[Scott Dorsey]

karle wrote:
Thanks for the answers. I would appreciate it if people contributed to the
problem instead raising questions withou trying to answer those.

You haven't defined the problem yet. The first stage is to define the
problem.

I need
opinions here.

Anyone can give you opinions. I think the best ice cream in the world is
plum flavour. Others may differ. Is this useful information? Probably
not. That's the thing about opinions.

Treat this as if it were your own experiment and share your approach or
theory.

What am I trying to prove or disprove with this theory? Do I care about
rate conversion artifacts, ultrasonic perception, converter artifacts, or
something else? What am I trying to prove the audibility of?

Until you frame the question better, you can't learn anything

Then please tell me how you would approach the question.

We don't know what the question IS.

My recommendation, FWIW, is to begin by asking what, specifically,
is different in a modern recording-and-reproduction chain between
different "sampling rates". Defining "sampling rates" is the first
hurdle. It's *not* the simple number that your post implies, (and
Scott has strongly implied the solution).

It doesn't have to be complicated. It could be tedious, but not complicated.
First, we use the same brand DA or AD converter for everything. Let's assume
that I could manage to get a manufacturer to match several units to an
acceptable level. (Yes, we need to define what would be an acceptable error
between the different units)

Then, we have to decide if the following methods are acceptable:

A-downsample a sound file to different rates and then upsample back the
resultiung sound files to play everything at the same rate.
B-downsample and then play every file at its respective downsampled rate.
C-or use X number of computers (one for each sample rate to be tested),
record with the same brand of AD, feed a live performance to every computer
at the same time and those computer would each record at a different
sampling rate. And then play the files back at their native sample rate.
D-Any method someone can think of around here.

The problem with C is, what do I record, and how do I record it to make sure
that the source will be "Accepted" by the audio community as a "Benchmark"
recording?

It depends on what you're trying to measure. These procedures measure
different things. You need to know what to measure before you make a
measurement.

Scott and Paul gave good ideas, I would like to have more. I need to hear
all the potential issues and questions. Then, I will be able to design a
proper experiment.

You need a hypothesis first.

I first need to isolate the "Sampling Rate" thing . Then, investigate the
different elements in the chain.

For example, how will a mastering engineer approach a song, which would have
to be delivered at different rates (CD, SACD, whatever)

Depends on a lot of factors. SACD isn't just a different sampling rate, it
is a totally different encoding system. Many mastering facilities are using
analogue chains anyway, so they can encode with any format at any rate off
the analogue feed on the fly.

Then when a end user listens to the product, what are the different elements
in the " live source-recording-mixing-mastering-delivery medium-end user
playback unit" chain.

There are lots of them.

But first I need to isolate the sampling rate issue and decide which
processing artifacts are acceptable in such a test.

What issue? We still don't know what issue you are talking about.
--scott


--
"C'est un Nagra. C'est suisse, et tres, tres precis."