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Motu M4 - Tear down, bit of internals analysis and few in-house measurements

So, I should accept the current behaviour as normal and that there's not much they can do 'practically' to make the knob behave more like a normal smooth volume dial, correct?
Pretty much.

I wouldn't expect too much in terms of "easy fixes" in this price class with years worth of design experience on behalf of the manufacturer (or rather the engineering firm they're contracting out to, as seems to be the case for MOTU). You'd stand a better chance in the super budget class, which seems to be the Wild West of audio interfaces.
That's hardly the most outrageous phenomena I've seen either - a review of the ESI Neva Uno/Duo series had a vocal sample with a dynamic mic that sounded like genuine telephone quality (no bass, no highs and noisy), and while a Swissonic Audio 1 had decent noise level on the dynamic it was kind of distorted (not to mention that both Audio 1 and Audio 2 seem to feature a bass boost of sorts).
 
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So, I should accept the current behaviour as normal and that there's not much they can do 'practically' to make the knob behave more like a normal smooth volume dial, correct? This stuff is a bit beyond me so I was thinking that maybe it was a simple tweak/fix on their end, and that they just didn't really consider it since they weren't making a mixer but instead an interface (where you adjust your gain once or twice and move on to the DAW). I'm always finding different ways to use my gear and always reveal these types of problems as I go. Maybe I'm the one needing tweaking/fixing, haha! Thanks for your input! :)
I think it's both normal behaviour and what should be expected. The pots in question are labelled "Gain", not "Volume", indicating they are not meant to be used as volume controls. They should be set such that the signal from the connected source will not lead to clipping. As you have not said what you are using it for, I can't make any assumptions about your setup and workflow, but I set and forget the gain, and use volume controls in my DAW and mixer.

/Richard
 
This took some time; I have too many projects ongoing at once...

Voila, two 7-step precision attenuators to replace the gain pots. They should give -17, -10, 0, +10, +20, +30 and +40dB. Not exactly beautiful SMD work, but they'll do the job.

20240407_230653.jpg


I have butchered a couple of cheap pots to make mounting flanges; they need a bit of drilling and machining, and will then be glued on top of the switches.
20240407_230836.jpg


Nicely and securely mounted, now I need to shave off the flange that protrudes over the hex nuts, and make 4.2mm-6mm adapters so the knobs will fit.
20240407_230506.jpg


It fits in perfectly in place of the pot.
20240407_230422.jpg
 
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This took some time; I have too many projects ongoing at once...

Voila, two 7-step precision attenuators to replace the gain pots. They should give -17, -10, 0, +10, +20, +30 and +40dB. Not exactly beautiful SMD work, but they'll do the job.

View attachment 362119

I have butchered a couple of cheap pots to make mounting flanges; they need a bit of drilling and machining, and will then be glued on top of the switches.
View attachment 362120

Nicely and securely mounted, now I need to shave off the flange that protrudes over the hex nuts, and make 4.2mm-6mm adapters so the knobs will fit.
View attachment 362121

It fits in perfectly in place of the pot.
View attachment 362122
Wow that took some work!
Probably too late now, but for these kind of small mods, I would recommend you jlcpcb. You can get a bunch of those PCBs printed any shape you want and shipped to you, probably for ca. 5 euros or less, being so small.
 
Mission completed. Or, at least functionally. Aesthetically not quite there; I have to make the original knobs fit the tiny shafts of the rotary switches. I'll butcher the potentiometers they replaced, drill 4.2mm holes into the 6mm shafts from the back end, and fit set screws.

20240410_215310.jpg


Output from the signal generator left channel (both channels read the same):
20240410_220822.jpg


And reading in Audiotester:
AudioTester.JPG


Which is spot on.

Now I know I can just hook things up, fire up Audiotester and the M4, and AudioTester will display the absolute value of the input signal. With the original pots for gain control, I would have to calibrate Audiotester every time the pots had been touched.

@AnalogSteph you were right about the resistor values, even though I was lucky and hit exactly 0dB. The attenuation did not turn out exactly as planned; the next lower gain step was supposed to be -10dB, but it turned out to be -11.05dB. No big deal, I'll use the 0dB setting 99% of the time anyway.

@AnalogSteph you were right about the resistor values, even though I was lucky and hit exactly 0dB.

If it's really a pot rather than a mechanical encoder, I would assume it's being used to tap off a varying percentage of a DC voltage which is then filtered and fed into an ADC (8 bits as commonly available on micros would do and provide 256 discrete values, in which case I'd guess it would be a linear pot). Prototyping may involve fiddling with a bunch of pots to get the set points just right, followed by playing with resistor ladder values in simulation.

Gain control itself is probably a hybrid of THAT6263 control in 3 dB increments + digital gain adjustment for the intermediate steps plus extension on top (the chip itself only has a 42 dB gain range, -8 to +34 dB).

But why use THAT6263 in 3dB increments? According to the data sheet, it has a mode with 1dB increments.

Richard
 
I would like to modify inputs 3/4 to be DC coupled. Before I crack the case open I thought I would ask if anyone as looked at this and identified the bypass capacitors in the circuit. It would be great to see the schematic first, but I have not seen one posted. Thanks
 
If you really need recording down to DC, I don't think this is going to be sufficient by itself... the ADC generally has a DC removal filter on the digital side that would have to be disabled by setting up its registers appropriately. Modifying the circuitry is one thing, but getting into the firmware is another matter entirely.
 
If you really need recording down to DC, I don't think this is going to be sufficient by itself... the ADC generally has a DC removal filter on the digital side that would have to be disabled by setting up its registers appropriately. Modifying the circuitry is one thing, but getting into the firmware is another matter entirely.
Brilliant. Thank you. It did not occur to me that there would be an internal digital filter. My measurement found a 2.4 Hz cut off that matches nicely with the spec for the internal filter. Looking at the data sheet for the AKD5554 I see there that holding pin 44 high enables the HPF (high pass filter). Apparently all I need to do is get out the microscope and cut the trace to pin 44 and ground it to disable the internal high pass filter. I'm assuming that the pin selection can't be over ridden by the software register.

I also measured an input impedance of around 5 kOhms single ended or 10 kOhms balanced for inputs 3 and 4. That is not in the MOTU data sheet for some reason.
 
@OlsonSystems, do keep us updated with your work. Also, could you let us know why do you want to do these modifications (for what exact purpose)? Thanks!
 
Apparently all I need to do is get out the microscope and cut the trace to pin 44 and ground it to disable the internal high pass filter.
It might be a bit tricky to do. If I see correctly it is connected to pin 45 with a very short trace, and you cannot bring 45 low. It is probably doable with a very good hand but wow!
The good news is that if you manage to cut it, you only need to short with a bit of solder to pin 43 that is probably low and that's it (unless you need a resistor to ground, but 43 doesn't seem to have one...)
 
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@OlsonSystems, do keep us updated with your work. Also, could you let us know why do you want to do these modifications (for what exact purpose)? Thanks!
Madness really. I want a high resolution DC coupled phono preamp that allows me to recreate the record groove profile without phase error at low frequencies. I have always wondered what it was that many people like about the sound of records, so I went deep down the rabbit hole investigating phono playback distortion for the past two months. Is it the mono bass below 100 Hz, the reduced separation, the high and higher distortion, the surface noise, wow and flutter, the limited dynamic range? I found www.pspacialaudio.com that has nice explanations of tracing distortion and "pinch" distortion due to non-zero stylus radius tracing a different path than the cutting head and large excursions causing an effectively narrower groove that modulates the tracking height of the stylus. Think of a wheel rolling over hills and valleys, the path traced by the wheel has the hill tops made wider and the valleys narrowed due to the wheel not contacting at the bottom on the slopes like it does on the peaks. So record playback has a 2nd harmonic distortion of 0.5 - 5% or more depending on the slope of the waveform traced. There is a way of calculating this distortion and compensating for it. I bought the software from that website. It only runs on a Mac computer. It rejects all my attempts to verify it works using test waveforms in files I feed it, so I am creating my own algorithm to remove the distortion from a record playback capture or add it to a digital audio source to simulate record playback. Anyone with a similar curiosity or vinyl record based mental illness can read my google docs. I documented the record creation and playback signal chain and my measurements of several cartridges using the Ortofon test record as a source. Without a test record or known signal source there is no way to measure anything. The files were too large, so here are links. They are poorly organized at the moment. Remember, it's madness, but these are measured results and unlike all the "talk talk talk" bs reviews in magazines and youTube videos about the sound of vinyl records, these tests can be run and the results can be replicated by anyone.





I built the simplest gain of 100 balanced buffer using two LT1037 op amps and driving inputs 3 & 4. There were problems using inputs 1 & 2 and there are problems with performance if the headphone amp is used while recording. The setup works so well that I was shocked to
see line noise in one of my test record rips. It turns out it was at 50 Hz, not my local 60 Hz, and the noise was in the recording. My 60 Hz noise is at the noise floor of the system.

So with DC coupling I should be able to integrate the signal from the phono cartridge and have a better chance of accurately reproducing the record groove displacement waveform for use by the distortion estimator.

Here's the MM phono buffer schematic. The R17, 18 C4, 5 are not in the build. They are there for the noise calculation to include the RIAA roll off that would be added to the playback after capture and processing.
1718138345769.png
 
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hello, sorry for asking on an old thread but since this is a tear-down thread i thought it would be the best place to ask before i pull the trigger for a motu M..

can anyone confirm that the hardware mixing/monitoring signal path of this interface is on all-through analog circuit?

i.e the analog input signals is not converted to digital domain by the A/D converter and then routed to output(s) by a D/A conversion on board, right?.. i read the manuals and it just says it is hardware direct-monitoring but some brands relatively fake this term by doing A/D-D/A conversion on-board with an unnoticable latency for easy routing and calling this zero-latency-direct-monitoring while techically it is not.. asking this since the manual doesn't have a signal flow diagram and not sure but the buttons/knobs seems like digital ones to me..

thanks
 
hello, sorry for asking on an old thread but since this is a tear-down thread i thought it would be the best place to ask before i pull the trigger for a motu M..

can anyone confirm that the hardware mixing/monitoring signal path of this interface is on all-through analog circuit?

i.e the analog input signals is not converted to digital domain by the A/D converter and then routed to output(s) by a D/A conversion on board, right?.. i read the manuals and it just says it is hardware direct-monitoring but some brands relatively fake this term by doing A/D-D/A conversion on-board with an unnoticable latency for easy routing and calling this zero-latency-direct-monitoring while techically it is not.. asking this since the manual doesn't have a signal flow diagram and not sure but the buttons/knobs seems like digital ones to me..

thanks
Any process (routing, mixing, monitoring) is managed in the digital domain.
Only the output after DA stages are analog, and of course the inputs before the AD stage.
But the audio interfaces all have been working like this for years.
Why are you looking for an analog domain operation?
If it's a matter of latency, don't worry it's well below the threshold of unusability.
 
Any process (routing, mixing, monitoring) is managed in the digital domain.
Only the output after DA stages are analog, and of course the inputs before the AD stage.
But the audio interfaces all have been working like this for years.
Why are you looking for an analog domain operation?
If it's a matter of latency, don't worry it's well below the threshold of unusability.
it is not about latency, it is about retaining a full analog signal path through the mixing/monitoring stage.
should i assume the same digital domain management happens on the so called analog mixers with usb interfaces?
and/or can you recommend me interface(s) with a full-analog direct monitoring/mixing feature which is at least in the same league or above the motu M-series
sorry if this is not the correct thread to ask
 
it is not about latency, it is about retaining a full analog signal path through the mixing/monitoring stage.
should i assume the same digital domain management happens on the so called analog mixers with usb interfaces?
and/or can you recommend me interface(s) with a full-analog direct monitoring/mixing feature which is at least in the same league or above the motu M-series
sorry if this is not the correct thread to ask
I find it a little hard to give you an adequate answer because it seems to my eyes that you come from another world.
I give you the most direct answer:
The audio world has been working digitally for years, and for reasonable reasons, so if you believe that analog has a tangible advantage over digital you should seriously review your ideas, because you won't find anything on the market apart from vintage components.
Full analog path in a mixer makes me shiver...
 
I find it a little hard to give you an adequate answer because it seems to my eyes that you come from another world.
I give you the most direct answer:
The audio world has been working digitally for years, and for reasonable reasons, so if you believe that analog has a tangible advantage over digital you should seriously review your ideas, because you won't find anything on the market apart from vintage components.
Full analog path in a mixer makes me shiver...
yes, it seems like i'm coming from a different world indeed.. you sound like there is not any analog product in your world..
it is not about digital vs. analog adventage-ing.. there are reasons i need to avoid adding "another" AD/DA conversion in the signal path
thanks anyway for informing me about the hardware direct monitoring is happening through the AD/DA converters assuming you are not mixing the info of the motu-M with your motu-ultralite which has a dsp mixer onboard
 
yes, it seems like i'm coming from a different world indeed.. you sound like there is not any analog product in your world..
it is not about digital vs. analog adventage-ing.. there are reasons i need to avoid adding "another" AD/DA conversion in the signal path
thanks anyway for informing me about the hardware direct monitoring is happening through the AD/DA converters assuming you are not mixing the info of the motu-M with your motu-ultralite which has a dsp mixer onboard

I don't understand why you say you need to avoid AD-DA conversion. It's typically a paranoia for audiophiles... unless there are concerns about latency, which as already mentioned are unfounded (speaking of direct monitoring).
The M2 or M4 works digitally like all audio interfaces. I'm not mistaken about this. There are also a thousand reasons why it cannot be otherwise.
 
I don't understand why you say you need to avoid AD-DA conversion. It's typically a paranoia for audiophiles... unless there are concerns about latency, which as already mentioned are unfounded (speaking of direct monitoring).
The M2 or M4 works digitally like all audio interfaces. I'm not mistaken about this. There are also a thousand reasons why it cannot be otherwise.
there are several reasons i need to avoid "another" "unnecessary" AD/DA conversion, maybe in an academic research perspective, that is long story..
but outside of that story, the conversion is still unnecessary cause we are not adding an effect or eq'ing the signal hence the "direct" monitoring.. also i don't understand why you are talking like it is technically impossible to implement all this within an interface.. like i can just put a full analog mixer infront of the interface, direct monitor from there while feeding the interface inputs.. it surely is technically possible doing all that within a single unit instead of running the cables between 2 products..
 
there are several reasons i need to avoid "another" "unnecessary" AD/DA conversion, maybe in an academic research perspective, that is long story..
but outside of that story, the conversion is still unnecessary cause we are not adding an effect or eq'ing the signal hence the "direct" monitoring.. also i don't understand why you are talking like it is technically impossible to implement all this within an interface.. like i can just put a full analog mixer infront of the interface, direct monitor from there while feeding the interface inputs.. it surely is technically possible doing all that within a single unit instead of running the cables between 2 products..
I am not saying that it is technically impossible, only that for audio interfaces it is more convenient to manage the routing (and volume) digitally, even without mixing and effects.
This practically satisfies all use cases... but I don't understand if yours is included or not...
 
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