Digitizing Vinyl. Help!

[Adrian]

Time has come to digitize my Vinyl collection. Having successfully
copied tape material to CD, I thought this would be easy!

My equipment is a Denon DP-35F Turntable with a Denon DL-300
Cartridge, a New ART "USB Phono Plus" interface and a Dell Latitude
D810 Notebook equipped with RIP Vinyl.

This past weekend I copied three albums. The signal is clean but not
strong. I have the gain on the USB Phono turned to the max. But, the
meter in RIP barely rises about the quarter way mark. If I look at
the signal in Audacity it is pretty "thin". I could comfortably use
at least 3dB more.

Do you have any thoughts on what is "wrong"? And, what can I do about
it? You guys have given me great advice in the past.
Many thanks

Adrian

[Todd H.]

Adrian > writes:

> Time has come to digitize my Vinyl collection. Having successfully
> copied tape material to CD, I thought this would be easy!
>
> My equipment is a Denon DP-35F Turntable with a Denon DL-300
> Cartridge, a New ART "USB Phono Plus" interface and a Dell Latitude
> D810 Notebook equipped with RIP Vinyl.

That's a moving coil cartridge. They're generally lower output that
moving magnet cartridges.

If I had to guess, without doing too much research for you, I'd wager
that the USB PHono Plus was designed with MMC's in mind, and not the
small output MCC that you have.


--
/"\ ASCII Ribbon Campaign | Todd H
\ / | http://www.toddh.net/
X Promoting good netiquette |
/ \ http://www.toddh.net/netiquette/ | http://myspace.com/bmiawmb

[Geoff]

Adrian wrote:
> Time has come to digitize my Vinyl collection. Having successfully
> copied tape material to CD, I thought this would be easy!
>
> My equipment is a Denon DP-35F Turntable with a Denon DL-300
> Cartridge, a New ART "USB Phono Plus" interface and a Dell Latitude
> D810 Notebook equipped with RIP Vinyl.
>
> This past weekend I copied three albums. The signal is clean but not
> strong. I have the gain on the USB Phono turned to the max. But, the
> meter in RIP barely rises about the quarter way mark. If I look at
> the signal in Audacity it is pretty "thin". I could comfortably use
> at least 3dB more.
>
> Do you have any thoughts on what is "wrong"? And, what can I do about
> it? You guys have given me great advice in the past.
> Many thanks

You need the transformer that (maybe) comes with your cartridge, or a phono
preamp with a specialised MC (Moving Coil) input. Your MC cartridge has a
very low output compared the the more usual MM (Moving Magnet) cartridges -
I doubt a USB phono interface would sport this.


geoff.

[Laurence Payne]

On Mon, 10 Dec 2007 12:47:33 -0800 (PST), Adrian
> wrote:

>Time has come to digitize my Vinyl collection. Having successfully
>copied tape material to CD, I thought this would be easy!
>
>My equipment is a Denon DP-35F Turntable with a Denon DL-300
>Cartridge, a New ART "USB Phono Plus" interface and a Dell Latitude
>D810 Notebook equipped with RIP Vinyl.
>
>This past weekend I copied three albums. The signal is clean but not
>strong. I have the gain on the USB Phono turned to the max. But, the
>meter in RIP barely rises about the quarter way mark. If I look at
>the signal in Audacity it is pretty "thin". I could comfortably use
>at least 3dB more.


Do you still have the amplifier you used to use when you played vinyl
all the time? It will have an input stage better suited to your
low-output cartridge. Use that as a preamp, feeding Tape Out into
Line In on the ART, switching out the RIAA stage.

[Dave W.]

On 10 Dec, 20:47, Adrian > wrote:
> Time has come to digitize my Vinyl collection. Having successfully
> copied tape material to CD, I thought this would be easy!
>
> My equipment is a Denon DP-35F Turntable with a Denon DL-300
> Cartridge, a New ART "USB Phono Plus" interface and a Dell Latitude
> D810 Notebook equipped with RIP Vinyl.
>
> This past weekend I copied three albums. The signal is clean but not
> strong. I have the gain on the USB Phono turned to the max. But, the
> meter in RIP barely rises about the quarter way mark. If I look at
> the signal in Audacity it is pretty "thin". I could comfortably use
> at least 3dB more.
>
> Do you have any thoughts on what is "wrong"? And, what can I do about
> it? You guys have given me great advice in the past.
> Many thanks
>
> Adrian

If, as you say, the signal is clean, then as long as you have it
digitised there is no problem. Before saving it to any 'lossy'
compression method, simply amplify it in Audacity. This is a
mathematical operation, and the 'clean signal' will be end up being as
loud as you want it to be.

Jack.

[Geoff]

Dave W. wrote:
>
> If, as you say, the signal is clean, then as long as you have it
> digitised there is no problem. Before saving it to any 'lossy'
> compression method, simply amplify it in Audacity. This is a
> mathematical operation, and the 'clean signal' will be end up being as
> loud as you want it to be.


However as you amplify it, yo will also bring up the quantisation noise. It
is best to optimise your recording level first.

The effect can be reduced by ensuring you are recording at 24 bits
resolution.

geoff

[Laurence Payne]

On Tue, 11 Dec 2007 11:30:45 +1300, "geoff" >
wrote:

>However as you amplify it, yo will also bring up the quantisation noise. It
>is best to optimise your recording level first.
>
>The effect can be reduced by ensuring you are recording at 24 bits
>resolution.

Waste of space really, off vinyl. Or off any other real-world source
where levels are under control.

[David Looser]

"Laurence Payne" <NOSPAMlpayne1ATdsl.pipex.com> wrote in message
...
> On Tue, 11 Dec 2007 11:30:45 +1300, "geoff" >
> wrote:
>
>>However as you amplify it, yo will also bring up the quantisation noise.
>>It
>>is best to optimise your recording level first.
>>
>>The effect can be reduced by ensuring you are recording at 24 bits
>>resolution.
>
> Waste of space really, off vinyl. Or off any other real-world source
> where levels are under control.

I agree. The background noise on the vinyl will dither the quantisation
quite effectively. Whilst ideally one would record with the peak signal just
failing to hit 0dBFS, in practice even with a 16 bit ADC when digitising
vinyl anything up to around 12dB of gain could be retrospectively applied to
the digital signal without audible quantisation noise becoming apparent.

David.

[Arny Krueger]

"geoff" > wrote in message


> Dave W. wrote:

>> If, as you say, the signal is clean, then as long as you
>> have it digitised there is no problem.

But what is digitized? Is it really a pure signal, or is there a noise floor
that can intrude on the faint signal coming into the RIAA preamp? A MC
preamp or transformer raises the signal above the noise floor of the MM
preamp. Since there is currently no MC preamp or transformer, the noise
floor of the MM preamp is probably the weakest link.

>> Before saving it
>> to any 'lossy' compression method, simply amplify it in
>> Audacity. This is a mathematical operation, and the
>> 'clean signal' will be end up being as loud as you want
>> it to be.

Problem here is that there's always an analog domain noise floor, if only in
the existing analog-to-digital converter. In this case I expect that the MM
RIAA preamp is the weakest link. I base this on many experiences with them.
Even with 16 bit converters, a MM RIAA preamp is the weakest link.

Let me give a real world numerical example. If I adjust a good MM RIAA
preamp so that the preamp clips at a slightly higher level than a
high-trackability cartridge mistracks on a test record, the needle-up noise
floor will be 70+/- dB down. Since the noise floor of a good 16 bit
converter is more like 96 dB down, the weakest link is the MM RIAA preamp.

> However as you amplify it, yo will also bring up the
> quantisation noise. It is best to optimise your recording
> level first.

Agreed. And that's why there are such things as MC pre-preamps and
transformers.

> The effect can be reduced by ensuring you are recording
> at 24 bits resolution.

A 24 bit converter does no more good - it just gives a higher resolution
rendition of the noise in the MM RIAA preamp.

[Richard Crowley]

"Dave W." wrote ...
> Adrian wrote:
>> This past weekend I copied three albums. The signal is
>> clean but not strong. I have the gain on the USB Phono
>> turned to the max.

> If, as you say, the signal is clean, then as long as you have
> it digitised there is no problem.

Lets review the bidding....

* Low-output MC cartridge feeding an inexpensive RIAA
phono preamp designed for MC.
* Gain on the preamp "turned to the max".
* Signal is "clean but not strong"

Therefore, by definition, the captured signal is NOT
"clean" after amplifying it (plus the noise) to the
nominal level.

Of course, Adrian could decide that it is good enough
for his purposes, and that is fine. But conventional
wisdom would suggest that the solution might be...
1) Use a conventional MM cartridge
2) Use a step-up transformer or pre-pre-amp for MC
3) Use a preamp designed for MC.

[Arny Krueger]

"Richard Crowley" > wrote in message

> "Dave W." wrote ...
>> Adrian wrote:
>>> This past weekend I copied three albums. The signal is
>>> clean but not strong. I have the gain on the USB Phono
>>> turned to the max.
>
>> If, as you say, the signal is clean, then as long as you
>> have it digitised there is no problem.
>
> Lets review the bidding....
>
> * Low-output MC cartridge feeding an inexpensive RIAA
> phono preamp designed for MC.
> * Gain on the preamp "turned to the max".
> * Signal is "clean but not strong"

> Therefore, by definition, the captured signal is NOT
> "clean" after amplifying it (plus the noise) to the
> nominal level.

Agreed.

> Of course, Adrian could decide that it is good enough
> for his purposes, and that is fine. But conventional
> wisdom would suggest that the solution might be...
> 1) Use a conventional MM cartridge
> 2) Use a step-up transformer or pre-pre-amp for MC
> 3) Use a preamp designed for MC.

I'd vote for solution number 1, more specificially this cartridge:

http://www.amazon.com/Shure-M97xE-High-Performance-Magnetic-Cartridge/dp/B00006I5SB/ref=pd_bbs_sr_1?ie=UTF8&s=electronics&qid=1197398346&sr=8-1

Cheapest way out and solves more problems.

[Adrian]

On Dec 10, 1:01 pm, (Todd H.) wrote:
> Adrian > writes:
> > Time has come to digitize my Vinyl collection. Having successfully
> > copied tape material to CD, I thought this would be easy!
>
> > My equipment is aDenonDP-35F Turntable with aDenonDL-300
> > Cartridge, a New ART "USBPhono Plus" interface and a Dell Latitude
> > D810 Notebook equipped with RIP Vinyl.
>
> That's a moving coil cartridge. They're generally lower output that
> moving magnet cartridges.
>
> If I had to guess, without doing too much research for you, I'd wager
> that theUSBPHono Plus was designed with MMC's in mind, and not the
> small output MCC that you have.
>
> --
That seems to be the consensus. Does anyone have a recomentation for
a more appropriate cartridge.

Thanks

Adrian

[Adrian]

On Dec 10, 1:21 pm, "geoff" > wrote:
> Adrian wrote:
> > Time has come to digitize my Vinyl collection. Having successfully
> > copied tape material to CD, I thought this would be easy!
>
> > My equipment is aDenonDP-35F Turntable with aDenonDL-300
> > Cartridge, a New ART "USBPhono Plus" interface and a Dell Latitude
> > D810 Notebook equipped with RIP Vinyl.
>
> > This past weekend I copied three albums. The signal is clean but not
> > strong. I have the gain on theUSBPhono turned to the max. But, the
> > meter in RIP barely rises about the quarter way mark. If I look at
> > the signal in Audacity it is pretty "thin". I could comfortably use
> > at least 3dB more.
>
> > Do you have any thoughts on what is "wrong"? And, what can I do about
> > it? You guys have given me great advice in the past.
> > Many thanks
>
> You need the transformer that (maybe) comes with your cartridge, or a phono
> preamp with a specialised MC (Moving Coil) input. Your MC cartridge has a
> very low output compared the the more usual MM (Moving Magnet) cartridges -
> I doubt aUSBphono interface would sport this.
>
> geoff.

So, time to find a Moving Magnet cartridge.

Thanks for helping.

[Adrian]

On Dec 10, 2:05 pm, Laurence Payne <NOSPAMlpayne1ATdsl.pipex.com>
wrote:
> On Mon, 10 Dec 2007 12:47:33 -0800 (PST), Adrian
>
> > wrote:
> >Time has come to digitize my Vinyl collection. Having successfully
> >copied tape material to CD, I thought this would be easy!
>
> >My equipment is aDenonDP-35F Turntable with aDenonDL-300
> >Cartridge, a New ART "USBPhono Plus" interface and a Dell Latitude
> >D810 Notebook equipped with RIP Vinyl.
>
> >This past weekend I copied three albums. The signal is clean but not
> >strong. I have the gain on theUSBPhono turned to the max. But, the
> >meter in RIP barely rises about the quarter way mark. If I look at
> >the signal in Audacity it is pretty "thin". I could comfortably use
> >at least 3dB more.
>
> Do you still have the amplifier you used to use when you played vinyl
> all the time? It will have an input stage better suited to your
> low-output cartridge. Use that as a preamp, feeding Tape Out into
> Line In on the ART, switching out the RIAA stage.

Yes I do have the receiver. However, it will awkward to set it up
just now. Since I see this as a project that will go on for some time
I amthinking about a new cartridge.

Adrian

[Adrian]

On Dec 11, 9:06 am, "Arny Krueger" > wrote:
> "geoff" > wrote in message
>
>
>
> > Dave W. wrote:
> >> If, as you say, the signal is clean, then as long as you
> >> have it digitised there is no problem.
>
> But what is digitized? Is it really a pure signal, or is there a noise floor
> that can intrude on the faint signal coming into the RIAA preamp? A MC
> preamp or transformer raises the signal above the noise floor of the MM
> preamp. Since there is currently no MC preamp or transformer, the noise
> floor of the MM preamp is probably the weakest link.
>
> >> Before saving it
> >> to any 'lossy' compression method, simply amplify it in
> >> Audacity. This is a mathematical operation, and the
> >> 'clean signal' will be end up being as loud as you want
> >> it to be.
>
> Problem here is that there's always an analog domain noise floor, if only in
> the existing analog-to-digital converter. In this case I expect that the MM
> RIAA preamp is the weakest link. I base this on many experiences with them.
> Even with 16 bit converters, a MM RIAA preamp is the weakest link.
>
> Let me give a real world numerical example. If I adjust a good MM RIAA
> preamp so that the preamp clips at a slightly higher level than a
> high-trackability cartridge mistracks on a test record, the needle-up noise
> floor will be 70+/- dB down. Since the noise floor of a good 16 bit
> converter is more like 96 dB down, the weakest link is the MM RIAA preamp.
>
> > However as you amplify it, yo will also bring up the
> > quantisation noise. It is best to optimise your recording
> > level first.
>
> Agreed. And that's why there are such things as MC pre-preamps and
> transformers.
>
> > The effect can be reduced by ensuring you are recording
> > at 24 bits resolution.
>
> A 24 bit converter does no more good - it just gives a higher resolution
> rendition of the noise in the MM RIAA preamp.

Excellent help. Thanks Arny. I am going to look for an appropriate
cartridge.

Any suggestions? :-)

Thanks

Adrian

[Adrian]

On Dec 11, 10:36 am, "Richard Crowley" > wrote:
> "Dave W." wrote ...
>
> > Adrian wrote:
> >> This past weekend I copied three albums. The signal is
> >> clean but not strong. I have the gain on theUSBPhono
> >> turned to the max.
> > If, as you say, the signal is clean, then as long as you have
> > it digitised there is no problem.
>
> Lets review the bidding....
>
> * Low-output MC cartridge feeding an inexpensive RIAA
> phono preamp designed for MC.
> * Gain on the preamp "turned to the max".
> * Signal is "clean but not strong"
>
> Therefore, by definition, the captured signal is NOT
> "clean" after amplifying it (plus the noise) to the
> nominal level.
>
> Of course, Adrian could decide that it is good enough
> for his purposes, and that is fine. But conventional
> wisdom would suggest that the solution might be...
> 1) Use a conventional MM cartridge
> 2) Use a step-up transformer or pre-pre-amp for MC
> 3) Use a preamp designed for MC.

Points well taken. I want to do this right. Option one is my currect
preference. I believe in KISS, keep it simple, stupid!

Adrian

[Adrian]

On Dec 11, 10:37 am, "Arny Krueger" > wrote:
> "Richard Crowley" > wrote in message
>
>
>
>
>
>
>
> > "Dave W." wrote ...
> >> Adrian wrote:
> >>> This past weekend I copied three albums. The signal is
> >>> clean but not strong. I have the gain on theUSBPhono
> >>> turned to the max.
>
> >> If, as you say, the signal is clean, then as long as you
> >> have it digitised there is no problem.
>
> > Lets review the bidding....
>
> > * Low-output MC cartridge feeding an inexpensive RIAA
> > phono preamp designed for MC.
> > * Gain on the preamp "turned to the max".
> > * Signal is "clean but not strong"
> > Therefore, by definition, the captured signal is NOT
> > "clean" after amplifying it (plus the noise) to the
> > nominal level.
>
> Agreed.
>
> > Of course, Adrian could decide that it is good enough
> > for his purposes, and that is fine. But conventional
> > wisdom would suggest that the solution might be...
> > 1) Use a conventional MM cartridge
> > 2) Use a step-up transformer or pre-pre-amp for MC
> > 3) Use a preamp designed for MC.
>
> I'd vote for solution number 1, more specificially this cartridge:
>
> http://www.amazon.com/Shure-M97xE-High-Performance-Magnetic-Cartridge...
>
> Cheapest way out and solves more problems

Thanks. Under serious consideration.

[Steven Sullivan]

In rec.audio.tech David Looser > wrote:
> "Laurence Payne" <NOSPAMlpayne1ATdsl.pipex.com> wrote in message
> ...
> > On Tue, 11 Dec 2007 11:30:45 +1300, "geoff" >
> > wrote:
> >
> >>However as you amplify it, yo will also bring up the quantisation noise.
> >>It
> >>is best to optimise your recording level first.
> >>
> >>The effect can be reduced by ensuring you are recording at 24 bits
> >>resolution.
> >
> > Waste of space really, off vinyl. Or off any other real-world source
> > where levels are under control.

> I agree. The background noise on the vinyl will dither the quantisation
> quite effectively. Whilst ideally one would record with the peak signal just
> failing to hit 0dBFS,

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.


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

[Geoff]

Laurence Payne wrote:
> On Tue, 11 Dec 2007 11:30:45 +1300, "geoff" >
> wrote:
>
>> However as you amplify it, yo will also bring up the quantisation
>> noise. It is best to optimise your recording level first.
>>
>> The effect can be reduced by ensuring you are recording at 24 bits
>> resolution.
>
> Waste of space really, off vinyl. Or off any other real-world source
> where levels are under control.

If he is recording at a lower than optimum level then it is well worth using
the extra bits, to reduce the detrimental effect of bringing up the level
once digitised.

It is a long time since amounts of data like that have been significant.
And once on CD he can delete the computer data anyway. Sheesh.

geoff

[Geoff]

Adrian wrote:

> So, time to find a Moving Magnet cartridge.
>
> Thanks for helping.

MC cartridges often offer benefits over MC. Why not buy a phono preamp with
a MC/MM switch ? Probably a cheaper option.

geoff

[jakdedert]

Adrian wrote:
> On Dec 10, 2:05 pm, Laurence Payne <NOSPAMlpayne1ATdsl.pipex.com>
> wrote:
>> On Mon, 10 Dec 2007 12:47:33 -0800 (PST), Adrian
>>
>> > wrote:
>>> Time has come to digitize my Vinyl collection. Having successfully
>>> copied tape material to CD, I thought this would be easy!
>>> My equipment is aDenonDP-35F Turntable with aDenonDL-300
>>> Cartridge, a New ART "USBPhono Plus" interface and a Dell Latitude
>>> D810 Notebook equipped with RIP Vinyl.
>>> This past weekend I copied three albums. The signal is clean but not
>>> strong. I have the gain on theUSBPhono turned to the max. But, the
>>> meter in RIP barely rises about the quarter way mark. If I look at
>>> the signal in Audacity it is pretty "thin". I could comfortably use
>>> at least 3dB more.
>> Do you still have the amplifier you used to use when you played vinyl
>> all the time? It will have an input stage better suited to your
>> low-output cartridge. Use that as a preamp, feeding Tape Out into
>> Line In on the ART, switching out the RIAA stage.
>
> Yes I do have the receiver. However, it will awkward to set it up
> just now. Since I see this as a project that will go on for some time
> I amthinking about a new cartridge.
>
> Adrian
You don't have to 'set it up' per se...just plug it in, in the vicinity
of the turntable and computer...ie you don't need speakers, antenna or
anything else. Put it under your monitor....

Space might be an issue, but you're just setting it up as a preamp for
your computer...one cord for power, one double RCA to the preamp, and
start ripping while you're figuring out your cartridge issues.

jak

[jakdedert]

Adrian wrote:
> On Dec 11, 10:37 am, "Arny Krueger" > wrote:
>> "Richard Crowley" > wrote in message
>>
>>
>>
>>
>>
>>
>>
>>> "Dave W." wrote ...
>>>> Adrian wrote:
>>>>> This past weekend I copied three albums. The signal is
>>>>> clean but not strong. I have the gain on theUSBPhono
>>>>> turned to the max.
>>>> If, as you say, the signal is clean, then as long as you
>>>> have it digitised there is no problem.
>>> Lets review the bidding....
>>> * Low-output MC cartridge feeding an inexpensive RIAA
>>> phono preamp designed for MC.
>>> * Gain on the preamp "turned to the max".
>>> * Signal is "clean but not strong"
>>> Therefore, by definition, the captured signal is NOT
>>> "clean" after amplifying it (plus the noise) to the
>>> nominal level.
>> Agreed.
>>
>>> Of course, Adrian could decide that it is good enough
>>> for his purposes, and that is fine. But conventional
>>> wisdom would suggest that the solution might be...
>>> 1) Use a conventional MM cartridge
>>> 2) Use a step-up transformer or pre-pre-amp for MC
>>> 3) Use a preamp designed for MC.
>> I'd vote for solution number 1, more specificially this cartridge:
>>
>> http://www.amazon.com/Shure-M97xE-High-Performance-Magnetic-Cartridge...
>>
>> Cheapest way out and solves more problems

Nope, cheapest is to simply use your receiver for a preamp...nothing
else to buy.

jak
>
> Thanks. Under serious consideration.

[David Looser]

"Steven Sullivan" > wrote in message
...
> In rec.audio.tech David Looser > wrote:
>> "Laurence Payne" <NOSPAMlpayne1ATdsl.pipex.com> wrote in message
>> ...
>> > On Tue, 11 Dec 2007 11:30:45 +1300, "geoff" >
>> > wrote:
>> >
>> >>However as you amplify it, yo will also bring up the quantisation
>> >>noise.
>> >>It
>> >>is best to optimise your recording level first.
>> >>
>> >>The effect can be reduced by ensuring you are recording at 24 bits
>> >>resolution.
>> >
>> > Waste of space really, off vinyl. Or off any other real-world source
>> > where levels are under control.
>
>> I agree. The background noise on the vinyl will dither the quantisation
>> quite effectively. Whilst ideally one would record with the peak signal
>> just
>> failing to hit 0dBFS,
>
> 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.

David.

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

[David Looser]

"geoff" > wrote in message
...
> Laurence Payne wrote:
>> On Tue, 11 Dec 2007 11:30:45 +1300, "geoff" >
>> wrote:
>>
>>> However as you amplify it, yo will also bring up the quantisation
>>> noise. It is best to optimise your recording level first.
>>>
>>> The effect can be reduced by ensuring you are recording at 24 bits
>>> resolution.
>>
>> Waste of space really, off vinyl. Or off any other real-world source
>> where levels are under control.
>
> If he is recording at a lower than optimum level then it is well worth
> using the extra bits, to reduce the detrimental effect of bringing up the
> level once digitised.
>
It's very unlikely that there will be any audible difference using 24-bit
unless the level is increased significantly (20dB or more) because the
background noise from the vinyl will dither the quantisation. What matters
far more is to use a high-quality RIAA amp and a low-distortion ADC.

> It is a long time since amounts of data like that have been significant.
> And once on CD he can delete the computer data anyway. Sheesh.
>
But many audio recording programs only work in 16 bit. There is no advantage
to using 24-bit for this purpose so it's not worth the extra hassle.

David.

[Geoff]

David Looser wrote:
>>
> It's very unlikely that there will be any audible difference using
> 24-bit unless the level is increased significantly (20dB or more)

That's exactly what the sceanrio was, I think. Whatever the 'quarter way
mark' on his meter or waveform display is.

geoff

[Arny Krueger]

"jakdedert" > wrote in message

> Adrian wrote:
>> On Dec 11, 10:37 am, "Arny Krueger" >
>> wrote:
>>> "Richard Crowley" > wrote in message
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>> "Dave W." wrote ...
>>>>> Adrian wrote:
>>>>>> This past weekend I copied three albums. The signal
>>>>>> is clean but not strong. I have the gain on
>>>>>> theUSBPhono turned to the max.
>>>>> If, as you say, the signal is clean, then as long as
>>>>> you have it digitised there is no problem.
>>>> Lets review the bidding....
>>>> * Low-output MC cartridge feeding an inexpensive RIAA
>>>> phono preamp designed for MC.
>>>> * Gain on the preamp "turned to the max".
>>>> * Signal is "clean but not strong"
>>>> Therefore, by definition, the captured signal is NOT
>>>> "clean" after amplifying it (plus the noise) to the
>>>> nominal level.
>>> Agreed.
>>>
>>>> Of course, Adrian could decide that it is good enough
>>>> for his purposes, and that is fine. But conventional
>>>> wisdom would suggest that the solution might be...
>>>> 1) Use a conventional MM cartridge
>>>> 2) Use a step-up transformer or pre-pre-amp for MC
>>>> 3) Use a preamp designed for MC.
>>> I'd vote for solution number 1, more specificially this
>>> cartridge:
>>> http://www.amazon.com/Shure-M97xE-High-Performance-Magnetic-Cartridge...
>>>
>>> Cheapest way out and solves more problems

> Nope, cheapest is to simply use your receiver for a
> preamp...nothing else to buy.

His receiver has a MC input?

That would be pretty rare!

[Arny Krueger]

"Steven Sullivan" > wrote in message

> In rec.audio.tech David Looser
> > wrote:
>> "Laurence Payne" <NOSPAMlpayne1ATdsl.pipex.com> wrote in
>> message
>> ...
>>> On Tue, 11 Dec 2007 11:30:45 +1300, "geoff"
>>> > wrote:
>>>
>>>> However as you amplify it, yo will also bring up the
>>>> quantisation noise. It
>>>> is best to optimise your recording level first.
>>>>
>>>> The effect can be reduced by ensuring you are
>>>> recording at 24 bits resolution.
>>>
>>> Waste of space really, off vinyl. Or off any other
>>> real-world source where levels are under control.
>
>> I agree. The background noise on the vinyl will dither
>> the quantisation quite effectively. Whilst ideally one
>> would record with the peak signal just failing to hit
>> 0dBFS,
>
> 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.

The simple solution is to set levels by recording something with your
recording software. Most recording software gives a quasi-real time display
for coarse level setting. Record the loudest passage and double check the
sample recording once made.

3-10 dB are good numbers for setting headroom. 3 dB is more appropriate for
static events like transcriptions, while 10 dB is more appropriate for level
setting during a rehearsal for a live event that you wish to record.

[Laurence Payne]

On Tue, 11 Dec 2007 13:13:11 -0800 (PST), Adrian
> wrote:

>> Do you still have the amplifier you used to use when you played vinyl
>> all the time? It will have an input stage better suited to your
>> low-output cartridge. Use that as a preamp, feeding Tape Out into
>> Line In on the ART, switching out the RIAA stage.
>
>Yes I do have the receiver. However, it will awkward to set it up
>just now. Since I see this as a project that will go on for some time
>I amthinking about a new cartridge.

What's to set up? Just put it somewhere near the turntable and
computer. It dousn't need speakers or anything.

[Arny Krueger]

"geoff" > wrote in message

> Adrian wrote:
>
>> So, time to find a Moving Magnet cartridge.
>>
>> Thanks for helping.
>
> MC cartridges often offer benefits over MC. Why not buy a
> phono preamp with a MC/MM switch ? Probably a cheaper
> option.


Whether MC cartriges have any inherent benefits over MM cartridges has
always been controversial.

One of the finest MM cartridges ever made still costs less than $100. It's
hard to get a good MM preamp for $100, and MC preamps are generally far more
expensive.

[tony sayer]

In article >, Arny Krueger
> scribeth thus
>"geoff" > wrote in message

>> Adrian wrote:
>>
>>> So, time to find a Moving Magnet cartridge.
>>>
>>> Thanks for helping.
>>
>> MC cartridges often offer benefits over MC. Why not buy a
>> phono preamp with a MC/MM switch ? Probably a cheaper
>> option.
>
>
>Whether MC cartriges have any inherent benefits over MM cartridges has
>always been controversial.
>
>One of the finest MM cartridges ever made still costs less than $100.

Which is please?...


--
Tony Sayer

[Peter Larsen[_2_]]

Steven Sullivan wrote:

>> I agree. The background noise on the vinyl will dither the
>> quantisation quite effectively. Whilst ideally one would record with
>> the peak signal just failing to hit 0dBFS,

> 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.

The peak signal when recording vinyl comes frm the clicks, usually 6 dB
higher than the signal for the large ones. Grammophone records and quality
playback of them can provide a very high quality sound IF and only IF all
links are good.

I have found it most useful to sample them at 96 kHz 16 bit so as to save
disks space, I don't see any logical reason in wasting it for writting 16
binary ones for each sample, but I want a good sharp and undistorted clicks
in case automated click removal is relevant. Mostly I just take the big ones
out with fix single click functionality, but there may be zones - outmost 3
millimetres come to mind and certainly the bands between tracks - where
automated removal _is_ relevant. Some of the time I also go for modest
overall noise reduction, but it is a case by case decision. What I always do
is to convert to 96-32 first, I then downsample to 44.1-32 prior to other
processing, such as compensating for combined cartridge and riaa frequency
response error.

<IDEA!>
I am fortunate to have received a B&K rest record at some event may years
ago, but there ought to be a market for playback calibration disks, a 45
might do fine and be reasonably shippable.
</IDEA!>

My main reasons for not staying at 96 kHz sample rate for the eq is
processing speed, an additional reason is that the relevant display window
in my preferred software is easier to use with a lower sampling rate because
it needs to display less. I usually end up with fixing overall channel
balance issues - mostly both sides of an album has the same error, some of
the time it is different on per side basis, I hardly ever fix balance on pr.
track basis, assuming that single track oddities are artistic intent but
that overall oddities are caused by alignent errors.

Then and first then I convert to 44.1-16 with my preferred dither. I have
been a great fan of high output MC cartridges since the first one hit the
market. There is a slight detail loss compared to the low output ones, but
the high output Ortofon I have fits my tonearm and my real world record
collection very well and it is nice to be able to skip the transformer or
the extra amplification stage. I found a 1980-ties preamp with MC stage a
couple of year ago, I couldn't buy the matching poweramp without also bying
the preamp, and the preamp was a very pleasant surprise ... hmmm ....


Kind regards

Peter Larsen

[Peter Larsen[_2_]]

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?

> David.


Kind regards

Peter Larsen

[David Looser]

"geoff" > wrote in message
...
> David Looser wrote:
>>>
>> It's very unlikely that there will be any audible difference using
>> 24-bit unless the level is increased significantly (20dB or more)
>
> That's exactly what the sceanrio was, I think. Whatever the 'quarter way
> mark' on his meter or waveform display is.
>
> geoff
>

He also said he could use "at least 3dB more", so I've no idea what levels
he was actually getting. In any case as has been pointed out before if the
problem is that a pre-amp intended for MM cartridges is being used with an
MC one then the noise generated in the pre-amp will be the real problem, not
the quantisation noise from the ADC.

David.

[David Looser]

"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.

David.

[Silk]

On Tue, 11 Dec 2007 13:11:02 -0800, Adrian wrote:

> That seems to be the consensus. Does anyone have a recomentation for a
> more appropriate cartridge.

Audio Technica AT95. If you can still get them.

[David Looser]

"Peter Larsen" > wrote in message
...

>
> I have found it most useful to sample them at 96 kHz 16 bit so as to save
> disks space, I don't see any logical reason in wasting it for writting 16
> binary ones for each sample, but I want a good sharp and undistorted
> clicks in case automated click removal is relevant. Mostly I just take the
> big ones out with fix single click functionality,

Does anyone remember the Garrad "Music Recovery Module"? It was designed to
remove the big clicks in real time by briefly shunting the audio with a
light-dependent resistor when a click was detected. Click detection was
based on the idea that clicks were of large amplitude, had a fast rise-time
and had a significant out-of-phase component. It actually worked quite well,
but no match for a software solution.

David.

[Eiron]

David Looser wrote:
> "Peter Larsen" > wrote in message
> ...
>
>> I have found it most useful to sample them at 96 kHz 16 bit so as to save
>> disks space, I don't see any logical reason in wasting it for writting 16
>> binary ones for each sample, but I want a good sharp and undistorted
>> clicks in case automated click removal is relevant. Mostly I just take the
>> big ones out with fix single click functionality,
>
> Does anyone remember the Garrad "Music Recovery Module"? It was designed to
> remove the big clicks in real time by briefly shunting the audio with a
> light-dependent resistor when a click was detected. Click detection was
> based on the idea that clicks were of large amplitude, had a fast rise-time
> and had a significant out-of-phase component. It actually worked quite well,
> but no match for a software solution.

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.

--
Eiron.

[Dave Platt]

In article >,
Eiron > wrote:

>> Does anyone remember the Garrad "Music Recovery Module"? It was designed to
>> remove the big clicks in real time by briefly shunting the audio with a
>> light-dependent resistor when a click was detected. Click detection was
>> based on the idea that clicks were of large amplitude, had a fast rise-time
>> and had a significant out-of-phase component. It actually worked quite well,
>> but no match for a software solution.

>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.

Interesting. I had somewhat different results with the
pop-and-scratch filter in Diamond Cut's DC-ART.

Although the subtract-and-compare test I did seemed, at first, to
indicate similarly good results, comparison with the original
suggested otherwise. A lot of the "scratches" removed turned out to
be the leading edges of musical transients - e.g. from snare drums,
cymbals, and other instruments whose output has a very fast rise-time.

The subtraction test alone wasn't enough to notice this... the clicks
and pops would still appear against a background of silence, since the
filter would not alter the signal at all except when it was actually
removing a transient. It was necessary to (visually) compare the
original musical waveform, with the waveform of the subtracted result,
to notice the correlation and realize what was happening.

If I turned down the sensitivity far enough to keep this from
happening, I found that some *real* pops and scratches were missed.

It's quite possible that Goldwave's algorithms are more sophisticated,
and make better distinctions between scratches and real musical
transients. It might be worthwhile to double-check the results, for
any given piece of music, to make sure that only unwanted glitches are
being removed.

I ended up using a hybrid approach... on records with a significant
amount of popping and crackling, I'd use the automatic filter only on
the quieter parts, and identify the pops and ticks in the louder
sections by ear and select the damaged part of the waveform manually
(and then use the "fix a scratch" reconstruction command on just this
part of the waveform).

--
Dave Platt > AE6EO
Friends of Jade Warrior home page: http://www.radagast.org/jade-warrior
I do _not_ wish to receive unsolicited commercial email, and I will
boycott any company which has the gall to send me such ads!

[Geoff]

Peter Larsen wrote:
> Steven Sullivan wrote:

>
> The peak signal when recording vinyl comes frm the clicks, usually 6
> dB higher than the signal for the large ones. Grammophone records and
> quality playback of them can provide a very high quality sound IF and
> only IF all links are good.
>
> I have found it most useful to sample them at 96 kHz 16 bit so as to
> save disks space, I don't see any logical reason in wasting it for
> writting 16 binary ones for each sample, but I want a good sharp and
> undistorted clicks in case automated click removal is relevant.
> Mostly I just take the big ones out with fix single click

Record at 24 bits, then once you've got rid of your clicks, then you can
raise the overall level with less degradation.

Why 96/16 rather than 44k1/24 ? I don't follow that logic. The highest freq
recorded on most LPs was around 15KHz, apart from clicks of course...

geoff

[Geoff]

David Looser wrote:
> "Peter Larsen" > wrote in message
> ...
>
>>
>> I have found it most useful to sample them at 96 kHz 16 bit so as to
>> save disks space, I don't see any logical reason in wasting it for
>> writting 16 binary ones for each sample, but I want a good sharp and
>> undistorted clicks in case automated click removal is relevant.
>> Mostly I just take the big ones out with fix single click
>> functionality,
>
> Does anyone remember the Garrad "Music Recovery Module"? It was
> designed to remove the big clicks in real time by briefly shunting
> the audio with a light-dependent resistor when a click was detected.
> Click detection was based on the idea that clicks were of large
> amplitude, had a fast rise-time and had a significant out-of-phase
> component. It actually worked quite well, but no match for a software
> solution.

That would freak out on modern hip-hop stuff that has surface noise/clicks
as part of the 'music' !

FWIW I'm looking at my old 301/SME/SME spinning away, right now .

geoff

[Geoff]

Eiron wrote:
> David Looser wrote:
worked quite well, but no match for a
>> software solution.
>
> 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.

Waves X-Click/Noise/Crackle allow you to listen to the output audio, or the
'removed' audio realtime.

geoff

[David Looser]

"geoff" > wrote in message
...
> David Looser wrote:
>> "Peter Larsen" > wrote in message
>> ...
>>
>>>
>>> I have found it most useful to sample them at 96 kHz 16 bit so as to
>>> save disks space, I don't see any logical reason in wasting it for
>>> writting 16 binary ones for each sample, but I want a good sharp and
>>> undistorted clicks in case automated click removal is relevant.
>>> Mostly I just take the big ones out with fix single click
>>> functionality,
>>
>> Does anyone remember the Garrad "Music Recovery Module"? It was
>> designed to remove the big clicks in real time by briefly shunting
>> the audio with a light-dependent resistor when a click was detected.
>> Click detection was based on the idea that clicks were of large
>> amplitude, had a fast rise-time and had a significant out-of-phase
>> component. It actually worked quite well, but no match for a software
>> solution.
>
> That would freak out on modern hip-hop stuff that has surface noise/clicks
> as part of the 'music' !

It could be switched to "bypass", in which case it became merely a
high-quality RIAA pre-amp.
>
David.

[David Looser]

"geoff" > wrote in message
...
> Peter Larsen wrote:
>> Steven Sullivan wrote:
>
>>
>> The peak signal when recording vinyl comes frm the clicks, usually 6
>> dB higher than the signal for the large ones. Grammophone records and
>> quality playback of them can provide a very high quality sound IF and
>> only IF all links are good.
>>
>> I have found it most useful to sample them at 96 kHz 16 bit so as to
>> save disks space, I don't see any logical reason in wasting it for
>> writting 16 binary ones for each sample, but I want a good sharp and
>> undistorted clicks in case automated click removal is relevant.
>> Mostly I just take the big ones out with fix single click
>
> Record at 24 bits, then once you've got rid of your clicks, then you can
> raise the overall level with less degradation.
>
> Why 96/16 rather than 44k1/24 ? I don't follow that logic. The highest
> freq recorded on most LPs was around 15KHz, apart from clicks of course...
>

I'm not sure I follow yours. Whilst the highest *recorded" frequency was
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.

In the old days of 405-line TV the (AM) sound channel would have a simple
impulse interference reduction limiter fitted. This worked on the fact that
the impulse would have a far faster rise-time than any audio content. For
this reason the band-width of the sound IF channel was kept far wider than
need for the audio, around 100kHz.


David.

[Peter Larsen[_2_]]

David Looser wrote:

>> 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?,

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

> we are talking about
> setting the analogue level into the ADC.

And?

> 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.

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

> David.


Kind regards

Peter Larsen

[Arny Krueger]

"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.

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!

[Peter Larsen[_2_]]

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, except that
there is incredibly poor audio around that will become less offensive via
good automated declick.


Kind regards

Peter Larsen

[David Looser]

"Peter Larsen" > wrote in message
...
> David Looser wrote:
>
>>> 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?,
>
> 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!!!

>> we are talking about
>> setting the analogue level into the ADC.
>
> And?
>
BEFORE you make the recording (so you haven't got a screen image from the
audio editing package to look at yet!)

>> 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.
>
> 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?

David.

[David Looser]

"Arny Krueger" > wrote in message
. ..
> "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.

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

David.

>

[Arny Krueger]

"David Looser" > wrote in
message
> "Arny Krueger" > wrote in message
> . ..
>> "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.

> 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.

[Eiron]

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.

[Peter Larsen[_2_]]

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

[Peter Larsen[_2_]]

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

[Peter Larsen[_2_]]

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

[Peter Larsen[_2_]]

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

[Steven Sullivan]

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 here:

http://www.cadenzarecording.com/papers/Digitaldistortion.pdf


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

[Steven Sullivan]

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

[Steven Sullivan]

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

[Steven Sullivan]

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

[Peter Larsen[_2_]]

Steven Sullivan wrote:

> 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....?

I stated an opinion. I do not waste time proving recording choices, I make
them based on what sounds best.


Kind regards

Peter Larsen

[Peter Larsen[_2_]]

Steven Sullivan wrote:

> 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 here:

> http://www.cadenzarecording.com/papers/Digitaldistortion.pdf

IF there is an inter-sample peak issue, then there will be two consecutive
samples at 0 dB FS. Quality audio software considers such samples to be
clipped and can provide a count of probably clipped samples.


Kind regards

Peter Larsen

[Peter Larsen[_2_]]

Steven Sullivan wrote:


[someone typed]

>> 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

For the wordlenght difference to matter in RECORDING there has to be signal
that is either truncated or hit by converter unlinearities. The latter
hardly applies in case of the same converter and the former does not apply
for vinyl. The only advantage in sampling at 24 or 32 bits is in the
workflow because a file format conversion can be skipped.

> -- 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.

First of all it will not sound like grammophone playback that is influenced
via the grammophone hearing the loudspeakers. This is not a new issue. For
quality playback the grammophone has to be in another room than the one you
listen in. Next there is the issue of number of analog components the signal
passes through prior to being digitized.

The issue that matters is that the digitized recording does not deteriorate
further and that the life of the backup version, the vinyl disk, is greatly
extended by it not being played back.

> -S


Kind regards

Peter Larsen

[Arny Krueger]

"Steven Sullivan" > wrote in message

> 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?

Yes.

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

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

> Scroll down to the 'TL Mastermeter'

Adobe Audition, AKA CoolEdit is widely used for transcribing LPs, and
provides a scrolling real-time display.

[Arny Krueger]

"Steven Sullivan" > wrote in message

> 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 here:
>
> http://www.cadenzarecording.com/papers/Digitaldistortion.pdf

In the real world they happen, but they are relatively rare. If you record
with levels near FS, there might be one or two such clipped peaks every 3-5
minutes. Their duration is by definition very short. They are very unlikely
to be heard. They are more of an intellectual nuisance than anything else.

[Arny Krueger]

"Steven Sullivan" > wrote in message

> 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.

Agreed.

> 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

I was discussing the results of some of my recent tests of MP3 coders with a
friend who had a long, sucessful career transferring analog tape to movie
film optical sound tracks. When I described how modern MP3 coders tend to
reduce information content by bringing up the noise level between musical
tones, he said: "You mean like vinyl or analog tape"?

LOL!

[Arny Krueger]

"Peter Larsen" > wrote in message

> Steven Sullivan wrote:
>
>> 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....?
>
> I stated an opinion. I do not waste time proving
> recording choices, I make them based on what sounds best.

Almost right. When we record we make choices based on what sounds best to
us, or what we think is the best reasonble practice.

Admittedly, we don't subject every choice we make while recording to DBTs on
the spot.

However, a great amount of wisdom has forced itself into my life when I did
subject recording certain choices to DBTs.

[[email protected]]

On Dec 14, 3:17 am, "Peter Larsen" > wrote:
> Steven Sullivan wrote:
> > 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 here:
> >http://www.cadenzarecording.com/papers/Digitaldistortion.pdf
>
> IF there is an inter-sample peak issue, then there will be two consecutive
> samples at 0 dB FS. Quality audio software considers such samples to be
> clipped and can provide a count of probably clipped samples.

Why, two adjacent full-scale sample DO NOT a priori
mean that clipping occurred. Indeed, in the realm of
PCM stream, two such samples are perfectly legal
and represent v a completely valid waveform. Now,
yyour DAC may not be able to handle them, but that's
the DAC's problem, not a sample representation
problem.

MORE than two consecutive full-scale samples may be
another issue altogether.

Secondly, it does NOT require, as you claim, that the
two samples be at 0 dB FS. You could have two of them
a mere 1 LSB lower than 0 dB FS and STILL have the
same issue, only the peak will be proportionally lower,
i.e., not much. That's the difference between 0 dB
and -0.00027 dB FS.

Thirdly, if your "quality software" automatically assumes
that two adjacent o dB FS samples unambiguous clipping,
I'd posit it is NOT "quality software."

[Steven Sullivan]

In rec.audio.tech Peter Larsen > wrote:
> Steven Sullivan wrote:

> > 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....?

> I stated an opinion. I do not waste time proving recording choices, I make
> them based on what sounds best.

Figured as much.


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

[Steven Sullivan]

In rec.audio.tech Peter Larsen > wrote:
> Steven Sullivan wrote:

> > 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 here:

> > http://www.cadenzarecording.com/papers/Digitaldistortion.pdf

> IF there is an inter-sample peak issue, then there will be two consecutive
> samples at 0 dB FS. Quality audio software considers such samples to be
> clipped and can provide a count of probably clipped samples.

Did you read any of the link I posted? Here's another

0dbFS+ Signals in Digital Mastering
http://www.tcelectronic.com/media/Level_paper_AES109.pdf

The two consec samples needn't be AT 0 dBFS, for there to be intersample overs. With
contrived signals the ISOs can be as high as +6dBFS. Anecdotally, I see reports of +1 dBFS.
So consec samples at -0.5dB can still flank an ISO.


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

[Steven Sullivan]

In rec.audio.tech Peter Larsen > wrote:
> Steven Sullivan wrote:


> [someone typed]

> >> 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

> For the wordlenght difference to matter in RECORDING there has to be signal
> that is either truncated or hit by converter unlinearities. The latter
> hardly applies in case of the same converter and the former does not apply
> for vinyl. The only advantage in sampling at 24 or 32 bits is in the
> workflow because a file format conversion can be skipped.

Look, I'm just reporting what *they* claim. Not what I think.

> > -- 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.

> First of all it will not sound like grammophone playback that is influenced
> via the grammophone hearing the loudspeakers. This is not a new issue. For
> quality playback the grammophone has to be in another room than the one you
> listen in. Next there is the issue of number of analog components the signal
> passes through prior to being digitized.

If you record the output of the grammophone/cart/pre, you are capturing whatever the
grammaphone is 'hearing' from the loudspeakers.

> The issue that matters is that the digitized recording does not deteriorate
> further and that the life of the backup version, the vinyl disk, is greatly
> extended by it not being played back.


You seem to be confusing me with someone who actually *believes* the audiophile
nosnense.



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

[Steven Sullivan]

In rec.audio.tech Arny Krueger > wrote:
> "Steven Sullivan" > wrote in message
>
> > 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?

> Yes.

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

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

> > Scroll down to the 'TL Mastermeter'

> Adobe Audition, AKA CoolEdit is widely used for transcribing LPs, and
> provides a scrolling real-time display.

Are you sure Audition shows you when you're generating intersample overs?

It's not enough just to offer scrolling real-time display. The display has to 'know' what
consumer DAC/output stages do to a signal with high consecutive sample peaks -- most of the
ones tested by Nielsen and Lund had no headroom for properly reconstructing ISOs between such
sample peaks. Good peak monitoring should warn you when an 'illegal' reconstruction peak will
be generated between two 'legal' sample peaks.

Not saying Audition doesn't do it...just wondering how it's been determined
that it does.



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

[David Looser]

"Peter Larsen" > wrote in message
...
> 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.

You have *predictive* software? Just like the lifts in "Hitch-Hiker" it
knows what's going to happen *before* it happens? I'm amazed, does this
software also predict the numbers for next weeks lottery?

David.

[Arny Krueger]

"Steven Sullivan" > wrote in message

> In rec.audio.tech Arny Krueger > wrote:
>> "Steven Sullivan" > wrote in message
>>
>>> 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?
>
>> Yes.
>
>>> I don't know that those are so common. HEre's one:
>
>>> http://www.secaudio.ch/side122.html
>
>>> Scroll down to the 'TL Mastermeter'
>
>> Adobe Audition, AKA CoolEdit is widely used for
>> transcribing LPs, and provides a scrolling real-time
>> display.

> Are you sure Audition shows you when you're generating
> intersample overs?

Yes.

In fact I just pulled CE 2.1 up on this computer I'm typing one and made a
few instesample overs by hand, just to be zillion-times sure.

> It's not enough just to offer scrolling real-time
> display. The display has to 'know' what consumer
> DAC/output stages do to a signal with high consecutive
> sample peaks -- most of the ones tested by Nielsen and
> Lund had no headroom for properly reconstructing ISOs
> between such sample peaks. Good peak monitoring should
> warn you when an 'illegal' reconstruction peak will be
> generated between two 'legal' sample peaks.

I've been aware of intersample overs for years - based on experience with my
own work.

> Not saying Audition doesn't do it...just wondering how
> it's been determined that it does.

Real world experience. Just happened to have my eyes open while working on
some recordings that I made.

[Arny Krueger]

"Steven Sullivan" > wrote in message

> In rec.audio.tech Peter Larsen >
> wrote:
>> Steven Sullivan wrote:
>
>>> 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 here:
>
>>> http://www.cadenzarecording.com/papers/Digitaldistortion.pdf
>
>> IF there is an inter-sample peak issue, then there will
>> be two consecutive samples at 0 dB FS. Quality audio
>> software considers such samples to be clipped and can
>> provide a count of probably clipped samples.
>
> Did you read any of the link I posted? Here's another
>
> 0dbFS+ Signals in Digital Mastering
> http://www.tcelectronic.com/media/Level_paper_AES109.pdf
>
> The two consec samples needn't be AT 0 dBFS, for there to
> be intersample overs. With contrived signals the ISOs
> can be as high as +6dBFS. Anecdotally, I see reports of
> +1 dBFS. So consec samples at -0.5dB can still flank an
> ISO.

With 16 bit samples 1 and 4 at zero, samples 2 and 3 at about 24,000 will
just give an over between them. With samples 1 and 4 at -32767, samples at
2 and 4 need to only be about 14,400 for there to be an over between them.
There will also be unders to the left and right of samples 1 and 4.

Again, in the real world this is pretty much moot. The ear can tolerate a
few dozen microseconds of light clipping as long as it is an isolated event.

[Peter Larsen[_2_]]

Arny Krueger wrote:

> However, a great amount of wisdom has forced itself into my life when
> I did subject recording certain choices to DBTs.

DBT's are good at large differences. It is also an excellent point to make
that a difference by definition is not a major difference if it doesn't show
up in a DBT. It is also extremely worthwhile to remember the differences in
tonality and imaging caused by moving a main pair 2 inches ....


Kind regards

Peter Larsen

[Peter Larsen[_2_]]

Steven Sullivan wrote:

> Did you read any of the link I posted? Here's another

No.

> 0dbFS+ Signals in Digital Mastering
> http://www.tcelectronic.com/media/Level_paper_AES109.pdf

> The two consec samples needn't be AT 0 dBFS, for there to be
> intersample overs. With contrived signals the ISOs can be as high as
> +6dBFS. Anecdotally, I see reports of +1 dBFS. So consec samples at
> -0.5dB can still flank an ISO.

This is no doubt important to remember in an equipment design context. It is
considerably less important in the real world context of digitizing vinyl at
96 kHz sample rate with the clicks defining the record level setting such
that there are two unused bits above the audio. Also btw. the file is not
clipped if the samples are correct, strictly speaking this is a DA
conversion issue.

> -S


Kind regards

Peter Larsen

[Peter Larsen[_2_]]

Steven Sullivan wrote:

> If you record the output of the grammophone/cart/pre, you are
> capturing whatever the grammaphone is 'hearing' from the loudspeakers.

That would be an incompetent thing to do, it is indeed one of the many
errors I too have made, but it is is not new knowledge.

> You seem to be confusing me with someone who actually *believes* the
> audiophile nosnense.

I did get that impression yes, my apology.

> -S


Kind regards

Peter Larsen

[Peter Larsen[_2_]]

David Looser wrote:

> "Peter Larsen" > wrote in message
> ...

>> 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.

> You have *predictive* software? Just like the lifts in "Hitch-Hiker"
> it knows what's going to happen *before* it happens? I'm amazed, does
> this software also predict the numbers for next weeks lottery?

You DO realize that you are quabling about the possibility of inter-sample
overs in DA conversion of a file that is recorded at 96 kHz sample rate with
2 full bits of headroom above the audio signal to make room for the clicks.
Those large clicks are later removed. The file is eventually as previously
suggested by me normalized to -2.5 dB ref. full scale.

The issue you worry about no doubt is real, but there is no data loss as
long as the sample values are correct. What this is about is that the analog
stage in front of the AD converter needs to have headroom above 0 dB FS and
that the analog stage after the AD converter needs to have headroom above 0
dB FS. The digital file can not be considered clipped if the sample values
are correct no matter how high the intersample peak happens to be.

If you have commercial pop cd's with a large number of consecutive samples
at 0 dB FS, then you have something to worry about and someting to rant away
about.

If you have such productions mp3's without a preceding gain reduction of
some 2 dB, then you will have even more overs ....

> David


Kind regards

Peter Larsen

[Steven Sullivan]

In rec.audio.tech Arny Krueger > wrote:
> "Steven Sullivan" > wrote in message
>
> > In rec.audio.tech Arny Krueger > wrote:
> >> "Steven Sullivan" > wrote in message
> >>
> >>> 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?
> >
> >> Yes.
> >
> >>> I don't know that those are so common. HEre's one:
> >
> >>> http://www.secaudio.ch/side122.html
> >
> >>> Scroll down to the 'TL Mastermeter'
> >
> >> Adobe Audition, AKA CoolEdit is widely used for
> >> transcribing LPs, and provides a scrolling real-time
> >> display.

> > Are you sure Audition shows you when you're generating
> > intersample overs?

> Yes.

> In fact I just pulled CE 2.1 up on this computer I'm typing one and made a
> few instesample overs by hand, just to be zillion-times sure.

How? And what did it show?

Audition Help includes this warning

"If you're planning to put normalized audio on CD, you might want to normalize the waveforms
to no more than 96% as some audio compact disc players have problems accurately reproducing bits
that have been processed to 100% (maximum) amplitude."

And when I've normalize a music track to 0dBFS, I've never seen the Audition peak meter go
into the +0 zone. So I assumed that its peak meter does not model reconstructed output.




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

[Steven Sullivan]

In rec.audio.tech Peter Larsen > wrote:
> Arny Krueger wrote:

> > However, a great amount of wisdom has forced itself into my life when
> > I did subject recording certain choices to DBTs.

> DBT's are good at large differences.

Actually, they're really quite useful for verifying audibility of *small* differences.
Hence their use in codec development and psychoacoustics research.

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

[Steven Sullivan]

In rec.audio.tech Peter Larsen > wrote:
> Steven Sullivan wrote:

> > If you record the output of the grammophone/cart/pre, you are
> > capturing whatever the grammaphone is 'hearing' from the loudspeakers.

> That would be an incompetent thing to do, it is indeed one of the many
> errors I too have made, but it is is not new knowledge.


? How do *you* digitize an LP, if not from the analog output?

Or are you just saying that when you do, you make sure there is not acoustic feedback from
nearby loudspeakers to the turntable?




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

[Steven Sullivan]

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

> > "Peter Larsen" > wrote in message
> > ...

> >> 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.

> > You have *predictive* software? Just like the lifts in "Hitch-Hiker"
> > it knows what's going to happen *before* it happens? I'm amazed, does
> > this software also predict the numbers for next weeks lottery?

> You DO realize that you are quabling about the possibility of inter-sample
> overs in DA conversion of a file that is recorded at 96 kHz sample rate with
> 2 full bits of headroom above the audio signal to make room for the clicks.
> Those large clicks are later removed. The file is eventually as previously
> suggested by me normalized to -2.5 dB ref. full scale.

You don't need to record at 96 to capture the clicks sufficiently for them to be located later
by software. 44.1 is quite sufficient.

[Arny Krueger]

"Steven Sullivan" > wrote in message

> In rec.audio.tech Arny Krueger > wrote:
>> "Steven Sullivan" > wrote in message
>>
>>> In rec.audio.tech Arny Krueger > wrote:
>>>> "Steven Sullivan" > wrote in message
>>>>

>>> Are you sure Audition shows you when you're generating
>>> intersample overs?
>
>> Yes.
>
>> In fact I just pulled CE 2.1 up on this computer I'm
>> typing one and made a few instesample overs by hand,
>> just to be zillion-times sure.

> How? And what did it show?

A line going up to FS and disappearing for a while, and then a line starting
at FS and going down. Something like it for -FS.

> Audition Help includes this warning

> "If you're planning to put normalized audio on CD, you
> might want to normalize the waveforms
> to no more than 96% as some audio compact disc players
> have problems accurately reproducing bits that have been
> processed to 100% (maximum) amplitude."

I've been saying as much on RAP for years.

> And when I've normalize a music track to 0dBFS, I've
> never seen the Audition peak meter go
> into the +0 zone. So I assumed that its peak meter does
> not model reconstructed output.

It indicates it, when it exists.

[Arny Krueger]

"Peter Larsen" > wrote in message

> Arny Krueger wrote:
>
>> However, a great amount of wisdom has forced itself into
>> my life when I did subject recording certain choices to
>> DBTs.
>
> DBT's are good at large differences.

...and any difference that is audible.

> It is also an
> excellent point to make that a difference by definition
> is not a major difference if it doesn't show up in a DBT.

If it doesn't show up in a DBT then it is not audible.

> It is also extremely worthwhile to remember the
> differences in tonality and imaging caused by moving a
> main pair 2 inches ....

Which I interpret as showing the futility of obsessing over small
differences.

[Peter Larsen[_2_]]

Steven Sullivan wrote:

> In rec.audio.tech Peter Larsen > wrote:
>> Steven Sullivan wrote:

>>> If you record the output of the grammophone/cart/pre, you are
>>> capturing whatever the grammaphone is 'hearing' from the
>>> loudspeakers.

>> That would be an incompetent thing to do, it is indeed one of the
>> many errors I too have made, but it is is not new knowledge.

> ? How do *you* digitize an LP, if not from the analog output?

> Or are you just saying that when you do, you make sure there is not
> acoustic feedback from nearby loudspeakers to the turntable?

Indeed. It is that coloration that some of the vinyl enthusiasts miss. Other
just enjoy the larger actually produced dynamic range on vinyl.

> -S

Kind regards

Peter Larsen

[Steven Sullivan]

In rec.audio.tech Peter Larsen > wrote:
> Steven Sullivan wrote:

> > In rec.audio.tech Peter Larsen > wrote:
> >> Steven Sullivan wrote:

> >>> If you record the output of the grammophone/cart/pre, you are
> >>> capturing whatever the grammaphone is 'hearing' from the
> >>> loudspeakers.

> >> That would be an incompetent thing to do, it is indeed one of the
> >> many errors I too have made, but it is is not new knowledge.

> > ? How do *you* digitize an LP, if not from the analog output?

> > Or are you just saying that when you do, you make sure there is not
> > acoustic feedback from nearby loudspeakers to the turntable?

> Indeed. It is that coloration that some of the vinyl enthusiasts miss. Other
> just enjoy the larger actually produced dynamic range on vinyl.

Vinyl does not 'actually produce' a larger dynamic range, unless the CD's dynamic range has
been intentionally reduced.


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

[Peter Larsen[_2_]]

Steven Sullivan wrote:

>> Indeed. It is that coloration that some of the vinyl enthusiasts
>> miss. Other just enjoy the larger actually produced dynamic range on
>> vinyl.

> Vinyl does not 'actually produce' a larger dynamic range, unless the
> CD's dynamic range has been intentionally reduced.

Dataset of actual dynamic range on selected vinyl records and cd's has been
made available, but didn't pass the requirements for an AES paper, probably
because it was too long and because I tried to cover too much ground in one
paper with too many illustrations.

A subset was in these and nearby newsgroups as I recall things and the
lot was on my website until I felt it referred to without any
source-reference
pointing at it being provided by the author of another paper.

Anyway ... The producers produced the larger dynamic range when they
produced the records. A lot of vinyl sounds better than a lot of digital
because it is plain better sound engineering. Digital ought to sound best,
but THE LOUDNESS RACE HAS RUINED IT.

The simple issue is that the number of multiband-compressors pr. incompetent
operator has gone up drastically.


Kind regards

Peter Larsen

[Arny Krueger]

"Steven Sullivan" > wrote in message

> In rec.audio.tech Peter Larsen >
> wrote:
>> Steven Sullivan wrote:
>
>>> In rec.audio.tech Peter Larsen >
>>> wrote:
>>>> Steven Sullivan wrote:
>
>>>>> If you record the output of the grammophone/cart/pre,
>>>>> you are capturing whatever the grammaphone is
>>>>> 'hearing' from the loudspeakers.
>
>>>> That would be an incompetent thing to do, it is indeed
>>>> one of the many errors I too have made, but it is is
>>>> not new knowledge.
>
>>> ? How do *you* digitize an LP, if not from the analog
>>> output?
>
>>> Or are you just saying that when you do, you make sure
>>> there is not acoustic feedback from nearby loudspeakers
>>> to the turntable?
>
>> Indeed. It is that coloration that some of the vinyl
>> enthusiasts miss. Other just enjoy the larger actually
>> produced dynamic range on vinyl.
>
> Vinyl does not 'actually produce' a larger dynamic range,
> unless the CD's dynamic range has been intentionally
> reduced.

Agreed. However the natural dynamic range of many musical performances is
within the dynamic range of either. One of the properties of the human ear
can lead to the mistaken perception that vinyl has a wider dynamic range -
vinyl's nonlinear distortion rises rapidly beyond a certain modest level,
and distorted music tends to sound louder than undistorted music. Of course,
the nonlinear distortion in digital is identically zero at any point below
clipping.

[Arny Krueger]

"Peter Larsen" > wrote in message

> Steven Sullivan wrote:
>
>>> Indeed. It is that coloration that some of the vinyl
>>> enthusiasts miss. Other just enjoy the larger actually
>>> produced dynamic range on vinyl.
>
>> Vinyl does not 'actually produce' a larger dynamic
>> range, unless the CD's dynamic range has been
>> intentionally reduced.
>
> Dataset of actual dynamic range on selected vinyl records
> and cd's has been made available, but didn't pass the
> requirements for an AES paper, probably because it was
> too long and because I tried to cover too much ground in
> one paper with too many illustrations.

> A subset was in these and nearby newsgroups as I recall
> things and the lot was on my website until I felt it referred to without
> any source-reference
> pointing at it being provided by the author of another
> paper.

> Anyway ... The producers produced the larger dynamic
> range when they produced the records. A lot of vinyl
> sounds better than a lot of digital because it is plain
> better sound engineering. Digital ought to sound best,
> but THE LOUDNESS RACE HAS RUINED IT.

The loudness race is a matter of art and business, not science or
technological limits. Therefore it has no place in a technical discussion of
the performance of media formats. Any recordings that are altered for
business or artistic reason must be immediately excluded.

There are so many extant recordings that unless statistically significant
samples are chosen, reasonable conclusions can't be reached.

Vinyl advocates are well-known for their apparently unintentional
cherry-picking of samples.

Besides, commercial recordings are not laboratory tools for evaluating
recording formats.

It would be reasonable to have an all-star team of vinyl cutting experts do
their best posssible job of cutting a mutually-agreed-upon test file on
carefully-selected and hand-tuned vinyl cutting equipment.

We should compare that to a CD burned by a modestly-skilled middle school
student on a 19.95 CD ROM drive.

It's easy to predict that the middle-school student will confound the
vinyl-cutting experts after the unbiased evaluations of the performance of
the two disks is finished.

> The simple issue is that the number of
> multiband-compressors pr. incompetent operator has gone
> up drastically.

That is about art and business, not science and technology.

The better a recording medium is, the more susceptible it is to abuse
because it is simply more responsive to the needs and preferences of the
person doing the production work.

[Eiron]

Arny Krueger wrote:

> and distorted music tends to sound louder than undistorted music.

Which is why electrostatic speakers are louder than they sound.
Quad erat demonstrandum. :-)

--
Eiron.

[Peter Larsen[_2_]]

Steven Sullivan wrote:

>> Indeed. It is that coloration that some of the vinyl enthusiasts
>> miss. Other just enjoy the larger actually produced dynamic range on
>> vinyl.

> Vinyl does not 'actually produce' a larger dynamic range, unless the
> CD's dynamic range has been intentionally reduced.

Correct. CD's tend to be mastered for less dynamic range than what was used
on vinyl. In my opinion that is the single most important cause of the cd's
being experienced as having inferior sound quality by listeners who compare
with actual concerts.

> -S


Kind regards

Peter Larsen

[David Looser]

"Steven Sullivan" > wrote in message
...
> In rec.audio.tech David Looser > wrote:
>
>> 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
>

I try not to. I can only take so much of people obsessing over the
improvement in sound quality they get by replacing the mains leads with
silver-plated wire, or changing the make of GZ32 rectifier used, or some
other minor (but usually expensive) alteration.

IMO if a difference doesn't show up in a DBT it doesn't exist, whatever the
audiophiles may claim. But if you've just bought an expensive new gizmo of
course it's going to sound better *to you*.

I'm no longer astonished at the claims made in such forums, but it would be
straightforward to mount a DBT of CD transfers from vinyl made using 16 and
24 bit ADCs (everything else identical of course, including ADC
architecture). If the DBT showed a clear preference for the 24-bit version I
would be astonished, and withdraw my comments.

David.

[Silk]

On Wed, 12 Dec 2007 10:14:24 +0000, tony sayer wrote:

> In article >, Arny Krueger

>>One of the finest MM cartridges ever made still costs less than $100.
>
> Which is please?...

AT95E or even a Linn rebadged one in the form of the K9 if you really
must pay more ;-)

[Eiron]

Silk wrote:
> On Wed, 12 Dec 2007 10:14:24 +0000, tony sayer wrote:
>
>> In article >, Arny Krueger
>
>>> One of the finest MM cartridges ever made still costs less than $100.
>> Which is please?...
>
> AT95E or even a Linn rebadged one in the form of the K9 if you really
> must pay more ;-)

How does that compare to Arny's favourite, the Shure M97xE?

--
Eiron.

[Silk]

On Sun, 16 Dec 2007 13:38:43 +0000, Eiron wrote:

> Silk wrote:
>> On Wed, 12 Dec 2007 10:14:24 +0000, tony sayer wrote:
>>
>>> In article >, Arny
>>> Krueger
>>
>>>> One of the finest MM cartridges ever made still costs less than $100.
>>> Which is please?...
>>
>> AT95E or even a Linn rebadged one in the form of the K9 if you really
>> must pay more ;-)
>
> How does that compare to Arny's favourite, the Shure M97xE?

I've no idea. I just know, having owned and used many, that the AT95 and
clones sound very good for the price. I particularly liked the Linn K9
(named because of its uncanny resemblance to Dr Who's robot dog) because
it came in a very tasteful grey and had a nice Linn logo on the front;-)

[Steven Sullivan]

In rec.audio.tech Arny Krueger > wrote:
> "Steven Sullivan" > wrote in message
>
> > In rec.audio.tech Peter Larsen >
> > wrote:
> >> Steven Sullivan wrote:
> >
> >>> In rec.audio.tech Peter Larsen >
> >>> wrote:
> >>>> Steven Sullivan wrote:
> >
> >>>>> If you record the output of the grammophone/cart/pre,
> >>>>> you are capturing whatever the grammaphone is
> >>>>> 'hearing' from the loudspeakers.
> >
> >>>> That would be an incompetent thing to do, it is indeed
> >>>> one of the many errors I too have made, but it is is
> >>>> not new knowledge.
> >
> >>> ? How do *you* digitize an LP, if not from the analog
> >>> output?
> >
> >>> Or are you just saying that when you do, you make sure
> >>> there is not acoustic feedback from nearby loudspeakers
> >>> to the turntable?
> >
> >> Indeed. It is that coloration that some of the vinyl
> >> enthusiasts miss. Other just enjoy the larger actually
> >> produced dynamic range on vinyl.
> >
> > Vinyl does not 'actually produce' a larger dynamic range,
> > unless the CD's dynamic range has been intentionally
> > reduced.

> Agreed. However the natural dynamic range of many musical performances is
> within the dynamic range of either. One of the properties of the human ear
> can lead to the mistaken perception that vinyl has a wider dynamic range -
> vinyl's nonlinear distortion rises rapidly beyond a certain modest level,
> and distorted music tends to sound louder than undistorted music. Of course,
> the nonlinear distortion in digital is identically zero at any point below
> clipping.

A euphonic illusion of wider dynamic range is not the same as 'actual'
dynamic range, of course. And as a format, CD offers a wider actual
dynamic range than LP.

Offering, and providing in practice, are two different things, and if
mixers and mastering engineers *choose* to limit the dynamic range on CD,
that's not a deficit of the format.

I trust everyone here understands that at this late date.



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

[Steven Sullivan]

In rec.audio.tech Peter Larsen > wrote:
> Steven Sullivan wrote:

> >> Indeed. It is that coloration that some of the vinyl enthusiasts
> >> miss. Other just enjoy the larger actually produced dynamic range on
> >> vinyl.

> > Vinyl does not 'actually produce' a larger dynamic range, unless the
> > CD's dynamic range has been intentionally reduced.

> Correct. CD's tend to be mastered for less dynamic range than what was used
> on vinyl.

True for pop and rock, less so for jazz, much less so for classical.

Classical music recording has always set the standard for 'high fidelity'
sound. And there, the benefits of CD are still most apparent.



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

[Steven Sullivan]

In rec.audio.tech David Looser > wrote:
> "Steven Sullivan" > wrote in message
> ...
> > In rec.audio.tech David Looser > wrote:
> >
> >> 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
> >

> I try not to. I can only take so much of people obsessing over the
> improvement in sound quality they get by replacing the mains leads with
> silver-plated wire, or changing the make of GZ32 rectifier used, or some
> other minor (but usually expensive) alteration.

> IMO if a difference doesn't show up in a DBT it doesn't exist, whatever the
> audiophiles may claim. But if you've just bought an expensive new gizmo of
> course it's going to sound better *to you*.


IF an audiophile makes a claim of a certain difference, and then cannot
pass a DBT, then I consider it unlikely that he actually heard one.

As DBTs are scientific measures, with results analysed in terms of
probability, they never 'prove' in the vernacular sense, that no
difference could possibly exist. Science doesn't require that level of
'proof' anyway, to draw a reasonable conclusion. But 'audiophile' tend to
misconstrue this to mean that it's still likely that someone else could
hear a difference. IN fact, we don't know whether it's *likely*. We just
know that it is not ruled out. There is a huge difference there, one that
audiophiles gloss over when they criticize DBTs (and science
generally)--which they do with mind-numbing regularity on such forums.


> I'm no longer astonished at the claims made in such forums, but it would be
> straightforward to mount a DBT of CD transfers from vinyl made using 16 and
> 24 bit ADCs (everything else identical of course, including ADC
> architecture). If the DBT showed a clear preference for the 24-bit version I
> would be astonished, and withdraw my comments.

IIRC Bob Katz , a highly tech-savvy mastering engineer, has done REdbook
vs hi-rez rate comparisons with subjects in the engineering community, and
found that any differences were down to filters, not the rates themselves.
More recently, E. Brad Meyer and David Moran in JAES published results of
a long term, multi-subject, multi-gear blind comparison of SACD vs SACD
downconverted to Redbook rates, and found that even 'golden ear' listeners
cannot tell the difference, unless playback levels are very high.



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

[Arny Krueger]

"Eiron" > wrote in message

> Silk wrote:
>> On Wed, 12 Dec 2007 10:14:24 +0000, tony sayer wrote:
>>
>>> In article
>>> >, Arny
>>> Krueger
>>
>>>> One of the finest MM cartridges ever made still costs
>>>> less than $100. Which is please?...
>>
>> AT95E or even a Linn rebadged one in the form of the K9
>> if you really must pay more ;-)

> How does that compare to Arny's favourite, the Shure
> M97xE?

For the record, the M97xE is not my favorite, the V15-VxMR was.

[why bother]

"tony sayer" > wrote in message
...
> In article >, Arny Krueger

This is a UK froop, the last poster we want in here is
Arny Krueger, whatever the subject, no matter the
thread evolution AK will eventually resort to his
cyberstalking of John Aitkinson, and Arny's ******** about
ABX and his highly dubious contentions...yawn.
If you're placing any credibilty in Arny's opnions or views
then go ask him in the froops he haunts, he's not wanted
here in the land of the sane and knowledgeable.

[Adrian]

On Dec 11 2007, 10:37*am, "Arny Krueger" > wrote:
> "Richard Crowley" > wrote in message
>
>
>
>
>
>
>
> > "Dave W." *wrote ...
> >> *Adrian*wrote:
> >>> This past weekend I copied three albums. *The signal is
> >>> clean but not strong. *I have the gain on theUSBPhono
> >>> turned to the max.
>
> >> If, as you say, the signal is clean, then as long as you
> >> have it digitised there is no problem.
>
> > Lets review the bidding....
>
> > * *Low-output MC cartridge feeding an inexpensive RIAA
> > phono preamp designed for MC.
> > * *Gain on the preamp "turned to the max".
> > * *Signal is "clean but not strong"
> > Therefore, by definition, the captured signal is NOT
> > "clean" after amplifying it (plus the noise) to the
> > nominal level.
>
> Agreed.
>
> > Of course,Adriancould decide that it is good enough
> > for his purposes, and that is fine. *But conventional
> > wisdom would suggest that the solution might be...
> > 1) Use a conventional MM cartridge
> > 2) Use a step-up transformer or pre-pre-amp for MC
> > 3) Use a preamp designed for MC.
>
> I'd vote for solution number 1, more specificially this cartridge:
>
> http://www.amazon.com/Shure-M97xE-High-Performance-Magnetic-Cartridge...
>
> Cheapest way out and solves more problems.

Solution implemented. The results, to quote Pop Larkin are "Perfick",
or at least somewhat as close as we ever get to perfect in the world
of audio reproduction. :-)

Many thanks

Adrian