High Playback Sampling Frequencies, Why are they becoming popular? Options

[icstm]

OK, at the risk of sounding completely stupid?

Why do people listen to music with high sampling frequencies?
What is 88k+ providing them?
I understand why you might record at a high sampling rate, but why keep that for playback?

Looking through the FAQ, there are threads from 2003 that point out that the sampling frequency and bit-dept work in tandem. So the quantisation error of 16bit at 44.1k has the opportunity to be corrected sooner at a higher sampling rate, so in some ways is like a dithering pattern.

However given noise introduced in the analogue systems required to listen to music, a SNR within a 16bit signal of ~96dB seems pretty good.

So assuming that speakers struggle to produce the sounds that a 192k sampling frequency allow (eg 96kHz) and assuming that 16bits were sufficient when compared to the analogue equipment in the system, what have I missed in these high sampling playback formats?

[Walter_Odington]

Your right that my understanding is limited given the company I'm in, and I totally accept that, but am i the only person here who thinks quantisation error is rounding error? I'll endeavour to be useful and use correct terminology.

What I meant originally was that by converting to a new sample rate, some of the new samples will be have a position in time in between old samples, and as such require some computational decision to be made on their value (interpolation or rounding or prediction or whatever system is in place). Whatever the process involved, the new values will have a lesser accuracy and the signal will have been degraded. And so keeping a constant sample rate may be better in some instances, be because a step of processing (and resultant degradation) can be avoided. Counter to that, I believe there are processes from which working at a higher sample rate does improve the sound:

Ill try to elaborate on the suggestion that granular time stretching can benefit from a higher sample rate. When time stretching, the sound is split up in to little grains of the ms magnitude. The larger the grain, the more noticeable the granulation when the tempo is lowered - this is because of the grains starting from a very similar point with reference to the source sound relative to total grain duration ie the onset of a new grain results in going back in time above a critically significant quantity (some may cite haas effect for thresholds of perception on sound events). The result is a weird "gl-gl-gl-gl" sound you can hear on a lot of 90s dance tracks, Ripgroove by Double99 ft Top Cat being a prime example where the man says "Brock out". To counter this undesirable sound processing artefact the grain size needs to be reduced so that each grain onset is less perceptible. At a higher resolution you can get smaller grain sizes, get less perception of the granulation, and some may say a better sound. I'll return with some audio to illustrate this.

Similarly, when sampling a sound and then playing it back at a different pitch there is benefit to processing at a higher sample rate. Samplers re-pitch sound, and this is almost always done by resampling. If the sample contains audio up until half the sample frequency then it follows that a 48kHz sample played back at the original pitch has an upper limit of 24khz. If this is played back two octaves lower then the upper limit of the sound spectrum is 6khz. If we had a sample of 192khz then it would have an upper limit of 96khz, and played back two octaves below would still have frequency content wih an upper limit of 28khz. The sound sampled at 48khz will sound dull when pitched down, the sound at 192 will maintain more sense of the original full frequency sound. I'll return with examples to illustrate this too.

Edit: tried to make it clearer

This post has been edited by Walter_Odington: Feb 20 2012, 05:10