A caveat to start. I am fully aware that everything I will be saying will probably be overkill for the application at hand. I fully acknowledge that while I do know some things, my knowledge is limited and hope that this post doesn't come off in the wrong way. I'm merely an insomniac who needs something to do at the moment. My attempt is to be educational, and not a nag. And though it may seem like it, I am not criticizing the current setup in so far as it pertains to what it is being used to accomplish. I would personally do a few things differently, but I think that concept of overkill is going to appear quite often. If any of this comes across in a negative fashion, then please feel free to pull it down and I will adjust as necessary.
I will start by simply following the signal flow.
Sennheisers are fantastic mics. I use a variety of them at school and on the job when I'm with my bosses at the symphony. If you are wanting to measure the noise of that the given pc part makes, then using a super cardioid is a great way to zero in on it. Although, if you are making a recording to compare the sound produced by item and the ambient noise to the room, then an omni-directional mic would be more in order. This will address two of the issues previously mentioned. Having the super cardioid at a greater distance from the sound source does reduce "proximity effect." In other words, directional microphones have an increased low end response as they get nearer to the sound source. Simply put, they get boomy. Having an omni-directional microphone eliminates proximity effect since it theoretically picks up sounds equally in all directions. Having a cardioid microphone means you are rejecting the room reflections that are emanating from behind the microphone, which does have a profound affect on a sound. (Yes, I know I'm knee-deep in overkill).
A binaural Neumann mic would be superb, but your other options would be to set up a two cardioid mics in XY, ORTF, or set up two spaced omni's. XY setup invovles two microphone elements set as close to each other as possible, one pointing left and one pointing right with one microphone element on top of the other. This is called a coincident pair. It will tend to create a wide stereo image which might not be the most accurate for this purpose. An ORTF stereo pair is when the elements of the two microphones are set up on the same horizontal plane at about 6-7 inches apart at about a 110 degree angle or so. The technique was developed by French radio and television stations back in the day. It was designed to model the way humans hear. You do have to be careful about phasing/comb filtering issues (I will spare you that description) which can cause serious misrepresentations about the sound in a room.
One must also be aware of a microphone's frequency response. The me66 has two different frequency responses. The pdf on Sennheiser's site is hard to make out, but it looks like there is a bump in the high end frequency starting at about 5kHz, peaking at about 9kHz, and then rolling off after 10-12kHz. What this means is that sounds residing in the sibilant range (any hisses, the 'whoosh' of air) will be amplified between 3-6dB. To us, this means that it will sound about twice as loud as it really is. There is apparently another frequency response that the me66 can have that has a cut in the high frequencies. If I were approaching it, I would find a mic that possesses as flat a response as possible.
In other situations, cable would matter. This is more of an even geekier side-note, but CAT-5 cable is likely to become the next standard of audio cable and I predict that in less than 10 years XLR cable will no longer be the mainstream. I won't get into all that here, but basically they have transmitted 4 channels of analog audio up to a 1/3 of a mile without any signal degradation on a single CAT-5 run! Try running four XLR cables 1/3 of a mile and watch what inductive and capacitive coupling do to your signal. Anyway, since the cables you are using in this rig are "electrically short," then this doesn't matter.
We use M-Audio products on many of our ProTools HD rigs at school. My experience with them in comparison to other audio interfaces (RME or Apogee) is that the M-Audio stuff tends to sound a little more brittle in nature. But, we are also talking the difference between a $200 sound card and a $500+ sound card. 24-bit and above (although 64-bit is overkill, even for audio folk) is ideal for any and every application with the exception of storage based needs. 192kHz would be the ideal sampling rate as the Nyquist theorem shows that there is a "brick wall" filter at 1/2 the sampling rate. By sampling at 96kHz, we loose all frequencies above 48kHz. But, you say, humans don't hear above 18-22kHz on a good day. Well, we don't "hear" it, but our brain does process it in some way that does affect the harmonic content of a sound and therefore can play a major effect on our aural response. Psychoacoustic studies have shown that humans can detect (not necessarily hear) changes in frequencies up to 80kHz. Again, I realize that this is somewhat superfluous to recording computer noise, but I figure if you weren't interested in audio you would have stopped reading a long time ago. Please oblige my insomnia.
I honestly know nothing about the power amp being used, so I can't really comment on that.
I also do not know the monitoring system being used. Take the same theory discussed about microphones, flip the signal flow in reverse, and you get the same thing for speakers/headphones. Ideally, you want as flat a response as possible. Otherwise you will have an inaccurate representation of the sound. What's interesting is that with this equipment that tends to have a "brighter" response (more high frequency content) you get the image of hearing more noise than is actually in the room. On the other side, we are drawn to this sound because typically we've lost a lot of our high frequency hearing by our early 20's. I have tinnitus at 27 and its no fun. I have to be careful about adding in the frequencies I want to hear because they might actually already be there, I just can't hear them.
When it comes to codecs for posting the sound online, storage-wise, you cannot beat a compressed (lossy) codec. As far as accuracy is concerned, you can throw that out the window. Any codec that is lossy (Ogg, AAC, WMA, MP3) can have compression ratios upwards or exceeding 40:1! To minimize the audio information these codec do the equivalent of taking the bass and treble knobs on your stereo and turning them almost all the way down. If you are ever curious about this, take your favorite classical cd and import it as an mp3 and listen then A/B them. The reason why mp3's don't sound all that different from the radio is because modern music generally tends to have about 4dBs of dynamic range (the difference between the loudest and softest sounds in piece). Whereas classical music can attain a dynamic range of 24dB very easily. So, we may think mp3's aren't all that bad (and they are convenient), but unless you EQ the recording to compensate for that compression, then all of your hard work can't be fully appreciated. Also, the Fletcher-Munsen curve plays a big role in how we hear. What these guys said was basically that our hearing changes at different amplitudes. At low amplitudes, we hear less lows and highs, but retain our hearing in the mids. The reason for this is left to the evolutionists among us. So, not only is the mp3 getting rid of the highs and lows, we are trying to hear low-level sound at low levels. I would recommend that if you are going to use a compressed codec that 320kbps be your choice rather than 128kbps. If you think that it's worth the extra space and download time, then a lossless codec (Wav, FLAC, WavPack, Shorten, Monkey's Audio, OptimFROG) might be a more accurate choice. Ideally, 24-bit/192kHz lossless would be optimal.
Like I said, overkill. But, in case someone wanted a little bit of audio theory, there it is. I won't rant about how mp3's are destroying the world and that ipods are killing everyone's hearing, but please, please watch your listening levels when gaming or anything else. Being kept up at night because your ears ring too loudly is not a happy thing. If you are an avid gamer and like to have your face blown off, please invest in a decibel meter and follow the OSHA standards chart for listening levels. At least until they invent nano-machines to restore all the cilia in our ears, then its all fair game.
Hope this hasn't seemed pompous or heavy-handed, neither of those things were my intention. I hope it contains some relevant information for someone. If any of the other knowledgeable folk see any mistakes, then please point them out. I would be much obliged.
Just one of those facts for fun, quality microphones can easily jump up $10-14,000 no problem. Between my professors and our collection at school I'd say we have about $200,000 worth of microphones. It's a weird feeling when you are holding a pair of microphones in your hands and your realize that you could buy a car for what they would sell for.