I'm getting more motivated to setup a proper ABX test, but that's somewhat challenging to do as I'd need to run identical recorders at seperate rates.
Maybe the easier way choose a band which releases in 24/96 (as FLAC download), and other formats like normal 16/44 for CD, and maybe lossy 320 Mp3. Buy one track in all that formats and do the test.
I could do that with the 24/96 files I've already recorded, which would be a better test as I know my recording situation had instrument created frequencies present >20kHz and know the setup, gear used, and all processing of the files. In that case, to eliminate as many other variables as possible, I would take a copy of the original 24/96 file, downsample it to 48 or 44.1 and upsample again to 96kHz. I'd then ABX test that file with the original. But that isn't the test I want to make as it would simply be duplicating the already tested and refuted
audibility of a standard 16-bit/44.1-kHz A/D/A loop inserted into a high-res playback chain. I accept the results of that double-blind study, which concluded the insertion is inaudible, so duplicating that doesn't interest me.
What I still wonder about is the
original capture conversion of 48 vs 96 kHz
using my particular equipment.. not because I think any ultrasonic information may be audible or that the high-quality resampling I can do on the computer may be audible, but because the recording equipment I'm using is modest and the ADCs in the recorders may perform better at 96 than 48 (or vice-versa) for a number of reasons. As I understand it, the main reason is that the nescesary anti-aliasing filtering can be less steep and easier to implement at 96kHz than 48kHz since the available transition band from 20KHz up to the half-sampling rate frequency is bigger. That's covered in the blog achalsey linked to above. Here's the illustrative image posted there-
Above: Whiteboard diagram from A Digital Media Primer for Geeks illustrating the transition band width available for a 48kHz ADC/DAC (left) and a 96kHz ADC/DAC (right).So it's really more a test of how well the modest ADCs in some of my equipment work. For example I wonder about the real-world performance of the analog low-pass filter before (or integrated in) the ADC chip used in the Tascam DR2d I use frequently (now selling for ~$100). I would wonder much less about high-quality ADC's like a Mytek, Lavry, etc. where far more resources are available for top quality analog filtering beore the ADC chip. It's somewhat ironic I suppose that it's likely to be the inexpensive equipment where recording at 96kHz may make a difference, whereas it's less likey to make a difference with higher end gear. But it makes sense when considering that the filtering must be done in the analog stage, and good quality high-slope analog filters are more difficult and costly to implement.
..48khz sampling already captures up to 24khz of high end, and no recorders I know of have analog capability above that, so what is 96khz capturing?
See the graphs above. The anti-aliasing lowpass filter must exclued all signal above half the sampling rate, so response tapers off as you approach that frequency. Practically, you can never really record all the way up to 24kHz at a 48KHz rate since the filter cannot be infinitely steep. The advantage of higher rates is easier to implement low pass filters with lower slopes, not capturing higher frequencies- one of the main points being made in the blog achalsey linked above.
