"David Looser" wrote in
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"Arny Krueger" wrote in message
. ..
"David Looser" wrote in
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"Peter Larsen" wrote in message
...
David Looser wrote:

Not necessarily ideal, due to the possibility of
intersample peaks. It's advisable to
record with peak samples a dB or three shy of 0 dBFS,
unless you have accurate peak monitors
that show you what the *output* level is.

That's really a measurement problem. If you actually
know exactly what the "peak of peaks" is, that can be
just shy of 0dBFS. I agree in practice a 3dB or so
margin between *apparent* peak and 0dBFS is advisable.

What millenium are you living in guys, look at the
waveform, does it hit 0 dB FS?


How do you look at an analogue waveform?, we are talking
about setting the analogue level into the ADC.

You can, of course, do a transfer, look at the resulting
digital waveform, and then re-do it if the levels are
way off, but generally it's easier to get it more or
less correct the first time.

Most audio capture software has a real time display.

But it's not predictive. If it indicates clipping you can
only go back and start again.

Strictly speaking, nothing is predictive. But here is something that is
indicative:

The loudest trackability track on a test LP.

I set levels using a trackability test track. If the
cartridge is mistracking, tain't no need for that much
more headroom in the preamp!

Fine, but bearing in mind that there are large variations
in maximim level from LP to LP you will probably need to
apply amplification to the digital file on some (most?)
of the recordings later.

Something I happily do with 20:20 hindsight.

Peter Larsen wrote:
Eiron wrote:

Scratch filters such as that in Goldwave are excellent; there is no
reason not to use them on a whole album. I tried subtracting the
'cleaned' version from the 'raw' one and was left with just the
clicks and scratches on a background of perfect silence, thus showing
that the filter didn't remove any music.

Take a known good recording with high quality transients, say a chamber
music recording made with DPA 4006 mikes. Pass it through a declicker,
notice the number of reported "fixed clicks". End of story,

Is there any particular LP you had in mind for this test?

--
Eiron.

geoff wrote:

Why 96/16 rather than 44k1/24 ? I don't follow that logic.

Because the treble sounds cleaner with better inter-transient silence, and
that really matters with decayed audio, it gets less splatty.

One more reason for this practice was that the cpu load on the Celeron was
20 to 25 percent when recording at 96-16 with CE2k, I didn't want to push
the issue, life is too short for worrying about clicks. That machine is now
en route to become volkswagens and mobile phones, the only thing that was
broken was the CPU fan .... really really bad noise occasionally, but it had
a 12 year active life.

The
highest freq recorded on most LPs was around 15KHz, apart from clicks
of course...

Exactly. It is only an asumption - ie. I haven't asked on the adobe forums -
but said asumption is that the better the clicks are recorded the easier
they are to identify correctly. I don't care much about this, simply because
I don't see much relevance of doing anything but fix single click's on most
vinyl. I have however encountered one disk that was so decayed that it
needed treatment as if 78 rpm ... ie. center channel extract with suitably
modified settings. It was a mono record and the result was amazingly fine.

geoff


Kind regards

Peter Larsen

David Looser wrote:

Surely the screen image of the audio editor package is good enough.

But you don't get to see that until *after* you've made the
recording!!!

Bad choice of software then. But it doesn't matter much. Take a record you
don't like and lower the cartridge on it with the lift while recording. When
that records cleanly the level adjustment is done.

Correct transfer means that the loudest click is not clipped, you
only need to make that adjustment once.

Yes of course, did I say anything different?

You do seem to consider it to be "an issue".

David


Kind regards

Peter Larsen

David Looser wrote:

Fine, but bearing in mind that there are large variations in maximim
level from LP to LP you will probably need to apply amplification to
the digital file on some (most?) of the recordings later.

Of course you will, you have to record clicks that are 6 to 8 dB above the
strongest cut audio signal cleanly so that the are easy to identify for
click removal software. That too is the good practice reason for using
longer than 16 bit wordlength when recording.

David


Kind regards

Peter Larsen

Eiron wrote:

Take a known good recording with high quality transients, say a
chamber music recording made with DPA 4006 mikes. Pass it through a
declicker, notice the number of reported "fixed clicks". End of
story,

Is there any particular LP you had in mind for this test?

I did it with a chamber music recording made with known minimalist
equipment. I don't think you got the point btw. ... "known good" was used in
the sense of a recording that is known to be good in terms of freedom from
clicks. That would not be something that was played back mechanically.


Kind regards

Peter Larsen

In rec.audio.tech Peter Larsen wrote:
David Looser wrote:

Not necessarily ideal, due to the possibility of intersample peaks. It's
advisable to
record with peak samples a dB or three shy of 0 dBFS, unless you have
accurate peak monitors
that show you what the *output* level is.

That's really a measurement problem. If you actually know exactly
what the "peak of peaks" is, that can be just shy of 0dBFS. I agree
in practice a 3dB or so margin between *apparent* peak and 0dBFS is
advisable.

What millenium are you living in guys, look at the waveform, does it hit 0
dB FS?

Look at the waveform when? After it's recorded? By then it's too late.

Are you familiar with the concept of intersample peaks? It's mainly a
monitoring problem. Read more he

http://www.cadenzarecording.com/pape...distortion.pdf


___
-S
"As human beings, we understand the world through simile, analogy,
metaphor, narrative and, sometimes, claymation." - B. Mason

In rec.audio.tech David Looser wrote:
"geoff" wrote in message
around 15kHz, the click spectrum would go much higher than that so
preserving the fast risetime of the clicks would be of value to automatic
click detection software. On the other hand the S/N ratio of no better than
70dB requires only a 13-bit ADC, leaving a margin of 3 bits (18dB) for click
headroom/ post digitising amplification even when using a 16-bit converter.
And it doesn't matter if high-amplitude clicks are clipped, as long as the
rise-time is preserved.

I would be astonished if anyone could tell the difference between an
original 24-bit digitisation and a 16-bit one when digitising vinyl.

You must not visit 'audiophile' forums much. Such claims are routine
-- as is the claim that neither digitization will sound as good as the
vinyl. They';re never backed up with anything like hard evidence, of course
but they're not at all uncommon. So if you ever feel like being thus astonished,
or perhaps depressed, visit audioasylum.com or stevehoffman.tv



___
-S
"As human beings, we understand the world through simile, analogy,
metaphor, narrative and, sometimes, claymation." - B. Mason

In rec.audio.tech Arny Krueger wrote:
"David Looser" wrote in
message
"Peter Larsen" wrote in message
...
David Looser wrote:

Not necessarily ideal, due to the possibility of
intersample peaks. It's advisable to
record with peak samples a dB or three shy of 0 dBFS,
unless you have accurate peak monitors
that show you what the *output* level is.

That's really a measurement problem. If you actually
know exactly what the "peak of peaks" is, that can be
just shy of 0dBFS. I agree in practice a 3dB or so
margin between *apparent* peak and 0dBFS is advisable.

What millenium are you living in guys, look at the
waveform, does it hit 0 dB FS?


How do you look at an analogue waveform?, we are talking
about setting the analogue level into the ADC.

You can, of course, do a transfer, look at the resulting
digital waveform, and then re-do it if the levels are way
off, but generally it's easier to get it more or less
correct the first time.

Most audio capture software has a real time display.

Yes, but is it accurate real time display, modeling
a reconstruction filter?

I don't know that those are so common. HEre's one:

http://www.secaudio.ch/side122.html

scroll down to the 'TL Mastermeter'



___
-S
"As human beings, we understand the world through simile, analogy,
metaphor, narrative and, sometimes, claymation." - B. Mason

In rec.audio.tech Peter Larsen wrote:
geoff wrote:

Why 96/16 rather than 44k1/24 ? I don't follow that logic.

Because the treble sounds cleaner with better inter-transient silence, and
that really matters with decayed audio, it gets less splatty.

Do tell. Your proof of this is....?




___
-S
"As human beings, we understand the world through simile, analogy,
metaphor, narrative and, sometimes, claymation." - B. Mason