Great content Dave!!! Using "plugins" on the x32/m32 can be tricky in that there's no guarantee you''ll maintain phase coherancy between channels. Each plugin seems to have a different amount of latency which can be a problem when bussing drums. This characteristic might be worth exploring. Thanks again for the content Dave!!!!!!
mentioned this in one of the previous videos, its a gotcha ive seen a few engineers not notice when paralell compressing with utimo comp. you get round it by dupicating. 1 subgroup for your uncompressed and another for smash.
@@DaveRat Tired of kicking a dead horse? Just used the M32-R short frame version last night with Andrea Bocelli in NYC. The mixer could not be physically touched without sending out huge phantom power-switching bombs. We never found out what channel was doing it but eventually stuck strictly to iPad control to avoid it. I offer no further comment on this.
Marshall, I’d vote this is applicable with any digital mixing system - when using plugins that are outside the box (mixer). Most digital platforms offer some form of ADC for processes that take place inside the native mix engine, but as soon as you go outside (SPRK, analog inserts, UAD, etc..) all bets are off. The Waves and UAD options do offer some latency compensation that can align WITHIN their host, but doesn’t account for any signal buffer going in and out of your mix system. My recommendation is to always have a measurement path setup with an FFT so you can verify that all impulses leave the desk at the same time.
I hope my question (about the xenxy) from some time ago didnt got picked up as annoying. You gave a very nice answer and I learnt something new that day. Keep up the good work! Love your vids
I know this one... Fader position digitisation. The rough part of the chirp is the rate at which the fader position is sampled. The squaek part is an artifact from the smoothing algorithm for the rough digitisation. I first met this when installing an alpha test version of the Amec Digital ( was supposed to be beta, but it really wasn't that ready). With no consideration of fader position sampling, the faders sounded scratchy. Yup very expensive digital scratchy faders.
This is such common, and old, ground in digital audio design I can’t believe anybody actually still misses it. It is truly 101. There are even complete circuits on a single surface mount chip available from Burr Brown, Analog Devices, Cirrus Digital, and others that handle the entire function, using either a generic linear analog pot or circular encoder to effect a full solution of extremely high spec. I’m not saying it’s a piece of cake, as power and ground must still be properly configured, but it ain’t exactly rocket surgery anymore.
@@artysanmobile There is a tradeoff between responsive faders and smooth faders all restricted by budget, both processing and hardware. The test case here could be considered a bit "bizarre" generating a sound and then nulling it. It would be handy to get a signal to chirp ratio. Another fun fact from the Amek desk... Occasionally it would incorrectly output 3 duplicate samples. It was barely audible on real programme audio, but stood out like a sore thumb on tone. I really never again want to spend 8 hours listening to 1kHz waiting for a click that was as loud as tapping 2 fingernails together.
@@dougaltolan3017 I don’t intend to argue with your direct experience, Dougal, but I do take exception to your statement that there is a gray area of fader operation quality. There is not. Signal to chirp ratio? Seriously? How about 140dB? Your comments on a prototype Amek don’t seem relevant. Mute switches are for muting. Faders are for attenuating. ZERO is the only acceptable level of artefacts for professional equipment. I hate to be so hard core, but exceptions would never pass reviewing muster and should be returned for refund. I own and have used digital mixers since the 1990s every day without any of these issues.
Great presentation Dave...I immediately thought about the benefits of better (i.e. Penny & Giles vs Alps) faders in the analog domain. (Obviously won't help here) Everything is a compromise in life. Some things are better in the digital domain, or class D amps, et. al, but there is always tradeoff, you're doing terrific job bringing light to exactly what that means through the signal chain. Thanks so much, Mike
Thank you Mike! And that's interesting. I don't know if the faders for digital are different or not. Are they linear encoders? Or are they resistive analog type faders that vary a DC voltage which the digital circuit interprets? Or? I must admit I assumed they were like a vca, where they slide between ground and a fixed voltage and a digital circuit reads that voltage and implements the associated amount of gain or attenuation. That said, the rotary knobs that endlessly spin are encoders without fixed end stops like the faders. Interesting! I love breaking assumptions!
Excellent deep dive into the chirp issue. You have proven I am not crazy because I have mentioned this to a couple colleagues who looked at me as if I’d completely lost it. Have you experienced any issues with the faders on either the x or m32? We have several of each and the M32s seem to have occasional dead fader symptoms. They still work, but are no longer motorized and is a pain when jumping from layer to layer. I do know the x32 faders are soldered in place vs wired with a small connector and two screws on the M32. Thanks again and I appreciate your efforts to go deep into these common use products in the industry.
Motorized faders do fail on all consoles. We don't own any m or x 32 other than purchased for the vid so I don't have experience with how the deal with time and usage
Here's the background information to understand what is happening here, from a digital console designer. I am interested to see the 6kHz, which is probably a very good result as this is the update rate of the gain coefficients. The effect of changing the gain (even in an analogue console) is to modulate the channel output with the gain change frequency, exactly the same process of frequency mixing used throughout electronics. The important part from an audio quality point of view is that the change is spread linearly across a large number of samples, not suddenly applied to one, and then keeping that value for the following samples until the next fader value is received. However, this smoothing of change rate is quite processing-intensive, and therefore is rarely perfect, and is less and less necessary as the update rate is increased. The 6kHz signal level Dave is seeing will be small compared to the 20Hz signal, and with modest fader position changes inaudible. The mute button obviously switches in one sample, so the 6kHZ is larger, but then the mute button is not usually used when a signal is present when there will be no 6kHz, so is much less important. Devices like the gates and compressors will not show this effect as the gain transitions are fully "smoothed" with an effective sample rate of 48kHz or whatever. How many dB down from peak is the 6kHz Dave, probably about -60?
Also I am told the Behringer Wing does not do it and none of the other consoles I tested do it. It seems to be unique to the m32 and x32 and other derivative designs by Behringer from that design I was told that the issue is related to the X32 and M32 processing in bloks of 8 due to lacking the processing to uses smaller blocks due to processing power shortages.whatever that means.
Now. Who was it that said digital would be the bee's knees? Really enjoying your tenacity in digging into these annoying things that the sales brouchers dont go into 👍. Happy holidays to you and your's Dave. Stay safe.
Hey Dave, love this short series, I've use the x32 extensively to mix my church's broadcast mix. I wonder if you have run into a this strange behavior where when you link two channels together, there would be a very small, but noticeable drop in volume. I noticed this and experimented by copying all the channel processing and routing and panned the channels left and right respectively, and the behavior still continued. Just something I found interesting and thought would be cool to see you dive into if you ever get around to it. You rock!
How do you mean? Do you pan them back together after you link them? What input material is there? It might just be a pan law thing you are experiencing pherhaps? :)
I saw another comment on this exact subject. Not sure if it was on this video or another. And also some useful responses. It's prob a subject is won't dive into but is interesting
Dave, I quickly replicated on my X32rack with no physical faders or mute buttons, only pure software control, and I hear basically the same thing. So it's not simply the physical interface or lack of proper mute button debounce/LPF, or physical fader resolution (Although I guess it could still be GUI fader resolution). I'm not sure how the muting is implemented - of course abruptly dropping or increasing the envelope of a signal will have momentary high frequency content, so I would expect muting/unmuting to have high frequency artifacts if they didn't make the envelope transitions slow and smooth (S curve, etc). You would think in a digital signal path it would be easy to make this muting envelope transition "nice". I may put in on my scope to see what's actually going on. But the fader should be much less severe in that respect, as the digital resolution should be quite good, so possibly something else is happening?
FYI, I did put it into my scope and triggered off the muting, and see that in the x32 rack, the muting function seems to use a simple 1st order LPF envelope function with a time constant of ~20ms. So it doesn't use any smoother fancy S curve type envelope. But other than the 1st order envelope LPF, I do not see any glitches or problems in the waveform, waveform is nice and smooth and continuous. Of course there is still high frequency information in an envelope change that is LPF'd, as the step response of a 1st order LPF does not have a continuous derivative. So it seems we are just hearing those artifacts.
actually, I can directly see the 6kHz problem on the scope. The LPF'd muting envelope is done with digital gain steps, and those occur at ~166us (-1/6kHz) intervals. By zooming I can clearly see the gain steps as the envelope increases, hence that 6kHz junk you are finding. I have interesting scope pictures if you are interested.
@@DaveRat I played a little more, and I can capture it carefully on the fader movement as well. Due to the 48kHz/8 sampling of the digital gain control signal, the low frequency waveform essentially is modulated up to around 6kHz, where you can hear it through your high pass filter. This effect will be directly proportional to the speed at which the fader moves. If the fader is moved very quickly, it will be loudest (should sound similar to rapidly & repeatedly pressing mute/unmute, hence your "zipper" effect). If the fader is moved very slowly, the effect will be much less. I verified this as well. The good news is that as it is directly proportional to both the signal level and the fader movement speed, you unlikely to hear this in normal mixing scenarios, as the signal itself will be MUCH louder than the artifact, especially with gradual fader adjustments.
Fascinating stuff Dave! I’d be interest to hear if the chirping appears with other processing like compressors or gates. I haven’t played with the internal “rack fx” compressors to know if they can sidechain but the channel strip comps definitely can
Could you do some latency and phase comparison using the LV1/dLive/Analog? Here in north Europe it seems to be to consoles of choice for most touring FOH techs. Maybe also put a Digico in the batch?
Really appreciate all these videos, super interesting. I'm just home hobby type of person and just kind of look at things really high level, like I can mix a song pretty easily in the analog world and it feels next to impossible in the box. Just way harder and I don't know why. Anyway, took your idea with the dual mono compressors on busses of like instruments, like vocals in a group, drums and bass in a group, other instruments in a group, light compression, a couple db. That really helps mixing a tune fast too. Really appreciate all the stuff you come up with. Maybe a comment on the muting. I have a A&H GL2400 which sounds decent, but those mutes as just switches drive me batty with the clicks. My A&H 4000 has the relay mutes which is just a million times better. It's like these digital board need to do that too, but obviously through programming and not relays. Happy Holidays to you and your family!
This was fascinating! I've never cared for these consoles and I've never been able to put my finger on it. I've heard the chirp many times. And I'm not saying this is the reason for my bias. It's a tool, use it. Watching his, being a fader rider at times. Has this been part of my experience? Dunno? I am very curious to see what you find with the QU16. Qu series are the most common install desk around here. I like them. A MixWizard on steroids. I don't think I've heard one chirp or something similar.
Nice digging into the M/X32 world Dave! Listen to the differance in linking 2 channels vs to separate (not linked) L/R Channels.Would like to hear what you think about that. ;-) Greetings from the Netherlands
@@DaveRat Level and sound quality change. Propably fase related. Kind of what post below is discribing about the fx inserts. Had some funny things with that also.
I have noticed the same, huge difference between the sound of 2 linked channels vs not linked. Of course both always panned hard left and right. It's very noticeable right away using music for instance and headphones
As a software engineer who has worked on audio consoles, MUTES should either happen at the zero-crossing point of a signal or be a fast fade. Unfortunately, with a low frequency signal such as you're using, this can be problematic as it means that the MUTE is too slow. This means there are trade-offs. If you performed the same test at 1kHz, you may not hear any artifacts. Fades are a similar problem, ideally you fade at very small dB change levels or fade at the zero crossing point of a signal. Again the low frequency of your source is showing up an inherent trade-off between speed of operation and zipper noise. I would guess that 96kHz or 192kHz sampling might make this better as you can have more 'attenuation points' during the fade/mute, but I think the noise will only be reduced by 6/12dB if that's the case. I'd be interested to see a graph of chirp level versus source frequency for several different consoles.
Consoles I have tested other than the x32 and m32 do not chirp. The SD11 and Yamaha CL1 are chirpless and run at 48k. Also analog consoles do not chirp. A mute click is pretty common. I will test zipper noise and such on the Yamaha CL1 I am testing
Dave, love the comparison series you’ve been doing on these consoles. I’m curious if these differences carry over into the digital stage boxes for these units as what’s being received by the mixer is a digital signal vs analog.
New test for you Dave! I wonder if you set these consoles to run over Dante and use high end I/O (Digico,Yamaha or Midas) will these systems take on some of the sonic attributes of those consols?
@@DaveRat I was able to replicate the zipper on my rack via a mix bus this weekend. Rapidly cutting a mix bus used for vocal delay sent that odd noise through my PA.
Wow! that's bizar!! It is really amazing of you know things about those cosoles no one ever gets to know... GREAT!!! Do you ever get feedback from Behringer about your findings? Are they learning from your findings...? (You should get paid for your deep-testing of their stuff! hahaha!)
Thank you Edwin. I am in touch with Robert who moderates the Behringer forums and he gives me info when I have questions and clarifies things. That said, I try not to get too close to manufacturers so I can avoid bias. I test and ask questions if there are things that puzzle me and I feel I may be off base in my tests.
I'm guessing that "sound of the fader resolution" (LOL) might not be heard while using the rack version with iPad on these units? This is a mechanical artifact yeah?
Haha this is very interesting! As someone else allready said. A signal to "chirp" ratio would have been nice info to have. How much are you amplifiying this noise for us to be able to hear it in the video? I´ve worked hundreds of shows on a M32 at this point and I have never experienced this in practise. Not even with very quiet source material and high gains settings on condendser mics like a Choir singing pianissimo dynamics over a few piano notes. I´d never touch the gain or mute in such a situation but i defintly ride the fader sometimes and I never heard it. You consider it a practical issue or just a fun fact? :) Been following you on and off ever since the papers preceeding your "orgasmatron" Sub array with the A. B, C and D stack of 3xSB28s each with different phasing and delays. You even had a carpenter cut out a board for you to set it up faster. That was cool :)
It just something to be aware of. I heard it when testing and muting low freq signals, then setup a test to isolate it. For most applications, it does not matter, just like a freq response over 16k or distortion over . 5% But when comparing to higher priced consoles or applications where perfection is more important, it's good to know.
Hey Dave! My question is a bit unrelated to the matter at hand - have you tried seeing how the digital aes50 signal could be passed in "reverse" through rat tails into a 4 channel xlr snake? I tend to work with x32/m32 with the aes50 stageboxes and usually tend to work places with existing xlr snake stage boxes. I've seen (binged is more accurate) you describe how your cat to xlr systems work, and the termination makes sense that it would work but the length of xlr running could be an issue with interference. I would love to hear your thoughts on the matter. Cheers from Haifa!
Actually yes, I have tested exactly that. Definitely loose lots of length, for short runs it can work but aes50 is raw digital without error correction so caution is advised. AES50 is very very poshy and temperamental and very set in travelling down familiar paths.
Not test wise. AES50 is a raw, non error corrected signal to the best of my knowledge, that keeps the latency super low but makes it less tolerant to issues and more likely to have very loud issues vs muting undesirable sounds.
do you have the gain really cranked to be abe to hear it? ever oticed it in norma gig envirements or even listenig with headphones. what firmware version is on the console?
@@DaveRat I replicated your results using the desks internal oscillator. Sent oscillator to a bus which was out of stereo. Used a channel to take source of the bus having tone sent to it, sent that channel to another subgroup and filtered out the 20hz tone. I was sending 20hz to subgroup at -18dB. The chirp is clearly audible with cranking the desk out or subgroup out up. I tried it with pink noise aswell, it's audible but becomes less and less audible the less that's filtered out.
Every M32 I've worked on has an audible "chirp" at what I believe is 16k. I haven't heard this same thing on an X32, but that doesn't mean that it's not there. Have you looked into it?
@@DaveRat I tried to file the edges off my snark. I noted a reading on your scope of approx 5mV. Assuming that was the amplitude of the chirping at that moment, it’s quite inaudible, well above the noise floor. Pretty alarming stuff. I’m so grateful you take the time to do these investigations, Dave. I will be using Calrec mixers in January and February for Olympics and will have the time to check them. I have already done some ultra deep dives on the Lawo consoles. With phones and a live Sennheiser HD27 dynamic mic headset in an empty soundproof announce booth, I have run the digital head amp gain up and down from 10 to more than 60dB, and absolute silence was the result. I mean DEAD silence in screaming hot phones, to the point I can hear myself clap 70’ away, outside the closed booth. Ditto mutes. Lawo is big money stuff but still the point is, it CAN be done. I believe you will find similar results with your SD11.
Fun and thank you. Yes, interesting stuff and for me it's all kind of stumbling around and see what shows up as I dive in. The aspects that seem most interesting are the least expected.
@@DaveRat I am most interested in learning which, if any, of the lower cost manufacturers have done their homework. Clearly the Midinger conglomerate has skipped right over some essential developement. I’m sad about that because I don’t know how far up the Midas Pro line this disease goes. I will be scrutinizing every mixer I encounter on the road time permitting and would like to share my findings on your channel. I promise to keep them brief. Mute, fader, head gain; quiet or not.
My tests comparing the x32 vs x32 with m32 as a stagebox shows that the audio advantages are determined by the the stagebox. The interface and durability advantages are determined by the whatever the engineer is operating. As I move into testing the Digico sd11, I see clearly that both the x32 and m32 are cheap toys as far as ruggedness, but the m32 is much less toy like than the x32
Imagine when you have two digital audio sampled systems feeding audio to each other, each using slightly different sample clocks..eventually a buffer gets starved or will overrun. I have designed systems that detect this plesiochronous condition and adjust samples rates accordingly.
True and in all these tests I use analog in and analog out so sample rate drift has zero impact. With aes50, the consoles clock over the aes50 connection. Chosing one console or the other to clock when they are digitally interconnected is a must and the norm. Once the signal is analog, whether it was clocked at one rate or another in the signal's precious life as a digital signal is as meaningless a whether a mic cable was stored in a red case or blue case before being plugged in.
Anyone who is questioning Dave and his credibility and credentials just shows that they really have no clue or don’t even work in this industry. I work on the periphery doing mostly Cinema sound and even I knew who Dave was before these videos started!
Dave, I've been in the broadcast business for 42 year's, I consider your knowledge incredible. Your real world experience is refreshing. Keep it up.
Thank you Ross and an honor to be of assist!
Great content Dave!!! Using "plugins" on the x32/m32 can be tricky in that there's no guarantee you''ll maintain phase coherancy between channels. Each plugin seems to have a different amount of latency which can be a problem when bussing drums. This characteristic might be worth exploring. Thanks again for the content Dave!!!!!!
+1 on a phase check with fx!
Interesting, I will ponder that adventure. Gonna try and switch gears to newer and/or higher end consoles for a while though
mentioned this in one of the previous videos, its a gotcha ive seen a few engineers not notice when paralell compressing with utimo comp. you get round it by dupicating. 1 subgroup for your uncompressed and another for smash.
@@DaveRat Tired of kicking a dead horse? Just used the M32-R short frame version last night with Andrea Bocelli in NYC. The mixer could not be physically touched without sending out huge phantom power-switching bombs. We never found out what channel was doing it but eventually stuck strictly to iPad control to avoid it. I offer no further comment on this.
Marshall, I’d vote this is applicable with any digital mixing system - when using plugins that are outside the box (mixer). Most digital platforms offer some form of ADC for processes that take place inside the native mix engine, but as soon as you go outside (SPRK, analog inserts, UAD, etc..) all bets are off. The Waves and UAD options do offer some latency compensation that can align WITHIN their host, but doesn’t account for any signal buffer going in and out of your mix system. My recommendation is to always have a measurement path setup with an FFT so you can verify that all impulses leave the desk at the same time.
Hi Dave. Don't be discouraged with negative comments. You knowledge is amazing. Keep up the good work.
Thank you thank you
@@DaveRat You are very welcome
every time dave posts ... im like.... thank god for this scientist
Thank you Ankit!
Dave I met you here in PHX at a festival about 20 years ago. Ive always admired your work and knowledge in our industry. Keep up the great work!
Hey hey Artie! Thank you!
I hope my question (about the xenxy) from some time ago didnt got picked up as annoying. You gave a very nice answer and I learnt something new that day.
Keep up the good work! Love your vids
All good! The the annoying comment human went on to get himself deleted by being rude to others of the cool people here
Thanks Dave, keep doing what you're doing.
Thank you Kwame!
I know this one...
Fader position digitisation. The rough part of the chirp is the rate at which the fader position is sampled. The squaek part is an artifact from the smoothing algorithm for the rough digitisation.
I first met this when installing an alpha test version of the Amec Digital ( was supposed to be beta, but it really wasn't that ready). With no consideration of fader position sampling, the faders sounded scratchy. Yup very expensive digital scratchy faders.
Interesting and thank you Douglas!
This is such common, and old, ground in digital audio design I can’t believe anybody actually still misses it. It is truly 101. There are even complete circuits on a single surface mount chip available from Burr Brown, Analog Devices, Cirrus Digital, and others that handle the entire function, using either a generic linear analog pot or circular encoder to effect a full solution of extremely high spec. I’m not saying it’s a piece of cake, as power and ground must still be properly configured, but it ain’t exactly rocket surgery anymore.
@@artysanmobile There is a tradeoff between responsive faders and smooth faders all restricted by budget, both processing and hardware.
The test case here could be considered a bit "bizarre" generating a sound and then nulling it. It would be handy to get a signal to chirp ratio.
Another fun fact from the Amek desk... Occasionally it would incorrectly output 3 duplicate samples. It was barely audible on real programme audio, but stood out like a sore thumb on tone. I really never again want to spend 8 hours listening to 1kHz waiting for a click that was as loud as tapping 2 fingernails together.
@@dougaltolan3017 I don’t intend to argue with your direct experience, Dougal, but I do take exception to your statement that there is a gray area of fader operation quality. There is not. Signal to chirp ratio? Seriously? How about 140dB? Your comments on a prototype Amek don’t seem relevant.
Mute switches are for muting. Faders are for attenuating. ZERO is the only acceptable level of artefacts for professional equipment. I hate to be so hard core, but exceptions would never pass reviewing muster and should be returned for refund. I own and have used digital mixers since the 1990s every day without any of these issues.
@@artysanmobile ZERO eh? I challenge you to show me any piece of analog audio equipment with ZERO noise.
Gear heads tend to be hyper polarizing with their opinions, but I appreciate your level-headed approach to these videos!
Thank you Mark!
Great presentation Dave...I immediately thought about the benefits of better (i.e. Penny & Giles vs Alps) faders in the analog domain. (Obviously won't help here) Everything is a compromise in life. Some things are better in the digital domain, or class D amps, et. al, but there is always tradeoff, you're doing terrific job bringing light to exactly what that means through the signal chain. Thanks so much, Mike
Thank you Mike! And that's interesting. I don't know if the faders for digital are different or not. Are they linear encoders? Or are they resistive analog type faders that vary a DC voltage which the digital circuit interprets? Or?
I must admit I assumed they were like a vca, where they slide between ground and a fixed voltage and a digital circuit reads that voltage and implements the associated amount of gain or attenuation.
That said, the rotary knobs that endlessly spin are encoders without fixed end stops like the faders.
Interesting! I love breaking assumptions!
Another excellent and interesting video, thank you.
Thank you Sol!
Lovely 😊
👍
Thanks for the great content
Awesome thank you!
Excellent deep dive into the chirp issue. You have proven I am not crazy because I have mentioned this to a couple colleagues who looked at me as if I’d completely lost it. Have you experienced any issues with the faders on either the x or m32? We have several of each and the M32s seem to have occasional dead fader symptoms. They still work, but are no longer motorized and is a pain when jumping from layer to layer. I do know the x32 faders are soldered in place vs wired with a small connector and two screws on the M32. Thanks again and I appreciate your efforts to go deep into these common use products in the industry.
Motorized faders do fail on all consoles. We don't own any m or x 32 other than purchased for the vid so I don't have experience with how the deal with time and usage
It seems that every forum /video format has it’s trolls. I guess some folks just need to be jerks, eh? Keep up your good work, Dave!
Thank you Mike!
Great content. Very interesting and informative. Keep it up.
Awesome and thank you!
Intresting topic Dave, i've noticed a little ''chirp'' on multiple consoles too. most of them were just with the mute though.
Interesting. I am told that as processing power increases, the chirp should raise in frequency to beyond audibility in newer consoles.
Here's the background information to understand what is happening here, from a digital console designer.
I am interested to see the 6kHz, which is probably a very good result as this is the update rate of the gain coefficients. The effect of changing the gain (even in an analogue console) is to modulate the channel output with the gain change frequency, exactly the same process of frequency mixing used throughout electronics. The important part from an audio quality point of view is that the change is spread linearly across a large number of samples, not suddenly applied to one, and then keeping that value for the following samples until the next fader value is received. However, this smoothing of change rate is quite processing-intensive, and therefore is rarely perfect, and is less and less necessary as the update rate is increased. The 6kHz signal level Dave is seeing will be small compared to the 20Hz signal, and with modest fader position changes inaudible. The mute button obviously switches in one sample, so the 6kHZ is larger, but then the mute button is not usually used when a signal is present when there will be no 6kHz, so is much less important. Devices like the gates and compressors will not show this effect as the gain transitions are fully "smoothed" with an effective sample rate of 48kHz or whatever. How many dB down from peak is the 6kHz Dave, probably about -60?
Also I am told the Behringer Wing does not do it and none of the other consoles I tested do it. It seems to be unique to the m32 and x32 and other derivative designs by Behringer from that design
I was told that the issue is related to the X32 and M32 processing in bloks of 8 due to lacking the processing to uses smaller blocks due to processing power shortages.whatever that means.
Now. Who was it that said digital would be the bee's knees?
Really enjoying your tenacity in digging into these annoying things that the sales brouchers dont go into 👍.
Happy holidays to you and your's Dave. Stay safe.
Thank you Shane and happy holidays!
Hey Dave, love this short series, I've use the x32 extensively to mix my church's broadcast mix. I wonder if you have run into a this strange behavior where when you link two channels together, there would be a very small, but noticeable drop in volume. I noticed this and experimented by copying all the channel processing and routing and panned the channels left and right respectively, and the behavior still continued. Just something I found interesting and thought would be cool to see you dive into if you ever get around to it. You rock!
How do you mean? Do you pan them back together after you link them? What input material is there? It might just be a pan law thing you are experiencing pherhaps? :)
I saw another comment on this exact subject. Not sure if it was on this video or another. And also some useful responses.
It's prob a subject is won't dive into but is interesting
Dave, I quickly replicated on my X32rack with no physical faders or mute buttons, only pure software control, and I hear basically the same thing. So it's not simply the physical interface or lack of proper mute button debounce/LPF, or physical fader resolution (Although I guess it could still be GUI fader resolution). I'm not sure how the muting is implemented - of course abruptly dropping or increasing the envelope of a signal will have momentary high frequency content, so I would expect muting/unmuting to have high frequency artifacts if they didn't make the envelope transitions slow and smooth (S curve, etc). You would think in a digital signal path it would be easy to make this muting envelope transition "nice". I may put in on my scope to see what's actually going on. But the fader should be much less severe in that respect, as the digital resolution should be quite good, so possibly something else is happening?
FYI, I did put it into my scope and triggered off the muting, and see that in the x32 rack, the muting function seems to use a simple 1st order LPF envelope function with a time constant of ~20ms. So it doesn't use any smoother fancy S curve type envelope. But other than the 1st order envelope LPF, I do not see any glitches or problems in the waveform, waveform is nice and smooth and continuous. Of course there is still high frequency information in an envelope change that is LPF'd, as the step response of a 1st order LPF does not have a continuous derivative. So it seems we are just hearing those artifacts.
actually, I can directly see the 6kHz problem on the scope. The LPF'd muting envelope is done with digital gain steps, and those occur at ~166us (-1/6kHz) intervals. By zooming I can clearly see the gain steps as the envelope increases, hence that 6kHz junk you are finding. I have interesting scope pictures if you are interested.
Very interesting! Thank you!
@@DaveRat I played a little more, and I can capture it carefully on the fader movement as well. Due to the 48kHz/8 sampling of the digital gain control signal, the low frequency waveform essentially is modulated up to around 6kHz, where you can hear it through your high pass filter. This effect will be directly proportional to the speed at which the fader moves. If the fader is moved very quickly, it will be loudest (should sound similar to rapidly & repeatedly pressing mute/unmute, hence your "zipper" effect). If the fader is moved very slowly, the effect will be much less. I verified this as well. The good news is that as it is directly proportional to both the signal level and the fader movement speed, you unlikely to hear this in normal mixing scenarios, as the signal itself will be MUCH louder than the artifact, especially with gradual fader adjustments.
would love to see you do some analysis on the Allen and Heath SQ series mixers.
Will see what I can do
Fascinating stuff Dave! I’d be interest to hear if the chirping appears with other processing like compressors or gates.
I haven’t played with the internal “rack fx” compressors to know if they can sidechain but the channel strip comps definitely can
Interesting, I doubt it happens with internal non interface controlled levels. But curious to know
Could you do some latency and phase comparison using the LV1/dLive/Analog? Here in north Europe it seems to be to consoles of choice for most touring FOH techs. Maybe also put a Digico in the batch?
Working in a Digico test now
@@DaveRat cool! Thanks for replying. I'm using lv1 so I hope you will have a chance to test that too
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I’ve always heard that but thought it was a squeak in the button itself. Never thought of it as in the actual signal path.
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Really appreciate all these videos, super interesting. I'm just home hobby type of person and just kind of look at things really high level, like I can mix a song pretty easily in the analog world and it feels next to impossible in the box. Just way harder and I don't know why. Anyway, took your idea with the dual mono compressors on busses of like instruments, like vocals in a group, drums and bass in a group, other instruments in a group, light compression, a couple db. That really helps mixing a tune fast too. Really appreciate all the stuff you come up with.
Maybe a comment on the muting. I have a A&H GL2400 which sounds decent, but those mutes as just switches drive me batty with the clicks. My A&H 4000 has the relay mutes which is just a million times better. It's like these digital board need to do that too, but obviously through programming and not relays. Happy Holidays to you and your family!
Happy holidays and thank you Adam!
This was fascinating! I've never cared for these consoles and I've never been able to put my finger on it. I've heard the chirp many times. And I'm not saying this is the reason for my bias. It's a tool, use it. Watching his, being a fader rider at times. Has this been part of my experience? Dunno? I am very curious to see what you find with the QU16. Qu series are the most common install desk around here. I like them. A MixWizard on steroids. I don't think I've heard one chirp or something similar.
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Nice digging into the M/X32 world Dave! Listen to the differance in linking 2 channels vs to separate (not linked) L/R Channels.Would like to hear what you think about that. ;-) Greetings from the Netherlands
Interesting, what am I listening for or looking for? I only have the x32 now as I have the m32 back to my friend who loaned it.
@@DaveRat Level and sound quality change. Propably fase related. Kind of what post below is discribing about the fx inserts. Had some funny things with that also.
seconded. The level loss when stereo linking channels on the X32 is very obvious. Would be interesting to know what exactly is going on there
Ok, will ponder that and see how things go as I compare to higher end consoles
I have noticed the same, huge difference between the sound of 2 linked channels vs not linked. Of course both always panned hard left and right. It's very noticeable right away using music for instance and headphones
As a software engineer who has worked on audio consoles, MUTES should either happen at the zero-crossing point of a signal or be a fast fade. Unfortunately, with a low frequency signal such as you're using, this can be problematic as it means that the MUTE is too slow. This means there are trade-offs. If you performed the same test at 1kHz, you may not hear any artifacts. Fades are a similar problem, ideally you fade at very small dB change levels or fade at the zero crossing point of a signal. Again the low frequency of your source is showing up an inherent trade-off between speed of operation and zipper noise. I would guess that 96kHz or 192kHz sampling might make this better as you can have more 'attenuation points' during the fade/mute, but I think the noise will only be reduced by 6/12dB if that's the case. I'd be interested to see a graph of chirp level versus source frequency for several different consoles.
Consoles I have tested other than the x32 and m32 do not chirp. The SD11 and Yamaha CL1 are chirpless and run at 48k.
Also analog consoles do not chirp.
A mute click is pretty common.
I will test zipper noise and such on the Yamaha CL1 I am testing
Dave, love the comparison series you’ve been doing on these consoles. I’m curious if these differences carry over into the digital stage boxes for these units as what’s being received by the mixer is a digital signal vs analog.
To the best of my knowledge the stage box a to d and d to a are the same as the same branded consoles
New test for you Dave! I wonder if you set these consoles to run over Dante and use high end I/O (Digico,Yamaha or Midas) will these systems take on some of the sonic attributes of those consols?
Most likely. The noise issues are I/O related. Ping and zipper noise are control related.
I'm going to have to try this on my X32 rack. I wonder if it's a function of the encoder itself or if that's in the software.
I've heard it's related to the way the mixer processes the digital signal I've also been told the wing should not have the ping
@@DaveRat I was able to replicate the zipper on my rack via a mix bus this weekend. Rapidly cutting a mix bus used for vocal delay sent that odd noise through my PA.
Very cool and it's fun and useful to know what these artifacts are
Wow! that's bizar!! It is really amazing of you know things about those cosoles no one ever gets to know... GREAT!!! Do you ever get feedback from Behringer about your findings? Are they learning from your findings...? (You should get paid for your deep-testing of their stuff! hahaha!)
Thank you Edwin. I am in touch with Robert who moderates the Behringer forums and he gives me info when I have questions and clarifies things.
That said, I try not to get too close to manufacturers so I can avoid bias. I test and ask questions if there are things that puzzle me and I feel I may be off base in my tests.
@@DaveRat wow!!! very good answer!! thanks! :-)
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I'm guessing that "sound of the fader resolution" (LOL) might not be heard while using the rack version with iPad on these units? This is a mechanical artifact yeah?
Not mechanical and should be the same with an iPad. But also, as I mention, the lever of these issues is very low. I set up to make them audible.
Oooh Digico $$$$$$$ fancy smancy
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Hey Dave. Ide like your opinion on the presonus brand
Not a lot of exposure but have owned a few useful units that worked well at a good price
@@DaveRat Cheers Dave
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Haha this is very interesting! As someone else allready said. A signal to "chirp" ratio would have been nice info to have. How much are you amplifiying this noise for us to be able to hear it in the video?
I´ve worked hundreds of shows on a M32 at this point and I have never experienced this in practise. Not even with very quiet source material and high gains settings on condendser mics like a Choir singing pianissimo dynamics over a few piano notes. I´d never touch the gain or mute in such a situation but i defintly ride the fader sometimes and I never heard it.
You consider it a practical issue or just a fun fact? :)
Been following you on and off ever since the papers preceeding your "orgasmatron" Sub array with the A. B, C and D stack of 3xSB28s each with different phasing and delays. You even had a carpenter cut out a board for you to set it up faster. That was cool :)
It just something to be aware of. I heard it when testing and muting low freq signals, then setup a test to isolate it.
For most applications, it does not matter, just like a freq response over 16k or distortion over . 5%
But when comparing to higher priced consoles or applications where perfection is more important, it's good to know.
Hey Dave!
My question is a bit unrelated to the matter at hand - have you tried seeing how the digital aes50 signal could be passed in "reverse" through rat tails into a 4 channel xlr snake?
I tend to work with x32/m32 with the aes50 stageboxes and usually tend to work places with existing xlr snake stage boxes.
I've seen (binged is more accurate) you describe how your cat to xlr systems work, and the termination makes sense that it would work but the length of xlr running could be an issue with interference.
I would love to hear your thoughts on the matter.
Cheers from Haifa!
Actually yes, I have tested exactly that. Definitely loose lots of length, for short runs it can work but aes50 is raw digital without error correction so caution is advised. AES50 is very very poshy and temperamental and very set in travelling down familiar paths.
@@DaveRat Thank you for the quick and helpful answer!
Besides aes50, have you tried other digital protocols running on cat cable to see how they fare?
Not test wise. AES50 is a raw, non error corrected signal to the best of my knowledge, that keeps the latency super low but makes it less tolerant to issues and more likely to have very loud issues vs muting undesirable sounds.
Haaa! you can hear the LSBs / MSBs.
The two consoles look fantastic. Makes me want one and I don't need it.
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do you have the gain really cranked to be abe to hear it? ever oticed it in norma gig envirements or even listenig with headphones.
what firmware version is on the console?
going to try repicate this today whe im at an x32
Latest firmware. Yes, gain is hot
@@DaveRat I replicated your results using the desks internal oscillator. Sent oscillator to a bus which was out of stereo. Used a channel to take source of the bus having tone sent to it, sent that channel to another subgroup and filtered out the 20hz tone.
I was sending 20hz to subgroup at -18dB.
The chirp is clearly audible with cranking the desk out or subgroup out up.
I tried it with pink noise aswell, it's audible but becomes less and less audible the less that's filtered out.
Interesting and very cool. I am testing the sd11 and it is different but there is a chirp on the master but not the group or channel
@@DaveRat another venue i work has ah qu will give that a try
Every M32 I've worked on has an audible "chirp" at what I believe is 16k. I haven't heard this same thing on an X32, but that doesn't mean that it's not there. Have you looked into it?
Must be more clear... the "chirp" happens when switching layers.
Yep, both are the same chirp
I've heard a similar noise when turning up and down the volume on my phone while using app-based signal generators.
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Well, if you don’t move the faders, use the mutes, or change head amp gains, they are wonderful mixers. 🤯
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@@DaveRat I tried to file the edges off my snark.
I noted a reading on your scope of approx 5mV. Assuming that was the amplitude of the chirping at that moment, it’s quite inaudible, well above the noise floor. Pretty alarming stuff.
I’m so grateful you take the time to do these investigations, Dave. I will be using Calrec mixers in January and February for Olympics and will have the time to check them. I have already done some ultra deep dives on the Lawo consoles. With phones and a live Sennheiser HD27 dynamic mic headset in an empty soundproof announce booth, I have run the digital head amp gain up and down from 10 to more than 60dB, and absolute silence was the result. I mean DEAD silence in screaming hot phones, to the point I can hear myself clap 70’ away, outside the closed booth. Ditto mutes. Lawo is big money stuff but still the point is, it CAN be done. I believe you will find similar results with your SD11.
Fun and thank you. Yes, interesting stuff and for me it's all kind of stumbling around and see what shows up as I dive in. The aspects that seem most interesting are the least expected.
@@DaveRat I am most interested in learning which, if any, of the lower cost manufacturers have done their homework. Clearly the Midinger conglomerate has skipped right over some essential developement. I’m sad about that because I don’t know how far up the Midas Pro line this disease goes. I will be scrutinizing every mixer I encounter on the road time permitting and would like to share my findings on your channel. I promise to keep them brief. Mute, fader, head gain; quiet or not.
Well, let's see how these other consoles test out. I am remiss to make assumptions and have seen even the best manufacturers stumble.
Does it still do the chirping when the channel is affected by a DCA?
Have not tested but to the best of my knowledge the chirp happens whenever there is a level change regardless of what you use to change the level
I would expect the DCA to have an identical result.
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Sonic Youth!
Hell yeah! I toured with them on Goo and Dirty tours as a system tech and pa provider. Love them!
I think a great test would be to compare the Behringer with the midas stage box vs just the midas.
I genuinely wonder if there'd be any difference.
My tests comparing the x32 vs x32 with m32 as a stagebox shows that the audio advantages are determined by the the stagebox.
The interface and durability advantages are determined by the whatever the engineer is operating.
As I move into testing the Digico sd11, I see clearly that both the x32 and m32 are cheap toys as far as ruggedness, but the m32 is much less toy like than the x32
I have a Wing. I’ll see if I can replicate this test on it and report back with my findings.
Cool an yes, the wing should not ping and curious to know.
Imagine when you have two digital audio sampled systems feeding audio to each other, each using slightly different sample clocks..eventually a buffer gets starved or will overrun. I have designed systems that detect this plesiochronous condition and adjust samples rates accordingly.
True and in all these tests I use analog in and analog out so sample rate drift has zero impact. With aes50, the consoles clock over the aes50 connection.
Chosing one console or the other to clock when they are digitally interconnected is a must and the norm.
Once the signal is analog, whether it was clocked at one rate or another in the signal's precious life as a digital signal is as meaningless a whether a mic cable was stored in a red case or blue case before being plugged in.
Anyone who is questioning Dave and his credibility and credentials just shows that they really have no clue or don’t even work in this industry. I work on the periphery doing mostly Cinema sound and even I knew who Dave was before these videos started!
Very cool discovery, very not cool feature... I will have to try this on every board I get access to.
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