Question for recording artist/engineers


Let's say you have a jazz band who wants to sell cds of their music with the best quality of sound they can achieve at the lowest out-sourced cost or do-it-yourself. If one wants to do a just-in-time type of manufacturing of their cd, how can they improve things?

Currently they are recording at 48k in Pro-tools, mastered in Sonic Solutions by Air Show Mastering, and then they use top of the line cds (Taiyo Yuden) with a Microboards Orbit II Duplicator. This has produced average cds but we want to do better.

What would you engineers do to improve this so it gets closer to audiophile quality? Would you recommend using a different mastering house, different cds, or a different Duplicator? Or would you just bite the money bullet and go directly to a full-scale manufacturer? We are trying not to have that much money tied up in inventory.

If this is the wrong place to post this question, please suggest another message board to post.

Thank you for your feedback and assistance.
lngbruno
That's not correct Piedpiper. 88.2k is not just halved to 44.1k. Conversions from 96k and from 88.2k to 44.1k both need a low pass filter and rate conversion algorithm. Also the conversion from 96k to 44.1k involves no losses relative to 88.2k to 44.1k. It just needs the right high quality conversion algorithm, which is present in the Sonic Solutions workstation.
Thanks for the input. The low pass filter makes sense but why wouldn't they just throw out every other sample? Why go with higher sample rate then? Archival only?
If you throw out every other sample of an 88.2khz signal, you will have a 44.1khz data set, but with all the frequencies over 22.05 khz aliased into the audio band. The low pass filter is used to attenuate signal energy above 22.05khz (Nyquist) before reducing the sample rate.

Not just archival; it actually sounds better to record at higher frequencies and then downconvert to redbook. Current belief about high resolution says it sounds better than redbook mainly because of reduction of the distortions caused by filtering, especially steep brick wall low pass filters. When you start with an 88.2/96khz/24b signal, you still need a steep filter at the downconvert stage, but there are steps in the filter design that can be taken to roll the filter off more gently, keep ripple very low, and dither the 24b signal down to 16b. I can recommend papers if you're interested.
Thanks for the info! You obviously know your stuff. Are you saying that 88.2 has no advantage at all over 96k. Would it not still be a simpler conversion. I'll do some listening.
Flex - I don't know the answer to this myself, but if there were no advantage to using the simplest possible algorithm (i.e., "throw out every every other sample") as opposed to something more complex, than why do you suppose we have inherited standards that represent frequency doubling (48KHz to 96KHz to 192KHZ, 44.1KHz to 88.2KHz) and tripling (44.1KHz to 132.3KHz)? Are these just remnants of a time when computing power was more precious?