Which Sounds Better Midas M32 vs Behringer X32 vs Copper Wire

No matter how great digital audio is, a short copper wire blows it away for pure transmission of wide frequency response, low noise, reduced latency and pretty much every spec we can measure. The question is whether those differences are relevant.
Lets do some tests and take a look.
00:00 Intro
00:35 Digital vs analog
01:37 The setup
02:05 Inexpensive digital vs analog
02:36 Sine wave direct vs Xenyx
03:45 Sine wave X32 and latency
04:29 M32 sine wave and latency
04:45 100hz square wave direct
05:18 100hz square wave xenyx
05:50 X32 and M32 100hz square wave
06:16 Compare the 4 square waves
07:43 M32 vs X32 latency
08:52 Nyquist freq M32 vs X32
10:00 Upper freq limits of X32, M32, and Xenyx
11:20 20K square wave all three consoles
12:25 Digital is not a piece of wire
14:12 Relevance, swimming and perception
16:30 Outro
#1 Behringer X32 Compare Digital vs Analog Metering & Output Levels
Members - • Behringer X32 Compare ...
Public - • Behringer X32 Digital ...
#2 Midas M32 Vs Behringer X32 Sonic Differences
Members - • Comparing Midas M32 an...
Public - • Midas M32 Vs Behringer...
#3 Midas M32 vs Behringer X32 mic pre noise comparisons
Members - • Midas M32 vs Behringer...
Public - • Midas M32 vs Behringer...
#4 Midas M32 vs Behringer X32 LF Overload and Muting
Members - • Midas M32 vs Behringer...
Public - • Behringer X32 Vs Midas...
#5 X32 vs M32 Preamp Noise & Gain vs Trim
Members - • X32 vs M32 Preamp Nois...
Public - • X32 vs M32 vs Xenyx No...
#6 X32/M32 AES50 Latency and Chat
Members - • X32/M32 AES50 Latency ...
#7 X32/M32 AES50 Cable Type and Length
Members - • X32/M32 AES50 Cable Ty...
#8 Midas Behringer X32/M32 Digital Dreams of Being a Short Copper Wire
Members - • Copper Wire vs Midas M...
#9 X32/M32 Fader and Mute Pings
Members - • Behringer X32 Midas M3...
#10 X32/M32 Scoping and Loading the Outputs +26
Members - • Scoping & Loading Down...
#11 X32 vs QU32 Metering, clip and max output
Editing
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[@josevillanueva9091]

Haha! I love how you literally schooled the “why are you using a cheap board” people with math, science, and your gear. Best part is you always stat humble. I love your videos man! I have a lot to learn and I always appreciate your knowledge.

[@DaveRat]

Thank you Jose!

[@ziofrenko]

the fact is that with a cheap digital board only I can make a band sound better that with a cheap analog board only... no doubt on this.

[@DaveRat]

Interesting. I have no problem with either, analog just make things easier to access and see what's going on and setups up faster and it's easier to find routing issues. How about we have a challenge? You're on a digital board and I go inside and change random things on various levels and I'm on an analog board and you can change whatever you want. Then we see who can fix things faster! Also I can mix in the dark on analog using no board labels. Are you able to do that on digital?

[@ziofrenko]

@@DaveRat digital win: is enaugh to save the scenes :P. Sorry :) I love mix with analog board... I'm with you on this kind of question. but you can mix with dozen of racks of luxury outboard, and i'm not. I have to bring board in place... an x32 rack is better than a GB8 with another case of outboard (if they make it available to you).

[@randomschittz9461]

@@ziofrenko You didn’t answer the question directly, you just tried to divert to other features. That kind of proves that you’re not familiar with analog and are merely trying to justify your purchase to someone who has used both mediums for decades.

[@parkeranderson1172]

Adding on what’s already been said, the reason the Digital’s don’t output the 20k square is because all the higher harmonics for the Fourier series are being filtered out (above nyquist). Great vid!

[@DaveRat]

Yes. Having the low pass Nyquist filter as far as possible or at least relatively far from the upper passband is desirable.

[@TomiBonTomi_2.0]

Man, this is phenomenal! Thank you for doing this! Instant sub, easily earned. I'm gonna binge watch you now, no joke! Looking forward to these KZbin highs and lows!

[@DaveRat]

Thank you Toni!

[@Edwin-van-der-Putten]

Another great video with unexpected outcomes! Great! Thanks, man! 🙂

[@DaveRat]

Thank you Edwin!

[@artysanmobile]

By the time you’re done with this series, we’re all going to be using banks of Xenyx mixers, sharpies and paper tape. Viva old school!

[@DaveRat]

Ha!!

[@waynerowlands1224]

Mr Dave! I’m a keen soundo, well I started playing bass in the 80s then from the early 90s have been involved in sound production too. I didn’t study as a tech but have worked a few good ones. I understood the structure & reasoning of your tests & comparisons. Was impressed at how you led the explanation of comparing equipment because it was clear & objective. I was talking out loud about the scope readings as you showed them. I can’t operate a scope but I get the language. Mate, at the end your summation of human perception was marvellously elegant & stunning! Wow… Thanks from a crusty old bloke that’s now buzzing like a teen!?!

[@arnoudjanschut]

Fun these tones you put on the end of the video.

[@DaveRat]

👍👍👍

[@jakewilliams2263]

Another great video from Dave!

[@DaveRat]

👍👍

[@peehandshihtzu]

So much to learn here! Great video Dave, whomever you are, LOL (wink, wink, love the shirt). Hey Man, thanks for doing what you do and happy holidays! :)

[@DaveRat]

Happy Holidays Peehand!

[@shaneelson]

"... does it need to be fun, intense or memorable." Exactly! Wishing you and your's a safe and happy holiday Dave.

[@DaveRat]

Happy holidays and thank you Shane

[@johnviera3884]

another amazing video. you’re the best

[@DaveRat]

🤙👍🤙

[@dngrusd]

Great video! It's questionable how much of that our ears actually discern, but knowing is better than not knowing. Things aren't always as they appear. Seeing the waveforms on the scope makes the difference between analog and digital crystal clear.

[@DaveRat]

Cool cool Danny

[@whiteclash]

Words of wisdom here in your conclusion. Thank you Dave!

[@DaveRat]

👍🤙👍

[@asilva_media]

love the shirt right off the bat 🤣🤣 also these consoles are surprisingly faster than the CL1

[@DaveRat]

🤙👍👍👍

[@TheLordcasio]

Holy Cow! I've been using a RAT Sniffer on the job (Reality TV house builds) for 16 years and I never knew, that beyond creating an amazing tool, you are such a prolific sound engineer. I've been going down the rabbit hole of your KZbin vids and have (am still) learning so much. Ass kiss-cry aside, I would be curious to see how much further latency and over modulation occurs on these digital (or even analog) mixing consoles, once you start adding effects (both real time-compression/EQ and time based). Thanks for sharing your smarts! ps:: when are you going to do a 1/4" TRS interface for your SoundTools CAT system?

[@stehfreejesseah7893]

Love the sniffer sender.

[@DaveRat]

So cool and thank you!!!

[@ricardohavinga]

Loved your closing statements, very insightful. Considering a Wing board and wonder if they have made any improvements. I would assume the sampling frequency was the major cause to the disappearance of higher frequencies on the digitals. The ringing on the Squarewave really had my curiosity as to how this would show up in live situations. The best part was using the cheap analog board as a comparision. Brilliant. I would assume a 96 khz sample rate would have less roll off as the multiplexing of the channels is on a broader width. Would love to see an A&H performance test on these tests.

[@mrpagliero]

nice work ,keep it up!

[@DaveRat]

👍

[@57kod]

In the synthesis modular world we use a lot of square waves. So it is nice to see the results, again, awesome video. Just fir curiosity, have you done a similar test on a Digico or something similar? Thank you!

[@DaveRat]

Cool cool, will test other boards as I get access and they come back from gigs

[@68Snaps]

Great video, super interesting! But I really wonder if switching the M32/X32 to 96K sampling would have changed the square wave response... It could be a nice test (48K vs. 96K). Anyway, thanks for the really interesting video! Happy new year to you and all your subscribers :-)

[@DaveRat]

They don't switch to 96k. Only 44.1 and 48k and thank you!

[@68Snaps]

@@DaveRat That does explain why you didn't test it... ;-)

[@DaveRat]

👍👍👍

[@LucasWasson]

Another really interesting video. Is there a story behind the shirt?

[@wesleychapman8842]

I knew it your a legend 🤘🤘🤘🤘🤘🤘 your still on my list of people that make shows happen to get a autograph the un sung heros that fix and make a show happen thank you

[@DaveRat]

Thank you Wesley and honored!!!

[@daleclyne6057]

Thanks again Dave, very interesting. How about a video compairing the X and M32 against a high end Digital console?

[@DaveRat]

Working on testing an SD11

[@wilcandou]

Love your inquisitive nature mate. 👍

[@DaveRat]

Thank you Wil!

[@robsidle5200]

Wonderful series, Dave. Thank you so much! Would you try this same 20k square wave test with TWO Analog to Digital to Analog conversions. Tone Generator > X32 > XLR> M32>Scope. I would be curious what two unlocked sample rates do to the signal. What about different clock rates? So many places have a digital board, feeding analog to a signal processor, and then analog to the amps, which have DSP themselves. Thats 3 A/D/A conversions between the mic and the speakers! That must have some effect on the signal.

[@DaveRat]

Unlike digital, analog signals do not benefit from being sync clocked together. Syncing clocks is to get the digital samples aligned, but once the signal is analog, there are no digital samples to align, even if the signal was previously digital or will be digital. Syncing clocks is for two digital units to align when transferring a digital signal between them.

[@Twongo]

That's it. I'm not splitting my overheads between consoles any more! This series is really great, Dave. I just got a gig in a theater that uses an M32 and the info you cover really can't be found anywhere else. (Other than doing the tests ourselves.) Because of these videos I have a more intimate relationship with the hardware than I would otherwise. Thanks! As far as compensating for the gear goes, I just twist knobs until it's right. But I feel that the knowledge is great to have at the edge of the mind. For instance, it's easy to hit a path too hard and wonder why I'm struggling to get what I want out of it. Then I remember how much headroom I have in the digital circuit and back the gain down and find that everything after that acts much more civilly. I'm also developing a habit of leaving some silence before changing channel states. These desks do like to exaggerate certain noises. At some point I may find myself pondering how close I am to the Nyquist freq. I'll know where I got the idea.

[@DaveRat]

Cool stuff!

[@RaymondSolo]

One of the best presentations in this series Dave! Amazing what a simple piece of wire can reflect! Just want to add, I’m betting that they capped the response at 24k to maximize processing power and capabilities in the common 20-20k frequency range.

[@DaveRat]

Thank you. The 24khz limit is due to the 48 k sampling rate. A 96k sample would force a 48khz limit.

[@Labaris]

Great video! Thanks

[@DaveRat]

🤙👍🤙

[@owenjbrady]

I am glad behringer is out there putting out good affordable gear, everything I have ever used by them as been great I can't complain at all. hope they stick around!

[@DaveRat]

I do like affordable gear that works well. My main issue with them is when they copy other gear and sell cheap knock offs. Make good gear. Make cheap gear, don't copy other companies gear.

[@randomschittz9461]

Yeah, I think I’ve had one bad piece of gear from them over the years, and Dave is right.. they got popular from copying DBX and got sued as well.

[@DaveRat]

🤙👍🤙

[@ytscksdabig1]

Your videos are amazing.

[@DaveRat]

Awesome and thank you!

[@dionaldtubang2894]

That analog rounding off the edges for hf square wave is just sum slew rate of all the solid-state parts in the circuit. Thanks for the video showing these differences. Very helpfull

[@tasteapiana]

Can't wait for you to get your hands on a set of top tier AD/DA converters; Prism, Lynx, SSL, Crane Song, etc. I think a lot of people would be shocked to see what is (and is not) actually happening in those $3,000 to $10,000+ units.

[@DaveRat]

👍

[@squidcaps4308]

Not a lot, not enough that it is worth the investment.

[@DaveRat]

Its tough to test but will ponder doing so at some point

[@stehfreejesseah7893]

I also would love to see some of the higher end gear in this situation, although I have a pretty good idea of how much it matters...

[@marcobalducci4789]

wonderful video, Mr. Rat. thank you.

[@DaveRat]

Thank you Marco!

[@COGSCNY]

Universe: Dave is muddying up the signals with that analogue board. Dave: Hold my Scope

[@DaveRat]

👍👍

[@adrianscherer8556]

Thanks for the overall intel. Looking forward to use this infromation in real world application. Cheers!

[@DaveRat]

👍

[@joshlustig3327]

I’d love to see the effect that gain has on all of this. The biggest pitfall of the M32/X32 (at least to my ears) is the sound of the head amps. They seem very inconsistent, I often find myself getting very different frequency responses with different gain (with the same source, even with IEMs/no other processing/all other variables removed).

[@kevinpetit9886]

Great video. 😃👍♥️

[@DaveRat]

👍👍

[@paulmorrey733]

Thanks Dave

[@DaveRat]

Cool cool Paul!!

[@juneaftn]

Great video! I'd also like to see any differences in the sub-bass frequencies like 50-60hz. I guess there might be between analogue and digital.

[@DaveRat]

Near identical except the digital board tend to reproduce frequencies below 10hz better

[@juneaftn]

@@DaveRat Oh, I thought most analogue mixers have high-pass filter to block DC at around 20hz or so and you will see phase issues above it.

[@nickwallette6201]

@@juneaftn I would imagine the digital boards probably do, too. For one thing, a lot of ADC/DAC ICs are single-supply, so there's a virtual ground at 1/2Vcc. That requires high-pass filtering to separate signal from supply. I don't know what the M32/X32 uses, but I would be surprised if it were anything other than +3.3V or +5V supply.

[@DaveRat]

Digital boards are really good at reproducing very low frequencies.

[@patdry]

Enough about the consoles… what a great shirt!! 😂

[@DaveRat]

Ha!! Thank you.

[@petruciucur]

That is quite amazing

[@DaveRat]

🤙🤙🤙

[@SuperMcgenius]

Thank you, what my ears told me years ago.👍😇

[@DaveRat]

🤙👍🤙

[@laurencevanhelsuwe3052]

As a "synth head" (synth nerd) I can assure you that the exact frequency spectrum of square waves matter to our ears quite a lot. Most synthesizers allow the output of pure square waves, so in lots of synth sound production, square waves are really bread and butter starting points (along with triangle, saw and noise). Your video is a real eye-opener on this topic. Thanks for making and sharing!

[@DaveRat]

👍👍

[@ferociousmullet9287]

Sorry but you are wrong, not a single analogue synth has ever out put a perfect square or sawtooth or triangle waveform. Not once. Most digital synths do not do this either due to limitations in sample rates and filtering. Which is the same problem we are seeing on these mixers. The 'gibbs' phenomenon. You are objectively wrong.

[@laurencevanhelsuwe3052]

@@ferociousmullet9287 Shot down in raging flames for the use of the "pure" adjective. You are quite possibly objectively right (cfr. Carlsberg advert)

[@caseykittel]

I’m picturing dave and flea having really deep conversations from time to time.

[@DaveRat]

👍

[@joelcelso4455]

love that tees too

[@DaveRat]

🤙👍🤙

[@caseykittel]

This is exciting stuff dave! I love it. Side question - I can’t find a good tutorial on how to properly coil a 150’ snake for transport and storage. I know this is simple stuff, but I can’t find a good video on it! I don’t want to damage the snake, or almost more importantly, I don’t want to waste time during setup fighting with a tangled, hard to lay flat cable. Help! Thanks.

[@DaveRat]

Cool cool, I will ponder that.

[@devin1leon]

Hello Dave, I thoroughly enjoy your videos. Thank you for doing them. Would it be possible for you to compare an M32 to a Wing console? Also a 338 to an S6L? Sincerely, Devin Clayborn

[@DaveRat]

Hmmm, will see. I am more into disproving or confirming myths and and clarifying misunderstandings than I am into product review or comparison. I did the M32/X32 comparison because there is so much false info and conjecture floating around out there, thought I'd add some clarity

[@lightbonkey]

​@@DaveRat But I am still confused, which good console can we use between M32 vs X32?

[@adamcoe]

Your last comment is VERY important. Yes, it's demonstratable that analog gear surpasses the capabilities of digital in terms of frequency response, but how does it translate in a real world way when you're trying to make a band sound good. It might matter a lot for some things, it might not matter at all for others. That being said I would imagine the vast majority of applications are in the latter camp. And I'd even go so far as to say that's not only live but in the recording world as well. As you say, there's a tiny sliver of listeners that are in a position to even experience the extreme extreme high fidelity of the most high performance systems, like what you'd have at say, Skywalker Sound or something of this nature, but in all probability, as long as you know the limitations of your system (whatever they might be), there's always a best case plan to simply make things enjoyable for the audience, which is the goal of any live sound engineer or any record.

[@jedi774]

I love the stuff Dave does. He takes theory... and shows it’s relevance to real world applications. I’d rather have more flexibility and relative control over absolutely perfect reproduction... but that’s just because that’s what I need and want in my real world application. Others may need a more perfect representation... I love that Dave doesn’t really throw judgement at the results. Just measures and reports and talks about the facts.

[@DaveRat]

👍

[@zambotv8150]

Big love from Scotland Dave, please run for president

[@DaveRat]

Thank you Phil and oh, that does not sound like fun gig. I'd rather head to Scotland for a drink and a surf!

[@zambotv8150]

@@DaveRat You would make a great president, much more capable and interesting than the guys that currently do it. You are welcome here anytime buddy, the kettle's always on for men of your stature !!!!

[@DaveRat]

I will stick with president of Rat Sound! Yay!

[@zambotv8150]

@@DaveRat Har!!!!! Probably a good choice,

[@DaveRat]

👍👍👍

[@sartoriusrock]

The increased rippling in the tops and bottoms of the square wave seen when the frequency is increased is an inherent distortion due to Nyquist and anti-aliasing filters. The higher frequencies needed to create the sharp edges of a square wave at ~11k and up are above Nyquist and are thus filtered out by the anti-aliasing filters, thus the ripple is seen. Comment added slightly before finishing the vid, sorry if I repeated something you already mentioned!

[@robertlofgren6834]

Yes, mathematically a square wave is an infinite number of sinewaves. Since all digital mixers are hipass limited only sinewaves below half the nyquist freq will show. A digital mixer can never reproduce a perfect square wave due to always having to work with a bandpassed signal.

[@kensmith895]

@@robertlofgren6834 True - but also could the anti alias filter be ringing?

[@sartoriusrock]

@@kensmith895 If you look at the resulting waveforms that result from various steps of a reimann sum equation that leads to a square wave, the 5th or 6th step yields a graph that looks a lot like the rippled waveform coming from the console. It is definitely possible that the anti-aliasing filter has ringing that can be measured, but not in the way that is presented here.

[@tompettytribute]

@@kensmith895 looks like the transient response of the very sharp digital filter after the delta-sigma ADC to me. Analog antialiasing filter before the adc unlikely to have that kind of response.

[@DaveRat]

👍

[@andrewhigdon8346]

Btw, I have a pretty badass rig in my FROG, and those last sub bass tones on your outro were nice, with being 16:45 heard as well. Honestly the rig is pretty flat from 25 hz to 16khz, but if I move the crossover up from 22hz to 27 hz, it really affects the harmonics above, and deteriorates the response badly. A friend recently gifted me an old but in new condition BSS MiniDrive, which will replace the old Driverack. I’m looking forward to even MORE quality and harmony.

[@bokajllensch661]

I have a 25W 8" speaker in a random sealed enclosure hooked up to a tiny old hifi amp and I can also hear those tones. Its only flat from 36-18 000 Hz tho.

[@DaveRat]

Good stuff

[@AirRifleSport]

What is the lowest tone produced at the end of 5he videos? My in ear headphones seem to be reproducing extremely low frequencies. Just curious.

[@DaveRat]

If I recall correctly the lowest note in the outro is E0 which is 20.6 hz

[@TylerMarletteNY]

I’m using the midas stagebox for all ins and outs and the x32 rack for processing, would there be any difference in switching to a m32 for processing? do they have the same amount of dsp power, plugins, etc

[@DaveRat]

DSP processing and software are the same. The main advantages of the M32 over the X32, that I can find is lower noise and more durability and some higher quality components and slightly less latency in the console The M32 is a little more pleasant to work on with the screen angle and better faders but for most applications there's not much difference. I think the M32 will last a little longer on the road as it's slightly better built

[@stevenherrell6668]

As someone very well acquainted with both desks and an owner of the smaller M32r this is super interesting. I use outboard compressors with these to try and get back some of what I lose going digital. I wonder how much latency I'm adding to to everything. Maybe I should be sticking to the FAC comp in the console....? Great stuff, thanks for sharing with us.

[@vedasticks]

it wont be addig much, 0.9ms pus 0.9ms going out and back in. isnt an issue uness your using it on doube miced sources or if using paralel processing

[@DaveRat]

Yes, just adding some latency is not really an issue for FOH. A millisecond is about 13 inches. So having the time delayed a foot or so is no biggie unless you have delayed and non delayed versions of the same signal added back together. Or added latency can be undesirable for monitors and in-ears. It is important to pay attention to and also, converting to digital and back to analog and back to digital and back to analog is undesirable as well. The summation of multiple issues is where we see degradations that become audible.

[@stevenherrell6668]

@@vedasticks Copy. Thanks for the insight guys. Always bus my monitor sends pre-eq so the artist never sees or hears the inserted comp.

[@DaveRat]

👍

[@DaveRat]

I would need to test the added latency but I thought the console has 0.8 Ms from analog input to analog output. So inserting should add the same, a total of an additional 0.8ms output to input.

[@TheAT5000]

This leaves me with a question.... Is this only applicable for live sound, or does it translate in to the recording world as well? Example, in the digital consoles, they run the input through an analogue preamp, then through a analogue to digital converter pre fader. If you are running all analogue, you can keep everything analogue through the faders, and even out your mix and master busses. But then it still has to go through an analogue to digital converter. Do different converters then have a large effect on the recorded sound? And what style of recording preserves the most accurate sound quality?

[@DaveRat]

It's about using the proper tool for the job. Digital adds a lot of features and power and control and is necessary for some things or many things. That said it's important to know that analog or just a simple piece of wire is better than converting to digital and back again if you don't have to There is a line of thinking that digital is better but in reality simplest solution that performs a job well is best

[@shaddoty]

It's cool seeing a square wave murdered on a digital board, that must be why I love analogue synthesizers so much

[@bockaudio]

Thank you for the vids and reminding everyone sound outside of 20hz-20khz is part of the human experience. I say this as someone who spends a LOT of time measuring mics professionally.

[@DaveRat]

Thank you David!

[@gary23jag]

Dave thanks for another great vid, again for me the one thing I take away from this experiment is just how good cheap Behringer gear really is, professional’s are to quick to dismiss Behringer because of price.

[@LucasWasson]

Cheap Behringer mixers and guitar effects. Their speakers and amps are a waste money.

[@rwj1313]

Professionals don't dismiss their gear. Gearsnobs that can't mix do. I've been doing FOH since 1981. I've never refused to mix a show because of the gear. That would be unprofessional IMO. I learned on a 16 CH Peavey MK IV. I've mixed on nearly every brand of console that's ever been sold in the US. And I've mixed on every desk at every price point as well. I've done clubs and arenas. I spent 2 years with the band 'Brother Cane' in the early nineties and got to mix on big Clair Brothers rigs through a Yamaha PM5000. We did a lot of festivals and I got to mix on large format Midas, Gamble, and Yamaha. I sound-checked a BC on an ATI Paragon but the show got rained out and I didn't get to use it. If someone can't get a mix on budget gear they won't be able to get a mix on high-end gear either. Would you really want a pilot flying your airliner that can't fly a Cessna 150?

[@DaveRat]

It's easy to make a decent or good sounding mixer with simple EQ for very low cost with off the shelf parts. I would not apply the quality aspects of one product across the product line unless a manufacturer has consistently shown shown that their products are based on high quality rather than low cost as the driving force

[@jedi774]

I think it’s important to note the pros don’t necessarily dismiss Behringer because of price. They dismiss it because I’d their perception of its quality. It’s ability to withstand the rigors of professional use, it’s ability to produce consistent results over time and between units... Some may just be price snobs... and some may just be ignorantly espousing untested theories they’ve heard... but most that I have met have come to this opinion by experiencing failures due to lower quality components or lower tolerances. Behringer has always been marketed as a “cheaper” read “more affordable” choice for people who want the features, but can not afford the more expensive gear... and for the most part, I think they deliver on that. For some folks, it meets their needs and their budget. I’d much rather have an x32 for a play at a community theater than a 50,000$ analog board with no outboard gear... the flexibility and control of the cheap, lo-quality digits board trumps the superior build quality and reliability of the analog for me.. that might be different for someone else... but that’s my pro/con assembly.

[@DaveRat]

👍

[@teabreakbeats]

Thanks - you proved no need for me to buy the SSL big six. Behringer will do!

[@DaveRat]

Interesting.

[@16LiveRecords]

Just thinking, how does it affect real speaker performance, if you drive him with square wave signal, this kind of sound waves. Maybe that coil is less acurate to that signal. But if you drive him with that little "ramp" time, output sound can be same.

[@DaveRat]

When I test the sd11 I will test a square wave at 48k and 96k sampling rates and listen with headphones as well as record the output and see if there is any audible differences. If so, I will share.

[@softsmoken]

Some of the things you mentioned are the reasons why Sony for example runs a tweeter functional up to I think it was 50khz? The perception even if not heard makes a difference in those devices and will provide a different experience it seems. Good stuff.

[@DaveRat]

👍👍

[@Aquatarkus96]

The Sony's with those tweeters aint too bad for their price either. Lovely smaller bookshelf speakers for the living room. A bit weak in the bass, but nothing too deficient.

[@torchlord11]

Are the Boards that have a USB digital recording interface output have an analog front-end mixer? If so how does being converted to digital degrade the analog signal?

[@DaveRat]

The conversion will take its toll. The best digital can do is have the degradations so small that it does not matter

[@nuvoid]

amazing how he addresses haters of his testing procedure, with test results proving them wrong.

[@DaveRat]

🔧🔧🔧 👍👍👍

[@NealMiskinMusic]

Both of the digital consoles must be sampling at 48khz, as things get weird at half that frequency, 24khz, so that's your Nyquist frequency. If the console was running at 44.1khz that same weirdness would happen at 22.05khz.

[@DaveRat]

Yes

[@AirRifleSport]

Can this also apply to analog emulation in plugins? Maybe a video on this?

[@DaveRat]

Hmmm, I really don't feel excited about analog emulation plugging adventures

[@WaterTimeLapse]

So those sines at the digital desk are obviously related to the Nyquist frequency. Could you for demonstration switch to a higher clock to illustrate that?

[@EmporioZuagroast]

The fact that the sines get wobbly at exactly 24k and disappear above that tells me that both consoles are already running at their maximum of 48kHz. Can't switch to higher clocks on either of these consoles.

[@DaveRat]

👍👍

[@WaterTimeLapse]

@@EmporioZuagroast ahh that's too bad. But thanks for the reply.

[@BojanBojovic]

Ah, the scientific method... :) Not sure how those flaws of 48khz digital can translate to the real world and how much of those limitations people can actually hear, but for for people I am playing live with, who spend their entire life behind the speakers and still religiously argue about what piece of gear is the best, cheap digital mixers are brilliant. I can compress their shouting in the microphone, and compress the guitarists who play solos while singers sing some quiet parts of the song, fix digitally the bad room and bad speaker placement, and the list just goes on... :) Long story short, I do not need to fight with them while explaining everything they are doing wrong, now I just use digital mixer and fix whatever I can. Everyone is happy. :)

[@DaveRat]

👍🤙👍

[@derickmcwilliams8089]

So can we conclude from this that the digital consoles produce degraded "transient responses" of the signals compared to that of analog?

[@DaveRat]

I believe so. The question is whether the reduced transient response is audible or relevant.

[@sickmessiah]

Can you do this test on the presonus studiolive series iii boards ? IM curious

[@DaveRat]

Probably not but hopefully as I test I am helping you and others learn to test and learn to look for things that help you choose the best gear for the job

[@magoostus]

did anyone else notice how the consoles reproduced true sine waves even at 24khz? d/a converters are not stair-steps and i'm glad videos like this exist

[@DaveRat]

Yes! The stair step is a fallacy as and also, perfect sine waves at just below 1/2 the sample rate! Just a hair over 2 samples per cycle and it is accurate

[@nickwallette6201]

@@DaveRat I have a hard time with this, because there are a lot of DACs from the earlier days of digital audio that CLEARLY have stair-steps. All the old multiplexed mono DACs like the Burr-Brown PCM stuff you would find in CD players, Roland synths, etc.. they would use one digital-to-analog channel into a sample-and-hold circuit, and then into low-pass filters that served as the analog reconstruction filter. These days, DACs are cheap enough to dedicate one per channel, but that reconstruction is still happening somewhere. With Delta-Sigma and the like, it's happening WAY above the Nyquist frequency, but... it's technically still there. :-) (Even if it's not so much stair-steps as PWM in a lot of cases.) Now, you could argue that a lot of this is moot. The reconstruction filter is ... either in spirit, or literally "part of" the DAC, and the result is a smoothed waveform. But, every time someone watches the Xiph video and then starts spreading the gospel of There's No Such Thing As Stair-Steps, I cringe a little.

[@DaveRat]

Hmmm, i think there is some confusion abkut stair stepping. For a sine wave to actually be outputted as a staircase version, that would take a tremendous amount of high frequency energy. Just over two sample per hz will give a smooth perfect sine wave, no stair steps. Now with low bit depth, there can be volume level "stair steps" where a signal of a certain level will be ever so slightly louder or softer than the original analog signal. But as far as stair steps and being able to see them or them existing on the analog output of of a signal that was digital, not only are they not there, but the digital console actually does the opposite. If you send a high frequency square wave into a digital console, it converts it to a sine wave. If you sent a stair stepper signal into a digital console, the digital console would actually smooth it out, due to limitations of the high frequency capabilities.

[@nickwallette6201]

@@DaveRat Yeah, for a perfect vertical rise, you need the sum of an infinite set of odd harmonics. :-) Which is difficult when there's any amount of bandwidth filtering! But that's not what the stair-steps argument is about. (At least, I don't think so!) If you have a sec, I'll start with the common ground of this argument, and then get to the point... AFAIK, the idea of this stair-steps problem is that digital sampling takes a point-in-time sample of the analog waveform's value at a regular interval. The sample will be quantized to a discrete value between 0 and 2^n - 1, where n is the bit-depth. So, 0 to 65535 where n=16 bits, and so on. (Since we're dealing with AC waveforms with a 0V midpoint, you might think of that as -32768 to +32767. Same set of values, different offset.) Of course, I'm sure you understand all of that already, but we have to start at a point where everything is a given. The crux is that, since the X axis of a waveform is time, where each sample is a discrete point, and the Y axis is amplitude, and also a discrete point, the data between any two points is lost -- there's no continuity like in an analog waveform. The value "jumps" from one X,Y plot to the next X,Y plot. In a typical resistor-ladder DAC (which is not what most of us use these days, but the principle holds true enough still), the amplitude value would be set by selecting a pattern of bits that correspond with the sample value, and outputting voltage on those pins, which drive current through a series of resistors in a logarithmic voltage divider to produce a summed voltage analogous to the original waveform at that point in time. I.e., we set a variable voltage according to the value of each sample, one after the other. The value _between_ the samples is fixed to the last sample value. So, that voltage extends from the beginning of one sample period to the beginning of the next -- essentially a DC value for 1/44100th of a second at CD audio rate. The value then shoots to the voltage of the _next_ sample, and stays there at DC level for another 1/44100th of a second. The result is a series of short DC pulses of varying voltage with a straight vertical line between them. Stair-steps. We're not talking about a _recorded_ vertical line -- again, that would require infinite bandwidth. Instead, that line is a consequence of the voltage source (the DAC) changing output values instantaneously (or as near as it can) at the clocking of each sample. Now, when you look at the output of such a DAC on a scope, you may or may not actually see stair-steps. The reason you _may_ see them is because the DAC is outputting DC, but part of the DAC design is to use a low-pass filter that blocks frequencies above Nyquist. The contract we signed for the Nyquist theorem is that you have to do this filtering on the way in, and the way out. If you do that, then two samples per cycle is adequate to represent a bandwidth-constrained analog waveform. This reconstruction filter removes those straight vertical lines as the voltage changes between sample values, and converts them to slopes. The slope is shallow enough, if the filter's cutoff is 1/2 the sampling frequency, to remove the jaggies and produce a smooth curve between adjacent samples. No more stair-steps. So the question is, do stair-steps exist? People will watch Monty explain how the output of a DAC is always perfectly smooth, and then insist that they do not. However, if you measure the output of a DAC _before the reconstruction filter,_ it will indeed have steps. But, _can you_ measure before the filter? In many DACs, no -- the filter is designed in as part of the package, and there's no exposed access to the pre-filtered waveform. So it's kinda fair to say there are no steps. But, particularly back when a stereo or multi-channel audio device had to leverage a single DAC, an analog switch with multiple outputs, and parallel S&H circuits with their own post-filtering, then you absolute definitely positively _can_ see the steps. They do exist! :-) It's an academic argument at best, but for some reason it's one of those things that bugs me when there's this dogmatic insistence that digital doesn't stair-step. It only doesn't because it's filtered out. The waveform in a digital audio editor mimics this filter by using splines to draw the curve between samples, but there _isn't_ a curve between samples. It's just a series of discrete values. Also, today, DAC designs are more exotic than a resistor ladder, so the stairs have been replaced by 1-bit pulses at many times the sample rate, and all sorts of other tricks that can produce higher precision at a lower cost with more consistency than laser-trimmed resistances, so it's probably even less worth arguing about now. But I still can't let go of it! haha..

[@DaveRat]

Hmmm, seems to me that we can easily prove the information is not lost. Since all signals are a summation of multiple sine waves, if a digital converter can reproduce those sine waves any or all of them then you should be able to reproduce the signal accurately up to the highest frequency. And one simple demonstration to show the information is not lost is to sample a 23.5 signal and sample it at 48k well you have just over two samples a sine wave which. If you follow the argument that information is lost due to samples missing bits of information, then it would not be possible 4 the 23.5 k sine wave to be reproduced accurately. Yet it is. So yes logically complexities can be created 2 convince yourself that the information is lost but when testing we see that it is not.

[@OriginallyInspired]

dave is god, we'd be so lost without him. the amount of old wives tales about these desks before dave actually went and ran the tests is ridiculous.

[@schattenheim]

I had both I still have the M32 in my home studio. The difference of the sounds are subtle... For the kind of music i do ... both are great consoles - the hardware and design of the M32 is great and I like the feeling/handling. The X32 is ok but M32 feels better (both are agreat deal for the money). Thanx a lot for your great work (and live mixes you do and have done :-)

[@DaveRat]

👍👍 thank you!

[@noelhynek955]

I have heard rumours that, as you add processing, aka eq, gate, comp, it adds latency, could you do a session on this

[@DaveRat]

I will give a look.

[@robertlofgren6834]

@Noel, Those rumors are not true. Only thing that adds to the system latency is the fx-rack and aux-inserts.

[@DaveRat]

Thank you Robert!

[@randomschittz9461]

I also like big copper wires with high voltage

[@DaveRat]

👍🤙👍

[@russellhodges3869]

Like the shirt

[@DaveRat]

👍👍

[@MrWoodstock35]

Unrelated to the video content but where can I get a "Who the Fuck is Dave Rat?" shirt? I'm sure it's custom but I would totally buy one!

[@DaveRat]

Ha!!! Oh my, it was a gift. They would be funny to make

[@tompettytribute]

so, the step response (square wave) is a nice visual way of highlighting the impact of a finite sampling rate and sharp digital filtering - the question is: could you hear any difference between those two square waves? Of course there will be likely be some additional filtering of that 20kHz ringing by the speakers/headphones. I would guess that if you cannot hear the difference between those two signals (which are kind of worst case scenarios), then with typical signal content where only a very small percentage of the energy is up in that frequency range, it will probably be irrelevant. If you CAN hear the difference, then that's a decent argument for higher sample rates where less steep analog filters (post DAC) can be use to remove the higher frequency sampling artifacts.

[@gary23jag]

I would love to hear the difference, but as a 70y old ex part time sound man, I can just about hear 12k, the funny thing is my tinnitus is way above what I can actually hear.This colours everything I hear,I guess what I'm really saying is treat your ears with respect. i i

[@DaveRat]

👍👍👍

[@jezclark4882]

I don't disagree with anything said here, but it's worth noting that in the world of live sound, these days we're usually running digital amps into PA systems that are only specced up to about 16KHz anyway, and chucking that sound 30m+ across a field with the wind blowing. Anything you gain from having a console that goes really high is completely lost in that scenario due to air absorption and the wind. Indoors can be just as bad with the heat of lots of bodies causing the top end to bend upwards. Mixing in a studio, however, is a different matter altogether (although we all know it all gets MP3'd afterwards so eventually you're in the same boat downstream).

[@DaveRat]

Agreed that for many applications the added HF is not relevant and when touring arenas with a pa and you get that pristine theater gig and the system sounds flawless rather than lackluster, the capabilities are appreciated. Alsoperhaps looking at things in terms of octaves rather that frequencies is useful? Having gear that easily reproduces a full octave above and below hearing is desirable. Then we have highly capable gear that exceeds any application. Like having a car that can easily exceed the speed limit.

[@jasvq7]

Like the t shirt good one for the haters

[@DaveRat]

👍👍👍

[@mwestcc]

cool

[@DaveRat]

👍🤙👍

[@JonAnderhub]

So if you are using a solid wire split how much are the varying impedances of the boards and the signal generator affecting the signal distortion at the inputs of the other boards?

[@DaveRat]

I also test direct without the split and make sure the results are the same That's how I found that the QU16 does exactly what you describe with voltages over 10vols on the input.

[@kensmith895]

Very interesting comments about the possibility of perception above the traditionally accepted 20K limit. I'm about to open a can of worms here but --- I have commented elsewhere that I have a vinyl recording of Aretha Franklin made at Atlantic in NY with such luminaries as Arif Mardin and Gerry Wrexler. On a spectrum analyser that album has strong content from the brass section well above 20K. I can hear the difference between a digital copy of that album made at 44.1K and 96K. That may partly be related to my equipment but I can't tell the difference between my 96K digital version and the original. Whereas I can with the 44.1 version. Interesting

[@stephenevans2518]

Are you sure it's content from the album and not harmonics from distortion due to the innacuracies of vinyl reproduction? Unless the album was recorded 100% analogue or sampled ridiculously high then a 96k master is going to be exactly what was mixed and so anything else in the vinyl version is likely to have been added by the vinyl process?

[@kensmith895]

@@stephenevans2518Vinyl tracking errors are also a possibility. The album was recorded in the late 1960's. As well as my 96K digital vinyl transcript, I subsequently purchased a 192K version produced from the original master tape. Both have comparable 20KHz+ content.

[@HazeAnderson]

I am just gonna throw out a crazy hypothesis that these frequencies above ~20K are what make something sound "real" ... that is ... if you closed your eyes would you be able to tell the difference between a live horn instrument playing in the room versus an electronic replica thru a loudspeaker. Me personally? I could care less how "real" it sounds if the performance is smokin hot!

[@DaveRat]

Haze, agreed, realism is one of many factors that may or may not be important or relevant and extended HF response is one of the factors to improve realism.

[@daleclyne6057]

@@kensmith895. Wasn't it also a direct to vinyl recording ?

[@isettech]

Don't forget the tests are valid at the impedance of the wire. Add a very high impedance source of 50,000 ohms into a 62 nominal impedance 100 foot snake, and the difference is very real.

[@DaveRat]

👍👍

[@ziofrenko]

a 96khz converter is enaugh to give us 40khz of audio frequency... the problem is that PA cabinet give us no more then 18khz...

[@DaveRat]

Hmmmm, movie tend to be shot in 4k yet people rarely view them in anything above 2K or 1080p. Should they just shoot the movies in 2K or 1080p?

[@ziofrenko]

@@DaveRat not good comparison, imho. on video you can zoom in with higher resolution shooting, and have the maximum quality to final resolution. but live mixing is not recording. on recording I am with you: 96k is the minimum. in live mixing, with board they do not introduce distortion (and so aliasing) with vst and with limits of PA cabinet is not so important to have extra resolution. I add: for many years we have recorded and mixed discs at 44.1 or 48k and we have not problem at all.

[@DaveRat]

Definitely a gray area. On one hand I understand the logic and all the math and specs. On the other hand recording never sound as clear and realistic as actual real time live. It is near impossible to record and replay so accurately that a person believe it is real time live. So what is the issue? The speakers? The bit depth? The sample rate? Analog vs digital? The mics? Or perhaps all of the above or some combination there of. I believe that if we could go the other way, record and playback something indiscernible from the live instrument that was recorded, we could then go the other way and reduce bit depth, reduce sample rate, reduce other factors until it then became noticeable that it is a recording. And then we could say with a higher degree of confidence what matters and what does not. Until then, we are stuck pushing specs higher and higher in hopes of getting remotely close to realistic recreation. Which seems a long ways off.

[@cesareferrari8056]

It's not clear to me if the two digital consoles have their clocks slaved together? The delay you were seeing is not quite 180 degrees at nyquist, so that < 1 sample as they both appear to be running at the same sample rate. If you were to designate one as the master and slave the others digital clock to it, i'd speculate that they will be perfectly in time, removing that phase offset.

[@DaveRat]

Interesting but keep thinking this through. I am using an analog input and output on both consoles. The signal goes in and comes out analog, whether the consoles are synced or even operating at the same sample rate is not relevant when interfacing using analog. You could have a monitor console operating at 48k, a foh console operating at 96k and a recording console operating at 192k all at a show and connected to the same analog splitter. They could all have differing latencies and clocking them together would not alter the latency differential. At least not by more than the clock drift or offset. If you have consoles with differing latencies, clocking them together will not make the latencies the same. Anyway, that's all is beside the point, the two consoles each actually use a different type of digital to analog converter. What we are seeing is the longer latency in the X32 which has slightly slower and less expensive D to A converter.

[@lombybedlam]

Many X32 users do not know about the internal signal delay))) I've been using the X32 for 15 years now and no longer want analog boards with tons of cables and huge boxes of external processing, digital makes my job easier.

[@iggyfritz7150]

Thanks Dave for proving what my ears have known for years, how bout a tshirt that reads " analog ears for years" or " wisper words of wisdom let it be analog 😂 ok I'll see myself out now.

[@DaveRat]

👍🤙🤙

[@jeremyh8833]

Wouldn't expect them to waste processing power on frequencies we not only can't really hear but most PA systems can't even produce.

[@DaveRat]

I personally like the equipment that I use to be able to reproduce frequencies above and below what I will use it for. Having a car that barely goes to speed limit is not the same as one that goes way faster than the speed limit even if you're driving the speed limit

[@jeremyh8833]

@@DaveRat I mean I would like for an X32 to sound as good as an S6L but you have to keep in mind the intention of the product. The X32 as you know is a budget console, and those previously stated factors makes it more than understandable even expected that they are going to cut costs where they can, and this seems like an obvious place to cut down on processing power.

[@DaveRat]

Yeah the The capability versus the cost of the X32 and M32 is pretty much as good as it gets. And they sound good enough and the sound quality is not the primary issue in my opinion. The biggest issue is that engineers clip the consoles and make them sound bad and there's not a lot of headroom

[@jetipama]

this is why the PRO series is @ 96k!

[@DaveRat]

👍

[@guitarz99]

i would buy a behringer console and a midas stage box, problem solved with the lower quality pre amps

[@DaveRat]

Yes, that would solve the noise aspects of the quality differentials. Still be stuck with lower quality faders and such

[@andyperrigo5000]

How would a $5000 or $10000 console compare to the copper wire?

[@DaveRat]

Testing an SD11 now, so I guess we will see

[@Gameboygenius]

Copper wire, pros: perfect signal reproduction. Mixer board, pros: can actually mix multiple signal and apply EQ and other effects.

[@DaveRat]

And analog mixer pros, has most of the advantages of copper wire, EQ and such, with the ability to patch digital effects without the messiness of converting the primary signal to digital. Advantages of digital is advanced complexity and recall in smaller lighter weight, lower cost packages and also has loads of copper wire.

[@fuckugplus]

At 12 hz on my headphone there is that weiiird thing that happens... I start to hear my heartbeat... Its like the frequancy are canceling the canceling noise you brain does (things brain does medigate constant input). Freaky thing. I tested on my cousin, she also experianced the same thing

[@DaveRat]

Interesting

[@simccaffrey]

yeah but even if your mixer is putting out content above 20kHz, and even if the loudspeakers can reproduce up there, can they do it for very long? Many loudspeaker processors/presets/dsp etc. are going to filter all that out to protect the drivers...also I've seen reports of high sample rate recordings with large quantities of noise shifted way up that could damage speakers/amps etc...and analog limiters aren't great at catching high frequencies either...

[@DaveRat]

Lots of variables. One scenario could be that many natural instrument sounds put out frequencies above 20K at significantly lower levels than frequencies below 20k. And that we as humans perceive this low level sounds as part of what makes something sound natural. We don't need much energy up there but a complete lack of energy above 20k is unnatural and reduces the realism. Yes, most speakers, amps and processors do not reproduce sounds above 20k and most if not all sound systems are a far cry from sounding real enough that it is not immediately obvious that we are hearing a recording and not the an actual real live instrument being played. The added hf response in itself is not relevant for most applications, just like having a car that exceeds 75mph is not relevant for most applications. But also having a car that will go 75 mph and absolutely will not for 76mph mat also be undesirable for many applications

[@simccaffrey]

​@@DaveRat yeah agree naturally there would only be a small amount of information up there...but with dsd and mqa etc. you can end up with an unnatural wash of noise up there; some guitar pickups also hash out odd stuff up there.... I don't think you can really perceive it when listening to music on speakers at moderate levels though...I run my monitor system in 96k, and have decent speakers (kh310a) which are fairly linear up to about 35k...honestly I've tested myself and can't say I could reliably tell when I have a lpf at about 24k, unless it's doing something measurable to the frequencies below...I know my ears are only reliable up to about 16/17k...I even have a butterworth lpf at 20k that I just leave on in the monitor controller on my normal preset, and a hifi preset that turns it off...it's not night and day, sure you notice a tiny change when switching...but I wouldn't put money that I could sit down and say it's on or off without an AB... I know what you mean but I don't agree with the car analogy, that absolute 75 could be the difference in avoiding an accident...sure all things being equal, it's better to have an extended response, but i think it's really way way down the list...at least in mono/stereo music listening anyway...maybe in a surround system things could change a bit more, that could be interesting to look into...

[@DaveRat]

Hmmm, well if you/we can't hear beyond 16k, why not limit at 16K? We already accept that the response must exceed our hearing, the question is "by how much?" Is 1/4 octave sufficient? One octave of buffer above hearing would put us at 40k. For me, once we have speakers can realistically reproduce sound accurately enough to where we can't tell the reproduction from the original, then we will be in a position to determine which parameters must be maintained and which can be shaved. For everyday life, for whatever application at hand, we scale the specs as needed. I don't need even to hit 12k for the Bose speakers around the house. For headphones and mixing I want the speakers to exceed the capabilities and response of the material being worked with

[@simccaffrey]

​@@DaveRat That was a bit of a drunk 4am rant TBH! And the problem with the "by how much" has a lot to do with how filters themselves work...like I said I can hear the 20kHz 24dB butterworth LPF...but imo that's probably more to do with it attenuating frequencies I can hear below, and maybe phase shift, than the cutting out of higher frequencies...there's always so many factors... Btw youtube it seems has raised its cutoff from around 16kHz to 20kHz for stereo also sometime recently...in chrome anyway, but not safari it seems...

[@DaveRat]

That is interesting. I will take a look at the KZbin filtering again

[@MadayMaday]

It took me watching for 5 minutes to realize who you were. I'm such a derp.

[@MadayMaday]

Would really like to see a comparison between analog Midas and digital Midas.

[@DaveRat]

I will do something down that line and do some testing on an H3000

[@MadayMaday]

@@DaveRat appreciate you sharing your wealth of knowledge brother. While I'm no longer doing live sound I still find this useful in the recording I do.

[@DaveRat]

Thank you Joe

[@zerokoolTV]

Why does Dave look like that european guy from Wayne Worlds He went to get some MMs so Ozzy would go onstage. They smart as hell but look crazy at the same time. Lol

[@DaveRat]

👍 sometimes striving to be average is not as fun as falling into eccentric

[@stevenjackson6360]

Reproducing a 20kHz square wave as a perfect sine wave is exactly what you want with a digital console, it demonstrates that the filters before sampling are sufficient to prevent aliasing We operate with a bandwidth of 20Hz - 20kHz. Anything outside of that is useless to audio engineers

[@DaveRat]

24k is less than 1/4 octave above what is slated as typical hearing. We hear about 10 octaves so it's less than 1/40th of our hearing range as a buffer. So for a freeway with a 70 mph speed limit and adding a 1/40th buffer, we should have cars that max out a 71.75 mph, right? Just having fun playing with assumptions here. I find it fascinating how we as humans strive for camera resolutions that far exceed is often needed, car speeds that far exceed what we will drive and yet when it comes to audio "naw, all good, barely above hearing all good" Do you follow the same theory with cameras? No need for a camera with over 4k resolution, right? I've got some old Nikon cameras cheap if you are interested in taking pics barely above the resolution of your phone, tv or computer monitor.

[@stevenjackson6360]

@@DaveRat it's great that you take the time to actually reply, good to chat with in real life. (I've worked as system tech for several companies that have provided l'Acoustics for you when you were with the Chillies) I get what you're saying but what practical benefits are achievable by extending the bandwidth above 20kHz? The highest playable notes usually have a maximum fundamental frequency of around 5kHz (C8 on a piano is just over 4kHz), giving 2 octaves of harmonics in the audible range. Where do you draw the line between audibility and possible benefit to a child under 12 years old? If there were significant benefits, then the standard sampling rate would have been increased long ago, driven by consumer demand. The benefit of 48 compared to 44.1 is the fact that the anti aliasing filter can be of a lower order and therefore will exhibit less resonance around the cut off frequency, which can be raised slightly. The benefit of 96kHz + is to avoid aliasing when introducing distortion through processing . With the camera analogy, increasing the bandwidth would be akin to extending the bandwidth of the sensor into infrared and ultraviolet, useful for some applications, but not for photography or cinema. Increasing camera resolution doesn't really have an audio analogue. An interesting experiment would be comparing 2 M32s with the 20kHz square wave, one at 44.1 and the other at 48 to see if the filter is good enough to reproduce the 20kHz sine without alias at 44.1. Keep up the good work.

[@DaveRat]

Hey hey Steven. I agree on the analogy to cameras of infrared and ultra violet. And to take it further, to recreate the experience of sunshine using purely visible light would lack the infrared warmth and edgy squint that infra red and ultra violet provides. And a visible light band limited version of sunshine recreation would always be lacking. For digital, it is feasible that each individual step having a range that barely exceeds human hearing by 1/4 octave or 1/2 octave for older humans, then often gets combined with multiple other steps, and one or more a to d and d to a conversion followed by a speaker system that tends to be even more limited in it's ability to reproduce the frequency extremes. It is my opinion that if one was to avoid letting tech limitations bias our goals, that shooting for maintaining a full octave above and below human hearing as far along in the signal chain as possible would not be not an unreasonable goal to shoot for. Then band limit as desired for the application at hand. Interestingly digital does solve the low freq side and even KZbin will process and deliver audio down to 5hz or lower. I did some vids testing KZbin audio which dies at around 14K or so depending on stereo or mono.

[@JaredBartimus]

Your discussions on 20khz+ sound being audible makes me wonder, would a 25khz sine wave from one speaker and a 26khz sine wave from another generate audible acoustic beats at 1khz? Also, if the same driver was playing a high intensity 2khz tone and a 20+khz tone would the doppler shift and other interactions bring the higher frequency tone into the audible range for half of the 2khz wave? Remembering the video you did with a driver playing a very low frequency tone causing audible shifts in a higher frequency tone being played. Lastly, would multiple sounds time offset by less than the nyquist frequency generate an audible effect that would be lost due to the sample rate? Thinking things like reverb with lots of time offsets All frequencies just examples.

[@DaveRat]

It's important to note that I believe I refer to above 20k being perceivable. Not necessarily audible. Audible refers to the ears hearing. And on the most literal level, we know above 20k is perceivable as we perceive light and infrared heat and people get burns from radio towers. All are frequencies above 20k. The question is not whether above 20k is perceivable as we know it is. But rather, at what level and 8s the perception relevant to the application or situation

[@JaredBartimus]

@@DaveRat then would the given scenarios potentially shift the sound from perceivable to audible?

[@DaveRat]

That is possible, would be interesting to test

[@JPHMP261]

its time to get out the S6L

[@DaveRat]

Yeah, I hope to test one at some point

[@vosperr]

9:12 :D

[@DaveRat]

👍👍👍

[@courgeonaute]

This is what would be expected from any digital board. Since the sampling rate is at either 44.1 kHz or 48 kHz. At 48kHz, the nyquist frequency is at 24 kHz. Meaning that the highest frequency you can reproduce is 24kHz. Think of it like the resolution of the signal in the time domain. Since you can't have a perfectly smooth digital signal, you have to set a sampling rate that can reproduced the entire human hearing range. But there's no reason to reproduce anything above 24K since we can't hear it anyway.

[@DaveRat]

Yet the 2 consoles are not the same around Nyquist. And though we can't "hear" beyond 20k does not mean we can not perceive beyond 20k. In fact your perception of light as you read this is a way way higher frequency than 20k, sunburn, heat from infrared light and headaches from RF antennas are all perceptions of frequencies above 20k. But that is not the point, we measure freq response to indicate the capabilities of an audio system much like we measure the top speed of a car to indicate the capabilities of a car. If one car can go 180mph and another can go 90mph max speed and you never will drive over 80mph, they are both the same, right? No difference between the cars. Since with the driving we do we will never feel a speed faster than 80. Basically things are more complex that the simplistic thought, "people told me it can't be heard, therefore it does not matter" and perhaps looking at what can be perceived by the body as a whole rather than just heard by ears. We can't see ulta violet light but we can get sunburned by it. Oh and also, can you hear 40khz? Can you perceive it? Are simple answers and assumptions to quantify the dynamics of the human body relevant? kzbin.info/www/bejne/qn_TgXl7rL1pj5Y Oh, and check this out

[@andrewhigdon8346]

Harmonics, harmonics, harmonics. Any manipulation we make, digital or analog, to any tone, waveform, noise, sound, call it what you want, but ANY manipulation changes not just the obviously manipulated bandwidth, but the upper and lower harmonics as well, especially analog gear. That’s why a parametric eq is always gonna sound better than a graphic equalizer, analog speaking. I learned some of this researching how transformers, in this case, direct boxes, use different types of transformers to eliminate noise, or at least reduce losses through the transformer. I used to say it was bullshit when a bass player would complain that a certain DI would take away some low end of the bass signal. I was wrong. An $18 Whirlwind EDB-1 direct box is vastly different than a Radial JDI. And that’s the big difference. Likewise, the Analog conversion to digital and vice versa is always going to have some fidelity loss simply because of the time shift, which equates to phase shift, which equates to a pitch shift, albeit very small, but it’s there, and how they mask these shortcomings seems to be what it’s all about. Honestly, the A/D and D/A conversion nitty gritty is a little over my head, but manipulation is manipulation. A good example is when you’ve got a great rig that sounds top notch with a flat analog graph. While the band, pink noise, music, whatever is playing through the rig, dump one band of eq completely, and the difference will astound you, as it sounds like there’s definitely MORE THAN just that, let’s say 1k missing. And there is. 500, and to some degree, 750, 875, and so on, with each harmonic off of the source having less loss, but again, 250, 125, 2k, 4K, and so on. Mega comb filtering, but because of the shortcomings of analog graphic eq circuitry, indeed, the very thing which separates a low cost graph from say a White, KT, or top notch dbx even. But here’s what really kicked me. I was running the sub aux out of my FROG(room over garage) rig through a nice mono dbx graph, a 2031, and even though I had the crossovers in the old Driverack PA, is also dumped everything from 100hz up out of the graph. Mistake. A HPF is better, as it’s a one sided slope, but bypassing the graph and punching it back in. Add a HUGE difference to my sub sound. Why? Because the desired manipulation can never be absolute or even remotely discreet, as the entirety of the graph functions as a complete circuit, whereas on a parametric eq, those filters are manipulated with much less interband interference or influence. I don’t know enough about that circuitry and am probably talking somewhat out of my ass, but organic sounds are composed of the whole frequency spectrum, including above and below what we can actually hear, but let’s say a graph went to 30k, and pull 25k negative 12 db, and you WILL hear a noticeable difference in the highs. Same thing with the straight copper analog to analog, the is very little if any manipulation, but with digital, lots, with great attempts at masking the issues from our ears. Even balanced to unbalanced signals can have lots of eq loss, as unbalanced signals going through a transformer aren’t so easy to flip back from the phase reversal of the transformer versus a balanced line. Reading about this recently made my head, swim, and I concede that I am pushing the envelope of my own understanding, HT the real lesson here is this: 25 years in, and I really thought I had sound figured out. I can mix the shit out of some shows, and my shows are known for always sounding great, but it turns out it’s as much the higher fidelity equipment in use nowadays more than my skills, and thusly knowing not to manipulate it TOO much. High passes on anything that doesn’t have bass content, same with the opposite end, but really careful attention to impeccable gain structure and level balance, not forcing things to work together that are fighting each other, like a snare and a vocal, or hi hats and the very highs of a soaring guitar lead. Pull something back, just a db or two, and the whole mix opens up. Balance, balance, balance, and with heavily altered harmonics, balance is more of a quest than reality. Why did all that 4K attack on the kick drum work back in the day on my Venice, but it just doesn’t happen on my X32? Harmonics. Why did sweeping out some 400-500 hz give it a “suck the air out of your gut” feel, but on digital it’s just not the same? Harmonics. We were getting more eq than we understood before in analog world, where now the eq is more accurate, kinda, but they play tricks to restore losses, so we get a facsimile of accuracy, less result than analog, but ultimately, a finer product in the bigger picture of the overall mix. Remember back when a dude would walk into your venue, “ring out(bullshit)” the singer’s 58 about as loud as they could get, and the prevailing result was lots of midrange removed. Now, when they check the rest of the inputs, all of that is missing, and they end up adding mid, which is usually ugly, or pulling lows and highs and turning it up, to get the program back. Is it the fault of the keyboard, bass, snare, and toms that the misguided demands of the vocal mic required so much manipulation? No, it’s not. So don’t treat them so poorly. Just make less movements to the eq, then try to transfer that to the strip, or even insert a graph if it’s that bad, but the bottom line is that so much “chopping” gives you that unnatural feel, while D/A and A/D converters are trying to restore losses, such that the cheap are quickly discerned from the costly. I had a Digico for awhile, and the preamps were pristine. Incredibly transparent. But TOO transparent, leaving no impression of “balls” on the input, as my Venice did so nicely, and the XL3 did much better. Oh, the price of quality. For live music, there’s nothing like Midas pres and JBL drivers with very nice Crown, Crest, or QSC amps in between, with BSS or KT processing. I ended up removing my dbx Driverack when the crossovers came with an X/M32 update. Before that I had used the internal crossovers in the amps, resulting in a mostly flat response, but I wanted compression for the PA at the crossover level, so I tried the Driverack. For the next year I wondered what had happened to the sound of my dead reliable PA. Tired 12” drivers? Passive crossovers getting cooked? Nope. Turned out it was the crappy Driverack PA. Sounded like shite all this time and I just neglected to acknowledge it. Converters, got me before I even knew what was afoot. Thanks Dave, good stuff. I didn’t realize how much using only Midas and nicer Soundcraft desks had spoiled me over the years, until I tried out various digital platforms when they were emerging. The O1V was pitiful. Awful sounding, and a clip on one of those was like the squarest of square wave nastiness. Ughhh. The little Mackie 1608 mixer that came out around the same time as the X32 was pretty bad too, and it took me several attempts to realize where the problem was. I sometimes would use that for the opening band at a house gig, where my FOH desk was a Verona. On the Mackie I was always fiddling with eq, and never really happy south the mix, and one time I just halted soundcheck, swapped back into the Verona, and those problems disappeared. Hmmm. Harmonics. Converters. Integrity. We hear more than we think we hear. What’s that? Did you say something? Couldn’t tell with all the noise around me.

[@HazeAnderson]

You should study electronics. Simple scenario ... I take the two wires from a microphone and hook them up to the two corresponding wires on a speaker ... what happens when you speak into the microphone? What is transmitted across the wires? Why do the wires need to form a circuit?

[@DaveRat]

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