In article , Rob
wrote:


That's really why I ask - I think. If there's more than one way to
downsample properly, I'm stuffed.

In principle 'downsampling' should be done 'properly' and will then lead to
a uniquely defined results - even if done in various algorithmic ways.

But in practice any downsampling or resampling can produce its own
(needless in theory) alterations that vary with the method used.

And in practice the vendors/creators may well add on other 'alterations'
they regard as an 'improvement' for each specific version. They may well
not admit this, or say how they did it. All part of the magic of
'mastering', etc. From the same people who brought us CDs that are clipped
and level-compressed to death because they "know" people "like" that. sic

But I have no idea what Naim have done. Might be able to tell once an
analysis has been carried out. I'd expect them to have avoided the insane
clipping, etc. But for all I know, they do other things because they judge
it gives 'better' results.

Slainte,

Jim

--
Please use the address on the audiomisc page if you wish to email me.
Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm
Armstrong Audio http://www.audiomisc.co.uk/Armstrong/armstrong.html
Audio Misc http://www.audiomisc.co.uk/index.html

In article , Michael Chare
wrote:
"Jim Lesurf" wrote in message
...


And as Arny has asked, can you say which particular files you (and
your daughter) compared? Might be best if I tried those if I can.


I used the pair of Jazz flac files and the pair of Classical flac files
on this page:

http://www.naimlabel.com/musicstore-test-files.aspx

The names are shown on the web page as:

Simple Psalm - Since Forever (Fred Simon) Beethoven - Symphony No.1 in C
(Iona Brown and the NCO)

However the tag data for the Fred Simon Jazz track shows it as: Simple
Psalm - Since Forever (Fred Simon)

Thanks. I'll put this on my list of things to do. Hope to do it soon as I
am quite interested to see how their files compare.

Slainte,

Jim

--
Please use the address on the audiomisc page if you wish to email me.
Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm
Armstrong Audio http://www.audiomisc.co.uk/Armstrong/armstrong.html
Audio Misc http://www.audiomisc.co.uk/index.html

"Arny Krueger" wrote in message
...
That wouldn't surprise me. Low clock rate processors are often high clock
rate processors with the lower clock speed simply enforced. Take a laser
and zap a link on the chip, or simpler yet take a bonding wire and jumper
two pads on the chip.

Or even simpler, get an HB pencil and draw a line to link the pads. I got
nearly a 50% increase in processor clock speed on one of my early Duron
CPUs.

On Fri, 9 Apr 2010 09:25:40 +0100, "David"
wrote:

That wouldn't surprise me. Low clock rate processors are often high clock
rate processors with the lower clock speed simply enforced. Take a laser
and zap a link on the chip, or simpler yet take a bonding wire and jumper
two pads on the chip.

Or even simpler, get an HB pencil and draw a line to link the pads. I got
nearly a 50% increase in processor clock speed on one of my early Duron
CPUs.

It's worth a try, and is sometimes well-documented as reliable. But
sometimes CPUs are jumpered at one speed because they failed testing
at a higher one.

"Laurence Payne" wrote in message
...
Or even simpler, get an HB pencil and draw a line to link the pads. I got
nearly a 50% increase in processor clock speed on one of my early Duron
CPUs.

It's worth a try, and is sometimes well-documented as reliable. But
sometimes CPUs are jumpered at one speed because they failed testing
at a higher one.

Indeed but I did my research and bought from a particular batch at a very
slightly higher price from an overclockers site. It lasted a good couple of
years before it was replaced, after which I took it upto and well beyond
it's limits.

"Rob" wrote in message
news
There are also various choices that could be made when
using one version to create the other, that then vary
the output. e.g. I understand that at one time Tony
Faulkner preferred a simplistic form of downsampling
that doesn't actually meet the sampling theorem. He
preferred the results, presumably because he thought it
made a 'change' that he liked. Or because it minimised
in-band filtering at the expense of aliasing.

That's really why I ask - I think. If there's more than
one way to downsample properly, I'm stuffed.

Not only are there many different downsamplers, with vastly different levels
of accuracy, but there is a time-honored process of simply starting out with
differently mastered recordings.

"Rob" wrote in message



The Java app looks fine, encoded the wav CD file, but
wouldn't encode the HD or mp3 files. I'm not sure why it
has to encode anything, and I tried it with the
'standardise' (gain and offset correction) on and off.

I haven't tested the Java app on a machine that can play HD files natively.

I've only tested it with 44/16 .wav files on a machine that can only play
44/16 .wav files.

You can convert MP3 files to .wav files a number of different ways that are
accurate. One is to load them into Audacity, and export them as .wav files.
MP3 files can also be accurately saved as .wav files using WinAmp. By
accurate, I mean that the resulting .wav files are represntative of what the
MP3 sounds like when played with a MP3 player.

Dirty little secret - all MP3 files are converted to .wav files during
playback, on-the-fly.

"Rob" wrote in message

On 08/04/2010 18:42, Arny Krueger wrote:
wrote in message


Rule number one is that when you do comparisons like
this, you take the high sample rate file and downsample
it yourself, which is easy to do with free software
that can downloaded from the web.

Why's that - are Naim not to be trusted?

Nothing specific about Naim, just that major producers
sometimes produce different technical renderings or
masterings of the same basic music work in different
formats. They may sound very similar, but never exactly
alike because they were slightly or significantly
different (it varies by work and format) prior to being
recorded in the various audio formats. It is common to
re-master musical works for distribution in a new
format.

I would have thought it was recorded in one, 'hig def'
format, and then downsampled.

Just wondered if there were any examples of distributors
meddling with the two versions.

I did a spectral comparison of the two "test-1" files. There were major,
multiple-dB, multi-octave-wide differences below 80 Hz and above 7 KHz..

Jim Lesurf wrote:
In article , Rob
wrote:


That's really why I ask - I think. If there's more than one way to
downsample properly, I'm stuffed.

In principle 'downsampling' should be done 'properly' and will then lead to
a uniquely defined results - even if done in various algorithmic ways.


That's not so.

Downsampling always involves a reduction in Nyquist frequency. It's
necessary therefore to filter the input to make sure frequencies above
this are sufficiently reduced. That filter can never be perfect, and
there will be various tradeoffs, involving extra loss of top-end,
in-band ripple and 'wrap-around' garbage from insufficient rejection of
higher-than-Nyquist signal. It's all down to what the person doing it
thought would be best (by some arbitrary criterion), and there is no
unique or 'right' answer.

--
Mike Scott (unet2 at [deletethis] scottsonline.org.uk)
Harlow Essex England

"Mike Scott"
wrote in message
Jim Lesurf wrote:
In article , Rob
wrote:


That's really why I ask - I think. If there's more than
one way to downsample properly, I'm stuffed.

In principle 'downsampling' should be done 'properly'
and will then lead to a uniquely defined results - even
if done in various algorithmic ways.

That's not so.

If you are defining "uniquely defined" as being some precise bit pattern,
then I am forced to agree.

Downsampling always involves a reduction in Nyquist
frequency. It's necessary therefore to filter the input
to make sure frequencies above this are sufficiently
reduced. That filter can never be perfect, and there will
be various tradeoffs, involving extra loss of top-end,
in-band ripple and 'wrap-around' garbage from
insufficient rejection of higher-than-Nyquist signal.

That would be one of those things that is true - theoretically, but from an
audibility standpoint, is not true.

The big difference is how sophisticated we have become in terms of designing
and implementing digital filters.

It's all down to what the person doing it thought would
be best (by some arbitrary criterion), and there is no
unique or 'right' answer.

If computational resources are highly estensible, it is possible to product
digital filters with very nearly ideal phase and amplitude characteristics.

The realm of perceptual studies have also improved - we now know that the
ideal phase characteristic for the required brick wall filter is neither
linear phase nor minimum phase. However, we base that knowlege on
experiments done at Nyquist frequencies well below 20 KHz, because sonically
innocious downsampling to 22 Khz has been routinely availble at a reasonble
cost for nearly a decade.