ProAudioGuy

Jadeclaw's response is largely correct but bears some expansion: First with SACD, Sony made sure from the beginning that digital copies would be difficult at best. Very few SACD players output any digital stream, and those that do are using, as Jadeclaw points out, downsampling to 16/44.1. That said, using analog I/O between units going through a single D/A to A/D cycle from SACD to MD should bring about results at least on par with digital dubs from 16bit PCM (i.e. via your S/PDIF input). DVD/Audio is in a similar fix, they were just more supportive of existing DVD standards. Most DVD/A discs have a regular PCM program outputing 5.1 compatible material (I think usually in DTS but that may be coincidental to my titles). The other thing to remember with DVD/A is the -potential- for analog watermarks in the signal. I have not personally been able to discern this but others say it is clearly audible onceyou know what tol listen for. More experimenting there for sure!

Lot's of good advice here. Let me throw in a a little more, as I've done a lot of MD recording of live music where quality was important. First, there -are- a lot of issues with your built-in preamp. It is fairly noisy, overloads pretty easy, has limited frequency response, and doesn't match up to the quality of an external preamp. However, just adding an external preamp doesn't solve all your problems. You now have to deal with an extra box, power issues, cords, more level setting, etc. Here are some hints to make the most of your unit as-is: 1) forget whatever noise you hear with nothing plugged in. Unterminated mic-preamps will pick up all sorts of electrical noise that goes away with a mic plugged in. 2) make sure you are setting the preamp gain to the lowest possible level to get a decent recording level. Since we desperately want to avoid digital clipping, you should be pretty conservative about the level setting anyway. 3) never, ever use the automatic gain (aka auto level) for mic recordings. It produces all sorts of artifacts (popping, "breathing", clipping, etc.). 4) get a better microphone. The Audio Technical A-T 822/824 are nice stereo condensor mics made especially for use with portable recorders and camcorders. They are self-powered, easy to position, fit a regular mic stand, and provide a good level signal to your MD unit. 5) forget noise reduction software or hardware. Reduce noise at the source, don't try to remove it later. All noise reduction methods are a compromise and are not suitable for most digital recording methods (unless you're recording in a physically noisy environment and need to remove that noise during editing). Alternatively, you can get into the area of using an external mic preamp. Sound Professionals makes one just for use with portable devices. A more budget minded, but less portable, solution is M-Audio's Audio Buddy mic pre. It is about $99 retail, has two decent preamps, phantom power for professional condensor microphones, and a selectable output level (-10db or +4db, -10 being the choice for your MD unit or any consumer gear). Also, with that in hand, you can buy a pair of inexpensive condensor mics that will shame your little Sony microphone. Oktava's MC1202 is a great choice and are often on sale for under $120 a pair. Hope this helps some, and good luck. Half the fun is trying to get the most out of what you have, so start with that. :cool:

Dex's responses are spot on regarding the subject of bit-depth and these different types of formats. Bit depth is a PCM concept, but we tend to apply it to non-PCM formats (such as DSD) as well, mostly as a point of reference. What is the bit-depth equivalent of DSD (SACD) for example? Well, somewhere well north of 24bits. But then, that -is- misleading, because DSD works so differently from PCM. The more important point Dex makes though is not driven home enough. Lossy formats such as MP3, ACC, and yes, ATRAC, all depend on conversion to a linear stream (PCM) first. I say linear in the sense of the time domain, if you want to get technical. So the conversion depth of the A/D and D/A converters becomes signficant. If your A/D front end on an MD unit is 16 bits, then you have a limitation right there. Now, you say, I'm using the digital input of my MD, so conversion isn't an issue. No? What is the input, native ATRAC? No, it is likely a S/PDIF connection passing at the most 20bit PCM audio through (you don't see a lot of 24 bit stuff on a S/PDIF connection folks). So again, conversion depth -somewhere- along the line has become a factor. So my MDS-JB940 has 20bit 44.1khz converters. I will get up-to (theoretically) 20bits of resolution on the other end of my ATRAC recording. You say wait, it is ATRAC compressed, how can you re-resolve 20bits? Well, ATRAC supports > 16 bit dynamic range, and that is largely what we are talking about. Then again, the noise floor in ATRAC recordings is probably less signficant than the ATRAC version and the encoding algorithm used. An interesting subject with many, many dimensions. Remember to consider, as Dex says over and over, the entire signal chain. And remember that in the majority of pop music recordings, at least some of the content has been through 16/44.1 PCM conversion (effects processors, digital recorders, digital mixing desks, etc. etc.). Happy MD'ing everyone!