MQA alternative - open source
 

I, and some others, have concerns about the 'closed' nature of MQA and the
way it might give big companies a level of control over what makers and
users of hardware and music may do. So we have been considering the
possibility of an open and free alternative.

This has been discussed on the 'pink fish' forum and I've released a 'bit
freezer' program as a part of this for experiment purposes. If anyone is
interested I'd suggest looking at the relevant forum threads. But the basic
idea is to have a system that people may prefer and will cost less in terms
of money *and* in terms of limitation of your choice.

Jim

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How is bit freezing different from simply converting 24-bit FLAC/ALAC
files to 16 bits (with appropriate dithering)?

Julf

MQA alternative - open source
 

In article , Johan Helsingius
wrote:
How is bit freezing different from simply converting 24-bit FLAC/ALAC
files to 16 bits (with appropriate dithering)?

Just to check: Have you already looked at the webpages I wrote on this a
while ago? If not, they do give some of the background, etc, which may
help.

The bitfreezing is discussed and exampled on
http://www.audiomisc.co.uk/MQA/cool/bitfreezing.html

but these two pages preceeded it to look at MQA and get this into context

http://www.audiomisc.co.uk/MQA/origa...reAndBack.html
http://www.audiomisc.co.uk/MQA/bits/Stacking.html

So reading those first might be useful.

More specifically, bitfreezing lets you choose how many bits per sample to
'freeze'. Simply converting down to 16 bits nominally means losing 8 bits
per sample if you start from 24 bit. That may be too much or too little for
a given recording.

But with bitfreezing you can remove, say, 6 bits per sample. Hence you can
tailor this optimally for the amount of excess noise bits in the given
input material. i.e. remove only over-specified noise.

Indeed, in some cases you may find that high sample rate material has a
wideband noise floor distinctly *above* the 16 bit level. So might choose
to freeze some of the LS bits in a 16bit recording.

Beyond that, it comes down to being a case-by-case decision.

Jim

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On 05/09/2016 17:12, Jim Lesurf wrote:
The bitfreezing is discussed and exampled on
http://www.audiomisc.co.uk/MQA/cool/bitfreezing.html

I hadn't. Interesting!

FYI you have a typo:

However there are some special cases where can’t make use of FLAC
for what the MQA patents call ‘legacy’ reasons. So let’s now look
briefly at those...

You need "we" after "where".

I remain happy with my LPs digitised at 88k2/24 and downsampled to
44k1/16 after click reduction. As good as I think I'll get.

Andy

MQA alternative - open source
 

Just to check: Have you already looked at the webpages I wrote on this a
while ago?

Yes, I had looked at them, thanks!

More specifically, bitfreezing lets you choose how many bits per sample to
'freeze'. Simply converting down to 16 bits nominally means losing 8 bits
per sample if you start from 24 bit. That may be too much or too little for
a given recording.

OK, so 24-16 (possibly dithered) truncation is a special case of
bitfreezing - bitfreezing being a general N-M (possibly dithered)
truncation?

If so, then yes, it probably gives you all the benefits of MQA, without
any patent/licensing issues.

Julf

MQA alternative - open source
 

In article , Johan Helsingius
wrote:
Just to check: Have you already looked at the webpages I wrote on this
a while ago?

Yes, I had looked at them, thanks!

More specifically, bitfreezing lets you choose how many bits per
sample to 'freeze'. Simply converting down to 16 bits nominally means
losing 8 bits per sample if you start from 24 bit. That may be too
much or too little for a given recording.

OK, so 24-16 (possibly dithered) truncation is a special case of
bitfreezing - bitfreezing being a general N-M (possibly dithered)
truncation?

If so, then yes, it probably gives you all the benefits of MQA, without
any patent/licensing issues.

Yes. It also avoids the problems inherent in MQA 'Origami' which tends to
spray anharmonic distortions into the results. No need to even downsample.

So far as I can tell, it works as well, or better, and is free and open.
Anyone who wishs can use it - as is, or modified to suit.

FWIW I also have my doubts about the MQA 'doctrine' which takes it as
'axiomatic' that human hearing "beats Fourier" and that very high "time
resolution" is required. So I'm looking into this at present and may add
another page to deal with it.

My personal suspicion is that the main reason companies may adopt MQA is
because they think it might given them a new IPR control mechanism and
platform. Adnd let them re-sell the same old content to us all, yet again.
Money for old rope. For them, 'sound quality' may be the bait they can use
to hook users.

But my basic view is that no-one actually needs MQA if they simply want
lower stream rates and smaller files for 'high resolution' audio. There are
alternatives. Bitfreezing is one. Based simply on realising that a lot of
the 'content' of high rate files may simply be over-specified noise bits!
These pad the files/streams to no useful audible purpose.

Jim

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In article ,
Vir Campestris wrote:
On 05/09/2016 17:12, Jim Lesurf wrote:
The bitfreezing is discussed and exampled on
http://www.audiomisc.co.uk/MQA/cool/bitfreezing.html

I hadn't. Interesting!

FYI you have a typo:

However there are some special cases where cant make use of FLAC
for what the MQA patents call 'legacy reasons. So lets now look
briefly at those...

You need "we" after "where".

Noted. :-)

I remain happy with my LPs digitised at 88k2/24 and downsampled to
44k1/16 after click reduction. As good as I think I'll get.

I've tended to use 96k/24 and then leave it as such after declicking. Saves
the bother of then resampling, and pushes any DAC reconstruction problems
up well above 20kHz. As things stand I have plently of disc space. But I
realise this is wasteful.

Jim

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FWIW I also have my doubts about the MQA 'doctrine' which takes it as
'axiomatic' that human hearing "beats Fourier" and that very high "time
resolution" is required. So I'm looking into this at present and may add
another page to deal with it.

Great! Seems most of the audio press has swallowed the MQA doctrine
hook, line and sinker...

My personal suspicion is that the main reason companies may adopt MQA is
because they think it might given them a new IPR control mechanism and
platform. Adnd let them re-sell the same old content to us all, yet again.
Money for old rope. For them, 'sound quality' may be the bait they can use
to hook users.

I agree.

But my basic view is that no-one actually needs MQA if they simply want
lower stream rates and smaller files for 'high resolution' audio. There are
alternatives. Bitfreezing is one. Based simply on realising that a lot of
the 'content' of high rate files may simply be over-specified noise bits!
These pad the files/streams to no useful audible purpose.

+1. Too bad nobody has the guts to actually state that in the
mainstream audio press.

MQA alternative - open source
 

In article , Johan Helsingius
wrote:
But my basic view is that no-one actually needs MQA if they simply
want lower stream rates and smaller files for 'high resolution' audio.
There are alternatives. Bitfreezing is one. Based simply on realising
that a lot of the 'content' of high rate files may simply be
over-specified noise bits! These pad the files/streams to no useful
audible purpose.

+1. Too bad nobody has the guts to actually state that in the mainstream
audio press.

Sadly, they tend to focus on "Well [I think] it sounds good" without even
asking if it sounds the same as the source material.

FWIW I tend to approach this by writing about the technical aspects which
may tend to 'undercut' claims made by companies, etc.

e.g. My column in the current (October 2016) issue of 'Hi Fi News' explains
bitfreezing and how it can cut away the 'fat' of over-specified noise to
get smaller files/streams. But doesn't mention MQA at all. People can then
draw their own conclusions... :-)

One good outcome of this has been that it made me, and some others, realise
just how much such 'fat' there is in many high rez files/streams! In some
ways this reminds me of the retail practice of putting things into a much
bigger box than actually needed for the content. Makes the item more
impressive at point-of-sale and tends to squeeze competitors off the shelf.
"Wow! A big file. I'm getting a lot for my money!" :-)

However sometimes you may get the same meat as in the past, but put into a
bigger bun. ;-

Jim

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Sadly, they tend to focus on "Well [I think] it sounds good" without even
asking if it sounds the same as the source material.

Ah, yes, the "audio is art, not science" school of thought.

e.g. My column in the current (October 2016) issue of 'Hi Fi News' explains
bitfreezing and how it can cut away the 'fat' of over-specified noise to
get smaller files/streams. But doesn't mention MQA at all. People can then
draw their own conclusions... :-)

That column is what made me look up your longer write-up. Hi Fi News
is the only Hi-Fi magazine I still subscribe to, mainly because of
stuff like the spectrograms of "hi-res" recordings, but with almost
every issue lately I been having the feeling that they are on a
rapidly slipping slope to pure subjectivism - and considered cancelling
my subscription.

One good outcome of this has been that it made me, and some others, realise
just how much such 'fat' there is in many high rez files/streams! In some
ways this reminds me of the retail practice of putting things into a much
bigger box than actually needed for the content. Makes the item more
impressive at point-of-sale and tends to squeeze competitors off the shelf.
"Wow! A big file. I'm getting a lot for my money!" :-)

However sometimes you may get the same meat as in the past, but put into a
bigger bun. ;-

Indeed. But bigger numbers must be better, right? :)

Julf

MQA alternative - open source
 

On 06/09/2016 09:24, Jim Lesurf wrote:
I've tended to use 96k/24 and then leave it as such after declicking. Saves
the bother of then resampling, and pushes any DAC reconstruction problems
up well above 20kHz. As things stand I have plently of disc space. But I
realise this is wasteful.

I'm not sure I really believe in Shannon; while it's theoretically
possible to sample a 20kHz wave at 44.1, and reproduce it, it's damn
hard at anything over Fs/3.

Sadly this no longer matters :( I've reached that time of life where my
hifi is better than my ears. 44.1 is enough.

Andy

MQA alternative - open source
 

On 07-09-16 22:28, Vir Campestris wrote:

I'm not sure I really believe in Shannon; while it's theoretically
possible to sample a 20kHz wave at 44.1, and reproduce it, it's damn
hard at anything over Fs/3.

It is one thing to say "While Shannon is true in theory, it is tricky
to make very steep filters, so in practice you need a bit of margin",
and another to state "I am not sure I believe in Shannon". It is kind
of like saying "a falling object is affected by air resistance, so
I am not sure I believe in Newton's laws of gravity".

Fs/3 is a rather conservative measure with modern knowledge about
filter design.

MQA alternative - open source
 

In article , Johan Helsingius
wrote:
On 07-09-16 22:28, Vir Campestris wrote:

I'm not sure I really believe in Shannon; while it's theoretically
possible to sample a 20kHz wave at 44.1, and reproduce it, it's damn
hard at anything over Fs/3.

It is one thing to say "While Shannon is true in theory, it is tricky to
make very steep filters, so in practice you need a bit of margin", and
another to state "I am not sure I believe in Shannon". It is kind of
like saying "a falling object is affected by air resistance, so I am not
sure I believe in Newton's laws of gravity".

Fs/3 is a rather conservative measure with modern knowledge about filter
design.

Agreed. Even early generations of CD players could produce reasonable 20kHz
sineusoids and waveforms that had components up to that. And the better the
recording and reconstruction filtering, the closer you can get to the
nominal ideal.

Jim

--
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On Wed, 7 Sep 2016 21:28:30 +0100, Vir Campestris
wrote:

On 06/09/2016 09:24, Jim Lesurf wrote:
I've tended to use 96k/24 and then leave it as such after declicking. Saves
the bother of then resampling, and pushes any DAC reconstruction problems
up well above 20kHz. As things stand I have plently of disc space. But I
realise this is wasteful.

I'm not sure I really believe in Shannon; while it's theoretically
possible to sample a 20kHz wave at 44.1, and reproduce it, it's damn
hard at anything over Fs/3.

Sadly this no longer matters :( I've reached that time of life where my
hifi is better than my ears. 44.1 is enough.

Andy

You don't sample at 44.1kHz, you oversample; 16 x would be typical. In
other words about 700kHz. It is trivially easy to make a filter that
gives you adequate rejection at that frequency, so aliasing is not an
issue.

Having done that, you filter digitally with a multiple tap filter,
which is easy to design, and gives you all the rolloff you need
between 20kHz and 22.05kHz.

Finally you decimate down to the wanted 44.1kHz sampling rate, still
with a perfect signal.

Playback is the reverse of this process. You recreate the intermediate
values at your oversampling rate with a reconstruction filter, pop it
out through the DAC and you have the same trivial lowpass filter to
give you your final analogue output.

As for Shannon. His equation is correct - it has worked for years, and
is showing no sign of giving up yet.

d

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On 08/09/2016 13:11, Don Pearce wrote:
You don't sample at 44.1kHz, you oversample; 16 x would be typical. In
other words about 700kHz. It is trivially easy to make a filter that
gives you adequate rejection at that frequency, so aliasing is not an
issue.

I read Jim's web page. Turns out there's a lot of high frequency mush in
supposedly high quality signals. It didn't get filtered out, which is
why "bit freezing" works.

snip
As for Shannon. His equation is correct - it has worked for years, and
is showing no sign of giving up yet.

OK, I handle digital data and I don't really have a problem there. It's
just when someone throws a 22kHZ signal at a 44.1kHz sampler I can't
really see how you don't get some kind of aliasing. The signal and the
sample rate will beat, just like a moiré fringe.

A quick rummage in my tracks seems to show brick wall at 16kHz.

Andy

MQA alternative - open source
 

On Thu, 8 Sep 2016 22:00:54 +0100, Vir Campestris
wrote:

On 08/09/2016 13:11, Don Pearce wrote:
You don't sample at 44.1kHz, you oversample; 16 x would be typical. In
other words about 700kHz. It is trivially easy to make a filter that
gives you adequate rejection at that frequency, so aliasing is not an
issue.

I read Jim's web page. Turns out there's a lot of high frequency mush in
supposedly high quality signals. It didn't get filtered out, which is
why "bit freezing" works.

snip
As for Shannon. His equation is correct - it has worked for years, and
is showing no sign of giving up yet.

OK, I handle digital data and I don't really have a problem there. It's
just when someone throws a 22kHZ signal at a 44.1kHz sampler I can't
really see how you don't get some kind of aliasing. The signal and the
sample rate will beat, just like a moiré fringe.

A quick rummage in my tracks seems to show brick wall at 16kHz.

Andy

Brickwall at 16kHz usually means lossy compression - MP3 or somesuch.
Limiting the top end this way makes compressing the rest a much easier
job.

d

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In article , Vir
Campestris wrote:
On 08/09/2016 13:11, Don Pearce wrote:
You don't sample at 44.1kHz, you oversample; 16 x would be typical. In
other words about 700kHz. It is trivially easy to make a filter that
gives you adequate rejection at that frequency, so aliasing is not an
issue.

I read Jim's web page. Turns out there's a lot of high frequency mush in
supposedly high quality signals. It didn't get filtered out, which is
why "bit freezing" works.

The frequency distribution of the 'mush' is largely irrelevant. It is the
total level of it summed over all frequencies that matters. So this simply
isn't really relevant to any argument about if 44.1k sample rate is
sufficient to carry 20kHz components accurately.


That the 'mush' level has higher spectral density at HF just tells you
something about the recording/ADC systems employed. But in general, the
old maxim rules: The wider you open the window, the more muck blows in!

However as above all that really matters for bitfreezing is the total
level of the 'sea of noise'.

The above *does* however show that starting off from using systems like
DSD is actually bad news as it inherently has a poor SNR when considered
wideband. So it is actually a lousy basis for being converted and used
wideband IMHO. But DSD and low-bit ADCs are often used for making the
source material that may then end up being sold as high rez lpcm.


snip
As for Shannon. His equation is correct - it has worked for years, and
is showing no sign of giving up yet.

OK, I handle digital data and I don't really have a problem there. It's
just when someone throws a 22kHZ signal at a 44.1kHz sampler I can't
really see how you don't get some kind of aliasing. The signal and the
sample rate will beat, just like a moiré fringe.

You started off talking about 20k, now you have moved camp to 22k. But the
principle is the same.

Shannon and the relayed maths shows us quite clearly that a series of
samples at 44.1k can represent signal frequency compoents up to a 'tad
under' 22.05k.

The meaning of 'tad under' is defined by the duration of the recording
(number of samples). So a recording lasting, say, a few minutes, is quite
capable of conveying components of 22k.

A quick rummage in my tracks seems to show brick wall at 16kHz.

Which tells you about how the people creating those tracks recorded them.
It doesn't tell you about what 44.1k can do.

Jim

--
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In article ,
Vir Campestris wrote:
A quick rummage in my tracks seems to show brick wall at 16kHz.

Just for info, 16kHz was regarded by the BBC as the upper limit needed for
the highest quality radio transmissions.

On any true programme material, there was no useful information above this.

--
*I finally got my head together, now my body is falling apart.

Dave Plowman London SW
To e-mail, change noise into sound.

MQA alternative - open source
 

On Fri, 09 Sep 2016 13:32:50 +0100, "Dave Plowman (News)"
wrote:

In article ,
Vir Campestris wrote:
A quick rummage in my tracks seems to show brick wall at 16kHz.

Just for info, 16kHz was regarded by the BBC as the upper limit needed for
the highest quality radio transmissions.

On any true programme material, there was no useful information above this.

And of course they had to leave room for an analogue notch filter to
take out the 19kHz stereo pilot tone. But of course the distribution
standard to all the BBC FM transmitters is NICAM, which is sampled at
32kHz, so 16kHz is very much the upper limit of possibility.

d

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In article , Don Pearce
wrote:

And of course they had to leave room for an analogue notch filter to
take out the 19kHz stereo pilot tone. But of course the distribution
standard to all the BBC FM transmitters is NICAM, which is sampled at
32kHz, so 16kHz is very much the upper limit of possibility.

Yes. Although I think I recall reading that in the early stereo days when
only Wrotham was occasionally used for stereo on Radio 3 they had to fit
filters for the first time to fix some problems. Before that they hadn't
explicltly added them. Probably relied on the feeds not to have much above
about 15kHz and the - then analogue transmission lines to TX - not actually
carrying it!

Jim

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On 09/09/2016 15:41, Don Pearce wrote:
On Fri, 09 Sep 2016 13:32:50 +0100, "Dave Plowman (News)"
wrote:

In article ,
Vir Campestris wrote:
A quick rummage in my tracks seems to show brick wall at 16kHz.

Just for info, 16kHz was regarded by the BBC as the upper limit needed for
the highest quality radio transmissions.

On any true programme material, there was no useful information above this.

And of course they had to leave room for an analogue notch filter to
take out the 19kHz stereo pilot tone. But of course the distribution
standard to all the BBC FM transmitters is NICAM, which is sampled at
32kHz, so 16kHz is very much the upper limit of possibility.


There is audio above 16kHz, and something must happen to it. Even if it
can't be 'heard' in the conventional sense, can it be experienced in
some other way?

The reason for asking is a little more involved than the fact that I can
no longer hear a 15kHz signal :-)

--
Cheers, Rob

MQA alternative - open source
 

In article , RJH
wrote:

There is audio above 16kHz, and something must happen to it. Even if it
can't be 'heard' in the conventional sense, can it be experienced in
some other way?

There are arguments and some experiments which indicate the answer *may* be
'yes'. *However* this really depends on the details of the case. And a lot
of the experimental reports may be mis-interpreting or erronious for all
kinds of reasons. And may only apply to some people in some specific
situations - which may not be equivalent to what people do at home with a
hifi system.

It is, however, generally accepted that younger people can hear tones up to
around 20kHz or above. Varies from one person to another, and with time and
exposure.

So your question would need to be more specific for us to say more than
'perhaps'. :-)

FWIW One of the 'axioms' (as stated by the authors) of MQA is the idea that
we can 'resolve' timing of the order of 5 microseconds and that this is
then presumed to mean we have to include the components that cue this for
human hearing. Alas, beyond that the definitions of what they mean that
I've so far seen seem to drift into being vague or hand-waving.

I have my doubts. Some of which should be plain from the URLs given
earlier.But I'm currently looking in more detail at the 'evidence' for this
'axiom'. :-)

Jim

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On Fri, 09 Sep 2016 16:38:52 +0100, Jim Lesurf
wrote:

In article , Don Pearce
wrote:

And of course they had to leave room for an analogue notch filter to
take out the 19kHz stereo pilot tone. But of course the distribution
standard to all the BBC FM transmitters is NICAM, which is sampled at
32kHz, so 16kHz is very much the upper limit of possibility.

Yes. Although I think I recall reading that in the early stereo days when
only Wrotham was occasionally used for stereo on Radio 3 they had to fit
filters for the first time to fix some problems. Before that they hadn't
explicltly added them. Probably relied on the feeds not to have much above
about 15kHz and the - then analogue transmission lines to TX - not actually
carrying it!

Jim

In the early days they used GPO lines, and they had a dreadful
roll-off above about 4kHz. They fed the light programme medium wave
transmitter. The Beeb allowed for that roll-off in their top end
filtering to maintain the 8kHz channel width. Anyway, at some point
the GPO replaced those lines, and the new ones were flat to a few 10s
of kHz. The BBC didn't change their compensation network for a while,
and for several months we had proper Hi Fi medium wave. It was AM (but
then so is FM with a Foster-Seely discriminator) but the signal was
loud enough that Top Gear on Sunday afternoon was great (That is the
John Peel version, Not Jeremy Clarkson).

d

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On Fri, 9 Sep 2016 17:21:39 +0100, RJH wrote:

On 09/09/2016 15:41, Don Pearce wrote:
On Fri, 09 Sep 2016 13:32:50 +0100, "Dave Plowman (News)"
wrote:

In article ,
Vir Campestris wrote:
A quick rummage in my tracks seems to show brick wall at 16kHz.

Just for info, 16kHz was regarded by the BBC as the upper limit needed for
the highest quality radio transmissions.

On any true programme material, there was no useful information above this.

And of course they had to leave room for an analogue notch filter to
take out the 19kHz stereo pilot tone. But of course the distribution
standard to all the BBC FM transmitters is NICAM, which is sampled at
32kHz, so 16kHz is very much the upper limit of possibility.


There is audio above 16kHz, and something must happen to it. Even if it
can't be 'heard' in the conventional sense, can it be experienced in
some other way?

The reason for asking is a little more involved than the fact that I can
no longer hear a 15kHz signal :-)

The problem with audio that is not heard is that it is probably not
controlled either. A great deal of it will make itself known by
aliasing back down into the audible band in unpleasant ways. Much
better to chop it out early in the signal chain so it can do no harm.

Anyway - what is your reason for asking? I'm intrigued now.

d

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There is audio above 16kHz, and something must happen to it.

There is signal content from many sources way beyond 16k, but
I am not sure you can call it "audio" if it isn't audible :)

Even if it
can't be 'heard' in the conventional sense, can it be experienced in
some other way?

Yes, definitely. A lot of the studies quoted by "hi-res" proponents
all show that yes, you can hear signals above 22 kHz - especially
when non-linearities in the audio chain cause reflected intermodulation
products much lower down in the frequency spectrum. Just not sure it
is a good thing...

Julf

MQA alternative - open source
 

In article ,
Don Pearce wrote:
In the early days they used GPO lines, and they had a dreadful
roll-off above about 4kHz. They fed the light programme medium wave
transmitter. The Beeb allowed for that roll-off in their top end
filtering to maintain the 8kHz channel width. Anyway, at some point
the GPO replaced those lines, and the new ones were flat to a few 10s
of kHz. The BBC didn't change their compensation network for a while,
and for several months we had proper Hi Fi medium wave. It was AM (but
then so is FM with a Foster-Seely discriminator) but the signal was
loud enough that Top Gear on Sunday afternoon was great (That is the
John Peel version, Not Jeremy Clarkson).

I had a mate who worked in lines at the BEEB, and he reckoned the only
truly wideband AM transmission was the Brookman's Park 247 one, up until
they were all limited to 4.5 kHz or whatever. Something to do with both
that frequency being a unique BBC one, and the land line being
particularly good. Sadly, he's no longer around to get the full story from
him.

--
*The best cure for sea sickness, is to sit under a tree.

Dave Plowman London SW
To e-mail, change noise into sound.

MQA alternative - open source
 

On Sat, 10 Sep 2016 00:42:44 +0100, "Dave Plowman (News)"
wrote:

In article ,
Don Pearce wrote:
In the early days they used GPO lines, and they had a dreadful
roll-off above about 4kHz. They fed the light programme medium wave
transmitter. The Beeb allowed for that roll-off in their top end
filtering to maintain the 8kHz channel width. Anyway, at some point
the GPO replaced those lines, and the new ones were flat to a few 10s
of kHz. The BBC didn't change their compensation network for a while,
and for several months we had proper Hi Fi medium wave. It was AM (but
then so is FM with a Foster-Seely discriminator) but the signal was
loud enough that Top Gear on Sunday afternoon was great (That is the
John Peel version, Not Jeremy Clarkson).

I had a mate who worked in lines at the BEEB, and he reckoned the only
truly wideband AM transmission was the Brookman's Park 247 one, up until
they were all limited to 4.5 kHz or whatever. Something to do with both
that frequency being a unique BBC one, and the land line being
particularly good. Sadly, he's no longer around to get the full story from
him.

That would have been during the period I remember. And Brookman's Park
sounds like the right transmitter too. Thanks

d

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In article , Johan Helsingius
wrote:

Yes, definitely. A lot of the studies quoted by "hi-res" proponents all
show that yes, you can hear signals above 22 kHz - especially when
non-linearities in the audio chain cause reflected intermodulation
products much lower down in the frequency spectrum. Just not sure it is
a good thing...

That sums up the area to me. Various experimental tests have been said to
'show' we can hear the presence/absence of related components well above
20kHz. But in general those experiments are open to various mechanisms that
cloud the reasons for this.

I've been reading the 'Kunchur' papers on this recently. These are amongst
those quoted for the MQA claims about '5 microsecond temporal resolution'.
Yet the results could be explained in various ways that don't require the
listener to actually be able to hear anything above just *7* kHz, let alone
22kHz!

e.g. the results may stem from something as simple as the listener becoming
'trained' by the tests into hearing signal level changes of the order of
0.2dB, say, when the signal level is switched abruptly. And - as Johan
points out - there are other possible reasons for the results.

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

MQA alternative - open source
 

In article ,
Jim Lesurf wrote:
There are arguments and some experiments which indicate the answer *may*
be 'yes'. *However* this really depends on the details of the case. And
a lot of the experimental reports may be mis-interpreting or erronious
for all kinds of reasons. And may only apply to some people in some
specific situations - which may not be equivalent to what people do at
home with a hifi system.

It is, however, generally accepted that younger people can hear tones up
to around 20kHz or above. Varies from one person to another, and with
time and exposure.

Surely the argument is not what the upper limit of some may be, but if
there is any need to reproduce those frequencies in practice?

--
*PMS jokes aren't funny; period.*

Dave Plowman London SW
To e-mail, change noise into sound.

MQA alternative - open source
 

In article ,
Don Pearce wrote:
On Sat, 10 Sep 2016 00:42:44 +0100, "Dave Plowman (News)"
wrote:

In article ,
Don Pearce wrote:
In the early days they used GPO lines, and they had a dreadful
roll-off above about 4kHz. They fed the light programme medium wave
transmitter. The Beeb allowed for that roll-off in their top end
filtering to maintain the 8kHz channel width. Anyway, at some point
the GPO replaced those lines, and the new ones were flat to a few 10s
of kHz. The BBC didn't change their compensation network for a while,
and for several months we had proper Hi Fi medium wave. It was AM (but
then so is FM with a Foster-Seely discriminator) but the signal was
loud enough that Top Gear on Sunday afternoon was great (That is the
John Peel version, Not Jeremy Clarkson).

I had a mate who worked in lines at the BEEB, and he reckoned the only
truly wideband AM transmission was the Brookman's Park 247 one, up until
they were all limited to 4.5 kHz or whatever. Something to do with both
that frequency being a unique BBC one, and the land line being
particularly good. Sadly, he's no longer around to get the full story from
him.

That would have been during the period I remember. And Brookman's Park
sounds like the right transmitter too. Thanks

He also said - and I've got no reason to not believe him - was the wide
bandwidth land line was in part due to the original TV sound one to AP. In
the early (pre WW2) days of TV, it was advertised as having better sound
quality than radio.

Of course the other thing is that the vast majority of AM receivers
restricted the bandwidth themselves. Although Quad and some others did
offer wideband designs.

--
*If you're not part of the solution, you're part of the precipitate *

Dave Plowman London SW
To e-mail, change noise into sound.

MQA alternative - open source
 

On Sat, 10 Sep 2016 12:27:27 +0100, "Dave Plowman (News)"
wrote:

In article ,
Don Pearce wrote:
On Sat, 10 Sep 2016 00:42:44 +0100, "Dave Plowman (News)"
wrote:

In article ,
Don Pearce wrote:
In the early days they used GPO lines, and they had a dreadful
roll-off above about 4kHz. They fed the light programme medium wave
transmitter. The Beeb allowed for that roll-off in their top end
filtering to maintain the 8kHz channel width. Anyway, at some point
the GPO replaced those lines, and the new ones were flat to a few 10s
of kHz. The BBC didn't change their compensation network for a while,
and for several months we had proper Hi Fi medium wave. It was AM (but
then so is FM with a Foster-Seely discriminator) but the signal was
loud enough that Top Gear on Sunday afternoon was great (That is the
John Peel version, Not Jeremy Clarkson).

I had a mate who worked in lines at the BEEB, and he reckoned the only
truly wideband AM transmission was the Brookman's Park 247 one, up until
they were all limited to 4.5 kHz or whatever. Something to do with both
that frequency being a unique BBC one, and the land line being
particularly good. Sadly, he's no longer around to get the full story from
him.

That would have been during the period I remember. And Brookman's Park
sounds like the right transmitter too. Thanks

He also said - and I've got no reason to not believe him - was the wide
bandwidth land line was in part due to the original TV sound one to AP. In
the early (pre WW2) days of TV, it was advertised as having better sound
quality than radio.

Of course the other thing is that the vast majority of AM receivers
restricted the bandwidth themselves. Although Quad and some others did
offer wideband designs.

The AM receiver I was using at the time was a two foot square frame
aerial, a variable capacitor and a diode - that was plugged in to a
Phillips tape recorder. It had plenty of bandwidth...

d

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I've been reading the 'Kunchur' papers on this recently. These are amongst
those quoted for the MQA claims about '5 microsecond temporal resolution'.
Yet the results could be explained in various ways that don't require the
listener to actually be able to hear anything above just *7* kHz, let alone
22kHz!

Even if the '5 microseconds' as true, even 44.1/16 achieves a temporal
resolution of less than 1 *nanosecond*.

e.g. the results may stem from something as simple as the listener becoming
'trained' by the tests into hearing signal level changes of the order of
0.2dB, say, when the signal level is switched abruptly. And - as Johan
points out - there are other possible reasons for the results.

Indeed. But when will we actually see that stated in an audiophile
(or even mainstream hifi) magazine?

MQA alternative - open source
 

In article ,
Don Pearce wrote:
Of course the other thing is that the vast majority of AM receivers
restricted the bandwidth themselves. Although Quad and some others did
offer wideband designs.

The AM receiver I was using at the time was a two foot square frame
aerial, a variable capacitor and a diode - that was plugged in to a
Phillips tape recorder. It had plenty of bandwidth...

True. But the problem with crystal sets was getting rid of unwanted
trassmissions. Unless you are pretty close to a single frequency
transmitter.

--
*The problem with the gene pool is that there is no lifeguard *

Dave Plowman London SW
To e-mail, change noise into sound.

MQA alternative - open source
 

On Sat, 10 Sep 2016 13:07:28 +0100, "Dave Plowman (News)"
wrote:

In article ,
Don Pearce wrote:
Of course the other thing is that the vast majority of AM receivers
restricted the bandwidth themselves. Although Quad and some others did
offer wideband designs.

The AM receiver I was using at the time was a two foot square frame
aerial, a variable capacitor and a diode - that was plugged in to a
Phillips tape recorder. It had plenty of bandwidth...

True. But the problem with crystal sets was getting rid of unwanted
trassmissions. Unless you are pretty close to a single frequency
transmitter.

That was no problem back in the sixties. Living in London, and only
really using it during the day, there were no interfering signals. I
could really only find two stations. It was a different story after
dark, of course, but I had a little tranny for that (Radio Luxembourg,
naturally).

d

---
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In article , Johan Helsingius
wrote:
I've been reading the 'Kunchur' papers on this recently. These are
amongst those quoted for the MQA claims about '5 microsecond temporal
resolution'. Yet the results could be explained in various ways that
don't require the listener to actually be able to hear anything above
just *7* kHz, let alone 22kHz!

Even if the '5 microseconds' as true, even 44.1/16 achieves a temporal
resolution of less than 1 *nanosecond*.

Yes. One of the problems here is that the proponents tend to get a bit
vague and sweeping about what they mean by terms like 'resolution' etc in
this context.


e.g. the results may stem from something as simple as the listener
becoming 'trained' by the tests into hearing signal level changes of
the order of 0.2dB, say, when the signal level is switched abruptly.
And - as Johan points out - there are other possible reasons for the
results.

Indeed. But when will we actually see that stated in an audiophile (or
even mainstream hifi) magazine?

Maybe I'll say it sometime. At present I'm (slowly) going though this area
in detail, running down references, etc. I'll then write a detailed webpage
saying what I found, and then probably write about it for HFN.

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

MQA alternative - open source
 

In article , Dave Plowman (News)
wrote:

Surely the argument is not what the upper limit of some may be, but if
there is any need to reproduce those frequencies in practice?

Agreed. However the MQA argument slips into a vaguely defined (so far as I
can see thus far) argument that we need 'temporal resolution' without
necessarily accurately reproducing the actual HF precisely.

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

MQA alternative - open source
 

In article , Don Pearce
wrote:

That was no problem back in the sixties. Living in London, and only
really using it during the day, there were no interfering signals. I
could really only find two stations. It was a different story after
dark, of course, but I had a little tranny for that (Radio Luxembourg,
naturally).

One interesting aspect of having been trawling 1930s and 1940s publications
is that there was a time when AM could deliver far higher quality than it
can nowdays. The spectrum was much less crowded, and TX filtering could be
more lax.

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

MQA alternative - open source
 

In article ,
Don Pearce wrote:
True. But the problem with crystal sets was getting rid of unwanted
trassmissions. Unless you are pretty close to a single frequency
transmitter.

That was no problem back in the sixties. Living in London, and only
really using it during the day, there were no interfering signals. I
could really only find two stations. It was a different story after
dark, of course, but I had a little tranny for that (Radio Luxembourg,
naturally).

Interesting. At school in Aberdeen schools broadcasts were recorded on a
Ferrograph using a Truvox radio jack. And being Aberdeen, quite a long way
from other than BBC transmitters. Ie the AM one at RedMoss. And you could
just hear something else burbling away. Sounded like another BBC prog.

--
*When cheese gets its picture taken, what does it say? *

Dave Plowman London SW
To e-mail, change noise into sound.

MQA alternative - open source
 

On Sat, 10 Sep 2016 13:48:04 +0100, Jim Lesurf
wrote:

In article , Don Pearce
wrote:

That was no problem back in the sixties. Living in London, and only
really using it during the day, there were no interfering signals. I
could really only find two stations. It was a different story after
dark, of course, but I had a little tranny for that (Radio Luxembourg,
naturally).

One interesting aspect of having been trawling 1930s and 1940s publications
is that there was a time when AM could deliver far higher quality than it
can nowdays. The spectrum was much less crowded, and TX filtering could be
more lax.

Jim

It certainly could. And I remember the first trials of stereo AM too.

d

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In article ,
Jim Lesurf wrote:
In article , Don Pearce
wrote:

That was no problem back in the sixties. Living in London, and only
really using it during the day, there were no interfering signals. I
could really only find two stations. It was a different story after
dark, of course, but I had a little tranny for that (Radio Luxembourg,
naturally).

One interesting aspect of having been trawling 1930s and 1940s
publications is that there was a time when AM could deliver far higher
quality than it can nowdays. The spectrum was much less crowded, and TX
filtering could be more lax.

I've got a twin track recording somewhere (or maybe not) made in the late
'60s of a simulcast with R1 AM from BP and R2 FM from Wrotham, both from
Quad valve tuners. On a Revox A77. Not a lot of difference - although AM
was slightly noisier. But not obviously lacking in top as you'd expect.

--
*For every action, there is an equal and opposite criticism *

Dave Plowman London SW
To e-mail, change noise into sound.