Really basic advice needed

[KevWind]

Since the definition of "point source" is
"noun Physics
1 a source of energy, such as light or sound, that can be regarded as having negligible dimensions."

Originally Posted by JanVigne

I'd agree with the comments regarding microphone placement. However, since a guitar is not a line source and the article claims it is not a point source, what exactly is it?

Does it matter? Perhaps multi source? The "point" is by definition it is not a point source instrument..... pun intended

I would still consider a grand piano to be a point source sine the fundamental frequency of the note being played originates at one point along the string being struck by the hammer. Harmonics exist at other locations along the string length.

The fundamental of a piano string may itself be a single source and it like a guitar can be accurately said to be producing in fact multiple single sources of sound. Those sounds do not exist in isolation instead the sound includes the Harmonics as well as the Fundamentals . Ergo what you hear is most probably not considered point source, but rather multi source.

The reasoning a guitar cannot be a point source since it has dimensions, is again logically faulty. Can you tell me of a musical instrument which has no dimensions?

Nothing faulty about it, that statement doesn't require that there has to be a no or negligible dimensional instrument to contrast it against, in order be true or logical.

Yes I agree , there isn't any acoustic instrument that I know of, that does not have some amount of physical dimension to it. Or conversely in practical terms, acoustic instruments do not fall into the category of "having no or even negligible dimension" which is by definition the criteria for a "point source" therefore it would not seem logical to consider any as being point source instruments.

[JanVigne]

"Does it matter? Perhaps multi source? The "point" is by definition it is not a point source instrument..... pun intended"





It matters in the discussion of theoretical points or those points which exist in the real world. In theory, a point source is infinitely small to the point of lacking dimensions beyond the molecular level. In a philosophical thought process, the question would be, how many times can you divide a length of string? The answer is, there are no limits to how many divisions can be made. Thinking of a vibrating string which represents a fundamental frequency, the harmonics of the fundamental frequency occur by the first, second, third, fourth, etc, division of the string into smaller and smaller vibrational components. We understand these divisions have no limit, harmonics - or smaller and smaller divisions - continue to occur into infinity. Obviously, the smallest divisions are well beyond our ability to apply dimensions to their existence. If you apply that definition of "point source" to a musical instrument, however, the body of the instrument has nothing to do with our discussion. The body is there to support the longest, lowest frequencies only and reacts with increasing disconnect to those smallest of vibrations.


In a real world of discussion of audio we talk more of "pseudo" point sources. Assuming we cannot even perceive the frequency which results from a string harmonic at 500kHz or above, we accept the real world requirement that any useful discussion must include objects with some dimensions. In other words, the "point of a needle" sized element cannot produce sufficient pressure on the surrounding air to allow a frequency we could detect. If we then assume there must be some dimension to the sound reproduction element, we then shift our definition slightly to say a point source is not completely without dimension and look at other qualities which define any point source transducer.

To that end we make the distinction between point source, line source or none of the above.

The most significant qualities of a point source transducer in real world audio, whether it is a musical instrument, a loudspeaker driver or a microphone capsule, is the time and phase coherence of the sound source. While the physical imperfections of material use impairs our ability to create a theoretically perfect point source transducer, we can come close with a device which is for practical purposes time and phase coherent across its frequency bandwidth. In audio, a single full range microphone capsule or loudspeaker driver which covers most of the frequency range useable to the human ear is said to be a "point source" since it will be time and phase coherent at all frequencies and at all distances away from the sound source. Within the known rules of perception, we can even create a further division of a point source by mounting low and high frequency drivers so close to one another that their immediate response as it occurs at the front baffle would be perceived to be that of a single full range driver. Problems arise, however, in the materials aspect of such a design since a large driver will have problems in the highest frequencies and the smallest driver will have issues with the lowest frequencies. Therefore, a filter is needed between the two drivers and filters are known to introduce time and phase errors in the electrical domain. The trick here is the electrical errors are largely ignored by our perceptions of the acoustic event and, when done with specific filter types, the result of the combined output is perceived to be another "pseudo" point source. Co-axial drivers are often termed real world point sources though by theory, they are not. What distinguishes these speakers as point source though is their time and acoustic phase coherence across their bandwidth and at any distance away from the front baffle of the enclosure which supports the drivers.

A line source lacks such coherence in the time and phase of the output. In other words, what comes in does not always perfectly match what goes out. If, for example, we consider a line source composed of multiple, identical, full range drivers in an array which exists in that 6' tall column, we should be able to envision the time and phase problems which result. If you were to stand 1' away from the plane of the drivers, you would hear an output jumbled in time and phase due to the varying distance of each individual driver away from your ear. It is only when we step back to a distance away from the drivers that our perception of the output allows our brain to meld the multiple sources into a time and phase shifted signal which we allow as acceptable though hardly an analog to the original input. Even this example ignore any inevitable reflections which would be added to our perceptions by nearby surfaces.

Obviously, if we begin to separate the drivers - or the location of the mic capsules - over a larger and larger distance, the less likely we are to achieve a coherent result as our perception of multiple, separate sound sources increases with the distance between transducers.


Let's then think about ... oh, say, ... a guitar. And a single full range loudspeaker driver. First, the speaker will be (within the limits of present day materials use) time and phase coherent since the diaphragm of the driver which is creating the full range signal is being driven from a single point. This and a full range mic capsule are the closest thing we have to a real world point source. However, if we place the driver on a baffle - which is useful for multiple reasons - we have now introduced a device which is similar to the resonant cavity of a guitar body. Again bowing to the limitations of materials, the transducer itself will remain time and phase coherent though there will ultimately be spurious output from the baffle and connecting materials. If we turn the baffle into an enclosure, we have now completed the transition to a point source transducer coupled to a resonant cavity which will have its own contributions to the output of the system. In other words, resonant cavities exist due to pressure zones and the compression and rarefaction of the enclosed air will result in ... well, resonances not related to the direct output of the point source transducer.

Take the driver and say it is the guitar string and you'll have identical systems. The vibrating guitar string is time and phase coherent at all distances away from the sound source. If we could isolate the vibrating string as we can by removing the driver from the enclosure, we would hear all frequencies created by the vibration of the string at all points along the string. It is only when we tie the vibrating string to a resonant cavity that we create spurious output which, unlike the speaker enclosure, we find desirable. The fact remains, however, the vibrating string itself is time and phase coherent at all distances away from the sound source. In the real world we accept this as the audio equivalent to a theoretical point source.

Of course, none of that denies the fact other parts of the guitar's system contribute to the output we call "guitar". And microphone placement will affect our perception of the timbre of "a guitar". In other words, though both employ vibrating strings as their sound source, we won't confuse the timbre of a guitar with the timbre of a violin or cello. All of those instruments do, however, operate as real world point source devices. Placing a pair of microphones at different locations along the guitar's "dimensions" will result in inevitable time and phase errors in the combined output of the two channels. This is the unusual case of our perceptions being able to discriminate between time and phase characteristics in, say, a spoken voice within a room and the same voice as it exists in one channel of a recording while the room sound exists in another. In a real world situation we have quite a bit of latitude in perceiving a single sound source. When those two elements of the original sound are separated into two or more channels our brain finds it more difficult to pull the two signals back into a single coherent voice.

So, yes, in theory a point source has no practical dimension. No one denies that. In the real world though we do not deal with the quantum physics of atomic construction and division. In the real world acoustic instruments act as "practical" point source devices.

[JanVigne]

" Ergo what you hear is most probably not considered point source, but rather multi source."



Nope. Play a 12th fret harmonic. Put your ear close to the string. You will hear all frequencies produced by the vibration of the string at any location along the string's length in time and in phase. Admittedly, the physical constrictions of the guitar's body and neck will cause reflections which will emphasize various frequencies at various locations along the string's length. But the string itself acts as a single point source in the real world. If the string were "multi-source", it would lack time and phase coherence at various distances away from the vibrating string. The issue here is that there are no perfect real world point source when we discuss audio. We accept that the real world means we must include something other than the string itself - at least two connecting pints, no? Therefore, we also accept there are no theoretically perfect point sources in physical reality. Yet to better understand how things operate in the real world, we make certain concessions to theory vs perceptions.


If you want to think your "expert" is correct due to the fact they have the authority of being in print, I can't stop you from doing so. I would accept the idea the differences we are discussing are largely semantic. If I think of the guitar as a point source when I make a time and phase coherent mono recording of a guitar and your "expert" thinks of the guitar as other than a point source when making a multi-channel recording which will by its nature include time and phase errors, then so be it. Arguing this point in this thread seems to be counter-productive to answering the op's questions.

We all agree that mic placement affects the final result of any recording. I would think we all also agree that the more channels we use to record a single sound source, the more chances we have of introducing time and phase errors. Time and phase errors create comb filtering effects. Since these are largely beyond the control of the recording engineer there are only a few ways to use microphones to capture a "realistic" single instrument sound. Either we think of the instrument as a single point source and strive to maintain those time and phase relationships or we teat the instrument as a multiple sound source and let time and phase go out the window.

[steve s]

Originally Posted by JanVigne

.... Arguing this point in this thread seems to be counter-productive to answering the op's questions.

We all agree that mic placement affects the final result of any recording. I would think we all also agree that the more channels we use to record a single sound source, the more chances we have of introducing time and phase errors. Time and phase errors create comb filtering effects. Since these are largely beyond the control of the recording engineer there are only a few ways to use microphones to capture a "realistic" single instrument sound. Either we think of the instrument as a single point source and strive to maintain those time and phase relationships or we teat the instrument as a multiple sound source and let time and phase go out the window.

Thanks for your concern, but as a Ph.D. in physical chemistry, I do find the discussion technically interesting, albeit irrelevant to my primitive immediate needs, like how to set the input level, etc. Having grown up around hypothetical abstractions like infinitely small mathematical points and physical point sources, I am comfortable with everything in the real world being to some degree an approximation to some ideal. Also, although in listening to some of the links that have been suggested above I've heard examples of two-miked recordings that sound nicer than one-miked, I'm committed to using my one mike. I'm thinking that its stereo setting does give me a bit of the two-mike advantage--if I can get it aimed right--without requiring dealing with phase issues or buying a mixing board or better software or who knows what else.

Thanks, again, guys for your time.

Steve

[JanVigne]

The advantage you have with a "stereo" mic - or one of the mic placements which places both capsules in close proximity to the other - is the time and phase errors are minimized. Our ears send the signal to our mind which then perceives a more or less "correct" signal. The production effects you introduce to the recording can destroy much of what the close proximity achieves, particularly if you pan each channel hard to each side.

[Fran Guidry]

Just for jokes I Googled up Harvey Gerst. He's been a busy man.

He owns and operates a recording studio: http://www.itrstudio.com/

He designed speakers for JBL, wrote songs with Roger McGuinn that were on the first Byrds album, designed amps for Acoustic Control, and more: http://tapeop.com/articles/bonus/harvey-and-alex-gerst

And he shares his recordings with us: https://soundcloud.com/harvey-gerst

I think I'll take Harvey's advice.

Fran

[frankhond]

Scrolling through the latest text masses I might have missed the answer to the question about clipping and levels. So here is my shot at it.

When recording digital, you want to set the equipment to 24bit recording (96khz is optional for a single instrument). This will give you good headroom. You want the general level to be around -12 and peaks around -6. You should never hit zero.

[Herb Hunter]

Originally Posted by Fran Guidry

Just for jokes I Googled up Harvey Gerst. He's been a busy man.

He owns and operates a recording studio: http://www.itrstudio.com/

He designed speakers for JBL, wrote songs with Roger McGuinn that were on the first Byrds album, designed amps for Acoustic Control, and more: http://tapeop.com/articles/bonus/harvey-and-alex-gerst

And he shares his recordings with us: https://soundcloud.com/harvey-gerst

I think I'll take Harvey's advice.

Fran

Good point. What matters, within the context of the opening post request, is Gerst’s microphone placement recommendation - not how well he informally explains the physics behind his advice. Anyone doubting the efficacy of his microphone placement has only to listen to his readily available recordings.

[KevWind]

Originally Posted by Herb Hunter

Good point. What matters, within the context of the opening post request, is Gerst’s microphone placement recommendation - not how well he informally explains the physics behind his advice. Anyone doubting the efficacy of his microphone placement has only to listen to his readily available recordings.

Yes and the SOS article speaks somewhat to that also . And I would suggest that whole idea of placing a single mic at the 6" to 12" from the neck joint became a staple method from the use of an acoustic guitar as part of larger mix situation, with multiple instruments and was found to be very useful in helping the string attack cut through and not get lost in the mix. While helping to eliminate the the proximity boom of the sound hole.

[JanVigne]

"When recording digital, you want to set the equipment to 24bit recording (96khz is optional for a single instrument). This will give you good headroom. You want the general level to be around -12 and peaks around -6. You should never hit zero."




We've been through this recently; http://www.acousticguitarforum.com/f...d.php?t=308643


My advice remains, experiment. Learn. Don't be afraid to fail.



I'm not going to suggest anything regarding Mr. Gerst's advice but working at JBL in the recent past is, to some, a dubious honor. It would mean he was there with Sean Olive and under the directive of Sean Olive. And, since arguments from authority are big on this portion of the forum, that is an "authority" many listeners take as a precarious position.

In the high end audio community Mr. Olive is seen as one of the uber-objectivists who take little notice of those who listen for a living or for their own enjoyment of music. "Pedagogic" and "dictatorial" would be the more flattering descriptions of Mr. Olive's approach to audio if you ask a subjective listener.

That shouldn't be taken as a rejection of Mr. Gerst's skills in the studio. But I would say there are many ways to accomplish a successful recording of a single guitar. This is especially true when the desire is not for some highly processed, "studio" quality result but rather a simple recording of someone playing with care for the sake of capturing a sound they might soon miss. In such a case, I view all of this "you must do this or that" to be highly overwrought and unproductive. If the op should decide a single, mono mic will do all he desires, let him reach that decision. If he decides to try for a more complicated solution, then he'll arrive there on his own. He will learn more from his mistakes and trying again than he will from following some forum originated recipe for what not to do. I don't doubt that most of the advice provided in these threads is well intentioned but I do think the advice provided in these threads too often ignores the needs and desires of the op.




IMO recording in 24 bit or 96kHz is fine advice ... if that will be the final format used to store your data. Given the opinion of the uber-objectivists that CD rates (16 bit/44.1kHz) are "perfect" and that anything more than "perfect" is the act of an "audiophoool", what is the purpose of recording at 24 bit?

I consider myself an audio subjectivist, I listen and I take note of what I perceive. Since any digital format will, by its nature, be finite in its resolution, I would suggest to the newbie home recordist that they experiment with levels. For the very reasons I stated in the previous thread. And that 24 bit files are fine unless you intend to store the data at or transfer the data to a lower bit rate. In those cases, higher than needed bit rates become more of a hindrance than a help IMO.

While I don't advocate recording at the lowest MP3 rates, working in the accepted 16 bit/44.1kHz rates are fine for a single instrument and should not have a perceptible difference between higher bit rates when played over the vast majority of systems or for the vast majority of listeners. Once again knowing the uber-objectivists, "numbers tell me all I need to know" arguments for the lack of perceptual differences between 16 bit and 24 bit recordings of far more complex material than a single guitar, I would have to think in the case of the home recordist - particularly the home recordist with modest equipment and modest desires - recording in anything higher than 16 bit is folly. Moving a microphone a few inches will make far more dramatic changes in your results than will any audible differences between bit rates above CD standard. Recording in either WAV or FLAC formats should be fine for the op. Both formats include all the data which is present at the time of the recording - where MP3 and lower quality rates discard some information for the sake of quantity being stored - and there will be nothing lost even if the recordings are transferred at a later date to another format.

Do experiment with your levels. Learn your equipment. Do not follow anyone's "advice" as they try to dictate what you should think is a good recording. As we discussed in the previous thread, recording at lower levels but higher sampling rates is, IMO, adding coffee to your milk.

It's your equipment and there are very few rules which are universal among equipment manufacturers when it comes to home or even "semi-pro" (nice marketing speak, is it not?) audio gear. I have to believe in this thread the op is not about to do multiple layers of mixing, editing, over-dubbing or after the fact processing. "Ergo", there is every advantage to recording at the highest level as read by the meters on his own equipment - not some professional studio, we're gonna mess with this for a few weeks now mindset - which will not result in overload of the system. But don't take my word for it any more than I would suggest you take anyone else's word for it. Don't get caught up in the idea there is only one way to do this. if you do, who will eat all those bananas you have hanging around?

Each system and each recording is unique. Learn what your equipment can do and make the most of it to satisfy yourself, not someone on a forum who thinks you should do what they say.

My advice is still the same and still very simple, experiment. Listen to your results and make the appropriate changes to your next try. If need be, ask for a second opinion from someone in your own house. "Do your own thing!" If you go into this with the idea you have to do "this" and "this" and "this" but never "that", you've just placed more pressure on yourself than necessary when all you really want to do is have a nice recording of your guitar performances. Something you can enjoy in the coming years, not something you can criticize on its technical merits forever and ever.

Have fun with this and good luck with your recording and your physical issues. I hope I've understood what you've asked of us and that I've given you good advice for a pleasing result.

[rick-slo]

Moderators might need to move this thread to the "Open Mike" part of the forum.

[JanVigne]

Why? If it goes there, not many people are going to find it.

[alohachris]

Aloha Steve,

I PM'ed ya.

alohachris

[david_m]

Oh my goodness. What an interesting road this thread took.

Without getting into the specifics and details of the debate (which I didn't read in detail)........I close mic my guitar for recording (~12" away) because I'm recording in an ugly sounding, poorly corrected, spare bedroom. I DON'T want the room sound to be prominent in the recording.

If you're recording in a lovely sounding room then by all means far mic to your heart's content. If you're recording in a compromised space (like the majority of us home recording hacks) a close mic can be essential to getting good raw material.

As Jan said, experiment and don't be afraid to fail. Your ears will tell you what's right and what's wrong. If YOU like the recording, then it's a good recording.

[steve s]

David,
Thanks for the suggestion. Chris gave me tons of advice on fixing the recording room, which I mostly filed away. My main playing chair is an old Morris rocker, and there are definite limits to how much gear and wiring can be tolerated in the front room. Since my playing is no great shakes now, and I know it's destined to get only worse, the normal motivation to buy good equipment to amortized over time is somewhat lacking.

I started out assuming close-miking was the way to go, but even with the mic up around the 12th fret I got too much boom from the sound hole, plus all of the noise from my fretting insufficiencies. About 2 feet away seems to be working for me. Mike is still over the 12th, but its angle is crucial to avoid hole noise and distortion, and to keep the Yeti's stereo channels close to balance. I've had to set the gain so signal max is between -6 to -12 dB. So far, S/N on playback is ok.

My success-meter is watching for my wife to smile when I play it back through the stereo system. I could see on my first several attempts that she was struggling to be polite, then suddenly I found a combo that brought the smile and "Now, that sounds like you."

My next big decision point will be when I have to decide whether to stick with the WAV files or go with some level of MP3 in order to fit stuff onto a CD. Do I believe the experts who say that blind A/B tests prove that people can't tell the difference or the experts who say they can?

Steve