Topic: octamic2 or micstacy

the time has come for me to purchase some mics and pre.  I have never really used mics or preamps before ,but i find that i am needing to record vocals, and some live instruments. ..

WHat is the biggest difference between these two units, octamic2 and micstacy?... both have 8 channel a->d. and i presume have the same pres,
the functionality to me seems to be very similar yet one is twice the price of the other. Why would i purchase micstasy over octamic2?


I would also like to know please if anyone see any sync issues or voltage issues with this setup? Does it look ok?

8 x microphones--> octamic2/micstasy---> adatout--> adi648 ----> hdspmadi in (masterclock)
i should be able to record 8 channels simultaneously?

Re: octamic2 or micstacy

The Micstasy stands in a class of it?s own. Many high-class recordings - some even Grammy nominated - were done with it in the last time. It just won the Editors Choice Award 2008 of the Professional Audio Magazin. Besides the high-end quality of preamps and AD conversion, the Micstasy offers a unique flexibility and usability. Just take a look on the control elements on the front and such awesome features, like Autoset and the individual control and link the channel gain settings in different ways. The Micstasy M is completely remote controllable from a DAW via MADI. The MADI option and the chance, to use eight channels as digital return path, sending audio via MADI back to the Micstasy's ADAT and AES outputs, making also a useful difference to the OctaMic II. The Micstasy M with MADI option will perfectly fit in your setup, chaining both MADI devices to the HDSP MADI.

best regards
Knut

Re: octamic2 or micstacy

ill buy one of those then
sold
danke

Re: octamic2 or micstacy

hello knut
my system has 4 hdsp 9652 samplers feeding adat to a 648 - so each card is feeding 16 channels into the 648 at 24bit 48000 rate.  then the madi takes this audio into the daw via hdspmadi.  I was going to sacrifice one of these adat channels to fit in the micstasy 8 channels.

will the madi card option for micstasy-M be redundant for me?  Or does this open up 8 channels at higher sample rates by connecting to the hdsp and making a madi ring .... ??? im a bit confused there.
as mentioned, i was thinking of just using an adat connection into the ad648 from a micstacy, but there is a better way isnt there?

Re: octamic2 or micstacy

One MADI connection transports 64 channels with 44.1/48 kHz. With 4 x 16 you are using already all available input channels of a single HDSP MADI card. You are right, with the Micstasy on an ADAT Input or the MADI cable you will at least loose 8 channels of the sample inputs.

It?s not posssible, to use the Micstasy and all sample inputs on different samplerates at the same time (96 kHz Micstasy + 44.1 Samplers). The MADI card and the ADI-648 are locked to a single samplerate. Higher samplerates will reduce the amount of MADI channels in the chain. 96 kHz = 32 channels. 192 kHz = 16 channels.

Advantages of a Micstasy MADI connection over ADAT:

1. The MADI connection via a chain with the ADI-648 will not produce more channels, but allows a more flexible distribution of your input channels. The sampler inputs stay connected on the ADI-648 (with Matrix!) and you can swap - remote controlled - at any time between eight channels of one sampler and the Micstasy.

2. Recording with higher samplerates:
The Micstasy provides 96 kHz with all 8 channels over the two ADAT outputs. 4 channels with 192 kHz. This configuration will consume two ADAT inputs of the ADI-648. This means, you will loose one sampler completelely.
In the MADI chain (Micstasy M) of course all eight channels get the full 192 kHz. All samplers stay connected, but you have to power off them for the time of the 192 kHz session - they don?t support this samplerate.   

3. Remote control of ADI-648 and Micstasy over MADI. Swap your input channels from sampler to Micstasy as you need them - directly from your DAW remote control software.

4. Distance: ADAT cables will allow 10 - 15 m distance between ADI-648 and Micstasy. An optical MADI cable up to 2000 m.

5. Micstasy Wordclock sync: The Micstasy provides no ADAT input for a clock connection from the ADI-648. This means you have to use a seperate Wordclock cable from your clock master (DAW?) to the Micstasy. The MADI option provides a complete I/O with the input usable as clock source for the Micstasy.

6. Eight return channels: A MADI connection will provide you with eight digital back channels. Connect a simple ADAT (or AES) DA converter to the Micstasy and use eight outputs for studio monitors or headphone mixes for your artists.

7. Flexible position in the connection chain: As the Micstasy M uses only 8 channels, up to 56 channels can be passed through. The Micstasy can also stay in the chain before the ADI-648. With ADAT behind the ADI-648.

In the end, MADI is the more expensive, but by far the more flexible and convenient technology. The reason, why so much Micstasys with MADI option were used by our customers.

best regards
Knut

Re: octamic2 or micstacy

Greetings again from tasmania.

i am only moments away from purchasing my micstasy.
This means that there will be more questions from me. 

Your explanation are very good AdminKnut. Thanks for taking the time.
One question arises only.
QUote:
In the MADI chain (Micstasy M) of course all eight channels get the full 192 kHz. All samplers stay connected, but you have to power off them for the time of the 192 kHz session - they don?t support this samplerate.   

That does sound excellent.  But what does a DAW do that is set at 48khz 24 bit , with lots of music recorded in this clock and depth, how does it change over to record at a higher clock for some accoustic insttruments or voice? Put simply, within the one project in the DAW, how do i cope with audio recordings that are different animals? DO i need to re-sample up or down?

I have read the entire manual, and some questions have arisen from this too please , if i may.

1. Will I need 3 cable to make madi loop with adi 648 and madi card ?
2.Does The micstasy needs to be powered up to be part of the loop, even when not in use, to pass the madi along?  Will it be able to pass on without power? I am starting to think green and reduce carbon footprint here in my studio.
3.  Are the analogue inputs mirrored in the analogue outputs after the preamp? What would a use of these be?
4. Finally, my last question relates to function3 on micstasy.  M/S (D) I have no idea what mid/side encoding for digital inputs are? I thought this unit was a AD and not a DA? Can this unit decode adat to analogue? I'm very confused. Please enlighten me thanks.

Enjoy Musikmesse 2009.....
i wish i could be there too, and have some nice German beer.
best regards.
roon.

Re: octamic2 or micstacy

roon wrote:

That does sound excellent.  But what does a DAW do that is set at 48khz 24 bit , with lots of music recorded in this clock and depth, how does it change over to record at a higher clock for some accoustic insttruments or voice? Put simply, within the one project in the DAW, how do i cope with audio recordings that are different animals? DO i need to re-sample up or down?

Yes. Some DAWs will resample in real time (Samplitude). I'd just stick with the sample rate of the existing project, though...

1. Will I need 3 cable to make madi loop with adi 648 and madi card?

Yes. Goint from the Mics directly to the PC.

2.Does The micstasy needs to be powered up to be part of the loop, even when not in use, to pass the madi along?

Yes.

3.  Are the analogue inputs mirrored in the analogue outputs after the preamp? What would a use of these be?

You could use the analogue outs to feed a backup recorder or so....

4. Finally, my last question relates to function3 on micstasy.  M/S (D) I have no idea what mid/side encoding for digital inputs are? I thought this unit was a AD and not a DA? Can this unit decode adat to analogue? I'm very confused. Please enlighten me thanks.

The MS matrix works for the mic inputs, not digital inputs.

Regards
Daniel Fuchs
RME

Regards
Daniel Fuchs
RME

Re: octamic2 or micstacy

Ignorant to the MS matrix, i have done some research.  I need to understand how it works in the analogue world.

Let?s assume for sake of hypothesis that we are recording a sine from a mono point source.  We chose 2 microphones.  A direct or cardioid mic pointed directly at the source (MID - M).  And a figure 8 mic on a 90 degree axis point at the source of the primary reflections (SIDE). 

The direct mic will record the exact sound, and the off axis figure 8 mic will record reflections.  As it is a diaphragm mic, the positive side of the wave will be the left reflection and the negative amplitudes will be the right reflection.  Please feel free to correct my interpretation.

Some simple addition of ordinates of these amplitudes can isolate left and right and middle channels.  This is MS decoding.

Centre M-S decoding requires a sum-and-difference matrix, where you add one side signal to the mid signal to get the sum, and subtract the other side signal from the mid signal to get the difference. To do this, one of the side channels is shifted 180 degrees in phase: when the polarity-shifted signal is added to the mid signal, you get the difference between the two signals. Using M for mid, S for side, and -S for the polarity-shifted side,

M + S = left channel

M + (-S) = right channel

Of course the mid is already isolated in a track of its own.

That was not hard mathematics.   I have performed one FFT across a large domain on paper with a pencil in my lifetime. That was hard and a big waste of time.

SO how does this translate to the micstasy?  Do adjacent channels become the 2 microphone channels.  MID mic on channel 1 and SIDE mic on channel 2.  SO if MS is selected on both channels and phase invert and MS decode is on in channel 2 i would get the right hand side? Which output does the result come from? 1 or 2?

Perhaps i have even researched the wrong topic and this MS decoding is something completely different.

PS I am sorry for asking so many questions.  I do not want to come across as a LOUDMOUTH.  Indeed , the opposite is true.
with all due respects,
roon.

9

Re: octamic2 or micstacy

roon wrote:

Ignorant to the MS matrix, i have done some research.  I need to understand how it works in the analogue world.

Let?s assume for sake of hypothesis that we are recording a sine from a mono point source.  We chose 2 microphones.  A direct or cardioid mic pointed directly at the source (MID - M).  And a figure 8 mic on a 90 degree axis point at the source of the primary reflections (SIDE). 

The direct mic will record the exact sound, and the off axis figure 8 mic will record reflections.  As it is a diaphragm mic, the positive side of the wave will be the left reflection and the negative amplitudes will be the right reflection.  Please feel free to correct my interpretation.

This is probably the most common error of understanding that people have when they begin to learn about M/S.  The side mic is a single, monaural microphone - it has a single output, a single signal - it's mono.  Because it's a bi-directional mic, it JUST HAPPENS to have a pickup pattern comprised of two lobes of sensitivity - in this case, the mic is most sensitive to sounds arriving from the left and from the right.

Here's where it gets confusing:  one mic signal/channel, comprised of two lobes of sensitivity.  Even more confusing is the phase relationship between the two lobes, but if you think about the microphone's structure as having a diaphragm that is open to the air on both sides, and is made in such a way that diaphragm motion in one direction produces a positive voltage at the output, whereas diaphragm motion in the other direction produces negative voltage at the output.  In our M/S setup, if there was a pressure positive event (like an impulse or a balloon popping) happening on the left side, in the first instant, the diaphragm would move away from that, towards the right.  Let's say that we've set our mic so that when this happens, it produces a positive voltage at output (normally referred to as the "front" of the bidirectional mic).  If an similar pressure positive event happened to the right of our M/S array, the S mic's diaphragm would have to move towards the left, and following how the mic is set up, this would produce a negative voltage at output.  Perhaps you can also see that if you take our balloon and pop it somewhere in the same plane as the surface of the microphone, the pressure would reach both sides of the diaphragm at the same time and at the same intensity, so the diaphragm would not move - it's the most insensitive region of the mic's pickup pattern, or the null.

Remember that I'm talking about the instant that the impulse's wave front reaches the microphone diaphragm.  At any point in time afterward, different conditions will apply, but the underlying principles will still apply - I'm just trying to simplify things for the sake of the explanation.

The next stage is the route the signals from both our M and S microphones into a mixer.  In the good ol' analogue days, you'd actually use a splitter wire attached to the output of the S mic - one input, with two outputs.  Remember it's still just a single signal, but now we have two identical copies.  Now, we take one of those two outputs and invert the signal polarity (most easily done by swapping the wired connections in the XLR connector, exchanging pins 2 and 3).  Now we have one signal, with two outputs, and one of those is polarity inverted.  So now you have THREE inputs for the mixer:  One from the M mic which you then pan to the centre, and the S signal (not inverted) panned hard to the left, and the inverted S signal panned hard to the right.  You'd match the levels of the two S signals and then you'd vary the mix between how much of the M and how much of the Ss you'd like - more M and less Ss makes for more mono, and more Ss and less M making it wider and wider stereo.

The key here is to remember the function of the panpot.  The panpot for the M signal's channel is centre panned - meaning it is sending an equal intensity signal to both the left speaker and to the right - so yes, it's splitting the signal and sending it to two destinations, but in this case both splits are equal, and IN PHASE.  This give the listener the impression that the sound is emanating from a point in the middle of the speaker array.

Because we've split our S signal before even getting into the mixer, we're dealing with two channels - one with it's output to the speakers panned left and the other channel panned right.  But really, with a more clever mixer architecture, like what you can find in most modern digital mixers - why not simply put the polarity inversion inside the mixer - say, just before the panning.  Just as we saw with the M signal, we have one signal going to two speakers, with each split being equal and in phase.  For the S mic, let's just do the same thing with one exception - let's invert the polarity of the signal running to the right speaker, but still send an equal intensity of signal to both.

You laid out the math in your original post, but perhaps the math doesn't really show what's going on, so let's examine things closely.  A key item to remember is that two signals that are equal in intensity and IN PHASE, then they will combine constructively, and the result will be a louder output for that signal.  On the other hand, if you combine two identical signals, with identical intensity, but with one of them phase inverted, then they will combine destructively, and will cancel each other out, resulting in no output at the speakers.

So....

We have our M/S mic array set up pointing at a stage.  An event occurs towards the audience's (and the mic's) left.  When the wave front of the initial impulse reaches the diaphragm of the M mic, it will move backwards (towards the back of the hall), producing a positive voltage at its output.  At the same instant, the diaphragm of the S mic will move from the left to the right, also producing a positive voltage at its output.  At the mixer, the M signal will be sent equally to both the left and right speakers - as will the output of the S mic, with the all important distinction that the signal sent from the S mic to the left speaker will be in phase with the M mic's signal, and therefore will combine CONstructively, making a louder output from the left speaker - AND - the signal from the S mic will combine with the M signal DEstructively on the mixer's output to the right speaker, resulting in a quieter signal output from the right speaker.  Put a human with good hearing in front of those two speakers, and they will experience the output of the two speakers as emanating from a single, virtual or "phantom" position, somewhere between the two speakers, but mostly towards the left.



roon wrote:

Some simple addition of ordinates of these amplitudes can isolate left and right and middle channels.  This is MS decoding.

No, MS decoding doesn't isolate the three channels (but only two mic signals) - it combines them to create two signals, one for the left speaker, one for the right.


roon wrote:

Centre M-S decoding requires a sum-and-difference matrix, where you add one side signal to the mid signal to get the sum, and subtract the other side signal from the mid signal to get the difference.

No.  In common use, the terms "sum" and "difference" refer to those signals coming out of the speakers.  The "Sum" refers to those parts of the signals that are the same intensity and polarity from both sides.  The "Difference" refers to those parts of the signals that are NOT the same - that's all those terms mean in this instance.


roon wrote:

To do this, one of the side channels is shifted 180 degrees in phase: when the polarity-shifted signal is added to the mid signal, you get the difference between the two signals. Using M for mid, S for side, and -S for the polarity-shifted side,

M + S = left channel

M + (-S) = right channel

Of course the mid is already isolated in a track of its own.

You've stated the math correctly, but I can't help but feel that you have misinterpreted what's actually going on - I hope my long winded explanation has helped with that.



roon wrote:

SO how does this translate to the micstasy?  Do adjacent channels become the 2 microphone channels.  MID mic on channel 1 and SIDE mic on channel 2.  SO if MS is selected on both channels and phase invert and MS decode is on in channel 2 i would get the right hand side? Which output does the result come from? 1 or 2?

As all this applies to the Micstacy, I don't really know, as I've never even seen one, BUT, I expect that it has some sort of either hardware or software mixer associated with it, that can route signals from its various inputs to various outputs.  If it's truly clever, it might even have an M/S decoder built right into the mixer, as described above.  But you don't really need a Micstacy to use M/S - the principles remain the same no matter what mixer you're using - even if you have to resort to the old fashioned idea of using a splitter wire for the S signal and three channels on the mixer.

I think the confusion arises from the fact that you have two signals coming into from the mics - the M and the S, and you have two signals going out to the speakers - the L and the R - but they're not the same signals at all.  M and S are not the same as L and R, but you can make L and R out of M and S, and viceversa, through some clever combining and inverting of signals along the way.  The benefit is that doing things this way allows you to, in effect, MOVE the microphones to create a wider or narrower stereo image, without having to leave the mix position to move microphones - you can just adjust things to your preference while listening to the result.

M/S is used in many other applications as well - FM radio is broadcast in M and S, and then turned into L and R in your radio receiver.  LP disks have the audio signal cut into the grooves so that the vertical motion of the stylus produces the M signal, and the side to side motion the S, which in turn get turned into L and R in your amplifier.  There are lots of reasons they decided to do things this way, but perhaps this post is getting a little long already...


roon wrote:

Perhaps i have even researched the wrong topic and this MS decoding is something completely different.

PS I am sorry for asking so many questions.  I do not want to come across as a LOUDMOUTH.  Indeed , the opposite is true.
with all due respects,
roon.

No problems - I hope I've cleared up some of the confusion for you, and for anyone else interested in this topic.

Frank Lockwood
https://LockwoodARS.com
Fireface 800, Firmware 2.77
Drivers: Win10, 3.125; Mac, 3.36

Re: octamic2 or micstacy

Frank
Thanks so much for making that big effort.
I was almost at the understanding but not quite.  After considering the energies of wave fronts incident upon the side mic i can see exactly that this is a sum of these vectors to a mono channel rather than stereo channel . As it was a long post, i read it many times to ensure understanding.  Thanks for taking the time.

UPon reading the manual again i have discovered that the micstasy encodes M/S using the odd pairs.  It has an encoder built in.  It is truly clever.

I will never cease to be amazed at all these techniques and technology.  Thanks again for imparting your wisdom.

Re: octamic2 or micstacy

i am proud new owner of RME Micstasy M and new ADI -8ds.

i feel like a small boy at Christmas time when i open up the box to see blue 19" facade.
Also coming today was some wonderful microphones and EMES black 5.1 surround system.

Thanks RME for such wonderful masterpieces.

Now if you would be so kind, please tell me :
1. how to change device number of micstacy and adi 648 so that i can put them in a madi loop with madi card.

2. WHat is RME anacronym for?

Best to all musicians (and engineers too)
roon.

Re: octamic2 or micstacy

From a few hundred kilometres north  ...

Just be aware that the Micstacy only applies M-S decoding on the digital outputs of odd/even pairs ... the analog outs are not affected.

M-S micing technique is most commonly applied in classical/acoustic music recording.  You can find condensor mics such as the AKG Blueline system or Schoeps that have cardioid (end-address) and fig-8 (side-address) capsules that you can use together; likewise some one piece stereo mics like the Studio Projects LSD2 will let you set one capsule in cardioid and the other in figure-8 and place them at right angles to each other to give the classic M-S configuration.  The small capsule mics work best because you can get the two capsules closer together, and therefore the image placement is much more accurate

There are some who use the M-S configuration in the studio to give an adjustable stereo image - if the gain of the S mic is lower than that for the M mic, the stereo image will be narrower.  After decoding it can then be panned to any point in the soundstage.  This is useful for things like pianos or marimbas which benefit from a bit of width.  It is hard to do this with the ubiquitous 3-fader trick which can be very confusing.

The big advantages of the M-S technique is that it guarantees you a good mono signal, and it gives you a better centre image than X-Y or spaced mics.  You will understand the features of M-S if you give it a try some time.

Have fun with the Micstacy ... I might end up with one some day!

De gustibus - et sonus - non est disputandum

Re: octamic2 or micstacy

thanks PT
please do not forget my inquiry in the post above
help me knut
ive got the contraption working in 48khz over adat... but not madi yet. A most enjoyable experience so far.

Re: octamic2 or micstacy

You could change the ID with the MIDI remote software.


Regards
Daniel Fuchs
RME

Regards
Daniel Fuchs
RME

Re: octamic2 or micstacy

i was expecting this unit to have the latest firmware.  What do i need to do to get the upgrade?

Re: octamic2 or micstacy

Admin Knut wrote:

The Micstasy stands in a class of it?s own. Many high-class recordings - some even Grammy nominated - were done with it in the last time. It just won the Editors Choice Award 2008 of the Professional Audio Magazin. Besides the high-end quality of preamps and AD conversion, the Micstasy offers a unique flexibility and usability.
Knut

Hi,

Regarding only the preamp and AD converters, are OctamicII and Mictasy in separate classes ?

Thanks for your information,

Didier

Re: octamic2 or micstacy

didier wrote:

Regarding only the preamp and AD converters, are OctamicII and Mictasy in separate classes?

Yes.


Regards
Daniel Fuchs
RME

Regards
Daniel Fuchs
RME

Re: octamic2 or micstacy

best preeeez.
best ADs
simple and complete functionality.
wunderful audioware
i want another one!

Re: octamic2 or micstacy

I think we got an interesting Micstasy story: Today we were visiting the well-known Principal studio complex in Senden Germany. All studios mix completely native in the box - with Micstasys as exclusive frontend ("best sound"). The remaining big SSL desk in the biggest control room is still used - as table and to impress visitors. 6-7 Micstasys should follow soon to replace the desks MicPres and some still used ADI-8 Pro AD converters.

The whole studio complex is using MADI connections with RME hardware. Imagine to do such a routing with analog or ADAT connetions - impossible or very expensive! Every studio includes an ADI-8 QS as identical and most accurate sounding monitoring solution. So it?s possible, to transport a project computer with the included RME MADI card to another studio and simply plug it in! The studio owners and engineers got a very consequent view: No fancy or legendary analog hardware ("no esoterism") to compensate weak mix or mastering skills. Capture the sound as it is and mix it! Only RME Micstasys and RME AD/DA converters plus In-the-box mixing with RME MADI cards. Today there were recording sessions of a new "S?hne Mannheim" project and final mix sessions of the upcoming album for the rising german band BLIND and their impressive new single for an upcoming Disney movie. The big studio complex in a former farm house ensemble is situated in green fields and woods and includes appartments and a little cinema/billard hall for the artists. Booked out in the moment but highly recommended! Very innovative and succesful music business thinking. Great atmosphere and great engineers!

Very funny: As we were introduced to working artists as RME people, the immediate response was always the same: "I?ve got a Fireface in my home studio - I?am very satisfied!" or "I want also a Fireface!" Kim was running out of her business cards in minutes.

A very impressive and interesting visit. The video documentation will follow soon.

best regards
Knut

Re: octamic2 or micstacy

KICKASS!! smile

Re: octamic2 or micstacy

RME Support wrote:
didier wrote:

Regarding only the preamp and AD converters, are OctamicII and Mictasy in separate classes?

Yes.


Regards
Daniel Fuchs
RME

Can you explain "how" the converters and preamps are different?

22

Re: octamic2 or micstacy

Isn't that explained in detail on the product pages of our website?

Regards
Matthias Carstens
RME

Re: octamic2 or micstacy

As far as I know the Micstasy uses the Texas Instruments PGA2500 digital controlled preamplifier, which is in the meantime quite widespread in pro audio industry. (I think it's also used in the Prism Sound Orpheus as well as in Apogee Duet and in the RME FireFace400, among others). Convertes in Micstasy are RME's latest generation, this is indeed explained on the product website.

Octamic II is more traditionally designed with analogue potentiometers and single-chip preamplifiers (anybody here to comment on the chip used?). Converters are not the same as in the micstasy. I think Daniel commented earlier on the converters somewhere here in the forums.

Regards,
Ulrich

Re: octamic2 or micstacy

The PGA2500 has 1 dB gain step. Micstasy has 0.5 dB gain step. So I wonder whether the Micstasy preamp are really based on the PGA 2500 like my Fireface 400. I would like more preamp inputs, at least a goods as the one from of the FF400. The OctamicII seems being the right solution for me. But how are the Octamic II preamps with respect to the FF400 ones ? Would I need the Micstasy for the same preamp quality like the FF400 ? 

Thanks in anticipation for making this issue clearer to me,

Didier

25

Re: octamic2 or micstacy

Of course the Micstasy has an extended analog input section which also handles the additional 0.5 dB steps. A nice feature to separate from other companies that use the PGA2500 as well ;-)

Regards
Matthias Carstens
RME

Re: octamic2 or micstacy

Ulrich wrote:

As far as I know the Micstasy uses the Texas Instruments PGA2500 digital controlled preamplifier, which is in the meantime quite widespread in pro audio industry. (I think it's also used in the Prism Sound Orpheus ...

I was informed by the head of Prism that the Orpheus used its own digitally gain controlled preamp solution (maybe a case of "He would say that, wouldn't he?").  Unfortunately for this discussion, when I had it on evaluation, I did not (professional etiquette) lift the lid to check.

However, as an owner of a FF400 and currently eyeing off a Micstacy, may I offer the opinion that digitally controlled pre-amps offer a greater operational convenience (reset-ability) than any finely-judged argument about quality could overcome (especially as implemented by RME).

Quality gets only half the marks - operational features are equally important, IMHO.

De gustibus - et sonus - non est disputandum