Topic: A device (or mixer) that monitors in decoded M-S but records discrete tracks

[SonoOtoSound]

Hey - I'm new to the forum and apologize if this has been asked before (i searched for it, unsuccessfully)

I do a lot of recording for film and radio and am looking to upgrade to a Sennhieser M-S setup. But I need a new recording device too (I have a Sony D-50) and really want one that allows me to MONITOR in stereo (in other words, the M-S signals decoded) but RECORD the mid and side as discrete tracks (in other words, the M-S signals un-decoded). I know the Sound Devices 702, 722, etc... all do that, but I'm hoping to find a cheaper solution that still sounds good and is reliable.

So far the only thing that seems reasonable is trying to find a Sound Devices MP-2 (this came before the SD MixPre, which came before the SD Mix-Pre-D). The MP-2 seems to be able to do this and then I could run that into my Sony D-50. The problem is they're old and I can't find any used ones anywhere. Does anyone know where I can find one? (or does anyone have a 702 or 722 they're looking to get rid of??).

If I can't find a used MP-2 or 702, can anyone recommend a good recording device that has this feature???

Thank you so much for your help!
Mark

[page]

Hey - I'm new to the forum and apologize if this has been asked before (i searched for it, unsuccessfully)

I do a lot of recording for film and radio and am looking to upgrade to a Sennhieser M-S setup. But I need a new recording device too (I have a Sony D-50) and really want one that allows me to MONITOR in stereo (in other words, the M-S signals decoded) but RECORD the mid and side as discrete tracks (in other words, the M-S signals un-decoded). I know the Sound Devices 702, 722, etc... all do that, but I'm hoping to find a cheaper solution that still sounds good and is reliable.

So far the only thing that seems reasonable is trying to find a Sound Devices MP-2 (this came before the SD MixPre, which came before the SD Mix-Pre-D). The MP-2 seems to be able to do this and then I could run that into my Sony D-50. The problem is they're old and I can't find any used ones anywhere. Does anyone know where I can find one? (or does anyone have a 702 or 722 they're looking to get rid of??).

If I can't find a used MP-2 or 702, can anyone recommend a good recording device that has this feature???

Thank you so much for your help!
Mark

I was actually going to suggest the mp2. doh.

If you need it on short notice and don't mind the price, there is someone selling a 722 in the yard sale right now (or at least they were a day or two ago).

[Fried Chicken Boy]

I'm not very familiar with the deck, but maybe someone who knows the Tascam DR-680 can chime in: does that recorder have flexible monitoring options?

[SonoOtoSound]

Hey thanks for your responses - the Tascam DR-680 *does* offer the flexible monitoring option, but I've been warned about it's cheapness. I'm going to be use this in the field for film and radio work as well, and need something that will hold up.

My latest thought is doing something like this: M/S mics -> Sound Devices MixPre-D -> Sony D-50 -> *M/S decoder for headphones* -> headphones. That could be a simple solution if a M/S decoder for headphones exists. I've found this:

http://www.wesdooley.com/aea/ds_MS38_Lite.html

which seems *perfect*, but it's been discontinued. I'm not handy enough to build one myself. Anyone know of something like this AEA MS38??? That would be perfect.

Thanks again for your thoughts!
Mark

[jbell]

I think you can get a V3 with M/S option added on! 

[udovdh]

Which mics do you intend to use with the recorder you are looking for?
(might help the audience)

[page]

I think you can get a V3 with M/S option added on!

but it won't send discrete tracks and allow for monitoring decoded tracks simultaniously.

Hey thanks for your responses - the Tascam DR-680 *does* offer the flexible monitoring option, but I've been warned about it's cheapness. I'm going to be use this in the field for film and radio work as well, and need something that will hold up.

What we think of as "cheap" is sometimes different than what someone who's employment depends on the recording thinks of as "cheap" so I'll go out on a limb and say if this is a professional application and needs to be robust due to envioronmental factors, the only other thing I could think that might fit would be the Edirol R-44. It can do MS Decoding with seperate settings for recording or playback. So you could say "no, I want to record discrete tracks, but playback decoded tracks" but I don't know if their term "playback" encompasses monitoring. If it does, that's great, but I'd want to play with it a lot before I baught one for that reason. In the Team Board section of this forum there should be a thread for Edirol that you might find someone nearby to you who has a unit you could test with.

edit: http://taperssection.com/index.php?topic=149224.0 I feel mildly dumb now, that took all of 10 seconds to find the answer to. (I own a 722 and not the R44 so I didn't know) Scratch that plan.

After that, the most robust products (e.g. built like a tank and intended for professional situations) I can think of that are remotely economical and do the desired function are used machines from Sound Devices. Stuff that can compete with that like Sonosax or Nagra just isn't available at a competitive price.

[kirk97132]

The 680 isn't any "cheaper built" than the Sony D50 and I'd say it is as well built as the r-44 having owned an R-44.  I have a 680 that has over a year on it with no issues.    It's not tank like ala the 7xx series but then again not much else is.

[DigiGal]

Sound Devices 302 Mixer is designed for film/video applications and will do what you're looking for, however, they're not cheap.

[SonoOtoSound]

Hey everyone - thanks for all your responses. udovdh - to answer your question: I'm going to get a Sennheiser MKH 30 and 50. DigiGal - thanks for pointing out the 302 mixer.. I didn't realize that was an option. However, at that price ($1300) I think I would just rather get a 702 instead and have a more compact setup with only one set of batteries, etc. But it's good to know the 302 can do this too. So assuming there isn't a better solution - what do you guys think would be a better setup (and by better I mean sound quality, ease of use, reliable, etc... - obviously this is all super subjective but again, I'll be using this in the field a lot to record ambiences, nature sounds, sound effects, radio interviews, as well as musical performances):

a SD mp-2 -> Sony d50    OR      the Tascam 680? (This is assuming I can even find an MP-2)

Again, thanks for all your thoughts - I'm really grateful for everyone's generosity and helping me out!
-Mark

[jbell]

I use to run a MP-2>D50 and it is a small and durable setup(smaller than the DR680).  The MP-2 is bulletproof and I think the D50 is the most well built hand held currently out there.  I haven't heard of any reports of people having issues with their D50's, but even though people have had solid experiences with the 680 there have been reports of problems.  That being said I have no experience with the DR680.  You missed my MP-2 by a few weeks I sold it very recently.  If you keep your ISO in the YS an MP-2 will surface and since you already have a D50 I would hold out for an MP-2 for a bit. 

[hi and lo]

The 680 isn't any "cheaper built" than the Sony D50 and I'd say it is as well built as the r-44 having owned an R-44.  I have a 680 that has over a year on it with no issues.    It's not tank like ala the 7xx series but then again not much else is.

Couldn't disagree more. I am happy our 680 is finally working, but would never recommend it to a non-hobbiest.

[kirk97132]

hi and Lo I know you've had issues but to counter that I have two 680's and have had no issues at all.  I sold my R-44 for the first 680 and the build quality meaning what they are built with and their functionality is equivalent.  I love my 680 and would buy another in a heartbeat.  As for durability I have rode the first 680 pretty hard.  used it a ton and have not had a lick of any kind of issue.  at a $1000 price point I think it's the best bang for the buck.  I run an Mp-2 a USBPre2 and V3 with it. Yet when run alone it delivers a great sound. 

As for Mp-2 used....an ISO here and troll E-bay they show up every so often. 

[Nick's Picks]

question...
why is it important to "monitor" a L/R signal if you are recording discrete M-S tracks ?  I dont understand the point.  You still have to mix the two tracks in post, and listen / adjust while doing it.
wouldn't it be prudent to do that in a nice quiet controlled setting vs. a concert where the db is so loud you can't hear shit any way ?

If you're trying to nail it on the fly, then just let the deck/preamp do the mixing and record the L/R channels.  Monitor that in real time.

[runonce]

question...
why is it important to "monitor" a L/R signal if you are recording discrete M-S tracks ?  I dont understand the point.  You still have to mix the two tracks in post, and listen / adjust while doing it.
wouldn't it be prudent to do that in a nice quiet controlled setting vs. a concert where the db is so loud you can't hear shit any way ?

If you're trying to nail it on the fly, then just let the deck/preamp do the mixing and record the L/R channels.  Monitor that in real time.

I would think one reason might be - that it would be slightly annoying to have a raw M/S channels fed into headphones...a figure 8 in one ear and a card in the other...not a great setup if you are doing location stuff...plus I could see it helping you keep "spot on" in you are in motion or have to track a moving sound source.

[Ozpeter]

I do a lot of recording for film and radio and am looking to upgrade to a Sennhieser M-S setup. But I need a new recording device too (I have a Sony D-50) and really want one that allows me to MONITOR in stereo (in other words, the M-S signals decoded) but RECORD the mid and side as discrete tracks (in other words, the M-S signals un-decoded).

I'm struggling to think of a good reason for recording an MS signal as MS.  I can think of at least one reason why it's unwise but no reasons why it should be done.

[runonce]

I do a lot of recording for film and radio and am looking to upgrade to a Sennhieser M-S setup. But I need a new recording device too (I have a Sony D-50) and really want one that allows me to MONITOR in stereo (in other words, the M-S signals decoded) but RECORD the mid and side as discrete tracks (in other words, the M-S signals un-decoded).

I'm struggling to think of a good reason for recording an MS signal as MS.  I can think of at least one reason why it's unwise but no reasons why it should be done.

Thats not what he's doing...

[SonoOtoSound]

Yeah, my reason for wanting to monitor in stereo is basically what runonce said. In addition, I'll be using a Sennheiser MKH 50 for the mid, which is a little narrow and reportedly can sometimes create some weird, blank spots in the stereo image for some sounds. It's pretty easy to fix, just by getting the right distance from your source but you can only do that if you're listening to a decoded stereo mix.

The reason I'm recording it as two discrete tracks (the mid and the side) is so that I have total control *afterwards* of how wide or narrow I want the sound to be. If you record in stereo you're stuck with whatever amount of width you've chosen and I don't want to chose that in the field, because sometimes there is no *one* correct amount. For film work, sometimes you want the exact same sound in two or three different perspectives to match different shots in the film.

[Ozpeter]

The reason I'm recording it as two discrete tracks (the mid and the side) is so that I have total control *afterwards* of how wide or narrow I want the sound to be. If you record in stereo you're stuck with whatever amount of width you've chosen and I don't want to chose that in the field, because sometimes there is no *one* correct amount.

I think you're not clear on the relationship between MS and XY.  They are freely interchangeable in either direction.

If you record discrete MS, in postproduction in a DAW you normally use a VST 'effect' to handle the conversion from MS to XY, and that effect will indeed have controls to vary the proportion between mid and side to allow you to choose the effective width.  But even a freeware MS VST effect (eg Voxengo MSED) will have modes to convert XY back to the original MS, and XY > MS > XY ("inline" mode) allowing the width of an XY recording (whether originated from an XY pair or an MS pair) to be varied in post production.  MS and XY recordings can be width-varied in exactly the same way - you don't have to have the original M and S signals to do it.

Recording an MS pair as MS gives rise to the monitoring problems you are encountering and in some contexts carries the risk of someone not realising the format of the recording and using it 'as is' with very undesirable results.  Converting from MS to XY at the time of recording is, in any context I can think of, a better practice and it makes no difference to the ability of varying width in post production.  That's why I've been recording that way using my Sennheiser pair for about 25 years and I have never once wished I'd recorded discrete MS.

[SonoOtoSound]

If you record discrete MS, in postproduction in a DAW you normally use a VST 'effect' to handle the conversion from MS to XY, and that effect will indeed have controls to vary the proportion between mid and side to allow you to choose the effective width.  But even a freeware MS VST effect (eg Voxengo MSED) will have modes to convert XY back to the original MS, and XY > MS > XY ("inline" mode) allowing the width of an XY recording (whether originated from an XY pair or an MS pair) to be varied in post production.  MS and XY recordings can be width-varied in exactly the same way - you don't have to have the original M and S signals to do it.

I figured there was a way to convert XY > MS (since doing a mono sum of the XY would give you the Mid - I assumed there would be a pretty easy way to then get a discrete Side signal too) but I didn't realize that a plug-in could allow you to go XY > MS > adjust the level of side and then > XY. Can it do all of that in one step??? Or do you have to use the plug-in to convert the XY to MS and then run the plug-in again on the new MS version of the audio to bring it back to XY?? From your description it sounds like it can do all of that in one step, which would be really impressive. Thanks for pointing that out.

Still though, I'm leaning towards recording it to two discrete tracks because I'll be using this setup primarily to record FX for film and I think that very frequently I'll just want to use the Mid signal (maybe 60% of the time). So having a separate M and S right off would be advantageous to me. I'm a total newbie to M-S, so your 25 years of experience is nothing for me to discard. Do you record all kinds of stuff this way, including FX? I'm very curious what you think about recording discrete MS if it would be primarily for FX (hence, having a mono Mid file right off the bat would be advantageous)

Thanks for your help - I really appreciate hearing your opinion!
Mark

[Nick's Picks]

The reason I'm recording it as two discrete tracks (the mid and the side) is so that I have total control *afterwards* of how wide or narrow I want the sound to be. If you record in stereo you're stuck with whatever amount of width you've chosen and I don't want to chose that in the field, because sometimes there is no *one* correct amount. For film work, sometimes you want the exact same sound in two or three different perspectives to match different shots in the film.

but, thats my point exactly.  do it in post.  no need to monitor it at all to hear what it might sound like in realtime (decoded signal to your headphones, while reccording raw m-s).  You're still going to mix it in post after the fact, so why worry about the "real time sound" ...when the finished product is going to be different any way.  Trust me, having done this many, many times..., what you think sounds great on the fly is never as good as what you mix in post in a controlled environment.   Plus, if you record a mixed M-S signal as stereo L/R, you can still bring it back to raw-M-S and remix it.  There is no "one chance" with Mid-Side.  No matter how the stream is recorded.

[Javier Cinakowski]

I gotta imagine there is some degradation from going back and forth between M/S and L/R and vice versa.   Wouldn't the audio editing software or plugin have to "recreate" the .wav during each process.  It isn't like we are doing this in the analog realm, wouldn't there be some degradation or digital artifacts every time this is done?

I just don't see how this could be a free lunch.  It isn't like you can EQ (destructive editing) a recording and then EQ it back without degradation of the product...

Am I missing something?

[page]

The reason I'm recording it as two discrete tracks (the mid and the side) is so that I have total control *afterwards* of how wide or narrow I want the sound to be. If you record in stereo you're stuck with whatever amount of width you've chosen and I don't want to chose that in the field, because sometimes there is no *one* correct amount. For film work, sometimes you want the exact same sound in two or three different perspectives to match different shots in the film.

but, thats my point exactly.  do it in post.  no need to monitor it at all to hear what it might sound like in realtime (decoded signal to your headphones, while reccording raw m-s).  You're still going to mix it in post after the fact, so why worry about the "real time sound" ...when the finished product is going to be different any way.  Trust me, having done this many, many times..., what you think sounds great on the fly is never as good as what you mix in post in a controlled environment.   Plus, if you record a mixed M-S signal as stereo L/R, you can still bring it back to raw-M-S and remix it.  There is no "one chance" with Mid-Side.  No matter how the stream is recorded.

I think his desire stems from making sure he's setup correctly, it's the same argument I see for monitoring anything in the field; you want to make sure you're not getting absolute trash.

[Nick's Picks]

its field recording...there is only so much that you can do in "real time".  If the sound sucks, you'll polish your turd much better sitting at your desk.

[Nick's Picks]

I gotta imagine there is some degradation from going back and forth between M/S and L/R and vice versa.   Wouldn't the audio editing software or plugin have to "recreate" the .wav during each process.  It isn't like we are doing this in the analog realm, wouldn't there be some degradation or digital artifacts every time this is done?

I just don't see how this could be a free lunch.  It isn't like you can EQ (destructive editing) a recording and then EQ it back without degradation of the product...

Am I missing something?

I hear ya, but I guess it comes down to math at its core.  And that is absolute, so no loss.

[Gutbucket]

Plugins make it easier in the digital realm and do it better without potential artifacts (no noise, no gain balancing issues) than the analog way of patching channels and inverting polarity through a mixer.  Plus they are far less of a PITA!

Yes, some plugins have two matrices so you can input X/Y, adjust M/S ratio and output X/Y again all in one plugin.  I think Voxengo's MSser or whatever it's called does that, if not other free ones do.  Some DAWs have that functinality built into advanced pan function on stereo tracks- Samplitude does.

There are a few good reasons for recording the raw M/S. Quick access to the mono mid without decoding is probably the main one.  I did so last week for that very reason. Ability to vary the stereo spread using the playback M/S decoder built into the recorder without offloading the files into a DAW is another. 

[noahbickart]

get in touch with mshilarious. He now has a m-s decode board option for the littlebox. he is making me one with 2 sets of rca pairs, one undecoded (e.g. m-s) and the other decoded (e.g. x/y). I've got outboard headphone amps for monitoring, but he can also build a headphone amp into the unit with a switch for the monitor feed. See naiant.com

And Nick, I think the point is that he isn't a OTS concert tapir like you. He is doing pro work with the pair on a boom. He needs to monitor in real time to know where to place the microphones- something over which he does have full control.
-Noah

[George2]

I have the original MS38 from AEA. Line Level. No headphone out. Just one knob. You would have to connect to your line out from recorder and then connect MS38 to headphone amp.

[Gutbucket]

Note the matrixed to mono mid cat paw. 

[George2]

Cats come with.. all 3 of them.. no charge!

[runonce]

The reason I'm recording it as two discrete tracks (the mid and the side) is so that I have total control *afterwards* of how wide or narrow I want the sound to be. If you record in stereo you're stuck with whatever amount of width you've chosen and I don't want to chose that in the field, because sometimes there is no *one* correct amount. For film work, sometimes you want the exact same sound in two or three different perspectives to match different shots in the film.

but, thats my point exactly.  do it in post.  no need to monitor it at all to hear what it might sound like in realtime (decoded signal to your headphones, while reccording raw m-s).  You're still going to mix it in post after the fact, so why worry about the "real time sound" ...when the finished product is going to be different any way.  Trust me, having done this many, many times..., what you think sounds great on the fly is never as good as what you mix in post in a controlled environment.   Plus, if you record a mixed M-S signal as stereo L/R, you can still bring it back to raw-M-S and remix it.  There is no "one chance" with Mid-Side.  No matter how the stream is recorded.

Are you reading the thread? He IS going to do it in post...

He's gathering foley - and has specific need to monitor his raw M/S decoded...

,,,he's not recording bands...

[Javier Cinakowski]

NicksPicks Posted:

I hear ya, but I guess it comes down to math at its core.  And that is absolute, so no loss.

Yeah, I understand the M/S math in regards to sum and difference.  However, my concern is that the audio editing software has to actually make a new wave file every time the encode/decode is done.  Hence an additional digital conversion process.  Sure, it is all digital, but I don't see how it can make new channels without making a new representation of the bits.  Sure, it can be mathematical reversed, but I suspect there is loss, akin to an additional DAC/ADC stage.   I guess proof would be if you can run a data checksum program on a M/S that has been encoded/decoded and it would still match.   I suspect it wouldn't...

[page]

NicksPicks Posted:

I hear ya, but I guess it comes down to math at its core.  And that is absolute, so no loss.

Yeah, I understand the M/S math in regards to sum and difference.  However, my concern is that the audio editing software has to actually make a new wave file every time the encode/decode is done.  Hence an additional digital conversion process.  Sure, it is all digital, but I don't see how it can make new channels without making a new representation of the bits.  Sure, it can be mathematical reversed, but I suspect there is loss, akin to an additional DAC/ADC stage.   I guess proof would be if you can run a data checksum program on a M/S that has been encoded/decoded and it would still match.   I suspect it wouldn't...

Na, it wouldn't, but how much different is up for debate. If nothing else, sample conversion noise would be introduced and dither noise, but besides that I don't know how well the sum/diff algorithm is and how much noise it would introduce. It would be an interesting test to do it like 20 times and compare it (assuming you could get the same mix at the beginning and end).

[Gutbucket]

[snip..]my concern is that the audio editing software has to actually make a new wave file every time the encode/decode is done.  Hence an additional digital conversion process.  Sure, it is all digital, but I don't see how it can make new channels without making a new representation of the bits.  Sure, it can be mathematical reversed, but I suspect there is loss, akin to an additional DAC/ADC stage.   I guess proof would be if you can run a data checksum program on a M/S that has been encoded/decoded and it would still match.   I suspect it wouldn't...

No your misunderstanding what digital processing is happening.  There is no 'additional digital conversion', no sample rate conversion, no loss.  If you wrote a new file of the output of a plugin that did an X/Y to M/S to X/Y conversion (and didn't adjust the ratio in the middle) the output should match the input exactly.  Check it if you want to.  If it doesn't then something is wrong.  If you change the ratio in the middle, it changes the data and they won't match, but that's the whole point of making an adjustment.

The increased processing bit depth of the program you use to host the plugin, above that of your audio file, accommodates any rounding errors in the sum and difference processing.  It's simple addition and subtraction.  Any modern DAW can do it perfectly and far more acurately than an analog mixer.

But go ahead and try it to convince yourself.

[page]

[snip..]my concern is that the audio editing software has to actually make a new wave file every time the encode/decode is done.  Hence an additional digital conversion process.  Sure, it is all digital, but I don't see how it can make new channels without making a new representation of the bits.  Sure, it can be mathematical reversed, but I suspect there is loss, akin to an additional DAC/ADC stage.   I guess proof would be if you can run a data checksum program on a M/S that has been encoded/decoded and it would still match.   I suspect it wouldn't...

No your misunderstanding what digital processing is happening.  There is no 'additional digital conversion', no sample rate conversion, no loss.  If you wrote a new file of the output of a plugin that did an X/Y to M/S to X/Y conversion (and didn't adjust the ratio in the middle) the output should match the input exactly.

I'm mildly skeptical of that, just because resampling from 48 to 44.1 and then back to 48 yields differences well under 18khz depending on your SRC algo. Someone did a bunch of tests and you could see where noise was introduced (again, depending on algorithm). I havn't tried, I understand the principle and you're right, it's just math, but something seems real fishy about that once you step out of the theoretical and into reality, maybe it's just cause I've read enough academic theory papers to see that there is usually a hole or flaw, maybe I'm just cynical.

[Ozpeter]

I'm assuming that the recording is bound to go through some kind of DAW processing before actually being used, so most of the considerations about additional manipulation of the file become pretty moot.  It's going to be manipulated anyway.

If you did want mono only, having recorded the file as XY stereo, you'd simply turn the side level control of the VST effect to zero.  These days, popping a VST effect onto a track is so trivial that I'd regard that as neither here nor there, in proportion to the whole effort of post production.

I would continue to assert that simply recording the MS mics converted to XY at the outset solves so much in the way of monitoring complexities at the outset, with no comparable downside, that it's the obvious solution to the initially stated problem.  I would strongly suggest exploring and evaluating the free, low-complexity route on a couple of experimental recordings first and then decide whether it is really necessary to throw money at the problem.

Link to Voxengo MSED (yes, it has inline processing) - http://www.voxengo.com/product/msed/

[Erick del Valle]

can anyone recommend a good recording device that has this feature???

MArk, try this for your heaphones output!

http://www.canford.co.uk/Browse/21263

Saludos
Erick del Valle
Santiago de Chile

[Gutbucket]

[snip..]my concern is that the audio editing software has to actually make a new wave file every time the encode/decode is done.  Hence an additional digital conversion process.  Sure, it is all digital, but I don't see how it can make new channels without making a new representation of the bits.  Sure, it can be mathematical reversed, but I suspect there is loss, akin to an additional DAC/ADC stage.   I guess proof would be if you can run a data checksum program on a M/S that has been encoded/decoded and it would still match.   I suspect it wouldn't...

No your misunderstanding what digital processing is happening.  There is no 'additional digital conversion', no sample rate conversion, no loss.  If you wrote a new file of the output of a plugin that did an X/Y to M/S to X/Y conversion (and didn't adjust the ratio in the middle) the output should match the input exactly.

I'm mildly skeptical of that, just because resampling from 48 to 44.1 and then back to 48 yields differences well under 18khz depending on your SRC algo. Someone did a bunch of tests and you could see where noise was introduced (again, depending on algorithm). I havn't tried, I understand the principle and you're right, it's just math, but something seems real fishy about that once you step out of the theoretical and into reality, maybe it's just cause I've read enough academic theory papers to see that there is usually a hole or flaw, maybe I'm just cynical.

Maybe I missed something in this conversation. Where did the talk of sample rate converson come from? Why does it have anything to do with M/S?

[page]

Maybe I missed something in this conversation. Where did the talk of sample rate converson come from? Why does it have anything to do with M/S?

Exemplary mostly. I'm of the belief that all mixing/processing introduces noise, but how much, where, and whether you can hear it is the real question.

[Gutbucket]

We'll rest assured that sum/difference processing is about as simple as it gets in digital audio processing.   Computers rock at 1+2=3 and 2-1=1.  If you were to run a checksum on a file before and after processing with Voxengo MSED without adjusting ratio, it would match.  Try it or ask in the Voxengo forum about it if you have lingering doubts, the developer there has been very good at answering a few questions i've posted there in years past.

Sample rate converion is an entirely different beast, it is inherently more complicated than the simple addition, subtraction and sign change (polarity) of M/S processing.  SRC throws out information by definition when changing the rate downward, though its done so well now that I wouldn't worry much about that either.  There are plenty of things that make a far more signigicant difference to be concerned about. 

Priorities, overlooking forests for trees and all that..

[SonoOtoSound]

I would continue to assert that simply recording the MS mics converted to XY at the outset solves so much in the way of monitoring complexities at the outset, with no comparable downside, that it's the obvious solution to the initially stated problem.  I would strongly suggest exploring and evaluating the free, low-complexity route on a couple of experimental recordings first and then decide whether it is really necessary to throw money at the problem.

Hey Ozpeter, you make a really compelling argument. The only reason I'm still not 100% on board with your recommendation is the desire to be able to use just a mono file. Let's say I'm working on a film and I frequently need to pull in a... good car door closing. It would just be so much easier to drag in a mono Mid file to a mono track, than to pull in the X-Y into a stereo track and then use the plug-in to alter it. Panning a stereo FX can be a total bitch - even if the plug-in was making it mono, having to pan a stereo fader is usually much more difficult than panning a mono one - especially if the panning is automated and moving around within a scene. Then the 20% of the time I want to use a stereo version of the car door FX (let's say because the picture has the car off in the distance) I drag in the Mid and the Side to a stereo track and then put a plug-in on that.

I know, I know. You're not convinced and think I'm going about it this the wrong way. But I just found a 302 for a great price and I think I'm going to get that (which does allow to me to monitor in stereo while recording discrete M/S) for the same price of a MixPre-D, which I'd have to get anyway. So I figure it's worth a try. And if my way of doing it proves difficult, I can always just do as you suggest with the 302 too. Thanks for all your thoughts though

[Javier Cinakowski]

Hmm, interesting and educational responses, thanks everyone.  I am going to try the checksum using voxengo without any ratio adjustment.  I am encouraged by the fact that I may be able to record stereo on the fly and rematrix later, without loss.... 

[Gutbucket]

Cool, let us know what you find.

Something I feel does matter significantly with this is the actual physical starting point, the particulars of the M/S or X/Y array you use, which mics, which patterns, which angles, and how coincident the setup actually is.  The useful range of adjustment is in most cases rather limited, so starting with a M/S setup that is as close as possible to what you want with a straight 50/50 decode, or an X/Y setup that is as close to what you want before possibly tweaking things somewhat with an X/Y>M/S>X/Y plugin is one of the keys to goodness.

Consider that what most would think of as 'standard' M/S using a cardioid mid is not equivalent to X/Y cardioids and in no possible M/S ratio does it decode to X/Y cardioids.  All standard two mic M/S setups using a cardioid or supercardioid mid decode to virtual X/Y mic patterns that have some degree of rear lobe.. but to my thinking that's a feature not a bug. 

I'd think SonoOtoSound would probably want a supercardioid mid for his FX work, to help isolate the sound of interest.  A shotgun mid is often used for film dialog work, but that won't sound as natural for non-speech FX sounds.

[noahbickart]

There is one small, perhaps only theoretical, benefit to recording m and s, and not and encoded signal.

In most cases, because the sound source in on axis to the M signal, levels are higher on the M than the S. However, if you record the M and S separately, you have the option of recording both as close to 0db as possible.

Yes, in post you'll end up bringing down the level of S, but in the process you have maximized your signal to noise ratio, because as you bring down the signal of S you also bring down the noise on that signal.

You give up this sonic benefit when you encode before recording.

[Ozpeter]

I know, I know. You're not convinced and think I'm going about it this the wrong way.

Oh, I'm open to a persuasive argument related to your particular needs, and I can see where you are coming from - I guess it depends (amongst many things) on the software are using, and how easy that makes the extraction of the side signal.

At the end of the day, you knowing what your particular needs are, having reviewed the thoughts expressed in this interesting thread, are the only one who can decide what's best for you!  As so often in audio, there is no "always right" solution.

[RichT]

Lots of DIY MS decoding info here, including a couple of (really simple) Neumann circuits.  Might have a go at the Neumann Wheatstone Bridge and see if it'll fit in a jack

http://www.uneeda-audio.com/#ms

[Nick's Picks]

also, if it hasn't been mentioned, schoeps has a great vst plugin for their double MS rig.  works great for MS too

[Tommy-boy]

I have my rig set up to record in M-S, but to monitor the decoded signal.  I accomplish this with the following equipment:

R-44 
Art S8   http://www.artproaudio.com/products.asp?type=88&cat=11&id=137
Behringer Eurorack Pro RX1602    http://www.behringer.com/EN/Products/RX1602.aspx
Phase Inverters (I think they're whirlwinds) - hard to see behind rack at moment.
Cables that go from RCA to XLR (I think I have inexpensive Hosa cables for this).  This it to make connection from R-44 to S8.
Cables that go from XLR to TRS (Hosa).  This to make connection from Phase Inverters on S8 to Eurorack.
Cables that go from RCA to TRS (Hosa cables).  This is to make connection from R-44 to Eurorack.

I basically run the analog outputs of the R-44 into the Art S8 and Behringer Eurorack.  The only purpose of the Behringer Eurorack is to monitor.

Assumptions - Input 1 on R-44 is Mid (M) mic; Input 2 on R-44 is Side (S) mic.

Chain for Mic 1:  Into R-44 (Channel 1), out of R-44 analog out (Channel 1) to Eurorack Channel 1 (panned center)

Chain for Mic 2: Into R-44 (Channel 2), out of R-44 analog out (Channel 2) to Art S8 where channel is split.  We'll call these channels 2A and 2B.  2A goes to Eurorack Channel 3 (panned left).  Signal 2B hits phase inverter and then goes to Eurorack Channel 5 (panned right).  The reason for going to channels 3 and 5 with the side (S) signal is that the Behringer inputs are ganged in pairs of 2.  Using Channel 1 of the 1-2 pair causes 1 to be mono – which I can then pan center.  Using Channel 3 of the 3-4 pair causes 3 to be mono – which I can then pan left.  Using Channel 5 of the 5-6 pair causes 5 to be mono – which I can then pan right.

Voila - instant monitoring of decoded signal.  R-44, in the meantime, is recording a pure unadulterated M-S on channels 1 and 2.

Granted, The Eurorack is a little noisy.  However - It's not in the recorded signal path and helps me monitor what's going on.  I do the decoding of the M-S channel later in my Daw.  Just copy track 2 from R-44 (pan original left and copy to right).  I typically send the 2 side signals (L and R) to a group channel in Cubase so that I can raise the volume of the sides with one fader.  Not hard stuff.  By ear, I adjust the relative volumes (M versus S) to get the stereo spread to sound right.

By the way, you cannot accomplish this without a splitter that has a transformer isolated output.  A “Y” splitter won’t cut it (I learned this the hard way instead of thinking it through first).

One other tip for M-S recording: when I’m setting up, I have somebody walk from left to right in front of mics so I can immediately determine if I have the darned thing setup correctly.  I can do this as I mainly record school bands (always some time before concerts start to do this).

Hope this helps.
Tom

[Ozpeter]

I mainly record school bands (always some time before concerts start to do this).

Well, I have to say for that purpose it's really hard to justify recording separate MS signals, imho - I just don't see the point of the complexity of the rig you describe compared to the simplicity of recording your MS mics straight to XY in the R44 - nothing else required.  Vary width if required during post production using the method I've described.  I should add that most DAWs should be able to vary the width dynamically during the performance to taste.  For instance, say it's a fairly reverberant venue, and the conductor makes some introductions from the podium.  You might reduce the width close to mono for clarity's sake during the announcement.  Just as easy to do with an MS VST applied to an XY recording as with an MS recording - mathematically the same.