I usually do this in the studio to be able to capture chaotic jam sessions where people are just playing around while trying out stuff performance wise. This helps taking the edge off for people who are not professional musicians or studio musicians. People who are none of those things usually can have a lot to say musically but get put off by the sterile and serious energy that goes around professional studios. Some people love listening to their own voice or acoustic performance (processed) in the same room they are playing in instead of using headphones, specially when there's other people in the room besides you. If you have ever been in a casual recording situation with non-professional musicians you probably understand what I'm talking about. I always used this trick to avoid feedback while being able to crank the mics up so that they feel more their own performance, as well as everyone else in the room. I do it very subtly and throughout the entire frequency range, because even a small difference in pitch can give you so much more extra gain when combined with some other techniques like EQ. Obviously this creates a very particular sonic stile that is usually only suited for urban or experimental music but still, sometimes it is better to capture a heartfelt and energetic performance in any way possible than to not do it at all. I knew I was onto something when I discovered this little trick! Glad to know this can be done in other settings like this one! subscribed!
In the context of engineering, I've come to think of the word "trick" as meaning "solution (or tasteful approach) to a fairly niche/specific problem". Learn and combine enough "tricks" and hey ho you're an engineer i.e., someone who applies little bits of knowledge to solve lots of little problems and provide one big solution.
Dumbass observation. First of all a trick is very different than magic, 2 very diferent things. Second, nobody is talking about magic here but you. And third, if you dont undertand how its done, by definition it is a trick. Dumbass
Missing in the explanation is, maybe, pitch shifted feedback. This is how shimmer reverbs work. Here, they pitch shift down and then the repeats are continually shifting lower. That might be the missing information.
that's the part that was explained quick and that I had to replay a lot haha. And still not sure of the routing haha but my guessing is that the the pitchshifter goes into the mono sub??
Dear Sirs, I'm not sure about the signal flow. He created an AUX Buss (Mix 10), turned on the LPF (100Hz) and inserted the Pitch Shifter. How did mix the signal in the subs? Should the subs be plugged into an Aux path or are they on the PA system processor?
Great ideea, definately gonna take a crack at it, I wonder if, would it work with orchestras's and other more melodic instruments or would it mutate the sound? keep up the good work
Depends on how good the processing is. On the Yamaha console with that particular setting, there is some amount of distortion since it wasn't intended to be used this way. What's nice is that it does have very low latency compared to other pitch-shifting plugins. For melodic and higher-quality purposes, we would chose a more sophisticated plugin but one that also had low latency. After that, the only question is one of taste.
It would mutate the sound regardless of “processing power”. Forget the pitch-shift manipulation and just do a gradient array and tune the delay to keep your “null” heavy where the basses (generally about 45 degrees off axis on-stage behind PA) and cellos (between 90-45 degrees) usually are set with a traditional orchestra. Remember with one sub “component” behind another, out of phase, creates a dipole shape, and as you delay the rear component it pulls the rear lobe inwards and broadens that and the front lobe, but too much (approaching 200% of the delay distance front baffle to rear) and it starts over again but is really creating harmonic havoc.
I occasionally volunteer for the Mick Jagger performing arts centre in Dartford England and they have the same desk , the processing plug-ins on that console are absolutely amazing , you can do absolutely anything you want.
How much it cost to have your controller linked to that fancy ipad? Looks like a whole lotta work for something thats not going to be appreciated by many but a good showcase at least to prove things to yourself and learn from experience..
How do you deal with the effect that. The initial sound of the drums gets pitch shifted so it is going to sound different from original sound from the drum itself.
No. The pitch-shifting is to break the feedback loop, so that sound that is amplified is not coincidental to the sound that is picked up by the mics a second time. Every trip through the mic-speaker-mic loop is down-shifted by one octave, where it quickly slips beneath the subsonic filter and is low-cut filtered away entirely. The audible effect (other than the absence of feedback) is secondary, and may be desirable or not.
Presonus Air18s, I have a pair of them & I am curious if u think stacking them with top one facing away from dance floor and uses the DSP to set as a Cardioid it will do the magic of getting most of the bass response coming forward up to what frequency before there is some real loss of the cardioid effect?, 80 or 100, certainly at 150 I would suspect... Thanks, The Wedding DJ, LOL... 😁
Sub bass is omni directional *USUALLY* , so stacking and directing the speakers opposite directions (without processing) shouldn't change anything with regards to bass phase correlation, compared to putting them next to each other, except for the fact that the phase correlation pattern will now be turned 90 deg, so what you usually expect on a horizontal plane, will now be happening on a vertical plane instead and visa versa, but nothing much will change front to back. If I understood your question properly bare with me.... The directional nature of a sound driver depends on it's size, so for example an 18inch woofer (1.5 feet) will be directional down to 1KHz, where as a 3 foot driver (kinda unrealistic for most applications) can be directional down to 500Hz, and any lower than that, the speaker is unable to focus the sound energy properly and it starts to gradually become omni-directional from there on downwards. If you want to focus sub bass in a very specific direction and have the best rear cancellation with just 2 speakers, you need to look at things like an end-fire array set up with delay (putting one speaker behind the other on a few ms delay so forwards the waves sum, and backwards the waves cancel), or a gradient array. However having said all that, I believe the speakers you have, have a special setting for delay, phase, gain and filters that works in the configuration you said to make a cardioid pattern by making something like an advanced drive gradient array (as well as an endfire preset when the cabs are placed in the correct positions too), and this will take up less space than an endfire, as you need a 5 foot grill to grill spacing distance (facing the same direction) for endfire with your speakers and Presonus's endfire preset, so you loose some dance floor, but get better directional focused bass. I don't know how well this (the gradient array cardiod pattern) specifically works with the Presonus Air line up (it really depends on what exact processing they are using, so is very hard to opine on), but it's probably more suitable to your application due to the small spaces things like wedding receptions are held in. I'm assuming you have other speakers you play the low mid and upwards on etc.? With these very closely spaced array set ups, if properly set up with correct processing (like the one you're talking about), you're probably looking at very little lobing if any lobes at all (the break down of cardiod pattern into fingers of phase correlation and nulls in between) up to at least 100Hz, and even at 120Hz there should be a decent area without major amounts of lobing (mby 3 with 2 thin nulls at most). However the thing is that even if the edges of the pattern start to form bumps at this frequency range, front to back performance often will change higher up than the front pattern starts to break up (although some array set ups are more prone to falling apart at the back than others, generally with the ones we are talking about this is much less so). If Presonus are using frequency dependent delays (which I have a sneaky feeling they will be), then this can basically mitigate this falling apart of the pattern throughout the entire speakers range in all directions, as each freq has a specific delay set to get the best pattern possible for that frequency (and for comparison with endfire, the closer you have them and the shorter the delay the less overall level you get, but the wider a frequency response your pattern will be stable over, if setting it up manually, IDK if there's options to change this preset with your set up). It's worth looking into if they use frequency dependent delays, as this could be the deciding factor between end fire and advanced drive gradient pairs, as it may not be more beneficial to have the space saving of advanced drive gradient if the Presonus settings can't fully mitigate the breakup of the pattern from 120Hz upwards with it (I assume they know what they are doing though and got it right). If Presonus haven't released any info on how their presets work, you can always set them both up and listen to how it changes things, walk around the space in front and behind the stack playing different freq tones and see what one works best for your application. ;) This is only for subs though, and shouldn't really be used for mids etc. and you want other speakers to handle the mid range if you are going for one of these set ups as they do fall apart when you start getting much higher than 150Hz, no matter what processing they are using (as it will be tuned specifically for the low end). But then, even if you have a pattern that falls apart at 120Hz upwards, you can just crossover at 80Hz anyway, or 65Hz if your mids go that low and you're playing particularly bass heavy music (or if the mids don't go that low, 100Hz or where ever the mids go down to). I hope I managed to make that make sense lol. If I didn't explain something properly let me know and I'll try to explain it better. ;)
@@LaserChair Glad to help. I forgot to add. You'll sometimes find systems (useually med to larger ones) with subs low passed as low down at 65 or mby 80 then dedicated bass drivers filling in the space between the mids and subs. Usually dedicated to the punch range. As if you split your freq spectrum into 4 bands (sub/bass/mids/tops) this will give better clarity to both bass, sub and mids (just like a 3 way studio monitor with a sub, vs a 3 way monitor doing the full range, the main woofer of the monitor does not have to strain to produce both the bass and the sub frequencies so will have more clarity and punch in the bass/punch band and the subs will be cleaner too). 😉👍
Hello! I’m subscribed to your channel and watch u all the time. But hv u ever heard of Danley Sound Labs n Gainesville, GA? I think they hv the deepest bass and make the best subs out of any company
So you subs will ONLY get subsynth signals between 25Hz and 50Hz, right? You'll have NO punch at all using that strange setup!! I guess that is not allowed!! Music just NEEDS the 60Hz - 110Hz range!!
It will use whatever the instrument has as even harmonics to boost the fundamental and even harmonics. But you will only hear the effect in the 2 octaves from 25Hz to 100Hz. It may sound bad if: A subharmonic is out of character. The instrument has strong 3rd harmonic (that would come out as 1 1/2 harmonic bleagh)
yes but if used "tastefully" can really enhance the FOH sound of the instrument. one variation I use is to both low AND high pass the sound going to the pitch shifter.. effectively "selecting" note range you want to generate the artificial sub from. I usually chuck a high pass on the pitch shifter output too.. to really dial it in.
@@szaki95 everything that goes thru this signal chain is pitched down to half the frequency! As the overhead mics must be highpass filtered anyway to avoid wind noise, there will not be much original signal left below 100Hz coming from these mics. So the result will be that the frequency range from 100-200Hz of these mics will be pitch shifted down to half the frequencies (50-100Hz) and reproduced by the subs. For this specific application with percussive drum signals it may sound ok or even natural. But do not do this with melodic instruments playing „real tones“...
If you split your sub boxes equally and placed them in the right and left of the stage in mono you still get phasing issues? I don't understand why the subs are all the way to one side of the stage,, thanks!
This is called an endfire configuration. Both sides of the stage would have this same setup. The rear group of subs are fed a flipped polarity sinal and delayed by the amount of time it takes for the sound to travel from the front end of the front stack to the front end of the back stack. This created a cancelling effect of the sound towards the back and focuses the sound out the front.
So this is basically just adding sub harmonics an octave lower than the original Tyco drum sound like say, an octave bass pedal. I mean, why not just move the subs forward, high pass the Tyco's up to around 60 hz and just have that coming out of the PA without the sub harmonics with that close mic'd main low Tyco in the middle eq'd like a kick drum with a bunch of low mid pulled out. Couple of overheads for the tops mixed in and you'd still get a huge sound.
It's not about adding sub octave content. It's about taking the natural feedback loop in the bass and having it constantly fall below what the subs can reproduce as every time it passes through the loop it is pitch shifted down and so can never build up and "feed back"
i do the same thing in audacity for my car sub.. pitch shift the song apply a LPF, then copy the same song and apply an HPF at 100hz and up.. works great
So what do you do ? you use the effects of Audacity ? First you "Change pitch" of minus 12 semi-tones, then apply an LPF with the "Graphic Equalizer" at what ? 100 hz ? abruptly ? Then add the original again on a new stereo track and apply an HPF at 100 hz again and merge both right ?
That looks like an endfire sub array which would put the null point behind the cluster, not directed onstage where the mics are, so no apparent benefit there. Why are the subs only on one side of the stage? I understand putting subs in the middle to even out the coverage that you would get from a L/R deployment, but this setup would have more sub on house right than on house left.
That sub config is end-fire - so I disregarded everything else you said after that... since this business is about technical expertise and attention to detail...
Actually that set up is a gradient. Add a third or more “components” to the front or back, all delayed off of the rear most “component”, and you have “end fire”, which makes a more focused and long throw sub lobe. A gradient is usually one “component” behind another, sometimes three total parts including the front one, but the rear one is delayed to the front one, and flipped “out” of phase relative to the front one. The gradient is very efficient space and gear and result wise, as you can shape you lobe with delay time over “100%”, as in MORE than 4.65 feet based on a 60hz center frequency 1/4 wave distance of spacing and delay. 0 delay on the rear speaker or set of speakers which I would call a “component”, and out of phase from the front, would make a dipole, with nearly equal front and rear lobes, and the sides pinched in at the “hips”, where any delay added to the rear “pulls” the rear lobe forward and broadens the front lobe, where your actual distance front baffle to rear baffle being your delay time producing a cardioid pattern at that center frequency, my choice being 60 hz. Delay times above “100%” of that time/distance approaches a hyper cardioid and then a super cardioid, where generally about 150% or more of delay begins countering that shape, and then you realize you (myself included) don’t know hack shit about music, harmonics, sound wave propagation, and that a linear equation would be nice, but noooooo, it’s not that simple. I’ve implemented this “in the field” for about three years now, and one time recently I was mixing a reggae festival, where I was heavily requested to mix and engineer the sound by the artists. Partway through, a good friend grabbed me and asked where my “famous” “nut twitching” bass mix was. I told him I thought it sounded great where I was, front and center, and he agreed, but said where he was it was like no subs at all. I asked him where that was and we walked laterally to about 60 degrees off axis. And I said, “damn, you are totally right”. With my iPad I added inches of delay to the existing setting I had, and sure enough, it broadened the front lobe to include our position dramatically. Damn. It reduced the forward reach, but that was fine, as the park was more broad than deep. None of this is linear though, and you can get some serious anomalies at certain frequencies by distance from your “array” and THAT is where you get humbled. This is why an SVT bass cab makes TONS OF “bass”, but upon real inspection, that bass response is remarkably different as you move around in front of the cab. No free lunches. This guys “trick” is just that, where that mic he was tapping on was in FRONT of the subs, albeit not directly, and he had lots of cancellation there. BUT, how did that sound at that angle from the subs out in the audience? Hmmmm. So you have cancellation, but then synthetically BOOST what was a high passed mic channel, with two layers of pitch shift, to “restore” what you had “robbed”? I’m not buying it. Yeah, he acknowledged the nuances, but…….no free lunches.
I think I understand. So because each hit of the drum is only lasts less than a second, it makes it out of the subwoofer. Whereas, under normal circumstances, feedback continues to be produced by the system, therefore it quickly gets cancelled out by the lo-pass filter and pitch shift. Is that correct?
I usually do this in the studio to be able to capture chaotic jam sessions where people are just playing around while trying out stuff performance wise.
This helps taking the edge off for people who are not professional musicians or studio musicians. People who are none of those things usually can have a lot to say musically but get put off by the sterile and serious energy that goes around professional studios. Some people love listening to their own voice or acoustic performance (processed) in the same room they are playing in instead of using headphones, specially when there's other people in the room besides you. If you have ever been in a casual recording situation with non-professional musicians you probably understand what I'm talking about.
I always used this trick to avoid feedback while being able to crank the mics up so that they feel more their own performance, as well as everyone else in the room. I do it very subtly and throughout the entire frequency range, because even a small difference in pitch can give you so much more extra gain when combined with some other techniques like EQ. Obviously this creates a very particular sonic stile that is usually only suited for urban or experimental music but still, sometimes it is better to capture a heartfelt and energetic performance in any way possible than to not do it at all.
I knew I was onto something when I discovered this little trick! Glad to know this can be done in other settings like this one!
subscribed!
An engineering tutorial of excellence! Thank you!
freaking genius
Those drums sound insane!
What an amazing trick!
There are no "tricks" or "magic" in the world of audio engineering. Just things you don't understand yet.
Ah, but... "any sufficiently advanced technology is indistinguishable from magic." -- Arthur C. Clarke.
Aren’t you fuckin deep.
To those that don't know.... it's a trick.... until they understand it
In the context of engineering, I've come to think of the word "trick" as meaning "solution (or tasteful approach) to a fairly niche/specific problem".
Learn and combine enough "tricks" and hey ho you're an engineer i.e., someone who applies little bits of knowledge to solve lots of little problems and provide one big solution.
Dumbass observation. First of all a trick is very different than magic, 2 very diferent things. Second, nobody is talking about magic here but you. And third, if you dont undertand how its done, by definition it is a trick. Dumbass
when the engineer got on the plaid shorts u know it’s about to hit
That is such an awsome trick! Especially with drums which often dont have enough low bass anyway.
0:06 gives me the heebi geebies. Pushing up so high up the box is asking for the bottom to kick you in the gut.
Missing in the explanation is, maybe, pitch shifted feedback. This is how shimmer reverbs work. Here, they pitch shift down and then the repeats are continually shifting lower. That might be the missing information.
Thank you! Is it your short process description, isn't it?
So, the subs are fed just from that pitch-shifted aux, or you're blending the aux and the main mix?
that's the part that was explained quick and that I had to replay a lot haha. And still not sure of the routing haha but my guessing is that the the pitchshifter goes into the mono sub??
Brilliant solution... I'll have to try this!
Thanks for sharing 😎👍
The demo at the end was epic.
Brilliant simple solution. Applause
Dear Sirs, I'm not sure about the signal flow. He created an AUX Buss (Mix 10), turned on the LPF (100Hz) and inserted the Pitch Shifter. How did mix the signal in the subs? Should the subs be plugged into an Aux path or are they on the PA system processor?
mono sub?
Great use of the flexibility of digital front ends (and not particularly intuitive).
Great trick, thanks a lot! Also very helpful videos, much appreciated! 🍻
Great ideea, definately gonna take a crack at it,
I wonder if, would it work with orchestras's and other more melodic instruments or would it mutate the sound?
keep up the good work
Depends on how good the processing is. On the Yamaha console with that particular setting, there is some amount of distortion since it wasn't intended to be used this way. What's nice is that it does have very low latency compared to other pitch-shifting plugins. For melodic and higher-quality purposes, we would chose a more sophisticated plugin but one that also had low latency. After that, the only question is one of taste.
@@devinlsheets_alphasound
Thank you very-very much for the repply. 👍😀
It would mutate the sound regardless of “processing power”. Forget the pitch-shift manipulation and just do a gradient array and tune the delay to keep your “null” heavy where the basses (generally about 45 degrees off axis on-stage behind PA) and cellos (between 90-45 degrees) usually are set with a traditional orchestra. Remember with one sub “component” behind another, out of phase, creates a dipole shape, and as you delay the rear component it pulls the rear lobe inwards and broadens that and the front lobe, but too much (approaching 200% of the delay distance front baffle to rear) and it starts over again but is really creating harmonic havoc.
Great and informative video!
That deserves my subscription.
Thanks.
I occasionally volunteer for the Mick Jagger performing arts centre in Dartford England and they have the same desk , the processing plug-ins on that console are absolutely amazing , you can do absolutely anything you want.
I imagine the weirdness of sound you speak of would be a result of artifacts from the pitch shifter and possibly phase issues?
If I get this tricky right. The mic that's point in font of the sub picks the low end also sent back to sub aux fed
I laughed out loud! I love this trick.
That 'pitch shifter'.. was it a built-in effect in a console? Does the YAMAHA's only have that?
It was just a plug-in on the console, I think a lot of consoles come with a similar thing
Wow, that’s brilliant! Is that your creation or some sort of Production Illuminati knowledge?
How much it cost to have your controller linked to that fancy ipad? Looks like a whole lotta work for something thats not going to be appreciated by many but a good showcase at least to prove things to yourself and learn from experience..
It costs only as much as a wifi router costs. So at minimum about $30. This is a standard feature on most modern mixing consoles.
How do you deal with the effect that. The initial sound of the drums gets pitch shifted so it is going to sound different from original sound from the drum itself.
they only pitch down the low frequencies, the rest stays the same, so they keeps their unique sounds
Well by listening. Like at minute 3:00. I hear it working well
So you’re just pitch shifting the drums to make them hit harder in the sub bass frequencies.
No. The pitch-shifting is to break the feedback loop, so that sound that is amplified is not coincidental to the sound that is picked up by the mics a second time. Every trip through the mic-speaker-mic loop is down-shifted by one octave, where it quickly slips beneath the subsonic filter and is low-cut filtered away entirely.
The audible effect (other than the absence of feedback) is secondary, and may be desirable or not.
@@nickwallette6201 I see, thanks for the clarification 👍
Omg, very clear Sound 😲❤️❤️
Nicely done.
That is a very interesting solution and how did you even think of it?
I'm gonna show this video to might control systems engineering professor and ask him what order of system is this
Damn does that sound good even over the internet!! Nice!!
THIS.... is incredible.
I have Yamaha Ql1 how to use these kind of technique? I don't know
🇿🇦Gods of Sound reproduction "Nexo". I love how Alex leads the team will practical explained example 🙏🏾.
What a clever idea!!
Outstanding 👏👏👏
brilliant!
What a posh way of saying I love bass !
Soo softness in bass💕
WOW. MIND BLOWN. 🤯
Nice tip.
More mics at the freeparty
Sound guy!!
Thanks
Man you just have to be careful, you can burn voice coils by cranking frequencies that the drivers can't reproduce! But cool idea!
Its pretty uncommen tho. And it depands if they passive or active.
@@karsmemelink770
Exactly
Those active processed boxes have robust protection.
Surely he knows what he's doing.
it's pretty much standard to always run a 40hz high pass on the whole PA
That's what the DSP in the amplifiers or system processor usually handle and remove to protect the speakers.
awesome!
Can we make this with two 18' scoops and a dpx pa+ ? XD
Presonus Air18s, I have a pair of them & I am curious if u think stacking them with top one facing away from dance floor and uses the DSP to set as a Cardioid it will do the magic of getting most of the bass response coming forward up to what frequency before there is some real loss of the cardioid effect?, 80 or 100, certainly at 150 I would suspect...
Thanks, The Wedding DJ, LOL... 😁
Sub bass is omni directional *USUALLY* , so stacking and directing the speakers opposite directions (without processing) shouldn't change anything with regards to bass phase correlation, compared to putting them next to each other, except for the fact that the phase correlation pattern will now be turned 90 deg, so what you usually expect on a horizontal plane, will now be happening on a vertical plane instead and visa versa, but nothing much will change front to back. If I understood your question properly bare with me....
The directional nature of a sound driver depends on it's size, so for example an 18inch woofer (1.5 feet) will be directional down to 1KHz, where as a 3 foot driver (kinda unrealistic for most applications) can be directional down to 500Hz, and any lower than that, the speaker is unable to focus the sound energy properly and it starts to gradually become omni-directional from there on downwards. If you want to focus sub bass in a very specific direction and have the best rear cancellation with just 2 speakers, you need to look at things like an end-fire array set up with delay (putting one speaker behind the other on a few ms delay so forwards the waves sum, and backwards the waves cancel), or a gradient array.
However having said all that, I believe the speakers you have, have a special setting for delay, phase, gain and filters that works in the configuration you said to make a cardioid pattern by making something like an advanced drive gradient array (as well as an endfire preset when the cabs are placed in the correct positions too), and this will take up less space than an endfire, as you need a 5 foot grill to grill spacing distance (facing the same direction) for endfire with your speakers and Presonus's endfire preset, so you loose some dance floor, but get better directional focused bass. I don't know how well this (the gradient array cardiod pattern) specifically works with the Presonus Air line up (it really depends on what exact processing they are using, so is very hard to opine on), but it's probably more suitable to your application due to the small spaces things like wedding receptions are held in.
I'm assuming you have other speakers you play the low mid and upwards on etc.?
With these very closely spaced array set ups, if properly set up with correct processing (like the one you're talking about), you're probably looking at very little lobing if any lobes at all (the break down of cardiod pattern into fingers of phase correlation and nulls in between) up to at least 100Hz, and even at 120Hz there should be a decent area without major amounts of lobing (mby 3 with 2 thin nulls at most). However the thing is that even if the edges of the pattern start to form bumps at this frequency range, front to back performance often will change higher up than the front pattern starts to break up (although some array set ups are more prone to falling apart at the back than others, generally with the ones we are talking about this is much less so). If Presonus are using frequency dependent delays (which I have a sneaky feeling they will be), then this can basically mitigate this falling apart of the pattern throughout the entire speakers range in all directions, as each freq has a specific delay set to get the best pattern possible for that frequency (and for comparison with endfire, the closer you have them and the shorter the delay the less overall level you get, but the wider a frequency response your pattern will be stable over, if setting it up manually, IDK if there's options to change this preset with your set up).
It's worth looking into if they use frequency dependent delays, as this could be the deciding factor between end fire and advanced drive gradient pairs, as it may not be more beneficial to have the space saving of advanced drive gradient if the Presonus settings can't fully mitigate the breakup of the pattern from 120Hz upwards with it (I assume they know what they are doing though and got it right). If Presonus haven't released any info on how their presets work, you can always set them both up and listen to how it changes things, walk around the space in front and behind the stack playing different freq tones and see what one works best for your application. ;)
This is only for subs though, and shouldn't really be used for mids etc. and you want other speakers to handle the mid range if you are going for one of these set ups as they do fall apart when you start getting much higher than 150Hz, no matter what processing they are using (as it will be tuned specifically for the low end). But then, even if you have a pattern that falls apart at 120Hz upwards, you can just crossover at 80Hz anyway, or 65Hz if your mids go that low and you're playing particularly bass heavy music (or if the mids don't go that low, 100Hz or where ever the mids go down to).
I hope I managed to make that make sense lol. If I didn't explain something properly let me know and I'll try to explain it better. ;)
@@DaftFaderthanks for this reply, i learned quite a bit from it!
@@LaserChair Glad to help. I forgot to add. You'll sometimes find systems (useually med to larger ones) with subs low passed as low down at 65 or mby 80 then dedicated bass drivers filling in the space between the mids and subs. Usually dedicated to the punch range. As if you split your freq spectrum into 4 bands (sub/bass/mids/tops) this will give better clarity to both bass, sub and mids (just like a 3 way studio monitor with a sub, vs a 3 way monitor doing the full range, the main woofer of the monitor does not have to strain to produce both the bass and the sub frequencies so will have more clarity and punch in the bass/punch band and the subs will be cleaner too). 😉👍
Nexo Aint no Meyer or Hennessy battle axe subs…( huge horns for 21’s) the Dave has his 30” subs
Yep, that's a winner!
Hello! I’m subscribed to your channel and watch u all the time. But hv u ever heard of Danley Sound Labs n Gainesville, GA? I think they hv the deepest bass and make the best subs out of any company
They should send me some and I'll do a RUclips video on them!
@@devinlsheets_alphasound idk how to reach them like that. U might wanna look into them. Nobody beats them when it comes to BASS!!
Cardiod ❌ Endfire ✅
ouchhh . nice trick . its work
So you subs will ONLY get subsynth signals between 25Hz and 50Hz, right? You'll have NO punch at all using that strange setup!! I guess that is not allowed!! Music just NEEDS the 60Hz - 110Hz range!!
Genius
Thanks Men
What? You run the signal that goes to the subs thru a pitch shifter? So your signal is completely distorted, you are simply producing subharmonics ?!?
I agree, i don’t understand what he is trying to achieve, I only know about basic pa and small sound systems….
Listen carefully, the 2nd time it goes thorugh the signal chain it's so low in pitch the speakers can't reproduce it anymore, so it doesn't feedback.
It will use whatever the instrument has as even harmonics to boost the fundamental and even harmonics. But you will only hear the effect in the 2 octaves from 25Hz to 100Hz.
It may sound bad if:
A subharmonic is out of character.
The instrument has strong 3rd harmonic (that would come out as 1 1/2 harmonic bleagh)
yes but if used "tastefully" can really enhance the FOH sound of the instrument. one variation I use is to both low AND high pass the sound going to the pitch shifter.. effectively "selecting" note range you want to generate the artificial sub from. I usually chuck a high pass on the pitch shifter output too.. to really dial it in.
@@szaki95 everything that goes thru this signal chain is pitched down to half the frequency!
As the overhead mics must be highpass filtered anyway to avoid wind noise, there will not be much original signal left below 100Hz coming from these mics. So the result will be that the frequency range from 100-200Hz of these mics will be pitch shifted down to half the frequencies (50-100Hz) and reproduced by the subs. For this specific application with percussive drum signals it may sound ok or even natural. But do not do this with melodic instruments playing „real tones“...
If you split your sub boxes equally and placed them in the right and left of the stage in mono you still get phasing issues? I don't understand why the subs are all the way to one side of the stage,, thanks!
this is mono, baby 😍
This is called an endfire configuration. Both sides of the stage would have this same setup. The rear group of subs are fed a flipped polarity sinal and delayed by the amount of time it takes for the sound to travel from the front end of the front stack to the front end of the back stack. This created a cancelling effect of the sound towards the back and focuses the sound out the front.
cool trick.
Clever boy 🏆
Very cool
Perfect wonderful
So this is basically just adding sub harmonics an octave lower than the original Tyco drum sound like say, an octave bass pedal.
I mean, why not just move the subs forward, high pass the Tyco's up to around 60 hz and just have that coming out of the PA without the sub harmonics with that close mic'd main low Tyco in the middle eq'd like a kick drum with a bunch of low mid pulled out. Couple of overheads for the tops mixed in and you'd still get a huge sound.
It's not about adding sub octave content.
It's about taking the natural feedback loop in the bass and having it constantly fall below what the subs can reproduce as every time it passes through the loop it is pitch shifted down and so can never build up and "feed back"
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Is it Endfire or Cardioid ?
Yes this is technically an endfire configuration.
Ohhhh i will try that definitely….😮😮😮
Excelent!
i do the same thing in audacity for my car sub.. pitch shift the song apply a LPF, then copy the same song and apply an HPF at 100hz and up.. works great
So what do you do ? you use the effects of Audacity ? First you "Change pitch" of minus 12 semi-tones,
then apply an LPF with the "Graphic Equalizer" at what ? 100 hz ? abruptly ? Then add the original again on a new stereo track and apply an HPF at 100 hz again and merge both right ?
Cardioid Or end Fire Array this is the difference
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That looks like an endfire sub array which would put the null point behind the cluster, not directed onstage where the mics are, so no apparent benefit there.
Why are the subs only on one side of the stage? I understand putting subs in the middle to even out the coverage that you would get from a L/R deployment, but this setup would have more sub on house right than on house left.
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couldnt you just use a subharmonic synth
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Трюк классный, но блин, почему суена как из начала девяностых??
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That sub config is end-fire - so I disregarded everything else you said after that... since this business is about technical expertise and attention to detail...
Actually that set up is a gradient. Add a third or more “components” to the front or back, all delayed off of the rear most “component”, and you have “end fire”, which makes a more focused and long throw sub lobe. A gradient is usually one “component” behind another, sometimes three total parts including the front one, but the rear one is delayed to the front one, and flipped “out” of phase relative to the front one. The gradient is very efficient space and gear and result wise, as you can shape you lobe with delay time over “100%”, as in MORE than 4.65 feet based on a 60hz center frequency 1/4 wave distance of spacing and delay. 0 delay on the rear speaker or set of speakers which I would call a “component”, and out of phase from the front, would make a dipole, with nearly equal front and rear lobes, and the sides pinched in at the “hips”, where any delay added to the rear “pulls” the rear lobe forward and broadens the front lobe, where your actual distance front baffle to rear baffle being your delay time producing a cardioid pattern at that center frequency, my choice being 60 hz. Delay times above “100%” of that time/distance approaches a hyper cardioid and then a super cardioid, where generally about 150% or more of delay begins countering that shape, and then you realize you (myself included) don’t know hack shit about music, harmonics, sound wave propagation, and that a linear equation would be nice, but noooooo, it’s not that simple. I’ve implemented this “in the field” for about three years now, and one time recently I was mixing a reggae festival, where I was heavily requested to mix and engineer the sound by the artists. Partway through, a good friend grabbed me and asked where my “famous” “nut twitching” bass mix was. I told him I thought it sounded great where I was, front and center, and he agreed, but said where he was it was like no subs at all. I asked him where that was and we walked laterally to about 60 degrees off axis. And I said, “damn, you are totally right”. With my iPad I added inches of delay to the existing setting I had, and sure enough, it broadened the front lobe to include our position dramatically. Damn. It reduced the forward reach, but that was fine, as the park was more broad than deep. None of this is linear though, and you can get some serious anomalies at certain frequencies by distance from your “array” and THAT is where you get humbled. This is why an SVT bass cab makes TONS OF “bass”, but upon real inspection, that bass response is remarkably different as you move around in front of the cab. No free lunches. This guys “trick” is just that, where that mic he was tapping on was in FRONT of the subs, albeit not directly, and he had lots of cancellation there. BUT, how did that sound at that angle from the subs out in the audience? Hmmmm. So you have cancellation, but then synthetically BOOST what was a high passed mic channel, with two layers of pitch shift, to “restore” what you had “robbed”? I’m not buying it. Yeah, he acknowledged the nuances, but…….no free lunches.
I must be missing something. Can anyone explain why the sound of the drums doesn't get lost in the subs, in the same way that the feedback does ?
I think I understand. So because each hit of the drum is only lasts less than a second, it makes it out of the subwoofer. Whereas, under normal circumstances, feedback continues to be produced by the system, therefore it quickly gets cancelled out by the lo-pass filter and pitch shift. Is that correct?
Apa daya browog
AUX is Auxiliary ... FFS people stop saying AWKS. lol ... Sorry, pet peeve of mine.