Converting 96KHz flac files to play on Squeezebox

[WinterRacer]

Hi All,

I have a Squeezebox Duet & run Squeezecenter on a Netgear ReadyNAS Duo. The latest release of Squeezecenter (7.3.2) will convert files with a sampling rate higher than 48KHz down to 48KHz when running on a Mac or Windows PC but the ReadyNAS install doesn't support this. If have a few flac files sampled at 96Khz that I'd like to play, so does anyone know of a programme that will allow me to down-sample from 96 to 48KHz (or 44.1) - and preferably free

One other more open ended question... I understand why one would want to use a sample precision of 24bit instead of 16bit, but why are people interested in sampling rates higher than 2 * 20KHz? Is it because poeple think we can perceive frequencies higher than 20KHz when listening to our hi-fi?

Thanks,
Rob

 

[Anonymous]

PC: Foobar, Nero

Mac: XLD, Maxÿ

 

[WinterRacer]

Thanks, Foobar worked a treat.

Rob.

 

[fatboyslimfast]

WinterRacer:One other more open ended question... I understand why one would want to use a sample precision of 24bit instead of 16bit, but why are people interested in sampling rates higher than 2 * 20KHz? Is it because poeple think we can perceive frequencies higher than 20KHz when listening to our hi-fi?
Thanks,
Rob

Well, there has been research that suggests that whilst we cannot "hear" and recognise sounds above 20KHz, they do have an effect on our listening. Some reports suggest that recordings (and reproductive equipment) that go up to and beyond 100KHz sound more "pleasing" than ones that stop at 20Khz.

The other theory is that if the recording and equipment can resolve up to 40KHz or more, then the sounds that it produces at 20KHz should be far less distorted, as it is not at the limits of the technology.

 

[WinterRacer]

Thanks, I did a quick search around and found some interesting threads on this. It seems opinon is split (as usual!) as to whether the befefits are around being able to hear above 20Khz vs. benefits it brings to DSP, e.g.,

"The main reason why sampling rates > 44.1Khz is used is that at the hardware / player, the required transition bandwidth of the analogue low pass anti-aliasing filter is much - much wider and thus a simple low order analogue filter is sufficient. Building a low-order linear phase low pass filter is very simple compare to a high order filter!

It has nothing to do with psychoacoustics but electronics. If the sampling rate is 192 kHz, do not expect to hear above the usual 18 - 20 kHz range.

As for the case of 24 bits, I supposed 16 bits audio isn't enough to represent the entire dynamic range of the human hearing. At its most sensitive region, around 2 - 4 kHz, this dynamic range is actually greater than 100 dB"

What is reassuring is that not many people seem to say there is no reason for it other than marketing!