals [Bell Labs Technical Journal] and guys like [Manfred] Schroeder were saying that once you're in ones and zeros [digitized], you can do all manner of stuff. It was just a matter of the technology advancing the point where we could do things in real time at audio rate. Computers existed, but the challenge was for full spectrum audio. For 20 kHz, you want to sample at least at 50 kHz. That means every 20 microseconds, you've got a slot to do something. When Richard [Factor] designed the first shift register-based delay line, nothing was happening in the digital world. All you could do at that rate was move the bits along the chain. When I designed the first Harmonizer, RAM [Random Access Memory] had just come out, with 4 kilobytes of RAM. And all I could do at a 50 kHz or so audio sampling rate, was come up with one address to write and another address to read – two operations every sample period. When we got to the late '70s, we were still not at the point where a general purpose CPU [Central Processing Unit] could do anything at audio rate, but there were special purpose processors. There were these things called bit slice [processors]. But it came to a point where we could do about a hundred things every sample rate. And that's the point where you saw the first digital reverbs. If you have a hundred operations, real time at audio rate, you can build a bunch of delays and feedback around them – feedback networks – and simulate what a room sounds like.

Right. All the acoustics of a room.

We call that algorithmic reverb. Eventide and I did it with the SP 2016 [Signal Processor]. With the 2016 I went a little nuts, because I didn't want to just do reverb. I figured that I could do a hundred things. I could do band delays. I could do a vocoder. It was more of a general purpose processor. EMT did a digital reverb [EMT 250], the Lexicon 224, Quantec [Room Simulator] – all of those were algorithmic reverbs.

Right. They're not sampling spaces.

They're not modeling a given space; they're just giving you the sense of a real room. And what did that turn out to mean? If you built a Schroeder reverb, it doesn't sound like a room. Even if you get the early reflections right and have an FIR [Finite Impulse Response filter] for the early reflections, it still doesn't sound like a real room because he just had a bunch of delays with feedback in parallel. The echo density builds up linearly. But what happens in the real room is the echo density builds up exponentially. The way to make that happen in an algorithmic reverb is you want to interconnect delays. Today that's called an FDN [Feedback Delay Network]. I built it for the 2016, before it was called a feedback delay network, but that's what it was. I thought it was a matrix.

Sure. Technically it is.

So, now we get to the 1990s, and suddenly you could do a thousand things every sample period. Now you could excite a room, capture an impulse response of the room, and do convolution reverbs. But the reality is that we knew in the ‘60s, maybe even in the ‘50s, that you could measure a room, but there was never the processing power to allow us to implement convolution in real time. When convolution happened, I went, “Yeah, no surprise.” Was I interested? Not at all. Because with convolution reverb, you measure the room from here to here but that's not really the real room. Okay, it's a room; what can I do with that? I could EQ it, but I was not interested. I was waiting for the day that, at audio rate, I could do tens of thousands of things per sample. There's another way to model the room, something that was written about in the literature years ago, and it's called modal. A room has modes; a room treats every frequency differently.

Oh yeah. In different places and ways all around the room.

Now, instead of having the static impulse response, you now have thousands of modes. If you can control each of those modes, you can control its amplitude, its level, and its center frequency for each of those modes. Here's one of the keys: If you can control each mode's decay differently, and you can control the phase or the delay between each of those modes, you have this wonderful, magical world that you can f#$% with sound in ways never before possible. I've been waiting for that. I can tell you that that's what we're doing. Why was this interesting to me? I play guitar, I've studied cello, now I’m downsizing and playing ukulele and I'm still in a band. When I was studying cello in the mid ‘80s, my cello teacher, James Hoffman in Brooklyn, introduced me to a Russian immigrant named Isaak Vigdorchik. Isaak was brilliant: a luthier and played violin in the symphony orchestra, but he was also a scientist. He had emigrated from the Soviet Union, but in grad school he got his hands on a Cremona violin, and he told me that if he tapped the front or back plate up and down the violin he could hear the notes of the Western musical scale. It was his contention that the makers either lacquered or sanded the wood so it would ring out at the notes of scale. He wrote a book, The Acoustical Systems of Violins of Stradivarius and Other Cremona Makers. I said, “Isaak, I don't believe you.” At the time, I had just designed the world's first audio rate FFT [Fast Fourier Transform] analyzer. Isaak was connected with someone at a museum; we were going to work together, get our hands on a violin, and see if what he was claiming was true – get a Stradivarius or one of those violins. He would say, “Anthony, the notes are in the wood.” That made me think that if I had a modal reverb I could put the notes in the room. I could modify the room so the room would ring out at the notes I wanted – or not ring out at certain notes. Back in the day, when people had echo chambers or a bad sounding room that would ring out, you'd want to damp it to get that ringing down. I'm putting the ringing back in!

I've got an EMT 140 plate reverb, and I EQ going in because of what rings out and might become too prevalent.

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