microphones to record classical guitar at home

Submitted by Haris on Mon, 12/15/2014 - 02:41

Now I use Zoom H4n with the in-built microphones to record classical guitar at home. I use it as audio interface to record to Audacity, or record to SD and then import it to PC. Here is a recording example:

https://www.youtube…

I would like to upgrade my system and need some advise: I think it would make a difference if I added a RODE NT-4. What do you think?
Also, is it better to buy a RODE NT-4 or one better mono microphone like Neumann KM-184-MT?
Maybe some other idea?

Thank you

Comments

Haris, post: 422687, member: 48680 wrote: I ordered [[url=http://[/URL]="http://www.oktava-s…"]this[/]="http://www.oktava-s…"]this[/]. I hope I made a good choice. If you have an objection please tell (there is still time to cancel).

Thank you a lot for helping me!

Well done! I think you will hear quite a difference between the built-in mics of the Zoom H4N and the Octavas plugged into the Zoom's XLR inputs. The disadvantage is that it may want to make you to move on to the next level of pre-amps to use them to their full potential.

Here is a 1st recording of Dionisio Aguado - Leccion No 19 I made with the Oktavias MK-012. I put the mics in xy about 60cm in front of the sound hole and recorded with maximum loudness at -12. My mics have interchangeable capsules (cardioid, hypercardioid, omni). I made this recording with cardioid. I used Zoom as preamp.
What is your opinion? Can you compare it to the Zoom recording? Can you listen any problem from the room? Any advice would be appreciated!

http://www.guitarraclasicadelcamp.com/viewtopic.php?f=26&t=21988&p=164370#p164328

Thank you!

PS If it is not appropriate to use a link of another forum please tell me.

No tiene los permisos requeridos para ver los archivos adjuntos a este mensaje.

What ever that says, I don't find any link to your recording.
We often use soundcloud.com to share music. if you want to try it.

More than often, close mics to the sound hole give boomy recordings. (unless the guitar is laking bass)
The common way is to place your x/y aray in front of the 12'th fret.
Move around to different spots in front of the mics stop when it sound good to you. In the end how it sounds is all that counts ;)

It sounds nice - I think you could get a bit more "silk" and "air" out of it, though... the upper range where you could get a bit more of the texture of the fingers on the frets and finger picking.

Very nice performance, by the way. ;)

Don't get me wrong - it doesn't sound bad at all! I'd just like to hear a bit more silk, a little more presence... off the top of my head, I'd say around 4-6k... if you added a touch, say 1-2db in that range, it might "open it up" a bit , give it some silky presence.

Can you describe:

1. how you mic'd the guitar (mics used, preamp, distance to instrument, and positions)
2. what your EQ settings are on the track(s)
3. if you used any gain reduction

-donny

Thank you Donny!
I made with the Oktavias MK-012. I put the mics in xy about 60cm in front of the sound hole and recorded with maximum loudness at -12. My mics have interchangeable capsules (cardioid, hypercardioid, omni). I made this recording with cardioid. I used [[url=http://[/URL]="http://www.zoom.co…"]Zoom[/]="http://www.zoom.co…"]Zoom[/] H4n as preamp. No EQ, no gain reduction, no sound edit at all. I would prefer to have the best sound only with good mic placement. Do I have to add EQ settings?

The sound of the recording is more "empty" then the sound of my guitar. Why? Because of the mic placement? Because of the mics? Because of my PC soundcard/loudpeakers? Because of the preamp?

Haris, post: 425055, member: 48680 wrote: Do I have to add EQ settings?

Of course not. :)

You can try and get the sound you seek by simply adjusting the mic positions - and in fact, this is the way I was taught. This method is now considered to be "old School" engineering, and has fallen into rare use...and that's disappointing, because it's a very important step in recording with mics.

But... there's nothing wrong with using EQ, either... to either attenuate frequencies you don't like, or, to enhance those that you do. Now... if you find yourself making huge tonal changes to get the instrument sounding the way you want, then you probably need to go back to square one and find out why you require this heavy correction.

"The sound of the recording is more "empty" then the sound of my guitar..."

You have to remember that the room you are in will play a big part in the sound that you are recording. Condensers are sensitive, and will pick up much more extraneous noise - even in cardioid - than dynamics will. I am NOT saying that you should use dynamics for this. Acoustic instruments benefit from condenser mics, because condensers are traditionally more sensitive, and have the ability to pick up those beautiful nuances of the instrument, so in your case, with your instrument and your style, you want a mic that has this kind of sensitivity. Just keep in mind that, at the same time the mics are picking up those nuances of the guitar, they are also sensitive enough to pick up the sound of the room you are in as well...
And, because you are using a coincidental pair (XY) you are also adding a higher probability that the mics will pick up the sound of your room...which is why I mentioned the space you are recording in... is it just a room? Or is there acoustic treatment of some kind in place?

If it's an "untreated" room, meaning you have no acoustical treatment, then you'll likely get some problem frequencies, flutter echo, and maybe even phase issues.

All of the above being said... you don't need heavy correction!

In fact, you don't need much EQ at all. Your recording is very nice. I was just suggesting that you might want to add a touch of top end - to add a little more silk, air, presence. I'm not talking huge amounts here, Haris... I'm talking about adding around 6-8k by maybe 1.5 to 2.5 db. Just enough to bring out the presence a bit. Your original recording sounds great. I was just suggesting to you, as an engineer, what I'd like to hear.

BTW... are you using HPF on your way into your recording device, either by adjusting the mic (if it has this feature, many do) or through your preamp, or, through your DAW? If not, you might want to consider doing so, maybe around 100 hz or so to start... There's no point n recording any frequencies below that with your instrument, because there isn't much down there that would be of any sonic benefit.

Just a suggestion. Or two... Maybe three... ;)

Haris, post: 425042, member: 48680 wrote: Here is a 1st recording of Dionisio Aguado - Leccion No 19 I made with the Oktavias MK-012. I put the mics in xy about 60cm in front of the sound hole and recorded with maximum loudness at -12. .

It's a nice recording, very nice performance !
Do you have to change anything? No. specially that we don't know the context in which the recording will be presented. Will other instruments or voice be added ? If so I'd wait to hear the whole thing.
If the goal is to make a guitar solo recording. I'd experiment with some reverbs. it could mask a bit of the room sound and give it life. Then, I'd talk about EQ if needed ;)

Also - I like the thought of working with some different reverbs too, but, if he wants that very intimate and "natural" sound (that's kinda where I hear him leaning towards with this song), then he may want to consider adding some acoustic treatment in his space, to attenuate the sound of the room - particularly if he plans on further use of multi mic arrays...

He mentioned that he thinks that his guitar sounds "empty":

"...The sound of the recording is more "empty" then the sound of my guitar..."

Unless I'm mistaken, this sounds like he's getting more of the room than instrument, or, at least more of the room than what he prefers to hear in relation to the direct sound of the guitar.

But, I'm making assumptions here. Haris needs to tell us what he's after.

IMHO of course.

It's a nice performance and recording, and I think it is probably better than it would have been using the Zoom built-in mics. However, for me there is too much of the room in the sound, despite the mics being only 600mm from the instrument. I haven't put the recording through any analytic tools, but could the problem be echoes off the floor?

Thank you for all replies!

DonnyThompson, post: 425082, member: 46114 wrote: the space you are recording in... is it just a room? Or is there acoustic treatment of some kind in place?

It is just a room, a small accoustically untreated room.
Also the sound is totally untreated.

pcrecord, post: 425085, member: 46460 wrote: It's a nice recording, very nice performance !
Do you have to change anything? No. specially that we don't know the context in which the recording will be presented. Will other instruments or voice be added ? If so I'd wait to hear the whole thing.
If the goal is to make a guitar solo recording. I'd experiment with some reverbs. it could mask a bit of the room sound and give it life. Then, I'd talk about EQ if needed ;)

pcrecord, no other instruments or voices will be added. It is a slolo guitar recording and it will not be presented to anyone, except my online classmates at Delcamp online lessons, where the quality of Zoom H4n would be enough. So I am doing it mainly for me.

Boswell, post: 425102, member: 29034 wrote: It's a nice performance and recording, and I think it is probably better than it would have been using the Zoom built-in mics. However, for me there is too much of the room in the sound, despite the mics being only 600mm from the instrument. I haven't put the recording through any analytic tools, but could the problem be echoes off the floor?

The room has a window (but I close the curtains) and a closet with one glass door. Of course walls, 1 more closet with painted wooden doors and a desk at the wall which could provide conditions for the basses to boom.
My mics have interchangeable capsules, so I can remove carioid and put hypercardioid or omni. Should I try hypercardioid sometime? What setup?
If you can listen the room, I think I should also try to make a mono recording. Should I use hypercardioid here?

Thank you again!

Haris, post: 425130, member: 48680 wrote: The room has a window (but I close the curtains) and a closet with one glass door. Of course walls, 1 more closet with painted wooden doors and a desk at the wall which could provide conditions for the basses to boom.
My mics have interchangeable capsules, so I can remove carioid and put hypercardioid or omni. Should I try hypercardioid sometime? What setup?
If you can listen the room, I think I should also try to make a mono recording. Should I use hypercardioid here?

Thank you again!

I would stay with the cardioid capsules and stereo recording for now. What you should try is a thick rug or carpet on the floor covering underneath all of the player and the space between the player and the microphones, and having a width of at least 2m.

X/Y is a nice and easy placement that assure accurate phase relationship between the 2 signals. And you can get pretty close to the mics with the instrument.
Other placement could give wider stereo images but not without adding more room into the equation.
You could also use only on mic pointing strait to the 12th fret or closer to the hole, and use a nice reverb to make it a bit wider and fuller.

Most audio interfaces already have pre-amps inside them, at least for their microphone (XLR) inputs. It is, of course, possible to use external pre-amps and feed their output into the line inputs of an audio interface, but you have to be careful about how the signal is routed inside the interface. Most lower-cost interfaces attenuate the line inputs and feed them through their microphone pre-amps, so if you are not careful you degrade the quality of your external pre-amps by having to send the signal through the pre-amps in the interfce as well. One way of overcoming this is by using external pre-amps that digitise their outputs and send the audio digitally (using S/PDIF or ADAT formats) to a digital input on the interface, thus by-passing all of the interface's analog circuitry.

To make recommendations for you, we would need to know

(1) whether you are wanting (a) an audio interface that has suitable internal pre-amps, (b) separate external pre-amps and an audio interface with line inputs that by-pass the interface's internal pre-amps, (c) digitising pre-amps and a suitable separate interface or (d) some other configuration

(2) your budget

I think I need solution (a), otherwise I think things will become more complicated and maybe expensive than I want. My budget is at about 1k if I choose this solution.
Are there solutions of (a) category that have quality similar to combination of separate audio interface+preamp?
Could you also suggest some audio interfaces that could be uesd with preamps with analog outputs?

Thank you

Boswell, post: 428736, member: 29034 wrote: Most audio interfaces already have pre-amps inside them, at least for their microphone (XLR) inputs. It is, of course, possible to use external pre-amps and feed their output into the line inputs of an audio interface, but you have to be careful about how the signal is routed inside the interface. Most lower-cost interfaces attenuate the line inputs and feed them through their microphone pre-amps, so if you are not careful you degrade the quality of your external pre-amps by having to send the signal through the pre-amps in the interfce as well. One way of overcoming this is by using external pre-amps that digitise their outputs and send the audio digitally (using S/PDIF or ADAT formats) to a digital input on the interface, thus by-passing all of the interface's analog circuitry.

To make recommendations for you, we would need to know

(1) whether you are wanting (a) an audio interface that has suitable internal pre-amps, (b) separate external pre-amps and an audio interface with line inputs that by-pass the interface's internal pre-amps, (c) digitising pre-amps and a suitable separate interface or (d) some other configuration

(2) your budget

Boswell, post: 428746, member: 29034 wrote: I note you are in Greece, so does that mean your budget is €1K?

Take a look at the new [[url=http://[/URL]="http://babyface.rme…"]RME Babyface Pro[/]="http://babyface.rme…"]RME Babyface Pro[/] - it could do what you want.

Thank you for reply!

RME Babyface Pro is a solution I am considering.
Could you please suggest some (b) and (c) category solutions?

The option (c) that I mentioned (external digitising pre-amp) could feed the lightpipe input of an RME Babyface Pro or similar audio interface with optical input. If you are having to keep your total expenditure under the €1K figure, there would not be many quality external pre-amps with 4 or 8 channels available to fit in that price.

Under option (b), don't forget that the Babyface Pro has two line inputs as well as two XLR microphone inputs, so a dual-channel fully analog external pre-amp could be connected to those and get the benefit of the quality converters in the interface.

Tell us how many channels in total you are looking for, and we can make further suggestions.

Boswell, post: 428736, member: 29034 wrote: but you have to be careful about how the signal is routed inside the interface. Most lower-cost interfaces attenuate the line inputs and feed them through their microphone pre-amps, so if you are not careful you degrade the quality of your external pre-amps by having to send the signal through the pre-amps in the interfce as well.

Bos... can you provide insight in terms of the best case scenario for what you've mentioned above.... this being for those of us who might have a high end stand alone analog pre in our chain, but that has no computer i/o ?

When I use the ADK pre - the AP1 - ( this is the same one I used for the transformer shoot-out here on RO a few months ago), because it only has analog outs, I was having to come out of the XLR output of the ADK to a balanced ( TRS) 1/4" jack plugged into the balanced 1/4" line-in of my [[url=http://[/URL]="http://www.presonus…"]Presonus VSL[/]="http://www.presonus…"]Presonus VSL[/], in order to convert the analog audio to digital.

Because I'm using the line in of the Presonus, I was controlling the gain through the ADK pre, and not through the gain control of the Presonus input channel. In fact, I had the input level on the Presonus line-in at the minimum setting - with the input gain pot on the Presonus turned all the way down - because I wanted to get the least amount of possible further amperage coloration, and I wanted to get the best possible sound of the ADK, without skewing it through any further amping of any kind; so this is why I chose to use a line input instead of an XLR, and besides, the ADK pre already brings the signal up to line level, so further gain is unnecessary. But my suspicion is that in doing it this way, I'm still probably getting something frm the Presonus - either an added character, or maybe even a diminished signal quality, because - even though I'm using the line ins - it's still passing signal through the preamp stage before it hits the converter stage, right?

BTW, I'm only talking about one channel of conversion, as the ADK is a single channel pre. ;)

So... What would be the best conversion option for me in this scenario?

1. Coming out of the ADK to a dedicated, stand alone converter (just a converter, without any pre)?
2. Or, am I better off with using a PCIe Card-Based type of conversion?
3. Or ... do you think that my current routing - using the Presonus and its converters - will suffice in terms of quality?

(While I'm very happy with the sound I'm getting with my current routing, I haven't really had a chance to hear what the ADK sounds like on its own - for as little possible preamp gain coloration I might be adding by using the Presonus, I can't help but wonder if I am truly hearing the best possible sound that the ADK has to offer, and how much degradation I might in fact be getting by using the line ins of the Presonus...).

I don't recall getting any negative comments regarding the signal "quality" during that shoot-out... all the comments were geared towards XFO's "coloration" preferences - but should I have chosen another way, a better way, to convert the analog signal from the ADK to digital?

I'm also going to tag Chris on this same question... ( audiokid ) - and while I understand that he generally prefers preamps without transformers, that's not really the point here... this is about conversion quality, and along with your thoughts, I want to hear his thoughts also, because I know he's used more than a few converters with his rig. I'm just looking for the best possible way to go about converting the analog signal from the ADK pre, and getting the best quality possible..... if it is determined that using the converters in the Presonus VSL isn't the optimal way to convert.

:)
d.

[="http://www.google.c…"]

Haris, post: 428784, member: 48680 wrote: Thank you for reply!

Could you please suggest some (b) and (c) category solutions?

[/]
[="http://www.google.com/url?sa=t&rct=j&q=&esrc=s&source=web&cd=3&cad=rja&uact=8&ved=0CE4QFjAC&url=http%3A%2F%2Fwww.soundonsound.com%2Fsos%2Fnov12%2Farticles%2Ffocusrite-forte.htm&ei=CZtLVb-XB4XuoAT-n4C4Dg&usg=AFQjCNFxLoR2seaB-48Z97D--PI5OumOXw&bvm=bv.92765956,d.cGUhttp://www.google.com/url?sa=t&rct=j&q=&esrc=s&source=web&cd=3&cad=rja&uact=8&ved=0CEQQFjAC&url=http%3A%2F%2Fwww.soundonsound.com%2Fsos%2Fnov12%2Farticles%2Ffocusrite-forte.htm&ei=anRLVca6O5SzoQSuw4CQBA&usg=AFQjCNFxLoR2seaB-48Z97D--PI5OumOXw&bvm=bv.92765956,d.cGU"]Focusrite Forte[/]="http://www.google.c…"]Focusrite Forte[/], [[url=http://="http://www.google.c…"]RME Babyface[/]="http://www.google.c…"]RME Babyface[/] /[="http://www.google.com/url?sa=t&rct=j&q=&esrc=s&source=web&cd=2&cad=rja&uact=8&ved=0CCoQFjAB&url=http%3A%2F%2Fwww.soundonsound.com%2Fnews%3FNewsID%3D18286&ei=ynRLVZ3CDdGvogTWk4C4BQ&usg=AFQjCNHZgjI6tl7jEtItTZfkfMVYXmFemg&bvm=bv.92765956,d.cGU"]RME Babyface Pro[/]="http://www.google.c…"]RME Babyface Pro[/], [[url=http://="http://www.soundons…"]SPL Crimson[/]="http://www.soundons…"]SPL Crimson[/]. are what i have found at that price. $500 - $750.

all have decent enough preamps. the Crimson is more of a desktop controller and monitor switcher.

Boswell, post: 428786, member: 29034 wrote: Tell us how many channels in total you are looking for, and we can make further suggestions.

Two channels are enough for me. I will record only my guitar, stereo or mono.

Boswell, post: 428786, member: 29034 wrote: Under option (b), don't forget that the Babyface Pro has two line inputs as well as two XLR microphone inputs, so a dual-channel fully analog external pre-amp could be connected to those and get the benefit of the quality converters in the interface.

So could I start with Babyface Pro and in the future if I wanted, I could add a preamp and bypass the Babyface preamps?

pcrecord, post: 428788, member: 46460 wrote: with a stretch of the budget, there is the Apogee Quartet that seams to have nice preamps.. but they're not as good as outboards...
http://www.sweetwater.com/store/detail/Quartet

But that needs Macintosh. I have Windows and Linux but not Macintosh.

Thank you for replies!

Haris, post: 428792, member: 48680 wrote: So could I start with Babyface Pro and in the future if I wanted, I could add an preamp and bypass the Babyface preamps?

Yes, that would be the route for adding a pair of external analog-output pre-amps. If at some point in the future you were able to go for digitising pre-amps such as the Focusrite ISA428, then you could build up to more channels by connecting those optically into the Babyface Pro, by-passing its converters.

DonnyThompson, post: 428789, member: 46114 wrote: Bos... can you provide insight in terms of the best case scenario for what you've mentioned above.... this being for those of us who might have a high end stand alone analog pre in our chain, but that has no computer i/o ?

When I use the ADK pre - the AP1 - ( this is the same one I used for the transformer shoot-out here on RO a few months ago), because it only has analog outs, I was having to come out of the XLR output of the ADK to a balanced ( TRS) 1/4" jack plugged into the balanced 1/4" line-in of my [[url=http://[/URL]="http://www.presonus…"]Presonus VSL[/]="http://www.presonus…"]Presonus VSL[/], in order to convert the analog audio to digital.

Because I'm using the line in of the Presonus, I was controlling the gain through the ADK pre, and not through the gain control of the Presonus input channel. In fact, I had the input level on the Presonus line-in at the minimum setting - with the input gain pot on the Presonus turned all the way down - because I wanted to get the least amount of possible further amperage coloration, and I wanted to get the best possible sound of the ADK, without skewing it through any further amping of any kind; so this is why I chose to use a line input instead of an XLR, and besides, the ADK pre already brings the signal up to line level, so further gain is unnecessary. But my suspicion is that in doing it this way, I'm still probably getting something frm the Presonus - either an added character, or maybe even a diminished signal quality, because - even though I'm using the line ins - it's still passing signal through the preamp stage before it hits the converter stage, right?

BTW, I'm only talking about one channel of conversion, as the ADK is a single channel pre. ;)

So... What would be the best conversion option for me in this scenario?

1. Coming out of the ADK to a dedicated, stand alone converter (just a converter, without any pre)?
2. Or, am I better off with using a PCIe Card-Based type of conversion?
3. Or ... do you think that my current routing - using the Presonus and its converters - will suffice in terms of quality?

(While I'm very happy with the sound I'm getting with my current routing, I haven't really had a chance to hear what the ADK sounds like on its own - for as little possible preamp gain coloration I might be adding by using the Presonus, I can't help but wonder if I am truly hearing the best possible sound that the ADK has to offer, and how much degradation I might in fact be getting by using the line ins of the Presonus...).

I don't recall getting any negative comments regarding the signal "quality" during that shoot-out... all the comments were geared towards XFO's "coloration" preferences - but should I have chosen another way, a better way, to convert the analog signal from the ADK to digital?

I'm also going to tag Chris on this same question... ( audiokid ) - and while I understand that he generally prefers preamps without transformers, that's not really the point here... this is about conversion quality, and along with your thoughts, I want to hear his thoughts also, because I know he's used more than a few converters with his rig. I'm just looking for the best possible way to go about converting the analog signal from the ADK pre, and getting the best quality possible..... if it is determined that using the converters in the Presonus VSL isn't the optimal way to convert.

:)
d.

Your Presonus VSL is pretty much as good as it gets at its price level; my memory is that the line inputs go directly to the ADC drive amplifiers, by-passsing the pre-amps. In addition, it gives you the option of adding external digitising pre-amps via ADAT, so units such as the ISA 428 with ADC option I mentioned in the previous post can be interfaced with no detriment to the ISA's audio quality and no further interfacing hardware needed.

isn't ADAT lightpipe is limited at best to 96K ?.... because of that, i think it's a dying protocol. it's nice to have for backward compatibility and that's probably the reason we continue to see it implemented in new products.

i have seen reference to 192 being available in things like the ONYX preamps. does SMUX support 192 Bos?

EDIT: i found[[url=http://[/URL]="http://www.mackie.c…"] this[/]="http://www.mackie.c…"] this[/] so i guess i was wrong about that. looks like i need to rethink my forecast ...

Kurt Foster, post: 428808, member: 7836 wrote: isn't ADAT lightpipe is limited at best to 96K ?.... because of that, i think it's a dying protocol. it's nice to have for backward compatibility and that's probably the reason we continue to see it implemented in new products.

i have seen reference to 192 being available in things like the ONYX preamps. does SMUX support 192 Bos?

EDIT: i found[[url=http://[/URL]="http://www.mackie.c…"] this[/]="http://www.mackie.c…"] this[/] so i guess i was wrong about that.

Who uses 192khz now a day ? Most recordist don't hear the difference between 48 and 96 Lol !! ;)

pcrecord, post: 428810, member: 46460 wrote: Who uses 192khz now a day ? Most recordist don't hear the difference between 48 and 96 Lol !! ;)

i would go at least 96k. i hear the filters at 20k with 24/48. i think with PONO and other hi res formats emerging, higer sample rates will be the norm, not the exception.

there's great debate on benefits of 96 or 192 .... studies support the idea that even high freq content we don't hear can affect harmonics in the lower octaves we do hear. othes say they can feel it. Geoff Emrik heard oscillations at 40k on a channel of the Focusrite ISA console that he complained about to Rupert Neve about. Neve didn't believe him until they pulled the strip and found a fault.

i'm looking at it just for forward compatibility and for archival reasons. i think we will be at DSD in a few years. till then I'll feel beter knowing my archive is stored at the highest rate available.

The official data limit for standard ADAT lightpipes is 12288 bits/sec, which is 48K frames/sec, each frame containing 256 bits (192 bits of data plus 64 control, status and sync bits). How the data bits in a frame are allocated around the channels is up to the individual manufacturer, but clearly for 24 bit data could be 8 channels at standard rate, 4 channels at double rate or 2 channels at quad rate.

Many years ago, I designed a 24-channel industrial data capture system to send data to a computer from a battery-driven recording head that when operating was at more than 10KV above ground potential in a very electrically noisy environment. I chose to transfer the data using the ADAT frame format allocated as 12 channels at 24KHz, 16bit per frame because I knew I could make use of a standard ADAT optical lightpipe to achieve the necessary bit rate, electrical isolation and interference rejection. I used the control bits to keep the frames alternating between the low 12 and the high 12 channels and the data from getting out of sync. I remember thinking that Alesis did very well in the late 1980s to design a data transmission system that worked so reliably and was years ahead of its time.