Article doesn't mention one of the more interesting (to me) aspects which was how feedback was avoided. The solution is elegant: each vocal microphone is doubled, meaning there are two at each position. The phase is inverted on one of them, the singer sings into only one, and both are sent to the speakers via their channel's amp.
The effect of that setup is that only the difference between the two microphones is amplified; common signal in both (i.e. the sound coming out of the speakers) is nulled out, but the difference signal (the voice) makes it through. It apparently wasn't quite perfect but was absolutely a lot better than wailing feedback.
The thing that made it sound so good was that any given speaker only reproduces a single source, but the article touches on that. The mic arrangement I described is simply what makes it possible.
While it's true that they did that and why, I'd ultimately chalk it up as more of a flaw than a feature. Vocals never sounded great on wall of sound shows because they could never sing perfectly into one mic. This can be confirmed by listentng to soundboard tapes of the shows, and comparing them with ones a year or two either side - the full on Wall was only used for about a year.
While the WoS laid much of the groundwork for how modern PAs are designed and operated, it was more of a white elephant than anything, and many of it's actual ideas were discarded. It was totally impractical to tour with and they lost money doing so. The only real technical legacy it has is of using coherent phased line arrays.
Really it's whole reason for existence (getting a coherent, in phase, non-canceled signal at an extended distance from the stage) isn't even relevant, as these days secondary speaker arrays with delay lines (to sync them perfectly with the mains) is almost childs play. Literally plug and play. Modern PAs can self-tune the whole system just from playing a short burst of white noise through the system, and listening for the response.
"Modern PAs can self-tune the whole system just from playing a short burst of white noise through the system, and listening for the response"
A technology which was developed by/with the Grateful Dead, by the way.
From what I understand they essentially financed the modern PA industry by spending a ridiculous amount of money on sound equipment. People don't realize how much money they made - they were pretty much the top grossing tour band for about 15-20 years, playing about 90-100 shows a year. And they could (and did) use those shows to experiment with sound in a way that probably no other band has done since.
I haven't watched any D&C shows, but I expect their sound quality was just as good, if not better, than the Dead's.
(I assume you're aware, but for the larger audience)... the grateful released an album "Two From the Vault" which was a soundboard recording... but the original soundboard had huge phase cancellation errors due to microphone placement. To recover it, some 20+ years later, with digital tech, the sound engineers could recover the original signal using some clever FFT and phasing very similar to what you describe modern secondary arrays use to self-tune.
> It was totally impractical to tour with and they lost money doing so.
The whole system travelled in two articulated lorries; but it took so long to set up that eventually they used four lorries, and two systems. One set would be for tonight's gig, the other would be on the road to tomorrow's gig, or starting to set up.
I think the mikes for the vocals were three mikes, not two (you can see triple mikes in The Grateful Dead Movie). I can't remember the rationale for using three mikes. The movie also shows the WoS being set up, under the direction of Ramrod, with Garcia gently urging him to get the speaker stacks hauled up higher, because that would be "really cool".
Well, yeah, compared to today it's not great but no one had tried anything like that before. They delayed the sound to distant speakers with tape delay. It's cool as shit and was the groundwork for how we do things today.
It's like saying relay computers were dumb... Boolean logic was new and no one had ever attempted stuff like that before.
This is very similar to how noise cancellation works on cell phones. The secondary mic is typically on the back of the phone and picks up the ambient noise to be subtracted from the primary mic’s signal.
Here's a fun one. I'm involved in maintaining the audio system for an auditorium used by a non-profit. After a flood and remodel, including replacing some audio components (like microphones), it was observed that the microphone on the main podium always had a 60 Hz hum. The hum depended on where the microphone was facing. Sometimes it was there, sometimes it was not.
Being a non-profit facility, there are no fancy DSPs to notch out the hum or anything like that, so more creative solutions were investigated. It was determined through dumb experimentation that orienting an identical microphone 180 degrees to the one with the hum and setting the gain similarly would nearly eliminate the hum.
Eventually, the working theory became that a relatively new large pad transformer installed across the street was being picked up by the microphones.
Orienting one microphone 180 degrees from the other caused the hum to be picked up out of phase from the main, and thus could be mixed in to cancel out the main mic hum.
Ultimately the real solution was simply buying better microphones, but there was a period of some months while a microphone sat off stage, pointed backwards.
The history of balanced lines (common mode rejection, differential pair, etc.) is fascinating. Apparently the first twists (as in "twisted pair" to pick up external interference as similarly as possible on each conductor) were achieved not within a bundled cable, but between utility poles. Every two spans would constitute a full twist, with two single wires alternating from left to right on the cross member.
But as for the acoustic implementation, even that has a long history. The Dead borrowed the idea from fighter pilot headsets, the only difference being that pilots were contending with a noisy cockpit rather than feedback. Same general idea that the unwanted sound hits both mics somewhat equally while the voice hits both mics somewhat unequally.
Yep, you sing into one but not the other, so there's a big difference in the vocal signal, whereas spill from the speakers is going to hit both mics pretty well evenly.
Yeah one microphone was behind the other, though I was under the impression it was half a wavelength behind and thus created something quite akin to modern active noise-cancellation?
For the most part, each performer plugged into the "Wall of Sound" had their own vertical 1xN stack of drivers.
3 drivers in a vertical line away will have less distortion than 3 drivers in a horizontal array; the horizontal drivers will suffer from comb filtering for listeners who are not located dead center at the middle of the array. (This of course assumes your audience is dispersed horizontally as opposed to floating randomly in space)
Modern home loudspeakers hew to this philosophy as well to an extent. As opposed to big "monkey coffin" 70s speakers with a random array of drivers sprayed across the front of the speaker[1], modern tower speakers have a vertical array of 2 or more drivers whose centers are aligned in a vertical line[2].
Another cool thing you can do with line arrays is beam steering, you can direct the sound to a certain extent. There are tradeoffs, but I always thought that was pretty damn magical.
For those wanting to listen to free legal taped audio of Dead shows, head over to https://relisten.net/grateful-dead or install Relisten app for iOS. All fan supported and open source.
Hey! I created this website and have been maintaining it along with my friend Alec for the last decade. Fun seeing it pop up here on HN. Thanks for sharing!
thank you SO MUCH fam! relisten is such a great repository and i have shared it plenty. thank you too block_dagger for phish.in, much the same.. much love!
Th' Dead not only allowed taping, they encouraged it: at the shows I attended, there was invariably a small grove of microphones set up near the soundboard, in the middle of the audience.
I'd love to hear more about your experience and observations of taping at the shows.
Not only did GD (and particularly Jerry Garcia and John Perry Barlow) eschew the 'intellectual property' model of music, their thought-leadership has lived on to become much of what we today consider fundamental internet technology and methodology.
Early decentralized crypto-economics, peer-to-peer file sharing, and the founding of the Electronic Frontier Foundation were all developed by some combination of deadheads and musicians and tapers, particularly on an internet service called The WELL.
The Green Pill Podcast had an entire episode exploring the bluegrass roots of blockchain technology; I was humbled / psyched to be a guest and play several of my songs, as well as some traditionals that GD also played. It's here if you're interested: https://www.youtube.com/watch?v=Y3s9Fu4yu7o&t=2898s
It's almost impossible to talk about The Dead and not talk about LSD.
Owsley (wall of sound engineer) was one of the original major LSD manufactures... and a dead show was always where you went to score ACID if you lived on the east coast. This remained true well into the 90's.
I'm going to guess that all of those early internet pioneers that you mentioned also have fond stories of LSD.
The Dead, Bill graham, hells angles, peoples temple, Patty Hearst.... There is a continuum of culture that spills out of San Francisco to this very day.
I was a bit too young to get too many shows (I saw one in 1993 and a couple of JGB after that), but I have extremely fond memories of going to the local head shop in my town that had a massive tape collection. You would bring 6 blank tapes and pick 5 from their library that they would copy for you and keep the other blank as payment.
Dave Rat, of Rat Sound/RHCP/Bassnectar/etc fame, has some very interesting takes on the wall of sound idea using modern equipment.
The core of it is that speakers are bad at polyphony, so if you can avoid it, you can produce something that sounds more natural to human ears, and do so in a larger area. The way to avoid it is more speakers, more stacks/arrays/etc. You don't necessarily need an array per instrument, because modern loudspeaker arrays are indeed much better than they used to be, and modern loudspeaker processing fixes a multitude of problems.
Another advantage of individual speakers for each instrument is that it allows positioning them without the phase cancellation artifacts you get with a stereo setup. It's obvious with some sounds, e.g. try playing some mono pink noise on stereo speakers and move your head. Then try again after hard panning the audio to only one of the speakers. This is why adding a real physical center channel makes dialogue clearer in movies.
The center channel also helps by allowing the listener to localize the dialogue differently than the music/FX. When the dialogue comes from a different point in space, it doesn't need to be louder than the music/FX for sufficient intelligibility. When downmixed to stereo or mono or any other non-center-having configuration, the dialog must be mixed well above the music/FX to avoid being masked by it. Sometimes a downmix occurs without this step being given enough consideration, resulting in complaints about the mix, when the original 5.1 (or whatever) mix was perfect but an engineer wasn't involved for downmixing.
This is also what Dave Rat is on about, in addition to IM and combing concerns, when explaining why a speaker per instrument sounds better than a mix: not just that it's a single point source, not just that it's doing one single job, but that all the sounds come from discrete points nowhere near each other. Just like an acoustic band using no reinforcement.
I wonder how do they deal with the phasing over so many speakers? Wouldn't an array of speakers require you to do something about that? Specially so if you are stuck with 70s tech.
The core of it is that speakers are bad at polyphony
You can look at intermodulation measurements for some popular and affordable home speakers here. The TL;DR is yeah, you're going to go from something like -60dB distortion to -40dB when doing synthetic multitone tests, but I would not remotely characterize this as being "bad" at polyphony nor the primary reason for the Wall of Sounds primordial "one vertical speaker array per instrument" design.
The way to avoid it is more speakers, more stacks/arrays/etc.
If we're talking about his remarks here I'd characterize his take as "more vertical arrays" and not just "more speakers." I realize it's a bit pedantic of me, but some people think that simply adding more speakers equals more betterer sound and it's quite far from the case. In general, multiple loudspeakers arrayed horizontally (this includes MTM center channels in home theater setups) lead to comb filtering.
I'm sure you know that, just clarifying for others.
If he has written about this elsewhere I'd love to read more!
I'm simplifying things significantly because I don't expect most HN'ers to know the ins and outs of pro sound.
Dave Rat has a good youtube channel, he also has some sort of insider subscription thing that I've never bothered with. I wouldn't say his ideas necessarily translate to every situation, but, he really presents these ideas as tools to use in a toolbox, not as gospel.
One interesting thing is that, we generally view comb filtering as universally 'bad', when, in reality, our ears do an excellent job of sorting out comb filtering when it comes to natural sounds. In fact, comb filtering is how we can locate a sound in 3d space with only two reference points (and, if you try, only one reference point moved around a little). That's remarkable, and points back to how speakers comb filter instead of mere comb filtering itself.
In practice, say you have a rock band, and your sound system has two arrays with subs spaced 40ft apart. Now, you're going to get a less than ideal pattern from that in the ranges where the bass guitar and kick drum live. How do you fix it? The answer is fairly simple, you simply run bass/kick in stereo, then, you delay the bass on one side by a little, kick on the opposite side just a little, then add some kind of EQ difference to the delayed side of each, then play with the delays until it sounds right.
Why does that work? Or does it really work? It's odd, because the math says "no no, it'll sound bad", the reality is, it can tighten up that comb to the point you don't really hear it and you can get good bass coverage out of a fairly poor system design.
I remember chatting with a sound technician at a concert once and he told me that putting amplification in front of the performers only started happening in the late 60's (ish). Before that musicians were actually subjected to insane DB's by standing only a few meters in front of the amplification. (Don't take this is as fact, but this diagram suggests that he was correct)
Some techniques require this. Ted Nugent wouldn't have gotten the crazy howling feedback out of his semi-hollowbody Gibson Byrdland had he not been standing directly in front a pair of Fender Super Twins pushing 4 15" drivers.
If it doesn't make your pants flap in the breeze, turn it up!
I believe most musicians these days achieve this using a nearby monitor speaker, for example Trey Anastasio from Phish, although I believe he may have adopted newer technology (see https://treysguitarrig.com/2023/08/31/2023-summer/ for more details). He could sustain notes for a long time with his custom hollowbody (like, minutes at a time).
Readit News
The effect of that setup is that only the difference between the two microphones is amplified; common signal in both (i.e. the sound coming out of the speakers) is nulled out, but the difference signal (the voice) makes it through. It apparently wasn't quite perfect but was absolutely a lot better than wailing feedback.
The thing that made it sound so good was that any given speaker only reproduces a single source, but the article touches on that. The mic arrangement I described is simply what makes it possible.
While the WoS laid much of the groundwork for how modern PAs are designed and operated, it was more of a white elephant than anything, and many of it's actual ideas were discarded. It was totally impractical to tour with and they lost money doing so. The only real technical legacy it has is of using coherent phased line arrays.
Really it's whole reason for existence (getting a coherent, in phase, non-canceled signal at an extended distance from the stage) isn't even relevant, as these days secondary speaker arrays with delay lines (to sync them perfectly with the mains) is almost childs play. Literally plug and play. Modern PAs can self-tune the whole system just from playing a short burst of white noise through the system, and listening for the response.
A technology which was developed by/with the Grateful Dead, by the way.
From what I understand they essentially financed the modern PA industry by spending a ridiculous amount of money on sound equipment. People don't realize how much money they made - they were pretty much the top grossing tour band for about 15-20 years, playing about 90-100 shows a year. And they could (and did) use those shows to experiment with sound in a way that probably no other band has done since.
I haven't watched any D&C shows, but I expect their sound quality was just as good, if not better, than the Dead's.
The whole system travelled in two articulated lorries; but it took so long to set up that eventually they used four lorries, and two systems. One set would be for tonight's gig, the other would be on the road to tomorrow's gig, or starting to set up.
I think the mikes for the vocals were three mikes, not two (you can see triple mikes in The Grateful Dead Movie). I can't remember the rationale for using three mikes. The movie also shows the WoS being set up, under the direction of Ramrod, with Garcia gently urging him to get the speaker stacks hauled up higher, because that would be "really cool".
It was Donna.
It's like saying relay computers were dumb... Boolean logic was new and no one had ever attempted stuff like that before.
Being a non-profit facility, there are no fancy DSPs to notch out the hum or anything like that, so more creative solutions were investigated. It was determined through dumb experimentation that orienting an identical microphone 180 degrees to the one with the hum and setting the gain similarly would nearly eliminate the hum.
Eventually, the working theory became that a relatively new large pad transformer installed across the street was being picked up by the microphones. Orienting one microphone 180 degrees from the other caused the hum to be picked up out of phase from the main, and thus could be mixed in to cancel out the main mic hum.
Ultimately the real solution was simply buying better microphones, but there was a period of some months while a microphone sat off stage, pointed backwards.
But as for the acoustic implementation, even that has a long history. The Dead borrowed the idea from fighter pilot headsets, the only difference being that pilots were contending with a noisy cockpit rather than feedback. Same general idea that the unwanted sound hits both mics somewhat equally while the voice hits both mics somewhat unequally.
What drives this? Singers and speakers are both localized sources, so I'd expect the mics to pick up similar phases for each.
I bet it's distance! Falloff depends on distance to source, so there should be a larger difference in volume for closer sources.
Deleted Comment
Deleted Comment
Edit: Apparently this is not the case!
https://www.soundonsound.com/techniques/line-arrays-explaine...
For the most part, each performer plugged into the "Wall of Sound" had their own vertical 1xN stack of drivers.
3 drivers in a vertical line away will have less distortion than 3 drivers in a horizontal array; the horizontal drivers will suffer from comb filtering for listeners who are not located dead center at the middle of the array. (This of course assumes your audience is dispersed horizontally as opposed to floating randomly in space)
Modern home loudspeakers hew to this philosophy as well to an extent. As opposed to big "monkey coffin" 70s speakers with a random array of drivers sprayed across the front of the speaker[1], modern tower speakers have a vertical array of 2 or more drivers whose centers are aligned in a vertical line[2].
____
[1] https://www.reddit.com/r/BudgetAudiophile/comments/yburht/at...
[2] https://www.audiosciencereview.com/forum/index.php?threads/r...
But you cannot say that on HN. HN is the place where people believe soundbars are as good as high-end audiophile speakers!
Why those bands, and not others? What do they have in common?
https://en.wikipedia.org/wiki/Taper_(concert)
(a disadvantage to too much ethology reading: I can't remember "Bill Graham Presents" without thinking of baboon behaviour)
Not only did GD (and particularly Jerry Garcia and John Perry Barlow) eschew the 'intellectual property' model of music, their thought-leadership has lived on to become much of what we today consider fundamental internet technology and methodology.
Early decentralized crypto-economics, peer-to-peer file sharing, and the founding of the Electronic Frontier Foundation were all developed by some combination of deadheads and musicians and tapers, particularly on an internet service called The WELL.
The Green Pill Podcast had an entire episode exploring the bluegrass roots of blockchain technology; I was humbled / psyched to be a guest and play several of my songs, as well as some traditionals that GD also played. It's here if you're interested: https://www.youtube.com/watch?v=Y3s9Fu4yu7o&t=2898s
Owsley (wall of sound engineer) was one of the original major LSD manufactures... and a dead show was always where you went to score ACID if you lived on the east coast. This remained true well into the 90's.
I'm going to guess that all of those early internet pioneers that you mentioned also have fond stories of LSD.
The Dead, Bill graham, hells angles, peoples temple, Patty Hearst.... There is a continuum of culture that spills out of San Francisco to this very day.
The FLAC codec was developed by deadhead tapers.
The core of it is that speakers are bad at polyphony, so if you can avoid it, you can produce something that sounds more natural to human ears, and do so in a larger area. The way to avoid it is more speakers, more stacks/arrays/etc. You don't necessarily need an array per instrument, because modern loudspeaker arrays are indeed much better than they used to be, and modern loudspeaker processing fixes a multitude of problems.
This is also what Dave Rat is on about, in addition to IM and combing concerns, when explaining why a speaker per instrument sounds better than a mix: not just that it's a single point source, not just that it's doing one single job, but that all the sounds come from discrete points nowhere near each other. Just like an acoustic band using no reinforcement.
https://www.soundonsound.com/techniques/line-arrays-explaine...
https://www.erinsaudiocorner.com/loudspeakers/kef_r5_meta/
https://www.erinsaudiocorner.com/loudspeakers/sony_sscs3_tow...
more: https://www.erinsaudiocorner.com/loudspeakers/
If we're talking about his remarks here I'd characterize his take as "more vertical arrays" and not just "more speakers." I realize it's a bit pedantic of me, but some people think that simply adding more speakers equals more betterer sound and it's quite far from the case. In general, multiple loudspeakers arrayed horizontally (this includes MTM center channels in home theater setups) lead to comb filtering.I'm sure you know that, just clarifying for others.
If he has written about this elsewhere I'd love to read more!
Dave Rat has a good youtube channel, he also has some sort of insider subscription thing that I've never bothered with. I wouldn't say his ideas necessarily translate to every situation, but, he really presents these ideas as tools to use in a toolbox, not as gospel.
One interesting thing is that, we generally view comb filtering as universally 'bad', when, in reality, our ears do an excellent job of sorting out comb filtering when it comes to natural sounds. In fact, comb filtering is how we can locate a sound in 3d space with only two reference points (and, if you try, only one reference point moved around a little). That's remarkable, and points back to how speakers comb filter instead of mere comb filtering itself.
In practice, say you have a rock band, and your sound system has two arrays with subs spaced 40ft apart. Now, you're going to get a less than ideal pattern from that in the ranges where the bass guitar and kick drum live. How do you fix it? The answer is fairly simple, you simply run bass/kick in stereo, then, you delay the bass on one side by a little, kick on the opposite side just a little, then add some kind of EQ difference to the delayed side of each, then play with the delays until it sounds right.
Why does that work? Or does it really work? It's odd, because the math says "no no, it'll sound bad", the reality is, it can tighten up that comb to the point you don't really hear it and you can get good bass coverage out of a fairly poor system design.
Despite being an insane murderer, Phil Spector was a musical genius.
Why does every blog try to write a novel. Please just mainline me information.
But wait, then there wouldnt be room for 700 ads
If it doesn't make your pants flap in the breeze, turn it up!