Eric,
Nearly all instruments have a series of harmonics that go several octaves above the fundamental note. For that reason, you can still recognize the sound of a bass guitar as different than a tuba or a standard guitar even when played back on a clock radio's 3" speaker. Most acoustic, and even electric stringed bass instruments second harmonic (the doubled frequency) is actually louder than the fundamental.
Lobing causes areas of more or less SPL at different frequencies in different areas. Good sound engineers (all the bands you mention have great engineers) walk the venue and make sure the lobes are not offensive.
Unless you move around, you won't be aware the sound of the bass changes a bit in different areas, unless you happen to be in a particular area that happens to be a "one note wonder", which generally does not happen outdoors unless there are walls or structures in certain areas.
Most engineers worth their paycheck find the horizontal comb filtering in the upper range caused by multiple point sources to be a far more objectionable issue than bass lobing. For that reason, the Clair Brothers and Audio Analyst's thousands of S4 cabinets have either had their components recycled, or relegated to "C" list and lower acts, or sold to third world markets. The vertical line arrays now almost exclusively used at all major rock shows largely eliminate horizontal comb filtering in the upper range, and more importantly to the modern concert patron, allow unobstructed sight lines to the huge video displays that allow you to see the pimple on an artist's cheek blown up to the size of an ambulance.
Bass lobing is simply a fact of life, as there is no way to get enough bass from a single point source to fill a venue, and even if there was, you have the phase/time misalignment with the mains (unless the mains output was also emanating from the same single infinitely small point source) at other than one position in the venue. Bass lobing from multiple sources is less a concern than room modes and building reflections.
Art
Last edited:
Hey JAG,
I'll have to look a little bit more into the material you posted, don't want to go into the detail too much right now.
google books preview unfortunately prevents me from seeing the specific page you are linking to....
The section of Bob's book I'm citing actually starts with a thorough treatment of triangulation.
I stand by my claim, according to my sources lambda/2 spacing will also still be sufficient for constructive summation (+6dB) without lobing (but with increased directivity compared to a single acoustic source, or the lambda/4 case).
But I admit that having sub/mains spacing of lambda/4 at crossover frequency will make for a smoother transition in the octave around crossover frequency. lambda/2 would in this case occur one octave above crossover, so throughout the crossover region there should be no lobing. Also, at this point the level offset should be at least 10 dB or more, so the summation ripple will be less than +/- 3 dB (source: summation table in bob's book p.71). So when going further up in frequency and finally getting into spacing >lambda/2, the level offset will be so significant that in practice there will be no lobing anymore since the mains has overtaken.
In case you are unfamiliar, Bob McCarthys book is based on 30+ years of live sound experience, starting with system engineering for the Grateful Dead and a substantial career with Meyer Sound that continues to this day afaik.
Bob provides heaps of simulated examples. A main goal of his method outlined in the book is actually to be able to make reliable simulations for real world PA system design. The page I'm referring to contains illustrations about the effect of source spacing on sound field energy distribution as a function of frequency.
I think you might like the book, although it's quite a heavy read and I don't know if it's entirely relevant to you.
PS: I also appreciate and enjoy exchanging thoughts with you and others on here
Yes you're actually right, sorry about that.
I'll have to look a little bit more into the material you posted, don't want to go into the detail too much right now.
google books preview unfortunately prevents me from seeing the specific page you are linking to....
The section of Bob's book I'm citing actually starts with a thorough treatment of triangulation.
I stand by my claim, according to my sources lambda/2 spacing will also still be sufficient for constructive summation (+6dB) without lobing (but with increased directivity compared to a single acoustic source, or the lambda/4 case).
But I admit that having sub/mains spacing of lambda/4 at crossover frequency will make for a smoother transition in the octave around crossover frequency. lambda/2 would in this case occur one octave above crossover, so throughout the crossover region there should be no lobing. Also, at this point the level offset should be at least 10 dB or more, so the summation ripple will be less than +/- 3 dB (source: summation table in bob's book p.71). So when going further up in frequency and finally getting into spacing >lambda/2, the level offset will be so significant that in practice there will be no lobing anymore since the mains has overtaken.
In case you are unfamiliar, Bob McCarthys book is based on 30+ years of live sound experience, starting with system engineering for the Grateful Dead and a substantial career with Meyer Sound that continues to this day afaik.
Bob provides heaps of simulated examples. A main goal of his method outlined in the book is actually to be able to make reliable simulations for real world PA system design. The page I'm referring to contains illustrations about the effect of source spacing on sound field energy distribution as a function of frequency.
I think you might like the book, although it's quite a heavy read and I don't know if it's entirely relevant to you.
PS: I also appreciate and enjoy exchanging thoughts with you and others on here
Sorry, but that looks like utter bull to me. An amateur doing excessive bass EQ on 80s fusion music mixed and mastered bass-light, hardly representative or applicable to CDs in general.
A professionally mastered CD with good headroom should sound good on any big system. If the genre is geared towards loud playback (dance, rock etc.) engineers WILL check at high sound levels (that's what the big main control room monitors are for...). The only reason why CDs might not translate well on big systems is if the band and producers are stupid enough to jump on the loudness war train and have their music hyperlimited to ridiculously low crest factors
Ref.: Post #438/448
Hi entropy455,
I've been trying to find the time to look a bit more into the subject of building the horn from individual cells, and there seems to be no good reason for not doing it that way. In the example from Post #448 I'm using 10m horn length, and 60,000cm^2 mouth area; if we divide the length into 4 equal segments, and assign "8ft-material thickness" to each side of a square mouth we end up w/ a horn composed of four conical horn segments that can be simulated in Hornresp. The dimensions at the segment break points come from a Hyp system design, thus the individual segments are conical, but the overall horn has a Hyp flare composed of straight sections.
Build from plywood, this might be a starting point for your experimental horn. You will then be able to evaluate how close you are coming to the Hornresp simulation, and how many cells you need to build to arrive at your desired SPL, and if you want to add another 8ft section to increase length and mouth size. You can use the 'Multiple Speaker' tool to see the approximate total SPL for any number of multiples. Please, note that the summation will not give you +6dB throughout the passband of the horn, you'll get about +6dB @ the lowest frequencies (e.g.: 20Hz), and closer to +3dB @ the upper frequencies. (e.g.: 150Hz).
Obviously, you can build the final concrete horn any way you like: round, square, rectangular, as a single internally open horn or from individual stackable cells, 4 cells left and 4 cells right, whatever...it's your baby.
Regards,
P.S.: I hope to find the time to add a sketch later today.
Hi entropy455,
I've been trying to find the time to look a bit more into the subject of building the horn from individual cells, and there seems to be no good reason for not doing it that way. In the example from Post #448 I'm using 10m horn length, and 60,000cm^2 mouth area; if we divide the length into 4 equal segments, and assign "8ft-material thickness" to each side of a square mouth we end up w/ a horn composed of four conical horn segments that can be simulated in Hornresp. The dimensions at the segment break points come from a Hyp system design, thus the individual segments are conical, but the overall horn has a Hyp flare composed of straight sections.
Build from plywood, this might be a starting point for your experimental horn. You will then be able to evaluate how close you are coming to the Hornresp simulation, and how many cells you need to build to arrive at your desired SPL, and if you want to add another 8ft section to increase length and mouth size. You can use the 'Multiple Speaker' tool to see the approximate total SPL for any number of multiples. Please, note that the summation will not give you +6dB throughout the passband of the horn, you'll get about +6dB @ the lowest frequencies (e.g.: 20Hz), and closer to +3dB @ the upper frequencies. (e.g.: 150Hz).
Obviously, you can build the final concrete horn any way you like: round, square, rectangular, as a single internally open horn or from individual stackable cells, 4 cells left and 4 cells right, whatever...it's your baby.
Regards,
P.S.: I hope to find the time to add a sketch later today.
Attachments
Bob,
As for sounding good, and having decent bass levels, they are two completely different things.
Many master tape recordings were made for vinyl records, and were purposely high passed to both maximize signal to noise and volume level, and have little content below 50 Hz. The same master recordings were then used for the CD, even though there are virtually no restrictions from 1 Hz to 20 kHz on a standard CD, and CD SN is some 40 dB better than virgin vinyl.
Some albums by bands that care enough to pay for re-mastering or re-mixing have re-released CDs, DVDs and Blu Ray with a quite different sound than the originals.
Listening on a system with a sub-woofer flat to 16 Hz, my top speaker's 50 Hz response is completely adequate for reproduction of those "classic rock" CD releases, while the same re-recordings often have content an octave or more below, I can hear (and see on an RTA) content in the low 20 Hz range regularly.
It is also funny to hear 16-25 Hz air handling noise that was recorded by engineers mixing digital recordings that failed to monitor with speakers that would reveal the problem, or just don't have hearing capable of hearing low- I'm often surprised to find that others can't hear the low frequency from an idling truck, or car door slams from a block away, while the noise is as loud to me as their conversation.
Thanks for the earlier compliments.
I don't hold "6o6" McCarthy's engineering for the Grateful Dead and residence with Meyer Sound against him, he has written loads of good stuff, most of which I agree with, in spite of his use and contributions to John Meyer's "state of the art" over-priced, under-performing systems ;^).
Next time I'm up that way I should see if Will Lewis, one of Meyer's marketing guys could set up a meeting with John as he suggested would be fun a few years ago, and maybe "6o6".
I have not seen him since the god-awful sounding Dylan/Dead show at the Minneapolis "HumpDome". Actually, it was a good mix, but the different vintage Meyers MSL3 cabinets used to cover various portions of the arena all sounded different, and none of the guys at FOH could be bothered to walk the venue and correct the EQ, they just sat in their prime "power alley" position with SIM telling them all was good, when it wasn't most everywhere else. With most of the band, engineers, and crowd "baked", I was probably the only guy out of the 40,000 people there that noticed.
"Good enough", what a long, strange trip it's been
.Art
Last edited:
I heard similar unmonitored external noise during a TV interview. There was a truck idling outside !
Of course, per se the two are not correlated at all - or well, it depends on genre and personal taste.
again, depends on genre and time frame. Probably correct for commercial album releases (think loudness war! Who is louder?), but hardly a universal truth (granted you didn't imply so either
). You most probably wouldn't find such a bass cut on old R'n'B, Soul, Funk, HipHop, and certainly not on jamaican stuff Also agree on this one. The red part of your comment is the main reason that I (sorry, edit: ) got so triggered by hollowboys post. I wanted to ensure that OP doesn't get even more confused.
Aaargh, such a long post.... summa summarum: there is nothing wrong with the CD as a format, or even with vinyl for that matter.
Program material varies for diverse reasons, so if you feel like you want to / have to EQ, just do it.
Last edited:
nope. look at FOH side racks. Full of compressors and limiters.
EDIT: sorry, meant to say tft screens of digital mixers
EDIT2: let's say CDs don't NEED to be more compressed than concerts, but of course they CAN be, if the artist/producer/A&R are stupid enough
If genre and artistic vision allows, a studio mix can be crafted to surpass a live performance in dynamic range, because you can exploit the headroom to the very last decibel, and have in general more control while doing so. Even doing it the old fashioned way (like Queen, 10 hands on a mixer).
Of course if you are deaf like Ulrich&Hetfield&their producer, you will make stupid decisions in the mastering process *cough* *deathmagnetic*
Last edited:
Not all CDs sound good. I have a Jethro Tull CD that sounds awful – muddy lows, dull highs. This music was clearly taken straight off a record that was first run over by a truck, then used as a Frisbee by kids on the playground, then buried in the sand for 17 years, then run over by a truck again. . . . Actually, it sounds a little bit worse than that.
On my wife’s stereo, some Offspring music has a relatively “quiet” bassline. Whereas the new TRON soundtrack will rattle the windows on the same setting. It’s normally an easy fix – just a small gain adjustment on the crossover. I like CDs, because it's essentially an original signal every time (medium does not degrade with use - like tapes and records. . . .) System EMI is also significantly reduced - while within the digital domain. . .
On my wife’s stereo, some Offspring music has a relatively “quiet” bassline. Whereas the new TRON soundtrack will rattle the windows on the same setting. It’s normally an easy fix – just a small gain adjustment on the crossover. I like CDs, because it's essentially an original signal every time (medium does not degrade with use - like tapes and records. . . .) System EMI is also significantly reduced - while within the digital domain. . .
Bob.Originally Posted by weltersys
Many master tape recordings were made for vinyl records, and were purposely high passed to both maximize signal to noise and volume level, and have little content below 50 Hz.
again, depends on genre and time frame. Probably correct for commercial album releases (think loudness war! Who is louder?), but hardly a universal truth (granted you didn't imply so either
Sorry to tell you (you must be rather young) but regardless of the year and genre, commercial 33 1/3 RPM vinyl album releases seldom have much content below 50 Hz, nor do the CDs made from the same masters.
12" 45 RPM single records coveted by DJs are the exception, the extra speed and size of the cut allows much wider grooves, increasing low frequency headroom considerably, but still a long way short of CD frequency response.
There are a few artists like Pink Floyd that sacrificed volume for low frequency extension on their vinyl album releases, and did separate mixes for CDs from the get go.
Although there is no lack of bass on old R'n'B, Soul, Funk, HipHop, and Sly and Robbie's Jamaican stuff, it does not go low, unless you EQ it so. Also remember (or learn) that virtually no "old" bass players used five string basses, the low "E" used on a 4 string bass is 41 Hz, the low "B" now used by every Joe Shmoe 5 string bassist is 31.5 Hz.
As the pros that cater to the EDM scene say "25Hz is the new 50 Hz".
Back in the day of 50 Hz roll off on records, very few PA systems had much response below 50 Hz, but when you ran into a PA with 8 foot bass horns that could do a solid 120 dB at 35 Hz, it was a "wow" experience.
Now, that same 120 dB at 35 Hz seems kind of "ho hum" if you have experienced 145+ dB at 25 Hz...
The OP's quest to use ancient technology to do what can easily be done now with lots of power and excursion is fun to read about, and has really brought me down memory lane.
Long live Entropy Eric!
Art "Use what Works Best, Leave the Rest to Museums, or to Other Folks to Try" Welter
Last edited:
Back around the turn of the century I converted my live tape recordings dating back to 1978 to CD. Other than some print through, all the cassettes sounded pretty much like they did when they were recorded- most of them sounded great, since they were not band passed like recordings made for vinyl release.
Since the release of CDs, I have had to dispose of many of my favorites that became unplayable do to surface scratches. I now convert all CDs to hard drive as soon as I make or purchase them to have a better archival medium.
As much as I like the convenience of digital recording and processing (your previous suggestion about purchasing digital delays, rather than DSP was another trip down memory lane...) the many different formats and failure to adopt specific "Q" features has created a "digital Tower of Babel" (Bennett Prescott's phrase) that was largely not a worry in the days of analog gear, iron men, and wooden speaker cabinets.
Art
Proper design (fixing issues in the design phase before they are real physical problems) is not mission creep. The fact that large concerts by necessity have absolutely huge audience coverage areas and the need for consistent coverage and consistent spl are somewhat conflicting goals has nothing to do with OP's system. OP should be able to cover his relatively small audience with somewhat consistent spl with a minimum of seat to seat variation if it's designed properly in the first place.
Ok, here's the first installment of the coverage sims using Danley Direct. This program is super simple to use and it's actually kind of fun. It took me less than 1 hour to learn to use the program and to set up this model. (That includes watching all the tutorial videos.) It took me about as long to capture, edit, save, host and post these images as it did to learn the program and set up this model.
SO I've set up a 60 x 60 foot audience area, dead center in the middle (where all the axis lines converge) is OP's hot tub sweet spot, 30 feet out from the subs and on axis with the sub centerline.
I chose to show two configurations here, subs only. The first is OP's suggested 60 foot center to center sub distance with dual subs. The second is the same subs stacked in a central location. I used the BC 418 subs and I used four of them simply because that will give me a huge stack of subwoofers. In the case of the dual separated sub stacks I toed them in to point at the hot tub sweet spot. For these initial sub only sims I've raised the sub crossover to 200 hz so it isn't impacting the bandwidth too much so you can see the full extent of what's going on.
So here's the first configuration, OP's 60 foot center to center distance with dual subs. I've shown the coverage map at 20, 25, 31, 40, 50, 63, 80, 100, 125 and 160 hz. In the 20 hz map I've included the color legend, so you can see the difference between dark read and yellow is about 15 - 20 db. The 20 hz coverage map is a nice visual description of the power alley concept and also shows clearly defined lobes off to each side.
Up at 160 hz the comb filtering is so severe and so closely spaced that you have 20 db nulls approximately every 4 feet as you move horizontally across the map. That means that unless the hot tub is less than 4 feet wide, some of it (at either end) is going to be directly in a 20 db null at 160 hz. As frequency decreases the comb filters spread out a bit so at 100 hz you could probably get a whole 8 foot wide hot tub into the sweet spot and the nulls would be just past the edges of the hot tub.
This is not really what I would call ideal. If you walk slowly across that 60 foot area in real life listening to full bandwidth pink noise you would definitely hear the 20 db nulls.
So now let's look at the single sub in a central location concept. This sim has the same amount of subs, I just stacked them all. There's no need to show a lot of frequencies this time, because they are all pretty much the same. You get even coverage everywhere, no comb filtering, no nulls.
Now like I said, I don't know if my big stack of horns in this sim has anything even approaching the directivity of your full size horn and there's no real help file or instructions that can help me answer that, so I'll have to do some playing around to see if large stacks increase directivity in this program and a full size horn can be approximated properly. Either way, I think the directivity for your first full size horn sim was -6db at about 60 degrees, so these sims should be reasonably accurate for a large portion of the 60 x 60 coverage map shown.
Later on I'll do some sims that introduce the mains. If there's any specific DSL model you'd like to see let me know, otherwise I'll be sticking to the program default (I think it's SH50 maybe) and using 1 per side at a 6 ft height.
SO I've set up a 60 x 60 foot audience area, dead center in the middle (where all the axis lines converge) is OP's hot tub sweet spot, 30 feet out from the subs and on axis with the sub centerline.
I chose to show two configurations here, subs only. The first is OP's suggested 60 foot center to center sub distance with dual subs. The second is the same subs stacked in a central location. I used the BC 418 subs and I used four of them simply because that will give me a huge stack of subwoofers. In the case of the dual separated sub stacks I toed them in to point at the hot tub sweet spot. For these initial sub only sims I've raised the sub crossover to 200 hz so it isn't impacting the bandwidth too much so you can see the full extent of what's going on.
So here's the first configuration, OP's 60 foot center to center distance with dual subs. I've shown the coverage map at 20, 25, 31, 40, 50, 63, 80, 100, 125 and 160 hz. In the 20 hz map I've included the color legend, so you can see the difference between dark read and yellow is about 15 - 20 db. The 20 hz coverage map is a nice visual description of the power alley concept and also shows clearly defined lobes off to each side.
An externally hosted image should be here but it was not working when we last tested it.
Up at 160 hz the comb filtering is so severe and so closely spaced that you have 20 db nulls approximately every 4 feet as you move horizontally across the map. That means that unless the hot tub is less than 4 feet wide, some of it (at either end) is going to be directly in a 20 db null at 160 hz. As frequency decreases the comb filters spread out a bit so at 100 hz you could probably get a whole 8 foot wide hot tub into the sweet spot and the nulls would be just past the edges of the hot tub.
This is not really what I would call ideal. If you walk slowly across that 60 foot area in real life listening to full bandwidth pink noise you would definitely hear the 20 db nulls.
So now let's look at the single sub in a central location concept. This sim has the same amount of subs, I just stacked them all. There's no need to show a lot of frequencies this time, because they are all pretty much the same. You get even coverage everywhere, no comb filtering, no nulls.
An externally hosted image should be here but it was not working when we last tested it.
Now like I said, I don't know if my big stack of horns in this sim has anything even approaching the directivity of your full size horn and there's no real help file or instructions that can help me answer that, so I'll have to do some playing around to see if large stacks increase directivity in this program and a full size horn can be approximated properly. Either way, I think the directivity for your first full size horn sim was -6db at about 60 degrees, so these sims should be reasonably accurate for a large portion of the 60 x 60 coverage map shown.
Later on I'll do some sims that introduce the mains. If there's any specific DSL model you'd like to see let me know, otherwise I'll be sticking to the program default (I think it's SH50 maybe) and using 1 per side at a 6 ft height.
Now it should be noted that all the images I showed were 1/3 octave, the tutorial specifically says that 1 octave is a bit more representative of what we actually hear, and that will make things look better. Here's the dual sub shown at 160 hz again, this time at 1 octave instead of 1/3. Indeed it does look better (it's now a pretty little spewing volcano, you could frame that and put it on your wall), but the two prominent comb filter nulls about 2 feet off the on axis centerline are still particularly worrying. That's prime real estate, almost directly on axis, yellow lines in that area are a problem.
About the legitimacy of this simulator's results - I'm not sure how accurate it actually is, but this is the software that DSL uses to design systems for full size arenas with up to dozens of individual coverage zones (watch the tutorial videos for a peek into that world). And it's the software that dealers are supposed to be using to help customers design systems for their unique goals and environments to make sales.
I'm not sure I'll be able to simulate the directivity of a full size horn properly with this software but I'm not at all worried about the accuracy of the results in the middle of the map. If directivity is wrong the accuracy will get less and less moving from the middle out to the edges of the map, but for the OP's small audience area there shouldn't be much problem even if the sub horn directivity isn't accurately represented.
An externally hosted image should be here but it was not working when we last tested it.
About the legitimacy of this simulator's results - I'm not sure how accurate it actually is, but this is the software that DSL uses to design systems for full size arenas with up to dozens of individual coverage zones (watch the tutorial videos for a peek into that world). And it's the software that dealers are supposed to be using to help customers design systems for their unique goals and environments to make sales.
I'm not sure I'll be able to simulate the directivity of a full size horn properly with this software but I'm not at all worried about the accuracy of the results in the middle of the map. If directivity is wrong the accuracy will get less and less moving from the middle out to the edges of the map, but for the OP's small audience area there shouldn't be much problem even if the sub horn directivity isn't accurately represented.
Last edited:
Ok, moving right along let's add some mains and see what happens.
We are going to compare OP's proposed configuration vs mine.
OP - 60 foot sub center to center distance, mains 30 foot center to center distance
Me - central sub location, mains 16 feet apart
Both configurations shown at 63, 100 and 160 hz
Both configurations have a global crossover setting of 100 hz, which means all subs are low passed at 100 hz and all mains are high passed at 100 hz, 24 db/oct BW.
Now the frequency response of both configurations with the virtual mic placed right in the center of the hot tub 6 feet high.
Note that when you are in the sweet spot it makes little difference which configuration you choose, the frequency response is lovely. But as soon as you take a step away from the direct center of the coverage map, it starts to make an incredible difference.
Also note that the use of 4x BC 418 subs and 2x SH50 mains gives a nice 20 db bump in the bass, like OP wants, so I didn't adjust amounts of speakers to give a flat frequency response.
Also, this bumped up bass is the reason the 63 hz coverage map is a lot more red and the 160 map is a lot more yellow and green. That's because 63 hz is a lot louder than 160 hz. But the important part is the smoothness of coverage.
OP's proposed setup produces really pretty starburst patterns, while my proposed setup is pretty boring to look at, but in this case boring is VERY GOOD, it's very even coverage while OP's proposed setup is an unholy mess of hot spots and deep nulls.
How much does any of this matter? You have to decide for yourself by testing it, setting up an example system, playing full bandwidth pink noise and walking around while listening and/or measuring.
We are going to compare OP's proposed configuration vs mine.
OP - 60 foot sub center to center distance, mains 30 foot center to center distance
Me - central sub location, mains 16 feet apart
Both configurations shown at 63, 100 and 160 hz
Both configurations have a global crossover setting of 100 hz, which means all subs are low passed at 100 hz and all mains are high passed at 100 hz, 24 db/oct BW.
An externally hosted image should be here but it was not working when we last tested it.
Now the frequency response of both configurations with the virtual mic placed right in the center of the hot tub 6 feet high.
An externally hosted image should be here but it was not working when we last tested it.
Note that when you are in the sweet spot it makes little difference which configuration you choose, the frequency response is lovely. But as soon as you take a step away from the direct center of the coverage map, it starts to make an incredible difference.
Also note that the use of 4x BC 418 subs and 2x SH50 mains gives a nice 20 db bump in the bass, like OP wants, so I didn't adjust amounts of speakers to give a flat frequency response.
Also, this bumped up bass is the reason the 63 hz coverage map is a lot more red and the 160 map is a lot more yellow and green. That's because 63 hz is a lot louder than 160 hz. But the important part is the smoothness of coverage.
OP's proposed setup produces really pretty starburst patterns, while my proposed setup is pretty boring to look at, but in this case boring is VERY GOOD, it's very even coverage while OP's proposed setup is an unholy mess of hot spots and deep nulls.
How much does any of this matter? You have to decide for yourself by testing it, setting up an example system, playing full bandwidth pink noise and walking around while listening and/or measuring.
Last edited:
An adjustable digital time-domain delay circuit will be processor based. Is that not by definition a DSP? It seems that you are suggesting that I'm selecting obsolete technology for my project. What is the difference between a digital delay, and a DSP?
I was considering the Yamaha DME64N. I like the idea of a computer interface. I can load my music up onto a computer, and never touch the CD again. What is your opinion on this digital delay?
Mini dsp will do your delay, active crossover and parametric eq (and other stuff) for as little as $100 and there's several models to choose from at different price points and different feature sets. It's also computer interfaced but once set can be used with or without the computer.
I don't know anything about the Yamaha other than it looks really expensive.
I don't know anything about the Yamaha other than it looks really expensive.
This is the one I'm thinking about, it's the cheapest one and I think it will do everything I need but I still have to read the manual to confirm.
https://www.minidsp.com/products/minidsp-in-a-box/minidsp-2x4
https://www.minidsp.com/products/minidsp-in-a-box/minidsp-2x4
Last edited:
- Status
- This old topic is closed. If you want to reopen this topic, contact a moderator using the "Report Post" button.