Re: [linux-audio-user] Re: producing a drum sample library for hydrogen

I'm not an audio professional, nor have I ever played one on TV. But it seems to me that one should probably minimise resampling at all costs unless it's necessary. If you have lots of libraries at 48K/24-bit, you should probably record your new material in that format, do all of your processing and mixing in that format, and only when exporting to your end-destination format doing the requisite dithering and resampling. I guess Hell is defined as your project having vocals recorded in 44/16, drum samples in 24/96, and the rest of your electronica in 48/16. I think even "trivial" up/downsampling [1:2 or 2:1] ratios are frought with some peril, since in the downsampling direction you must be absolutely sure there is no spectral energy above Mr. Nyquist's limit. Every resampling event implies some filtering, and filtering seems like one of those you things you do only when you have to. Filtering removes content, and even the best of them probably remove something harder-to-measure than S/N ratios and frequency response. There is a great deal of chatter amongst the audio professionals about "vitality" and the "reality" of a given recording's sound, which cannot be but hurt by too much chewing and processing. Again, I'm an outsider to the MUSICAL/CREATIVE part of audio - <EMBARRASSMENT> up until shockingly recently, I thought Octave progressions followed a power-of-2 sequencing. </EMBARRASSMENT> I know only one "audio professional" personally, who is a rather cynical and grizzled guy who is a hardcore ProTools Snob [not the new host-based stuff]. "Wire speed is my latency, fool." "Straight wire with gain is all I need from you stupid engineers!" He responds to my discussions and questions about DSP, digital filter design, and all of that as such "forest besides the trees" talk... "But how does it SOUND?! Are you so busy with your digital processing mathematics that you're using your EYES instead of LISTENING?" Etc, etc. He's obviously not a Luddite as he makes extensive use of digital audio [PT is an all-digital system], but his attitudes about which tools to use and when are remarkably different than what might seem logical to a mathematician or electronics engineer. It seems reasonable that before setting out to decide on your recording format, you need to figure out what you're producing for. Movie soundtrack audio seems to be centered around 48KHz/16bit formats [I think they do field acquisition at 20-24 bits on their Nagras though, no?], with CD audio being 44.1/16 of course. If your album is going to be "heavily processed" anyway, I suppose there's probably little harm in recording and editing at the highest format quality you can use, since repeated iterations through DSP stuff will be damaged less at the more "precise" formats. 16 bits is such a shockingly small numeric range, really, when you start tossing it into floating point algorithms for DSP. That final downsampling is probably going to do alot less damage to your work than the accumulated round-off distortions going back and forth to/from floating point algorithms. Anyway, this is my 1/2 penny. As a computer science/mathematician-oriented person, I know full well the trap of "I'm an engineer, I can FIX that..." impulses which arise from a consciousness and a grasp that every "problem" just requires another algorithm or technique to evolve or tune. But sometimes, less is more, and as I listen more critically to various pieces of music, I find the more "intact" it is, the more "real" it is to me. =MB= -- A focus on Quality.