Surround Effects

[zbt]

Hi everyone,

I am newbie here, is there surround effects that can play guitar notes all around you, like dolby 7.1?

[Mark Hammer]

Welcome.

What you ask begins with stereo.  The question that prompts is "what *kind* of stereo?"

The simplest stereo would be a split between two versions of a signal, usually staggered in time in some way, or processed differently, or both.  This would include separate wet/dry outputs from pedals, "ping-pong" delays in which successive repeats come out of opposite channels, auto-pans where the tremolo moves across channels back and forth, and so on.

More complex soundfield-positioning, as in Dolby 5.1 or 7.1 will digitally assign sound-sources to separate amplified speakers.  The 3-dimensionality of the result we hear, comes from our perception of location by the manner in which the audible result reflects off surfaces and has some delay.  Bear in mind that, whether sound is coming from one, two, 5 or 7 speaker cabinets, it is still being heard with only two ears.

In the early 1980's there was an interest in producing more realistic stereo soundfields, and simulation of spatial location.  I purchased a few devices that did this, and even built one from a Radio Electronics project article.  The perceptual principle they worked from was that, in the "real" world, any sound source not directly in front of you would arrive at each ear differently, in a particular way.  If you were not in an anechoic chamber, any sound coming from your right side would also be heard in your left ear, at a lower amplitude, with a tiny bit of time delay and some loss of high frequencies, resulting from what happens when the opposite-side ear hears what has travelled farther and bounced off an imperfectly reflective surface.

To accomplish this, the circuit would subtract left from right channel, and right from left, to derive whatever was fairly unique to each channel.  The derived information would then be processed by a bucket-brigade chip to achieve a very short delay (a few milliseconds), and then cross-fed to the opposite channel, with some lowpass filtering, and mixed in with that opposite channel at a lower amplitude.  So, you ended up with a little bit of right-side on the left, and a bit of left-side on the right.  Again, ever-so-slightly delayed, and filtered.

The audible result was dramatic.  You wouldn't notice it all that much when you first engaged it.  But when you disengaged it, your first reaction was "Hey, where did the stereo go?!  Why is everything smushed together?".  The psychoacoustic impact of creating such a "sound shadow" was striking.  All of a sudden, everything in the mix seemed to be coming from an identifiable location.  You could close your eyes and imagine where each instrument in the mix was. And in some instances, depending on the original stereo mix, things could even sound like they were behind or above you.

The caveat was that much depended on the quality and mix of the original source material.  Since noise sources are generally different between channels, things like static or worn grooves in vinyl, pops and clicks, had their overall amplitude increased.  If the source material wasn't impeccable, you paid a price for this improved 3-dimensionality in enhancement of non-music sounds.

The general strategy, however, was eventually ported over to the digital domain in the form of "Q-Sound" that studios would lease for mastering purposes, and eventually the Hughes SRS option for Windows Media Player.  Both of those spread out a stereo mix in ways that let the listener imagine the positioning of instruments and voices.

I am unaware of any pedals that aim for this territory.  Some chorus and phaser pedals with provide simultaneous add/subtract outputs, with two versions of the effect.  But, played through two amps, these tend to cancel each other out, in air.  I imagine there may be some digital pedals that provide simultaneous dual outputs, which are time staggered in some manner.  But it may also be possible in the analog domain.  I know the old A/DA Stereo Tapped Delay used the MN3011 bucket brigade chip, that had 6 staggered delay taps, which could be assigned to individual outputs, in order to achieve "spatial positioning" of a mono input in stereo.  So, if a person could add just a little more delay time to one output channel of a chorus or flanger, a stereo-positioning effect could be achieved, without cancellation in-air.

[iainpunk]

yes, use a 7 way splitter and 7 delay and/or vibrato pedals, all with slightly different settings. this gives a moving and spacious effect.
but dolby 7.1 means 7 mid and high channels and one bass channel, this means thet you should also add a single bass amp, since the low end on the guitar amplifiers will interfere and get ugly/chaotic, so take it out of the 7 guitar amps.

on a more realistic note, i suggest a stereo chorus.
or maybe build a 3-way vibrato to make the sound rotate between 3 amps, that would be adequate to have moving directions.
this tripple oscillator, the red dotted circles are LED's. use LDR's to have 3 vibrato circuits parallel, out of phase.


cheers, Iain

[ElectricDruid]

In the early 1980's there was an interest in producing more realistic stereo soundfields, and simulation of spatial location.  I purchased a few devices that did this, and even built one from a Radio Electronics project article.  The perceptual principle they worked from was that, in the "real" world, any sound source not directly in front of you would arrive at each ear differently, in a particular way.  If you were not in an anechoic chamber, any sound coming from your right side would also be heard in your left ear, at a lower amplitude, with a tiny bit of time delay and some loss of high frequencies, resulting from what happens when the opposite-side ear hears what has travelled farther and bounced off an imperfectly reflective surface.

To accomplish this, the circuit would subtract left from right channel, and right from left, to derive whatever was fairly unique to each channel.  The derived information would then be processed by a bucket-brigade chip to achieve a very short delay (a few milliseconds), and then cross-fed to the opposite channel, with some lowpass filtering, and mixed in with that opposite channel at a lower amplitude.  So, you ended up with a little bit of right-side on the left, and a bit of left-side on the right.  Again, ever-so-slightly delayed, and filtered.

The audible result was dramatic.  You wouldn't notice it all that much when you first engaged it.  But when you disengaged it, your first reaction was "Hey, where did the stereo go?!  Why is everything smushed together?".  The psychoacoustic impact of creating such a "sound shadow" was striking.  All of a sudden, everything in the mix seemed to be coming from an identifiable location.  You could close your eyes and imagine where each instrument in the mix was. And in some instances, depending on the original stereo mix, things could even sound like they were behind or above you.

This sounds ideal for an FV-1 algorithm.

[Phend]

I bought one of these back in approx 1982 and still use it (everyday) for television and records.
I think it works great and can still be purchased used for ...$50.00.

[Mark Hammer]

I bought one of these back in approx 1982 and still use it (everyday) for television and records.
I think it works great and can still be purchased used for ...$50.00.

I have one of those, as well as the "car stereo" version.  The Carver Sonic Hologram was always a bit above my price range.  Radio Shack included the feature on some of their equalizers.  It made the various Quincy Jones-produced Michael Jackson albums spectacular.  It felt like you were standing in a room of 20 musicians and could see every single one.

[Mark Hammer]

In the early 1980's there was an interest in producing more realistic stereo soundfields, and simulation of spatial location.  I purchased a few devices that did this, and even built one from a Radio Electronics project article.  The perceptual principle they worked from was that, in the "real" world, any sound source not directly in front of you would arrive at each ear differently, in a particular way.  If you were not in an anechoic chamber, any sound coming from your right side would also be heard in your left ear, at a lower amplitude, with a tiny bit of time delay and some loss of high frequencies, resulting from what happens when the opposite-side ear hears what has travelled farther and bounced off an imperfectly reflective surface.

To accomplish this, the circuit would subtract left from right channel, and right from left, to derive whatever was fairly unique to each channel.  The derived information would then be processed by a bucket-brigade chip to achieve a very short delay (a few milliseconds), and then cross-fed to the opposite channel, with some lowpass filtering, and mixed in with that opposite channel at a lower amplitude.  So, you ended up with a little bit of right-side on the left, and a bit of left-side on the right.  Again, ever-so-slightly delayed, and filtered.

The audible result was dramatic.  You wouldn't notice it all that much when you first engaged it.  But when you disengaged it, your first reaction was "Hey, where did the stereo go?!  Why is everything smushed together?".  The psychoacoustic impact of creating such a "sound shadow" was striking.  All of a sudden, everything in the mix seemed to be coming from an identifiable location.  You could close your eyes and imagine where each instrument in the mix was. And in some instances, depending on the original stereo mix, things could even sound like they were behind or above you.

This sounds ideal for an FV-1 algorithm.

Possibly.  Certainly feasible.  I just wonder if it is optimally intended for a mix, rather than a single instrument.

[Mark Hammer]

Here's the magazine project I built, many years ago.  It used an SAD1024.  The construction article provides a decent explanation of how it works.

https://worldradiohistory.com/Archive-Radio-Electronics/80s/1982/Radio-Electronics-1982-06.pdf

The second half, including the PCB layout, is found here:  https://worldradiohistory.com/Archive-Radio-Electronics/80s/1982/Radio-Electronics-1982-09.pdf

[ElectricDruid]

This sounds ideal for an FV-1 algorithm.

Possibly.  Certainly feasible.  I just wonder if it is optimally intended for a mix, rather than a single instrument.

There's stereo in and out on the FV-1, so that's no problem. Feed your mix in, get an enhanced mix out. Seeing as no long delays are required, you could even run the FV-1 at a higher sample rate to get more "hi fi" quality. It's spec'd to go up to 48KHz, which would be perfect.

[zbt]

Thank you for your reply Sir,

I have to digest your response, but first I am not guitarist and second please forgive for my English. 
I am just hobbyist and don't have a degree, excuse me if I am wrong.

"what *kind* of stereo?"

I think guitar is mono source.
What I learned from guitar amp, all is one speaker although there is 4x10 but still is mono.
What I dont understand in guitar amp there is clean and dirt amp,
so I diggin deeper, my first understanding clean is solid state and dirt is tube, as my friend insist it has to be tube? 
And I look some amp like Roland JC120, there is left and right speaker which path is to reverb unit.
So we have two kind of source from original and fx(Reverb) as you explain two versions of a signal.

More complex soundfield-positioning, as in Dolby 5.1 or 7.1

As common people (my understanding by counting speaker) ,
if I can play in mono (one speaker) or stereo (two speaker), why not 7.1
My friend lend me his guitar, and yes I played it through my stereo gainclone amplifier is LM3866 chip
but the sound not loud enough, and I argue with my friend, he said at least I must have 100Watt
so I try to increase the power voltage to make around 50Watt much better, and I still argue.
After he lend me his amplifier a solid state cort, now it sound louder, so I check what kind chip the amp has, is TDA2040.
I believe my LM3866 is much better, so I feel strange, I guess speaker is different.

it is still being heard with only two ears.

You mean record by two microphone, within a chamber, for reproduction?


          Center  Woofer
FL                               FR
              \\|//
SL  Left Mic ( O O ) Right Mic   SR
               ---             
RL                               RR

To accomplish this, the circuit would subtract left from right channel, and right from left,

A pasive one like this? (from old magazine)

The general strategy, however, was eventually ported over to the digital domain

I mean play directly to amp, instead I have four speaker in front, why not 8 in cardinal directions

yes, use a 7 way splitter

Any way to control volume for each? 

and 7 delay and/or vibrato pedals

an echo around you?
FL -> fx(Delay) -> SL -> fx(Delay) -> RL -> fx(Delay) -> RR -> fx(Delay) -> SR -> fx(Delay) -> FR

So, if a person could add just a little more delay time to one output channel of a chorus or flanger, a stereo-positioning effect could be achieved, without cancellation in-air.

If I play with stereo one channel is clean and the other dirt, would it be still cancel each other?

I thank you Sir Mark, for sharing your knowledge, the rest I am still digest.

Thank you Mr Iain, Mr Tom, and Mr Paul, thank you Sir.

[Mark Hammer]

I don't know what your first language is, but your English is good enough.

The JC120 (and all the JC amplifiers) have two power amplifiers, with each amplifier feeding a separate speaker.  The chorus effect is produced by using time delay for only one channel.  I have a small Fender amplifier that does the same thing, but with much less power and smaller speakers.

Many pedals will allow a person to do the same thing, by having a dry/clean output jack and a wet/delay output jack.  If each output is fed to a different amplifier, you get the same rich chorus sound as a JC120.  Of course, the volume level and tone of each amplifier need to be similar to produce a pleasing chorus, and a person needs to stand in the "right" place if the amplifiers/speakers are spaced apart.

Dolby 5.1 and 7.1 separate the sounds digitally "at source", rather than processing a common stereo soundtrack in a way that turns 2 into 5 or 7 channels.  You can certainly have more speakers than 2, and place them around you, spaced apart, but having something different coming from each speaker will not be possible without doing something VERY very complicated.

An amplifier based on an LM3866 should be more than loud enough.  Using the same speaker/s and same input signal, the accepted rule is that TEN times the wattage is needed to produce volume heard as twice as loud.  A TDA2040-based amplifier should not be heard as much quieter OR louder than an LM3866 amp.  If the 2040 amplifier is much louder, then that suggests that either the speaker/s used are more efficient, or the preamplifier stage provides more boost/gain, or both.  It is quite common for speakers to be very different in efficiency.  Some can be easily 4-6 times as loud with 1W of input, as other speakers of the same size.  Power rating is not a useful indicator.  If speaker A is rated at 40W and speaker B is rated at 50W, that does not mean B is louder.  It means that B can probably handle a little more output from the amplifier before the voice-coil burns up.  Speaker A could be more efficient than B, and much louder with the same amplifier level.

If you have a way to split the mono guitar signal into more than 2 outputs, AND you have a separate amplifier and speaker for each output, YES you can do something different to each of those individual outputs, so that something "different" is being heard from each amplifier.  Will it be pleasant?  That is a very different question. 

Will you be able to hear all those different sounds, using only two ears?  That is another very different question.  In the case of movie soundtracks, using 7.1 allows for the sound you hear to match how the source of that sound is moving on the screen.  So the footsteps of someone walking across the hard floor of a big room will move from a rear speaker on one side of you to a speaker on the other side.  But in that case the movie has two very big advantages over your guitar.  First, those footsteps are digitally encoded in the digital soundtrack to BE on one side then on both then on the other side, to create that illusion.  Second, the image on the screen tells you that the source is moving.

[iainpunk]

I think guitar is mono source.
What I learned from guitar amp, all is one speaker although there is 4x10 but still is mono.
What I dont understand in guitar amp there is clean and dirt amp,
so I diggin deeper, my first understanding clean is solid state and dirt is tube, as my friend insist it has to be tube? 
And I look some amp like Roland JC120, there is left and right speaker which path is to reverb unit.
So we have two kind of source from original and fx(Reverb) as you explain two versions of a signal.

yes, the full guitar amp is mono, but there are some stereo guitar amps, they generally have a stereo chorus effect.
the clean and dirt part is simple, your signal is sent in to 2 circuits, one is free of distortion, and has some tone shaping to make clean guitar sound nice.
the dirt side just amplifies the signal a lot, until its to much for the circuit, cutting of the peaks and troughs of the wave, this is called clipping. how this is achieved can be a plathera of ways, the early amplefiers use tubes to get that effect, while modern amps can use different types of transistors and diodes. tubes have a more gradual clipping, while transistors can be more abrupt, and op amps are the most abrupt in its clipping, which i like the most of all!

cheers, Iain

[zbt]

I couldn't find sad1024 at my local store, but this stereo simulator take another approach
https://www.ahrdf.net/forum/attachment.php?aid=409

also this look promising, using phase-shifter
http://thecarversite.com/manuals/mandir/Carver%20C-9%20service%20manual.pdf

I'll try this thing first.

My friends are Marshall fans, and his amp is crate not cort (my mistake), the guitar is cort.
Last time I bought a second hand boutique from a guy, which he describe sound like Marshall.
Wow how come this box turn out the amp to the expensive Marshall (I am naive here, or stupid)
I also check from internet it could be true, but it turns out it is just a tweak Marshall Guvnor,
What? an alias name of Marshall wrapped up in marketing trick?
I am disappointed, I would rather buy Marshall and tweak it or if I can I build my self.

I am new in this guitar world, hi-fi is more clear for me, get low distortion amp.
See if it has a big transformer, a big capacitor, a big heat sink, it may be good, but not always.

By definition of Taste, at this time I only knew milk.
If Gordon Ramsay is Guitarist, may be the amp has Sour Bass, Sweat Middle, Spicy Treble
My friend said turn bass 10 o clock, mid 12, treble 3, as his receipt, and I become Numb.

For me guitarist is fascinating God given creature an eccentric person, a gift to make world a live.
The term crunchy, is it guitar sound like doritos chips? A buzzing bee?
And sound like, I am sorry to say "fart", as for me if is like a leak capacitor, but they kind like it?
May be, if it don't smell bad, it sounds good? I am confuse.

So it is hard to say, it is subjective matter.

What I like about stompbox is a way Artist, Engineer, and Audience met.
Paint Artist add comfort to the eye, Guitarist add saltiness, Audience become pungent, and if don't work Engineer got bitter taste.
Add texture when we all play in a bar with beer.

The 7.1 is still a long way for me, again thank you for your explanation.

Sir Mark.


Hi Iain, currently just finish some boost circuit, and some fuzz, from beavis.
The fuzz is simple circuit, but it took me a couple month to figure out,
It's so simple, that effect my ego, until it become some kind a tuner radio.
The circuit laughing at me, as if teaching me to be more humble.
Its good and satisfying to make, now I am breadboarding a bigmuff circuit,
after found site name ElectroSmash, may be I can do a distortion like D+ also.

Do you also mix the clean and dirt, how does it sound for you?

cheers

[Mark Hammer]

Believe me, NONE of us could find an SAD1024 in a local store...unless we lived in a building at the Akihibara. 

Many low-cost amplifiers will sound similar.  Not because one is "disguised" under a different brand name (though that can happen), but because they use a similar design approach.  Many of them also use similar power-amplifier chips (e.g., TDA2050) and use similar "no-name" speakers made for them by companies (e.g., Eminence) that produce speakers for many manufacturers.

Thanks for the link to the Carver C-9.  I was always curious about it, since it came out at the same time as all the other devices I've mentioned.  I'm surprised that it does not use any time delay.  One aspect that IS important for your interests is that it relies on what actually starts out as stereo.  That is, it only produces a "better" stereo field from an original stereo recording/source.  It will not produce stereo from a mono source.

[iainpunk]

Do you also mix the clean and dirt, how does it sound for you?

i generally don't like mixing clean and dirt straight, but it is nice if you use a filter to mix.

here's how i added a clean blend to a big muff, the filtering is mirrored for the clean signal, if you have it on full bass, the clean path has full treble, and vice verse. if you have it in the middle, there would normally be a mid-cut, (often called a ''scoop''), this adds clean mid-range to the fuzz.
the downside is that sometimes, with certain component values, not the originals, there are slight resonances when the clean level is at max.

cheers, Iain

[zbt]

One aspect that IS important for your interests is that it relies on what actually starts out as stereo.  That is, it only produces a "better" stereo field from an original stereo recording/source.  It will not produce stereo from a mono source.

note taken

Thanks for your preferences, I'll try it.

[anotherjim]

Roland championed a faked surround over stereo idea called RSS (Roland Sound Space!) a while ago, including it in their COSM based digital effects products. It could make the sound seem to fly over/around and behind your head. The downside was that it only worked properly over sealed over-ear headphones since it was actually exploiting the sensitivity of our hearing to small directional effects, each ear has to get each channel exclusively. Think of the difference in sound when an omni mic is moved on or off axis to the sound source.
Like a lot of Rolands great digital advances, I think they dropped the idea and buried it. It lives on in my old VS multitrack recorder.

[iainpunk]

Roland championed a faked surround over stereo idea called RSS (Roland Sound Space!) a while ago, including it in their COSM based digital effects products. It could make the sound seem to fly over/around and behind your head. The downside was that it only worked properly over sealed over-ear headphones since it was actually exploiting the sensitivity of our hearing to small directional effects, each ear has to get each channel exclusively. Think of the difference in sound when an omni mic is moved on or off axis to the sound source.
Like a lot of Rolands great digital advances, I think they dropped the idea and buried it. It lives on in my old VS multitrack recorder.

some PC and xbox games have this RSS spatial sound integrated in it as well, as a toggleable option, that you can hear the direction of footsteps, and explosions in 3D space, giving you a definite advantage over players without the 3d sound turned on.

cheers, Iain

[zbt]

https://www.youtube.com/watch?v=wbIKKc8zcOA

https://worldradiohistory.com/UK/Electronics-Today-UK/80s/Electronics-Today-1983-07.pdf

page 76 Holophonic Head

I thought if we record with two microphone, the brain would decode the effect.   

[Mark Hammer]

Binaural recording, as it came to be called, has been used by people such as Lou Reed (though I forget what album).  The principle of sticking a fake head between two mic capsules, in order to produce the ear-to-ear differences we normally hear, is a good one.  The problem is that such physical methods do not accommodate the variety of microphones that many musicians prefer to use for recording specific instruments.  In that respect, digital modelling of location-in-space can be more efficient, and allow the engineer to use whatever microphone best complements the particular sound source.

As always, I recommend reading up on "auditory scene analysis"; a concept and approach spearheaded by one of my best undergraduate professors, many years ago.  https://www.researchgate.net/publication/200045100_Auditory_Scene_Analysis_The_Perceptual_Organization_of_Sound/link/54384e4d0cf24a6ddb934692/download