greenmachine

Her mic (-41dB) is about 6dB more sensitive than yours (-47dB). That means you'll propably get even quieter results with a similar setup.

Two more suggestions for a better overall sound just popped into my head: -definitely place your microphones higher than floor height, just imagine how muffled it has to sound down there... -if you're skilled, get your microphones a proper voltage, the internal 1.5 volts battery may be sufficient for recording lectures, but not for loud music. A 9 volts block should do the job.

Instead of spending money for new equipment, spend some time to educate yourself about microphone placement, proper use can make the major difference if your recordings sound 'flat'. Besides, try to avoid bass roll-off filters whenever possible. To get rid of it with your current equipment, simply replace the capacitors in the battery box with some of higher capacity. Superior? No way, it's just some kind of a cheap makeshift with some serious drawbacks.

Since you have a mic with a built-in power supply, why don't you skip the attenuator and use mic-in directly for quiet to medium-loud and line-in for louder sounds? In my experience you can't reduce distortion any further when you use a lower level setting than approximately 10/30 (mic-in / Sony recorders), for these situations plug to line-in and use anything adequate up to 30/30. Just a suggestion, figure out what works best for you.

This maplins toy seems to lower the supplied voltage from the recorder to the microphones significantly.

d.) she confused headphone and microphone jacks, plugged the mic with the attenuator to headphone out, the headphones to mic-in and thus recorded through headphones

3 potential faults i can think of: a.) she used line-in instead of mic-in by mistake b.) she set the attenuator to minimum instead of maximum volume by mistake c.) she set the attenuator to minimum and used line-in I wouldn't get rid of the attenuator in loud situations, unless you want to try the directly to line-in setup.

Not true, -50dB is less sensitive than -42dB, while -35dB is more sensitive. This site seems to clarify some confusion: http://www.acoustics.salford.ac.uk/acousti...Microphones.htm

No, i don't roll off anything, i try to capture the sound as close as possible to 'as heard' and do adjustments afterwards if necessary. For quiet to medium sounds i use the built-in preamp of the md unit (virtually always low sensitivity setting - the high sensitivity setting seems dispensable for sensitive microphones), for loud to very loud situations a self-made 'battery box' (without bass-rolloff) and line-in. I've never experienced any distortion with this setup even at bass-heavy, ear-damaging indoor rock concert levels. The cheaper attenuator may work for medium to loud sounds, but failed to do it's job at extreme levels. You might want to take a look at my live recordings gallery and compare the m/a (music to applause) ratio ( ) to your recordings to get a rough idea of what i consider to be very loud. To be fair, i have to admit that i use modified microphones, no idea if yours can handle such levels.

Neither one of those reduces a limited frequency range exclusively (unless a 'battery box' has a built-in filter, be it selectable or fixed), but rather the whole spectrum (if used properly). I wouldn' t cut off frequencies selectively anyway, rather do some equalization afterwards if unavoidably necessary.

Regarding your editing / mastering skills: You seem to have boosted the whole thing by about 10 dB by using some kind of limiter / dynamics compressor to achieve a higher overall loudness - besides the fact that compressed dynamics of accoustical instruments / orchestral recordings in general sound unenjoyable to me, there are some other serious drawbacks: Since your left and right channel don't match in volume, caused by your recording position / mic placement (and maybe to some extent by the matching tolerance of the two mic capsules), you were compressing the louder (left) channel more than the other one, which sounds strange to me. Also you've introduced clipping at the loudest position (seconds 53 to 56 of the edited file). A simple normalization would propably have done the job in this case without sacrificing sound quality. The abrupt cut of the applause is annoying, maybe you could have faded it out gently. By the way, paradoxically the applause sounds far more realistic and involving than the music itself, maybe you've been too far away from the source.

...and can't reduce/eliminate it by adjusting the levels and the hi/low sensitivity setting on the recorder.

Microphone self-noise can get annoying in quiet passages of very dynamical sounds and for recording quiet sounds in general. For such situations it would be best to use mics with a good s/n (signal to noise) ratio together with an external higher quality / lower noise microphone preamplifier (and a power supply for the microphones, a.k.a. 'battery box' - if not built in to the preamp). Also consider that an analogue transfer from md to pc can introduce additional noise. The best sounding most versatile costruction for me is a omnidirectional stereo microphone construction, wired like most headphones, this gives a lot room to experiment, to try out different positions / distances / baffles in between. It looks roughly like that: I often place an 'artificial human head' in between, it gives an excellent stereo image, it can look like that (or similar): Or for less flexibility but less cable to carry around (?) a t-style omnidirectional stereo microphone with a fixed distance (about 5-7 inches) between the elements (nevertheless an extension cable is suggested to avoid picking up motor noise from the md recorder), it looks roughly like that: Please take a look at this recent thread, especially post #5/6 for some explanation of common microphone related jargon: http://forums.minidisc.org/index.php?showtopic=10218 p.s. Do you mean 'download', since you don't seem to have the mics or recordings so far?

Directly plugged-to-the-recorder microphones tend to pick up motor noise from the mechanical drive, which can be annoying in quiet situations, not so obvious for loud sources. With an extension cable and some physical separation from the recorder it can be avoided. I have to say i'm not a fan of cardioid mics with little to no separation between the elements - even if it may be convenient - in comparison to other mics this model sound pretty 'thin', that means they have a narrow frequency response, they cut off a lot of bass and some of the high frequencies - here's a little comparison with samples between different mics, mostly Sony's: http://infos.0db.net/micros/compare/indexe.php3 But consider that all of these are one point cardioids, you can achieve a much wider, more realistic stereo image with greater separation between the left and right channel elements, keywords here are 'binaural' and 'jecklin disc'. I don't want you to sell anything, just make you aware that there's room for improvement, but maybe they even satisfy your demands.(?)

And by the way, there is no such thing an an universal direct answer. Search around what suits your needs (and your budget) best.

The only drawback of a too low level is an increased noisefloor (after normalizing) AFAIK. If it's not too dominant and you can live with it, it's fine, at least better than agc artifacts or clipping.

Yes it does (at least with my device - mz-r909), just tested it with a standalone cd player as source. Propably his input signal was so low that the whole recording was below the point where agc kicks in.

For suggestions to record the piano, check out this thread. Also this one for microphone recommendations. For WAV to MP3 conversion you can use LAME.

An optical INput on your recorder doesn't really help for transferring to your puter, so the cable doesn't help you if you don't have a separate unit with an optical OUTput. There is no difference between optical and digital in this case, optical is just one method to transfer digitally, the other one is called coaxial (electrical). If you don't have such a deck with optical OUT, you have to transfer it the analogue way, which can still give good enough results although it's lossy method (unnecessary additional DA-AD conversion).

...where some editing can be essential, since you have virtually no room information (coloring of the sound, delays from walls...). It can be a bad idea to record from the mixing board sometimes though, because the mix is designed to sound good together with the room's accoustics, some instruments - like drums - can be even so dominant that they need no amplification at all and thus are virtually non-present in the mix from the board, it should work for purely electronic music; but there are others who can explain it better in these threads: http://forums.minidisc.org/index.php?showtopic=8788 http://forums.minidisc.org/index.php?showtopic=5908

Alright, i get your point, but am still strongly against trying to compensate for inadequate equipment by post processing, especially if a better solution seems to be obvious and easily available. I assumed we were talking about recording live music, a rough and ready mix of several instruments and room accoustics, where in almost all cases it's best to record it as close as possible to 'as heard' without touching it in any way in my experience. It's a different story if you record separate sound sources though, especially from a close distance with little room information. This is the point where creativity is desired for the final mix, even if that means to alienate a sound beyond recognition to create an illusion of an artificial world. Other opinions welcomed.

This is what you'll get if you'd phase a mono signal by 180 degrees:

What kind of language did i speak this time?

It works like that: http://www.creativepro.com/story/feature/19197.html, just imagine the whole procedure for audio. If there's ever a situation when you can't capture the whole dynamic range (the quietest and the loudest parts) at once without introducing either clipping/distortion at the high end or noise at the low end, it could be (theoretically, i've never tried it) a good idea to use multiple, differently set channels, you intentionally 'under- and overexpose' parts of the signal and combine them (there are different not equally good methods) in the end. But since i believe the audiophile world is much more sophisticated than the imaginary, this technique is propably useless because of the already quite high dynamic range of single components nowadays available. I'm not sure how to synchronize the recorders with your method and how the result would sound if done properly though. How would it sound if you stood at two different places at the same time anyway?

I assume you mean to use the same mics for both recorders with an adapter or similar gadget. It sounds much like exposure bracketing (and combining afterwards) in photography to expand the dynamic range. No bad idea at all, but somewhat difficult to realize, especially with two independent 2-channel recorders instead of one 4-channel recorder. You have to be very careful not to 'off-phase' the result, a slight time difference can ruin it if you don't combine it very carefully. Also you should be using a lossless recording mode (pcm/wav) to avoid possible differences introduced by an encoder. This method screams for perfection. I like the idea.