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Bernard Bernard is offline
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Default need help to set up a M-Audio Audiophile 2496

Hi to Everyone,

Could someone help me managing my new audio card ?

It operates under Debian Linux, but I also tested it under Windows XP,
with same results. The input volume for digitilizing vinyl records is too
low, and I can't enhance it. On the contrary, with my old audio card that
was part of my ASUS motherboard, I was getting to high input volumes and I
had to lower the input to 7 to 13% of the maximum so as to avoid
saturating, such low settings were not in the green area for recording.

Here is what I get with Alsamixer :

Alsamixer v1.0.13
Card: M Audio Audiophile 24/96
Chip: ICE 1712 - multitrack
View: Playback Capture [All]
Item : list follows:

Item: IEC958 set on H/W In 0

Item: IEC958 Multi. Was in 'L CAPTUR R' mode, but I de-activated it, to no
avail. I am not in multichannel mode, only in stereo.

Item: IEC958 Multi 1: same thing

Item: IEC958 1 set to 'H/W In 1'

other options for IEC958 et IEC958 1 were H/W In 2 ... H/W In 7,
IEC958 In L, IEC958 In R, PCM Out, Digital Mixer.

Item: ADC (Analog to Digital Controller ??) set to 55%, dB gain =
-18.5 (in the green area)

Item: ADC1: idem : dB gain=-18.5

Item: ADC Capture. dB gain=-23 (in the green)

Item: ADC1 Capture. dB gain=-23 (in the green)

Item: DAC (Digital to Analog Controller ??) dB gain=-23 idem

Item: DAC Capture -23 dB

Item: DAC1 -23 dB

Item DAC1 Capture -23 dB

Item: Deemphasis. Set to 48 kHz. Other options : 32, 44.1,
OFF

Item: H/W set to H/W In 0

Item: H/W1 set to H/W In 1

other options : H/W In 2, H/W In 3,
.... H/W In 7, IEC958 In L, IEC958 In R

Item: H/W Multi L CAPTUR R CAPTUR de-activated

Item: H/W Multi1 L CAPTUR R de-activated

Items: Multi, Multi1, Multi2... Multi9 set to Mute (MM)

Item: Multi Track Internal Clock : 44100

Item: Multi Track Internal Clock Default : 44100

Item: Multi Track Peak 5656

Item: Multi Track Peak 6969 (two homonymous items, same for the next two)

Item: Multi Track Peak. Here a single figure : 69

Item: Multi Track Peak. Same thing, 69

Item: Multi Track Capture. Set to 86, against my will. The green area
starts at 39, but if I rise the setting, it comes back to low values after
restarting 'alsamixer'. True enough, I am not in multichannel mode, so
chances are that this setting is obsolete here.

Items: Third and fourth 'multi track peak'

Item: Multi Track Rate Locking. 2 options : Mute ou 00.

Item: Multi Track Rate Reset: idem

Item: Multi Track Volume Rate : 67


Thanks in advance for comments and advises.
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Richard Crowley Richard Crowley is offline
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Default need help to set up a M-Audio Audiophile 2496

Bernard wrote:
Could someone help me managing my new audio card ?

It operates under Debian Linux, but I also tested it under Windows XP,
with same results. The input volume for digitilizing vinyl records is
too low, and I can't enhance it.


You must use an RIAA phono preamp between the pickup
cartridge and the line input to the sound card. AFAIK, there
are NO computer audio cards which have inputs appropriate
for phono cartrdges (47K input impedance, high gain, and
the RIAA equalization curve).


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Bernard Bernard is offline
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Default need help to set up a M-Audio Audiophile 2496

Le Mon, 13 Jul 2009 13:13:14 -0700, Richard Crowley a écrit*:

Bernard wrote:
Could someone help me managing my new audio card ?

It operates under Debian Linux, but I also tested it under Windows XP,
with same results. The input volume for digitilizing vinyl records is
too low, and I can't enhance it.


You must use an RIAA phono preamp between the pickup cartridge and the
line input to the sound card. AFAIK, there are NO computer audio cards
which have inputs appropriate for phono cartrdges (47K input impedance,
high gain, and the RIAA equalization curve).


Yes, I do have a RIAA phono preamp between the two.
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Richard Crowley Richard Crowley is offline
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Default need help to set up a M-Audio Audiophile 2496

"Bernard" wrote ...
Richard Crowley a écrit :
Bernard wrote:
It operates under Debian Linux, but I also tested it under Windows XP,
with same results. The input volume for digitilizing vinyl records is
too low, and I can't enhance it.


You must use an RIAA phono preamp between the pickup cartridge and the
line input to the sound card. AFAIK, there are NO computer audio cards
which have inputs appropriate for phono cartrdges (47K input impedance,
high gain, and the RIAA equalization curve).


Yes, I do have a RIAA phono preamp between the two.


Then either the phono preamp output is too low or the 2496
card input gain is too low. The common differential diagnosis
techniques would quickly identify which was the case.


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Dave Platt Dave Platt is offline
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Default need help to set up a M-Audio Audiophile 2496


In article ,
Richard Crowley wrote:

It operates under Debian Linux, but I also tested it under Windows XP,
with same results. The input volume for digitilizing vinyl records is
too low, and I can't enhance it.

You must use an RIAA phono preamp between the pickup cartridge and the
line input to the sound card. AFAIK, there are NO computer audio cards
which have inputs appropriate for phono cartrdges (47K input impedance,
high gain, and the RIAA equalization curve).


Yes, I do have a RIAA phono preamp between the two.


Then either the phono preamp output is too low or the 2496
card input gain is too low. The common differential diagnosis
techniques would quickly identify which was the case.


One possibility, I suppose, is that the OP is using a low-output
moving-coil cartridge. These require some form of additional
step-up device (either a transformer, or a "head amp") prior to the
normal RIAA preamplifier.

It might also just be a relatively low-level moving-iron or
moving-magnet cartridge.

Another possibility is that the RIAA preamp was designed for the older
consumer standard ("full output" being 1 volt peak-to-peak) but that
the Audiophine 2496 is designed for the newer (CD-era) consumer
standard ("full scale input" being 2 volts peak-to-peak). This would
result in a 50% reduction in apparent input level.

An additional question for the OP: just how much "too low" is the
signal level once you've digitized it?

If the highest peak volume level you're seeing on your LPs is, say,
up to 25% of full-scale, then I'd actually recommend that you not
fiddle at all with the input gain. If you need to normalize, do so in
the digital domain after capturing.

Here's why.

There can be a fairly wide variation in the peak signal level of LPs.
Different manufacturers "cut" the albums differently. There's a
complex tradeoff between maximum cutting level, play time per side,
background noise level, and distortion. I've had some particularly
aggressive albums whose peak levels seemed to be almost 10 dB above
that of albums with a quieter cutting (although this is a fairly
extreme case).

In my opinion, when digitizing LPs, the most important thing to do is
make sure that the highest peak level, on the "hottest" LP, doesn't
clip. Clipping sounds really nasty, and cannot be corrected properly.

The dynamic range of even a 16-bit audio capture is significantly
greater than the dynamic range of an LP... even that of a
mint-condition, never-before-played LP pressed on really good virgin
vinyl. You can afford to throw away a couple of bits of dynamic range
at the top end, by having your average input level down at half- or
quarter-scale, in order to avoid the risk of clippping; the quietest
passages on the LP, and the residual surface noise will still be
comfortably far above the 16-bit quantization threshold.

This is even more true if you have a "24-bit" converter, and can
capture audio in a 24-bit format. I say "24-bit" because such
converters usually don't have a true 24-bit dynamic range... last time
I looked the better ones were in the 21- to 22-bit range.
Nevertheless, there's plenty of range here - quite a bit more than an
LP requires.

Best recipe: capture in a 24-bit format (with the input gain adjusted
to keep the worst-case peak on a really loud record some distance
below full-scale), edit (clean up pops and ticks), then convert to
16-bit format (gain-boosting to normalize while still avoiding
clipping, if you wish) using a decent dithering algorithm.

--
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!


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Peter Larsen[_3_] Peter Larsen[_3_] is offline
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Default need help to set up a M-Audio Audiophile 2496

Bernard wrote:

Could someone help me managing my new audio card ?

It operates under Debian Linux, but I also tested it under Windows XP,
with same results. The input volume for digitilizing vinyl records is
too low, and I can't enhance it. On the contrary, with my old audio
card that was part of my ASUS motherboard, I was getting to high
input volumes and I had to lower the input to 7 to 13% of the maximum
so as to avoid saturating, such low settings were not in the green
area for recording.

Here is what I get with Alsamixer :


Use m-audio's driver software. You may prefer not to use the newest version,
the newest version does not let you record the output of the monitor mixer
as I recall this.

It is generally preferable to use a complete hifi-preamp (or "pre-out" from
an integrated with a patch point beteeen preamp and poweramp) and use the
output to power-amplifier, that gives you a record volume control. The next
best thing may be to use the headphone output. Tape out may however be ok
and it is likely to be the cleanest sounding choice. It doesn't really
matter with a 24 bit card if the recorded audio generally is at -10 to -6
dB, something that is quite likely with this card and a ""standard riaa amp
and cartridge"", it is still cleaner to adjust level afterwards AND you DO
need to record all scratches cleanly.

The concept of an AD conversion card as this is is that there is that it is
a line in card with no controls, the only settings are for matching the
output level.

Alsamixer v1.0.13


Unless this software is broken it should work to select the cards inputs in
the recording software and leave everything in their default settings. My
recollection is - it may be wrong - that there is linux driver and software
available on m-audio's site, perhaps you should use that.

Thanks in advance for comments and advises.


Kind regards

Peter Larsen




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Alan[_3_] Alan[_3_] is offline
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Default need help to set up a M-Audio Audiophile 2496

Bernard wrote:
Hi to Everyone,

Could someone help me managing my new audio card ?

It operates under Debian Linux, but I also tested it under Windows XP,
with same results. The input volume for digitilizing vinyl records is too
low, and I can't enhance it. On the contrary, with my old audio card that
was part of my ASUS motherboard, I was getting to high input volumes and I
had to lower the input to 7 to 13% of the maximum so as to avoid
saturating, such low settings were not in the green area for recording.

Here is what I get with Alsamixer :

Alsamixer v1.0.13
Card: M Audio Audiophile 24/96
Chip: ICE 1712 - multitrack
View: Playback Capture [All]
Item : list follows:

Item: IEC958 set on H/W In 0

Item: IEC958 Multi. Was in 'L CAPTUR R' mode, but I de-activated it, to no
avail. I am not in multichannel mode, only in stereo.

Item: IEC958 Multi 1: same thing

Item: IEC958 1 set to 'H/W In 1'

other options for IEC958 et IEC958 1 were H/W In 2 ... H/W In 7,
IEC958 In L, IEC958 In R, PCM Out, Digital Mixer.

Item: ADC (Analog to Digital Controller ??) set to 55%, dB gain =
-18.5 (in the green area)

Item: ADC1: idem : dB gain=-18.5

Item: ADC Capture. dB gain=-23 (in the green)

Item: ADC1 Capture. dB gain=-23 (in the green)

Item: DAC (Digital to Analog Controller ??) dB gain=-23 idem

Item: DAC Capture -23 dB

Item: DAC1 -23 dB

Item DAC1 Capture -23 dB

Item: Deemphasis. Set to 48 kHz. Other options : 32, 44.1,
OFF

Item: H/W set to H/W In 0

Item: H/W1 set to H/W In 1

other options : H/W In 2, H/W In 3,
... H/W In 7, IEC958 In L, IEC958 In R

Item: H/W Multi L CAPTUR R CAPTUR de-activated

Item: H/W Multi1 L CAPTUR R de-activated

Items: Multi, Multi1, Multi2... Multi9 set to Mute (MM)

Item: Multi Track Internal Clock : 44100

Item: Multi Track Internal Clock Default : 44100

Item: Multi Track Peak 5656

Item: Multi Track Peak 6969 (two homonymous items, same for the next two)

Item: Multi Track Peak. Here a single figure : 69

Item: Multi Track Peak. Same thing, 69

Item: Multi Track Capture. Set to 86, against my will. The green area
starts at 39, but if I rise the setting, it comes back to low values after
restarting 'alsamixer'. True enough, I am not in multichannel mode, so
chances are that this setting is obsolete here.

Items: Third and fourth 'multi track peak'

Item: Multi Track Rate Locking. 2 options : Mute ou 00.

Item: Multi Track Rate Reset: idem

Item: Multi Track Volume Rate : 67


Thanks in advance for comments and advises.


Bernard, you are treading a well worn path. I have been doing what you
describe for a number of years with a moving coil cartridge, NAD PP2 pre
amp and the software levels set at max. Regardless of operating system,
I use my analog mixer between the pre amp and M-Audio card to raise the
level and often to raise the treble slightly.

Best wishes,

Alan.
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Ken[_8_] Ken[_8_] is offline
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Default need help to set up a M-Audio Audiophile 2496

On Mon, 13 Jul 2009 13:13:14 -0700, "Richard Crowley"
wrote:

Bernard wrote:
Could someone help me managing my new audio card ?

It operates under Debian Linux, but I also tested it under Windows XP,
with same results. The input volume for digitilizing vinyl records is
too low, and I can't enhance it.


You must use an RIAA phono preamp between the pickup
cartridge and the line input to the sound card. AFAIK, there
are NO computer audio cards which have inputs appropriate
for phono cartrdges (47K input impedance, high gain, and
the RIAA equalization curve).



RIAA computer audio cards
http://www.terratec.net/en/products/...inyl_2064.html
http://www.hama.de/portal/searchSele...bySearch*40515

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Bernard Bernard is offline
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Default need help to set up a M-Audio Audiophile 2496

Le Tue, 14 Jul 2009 07:07:00 +0100, Peter Larsen a écrit*:

Bernard wrote:

Could someone help me managing my new audio card ?

It operates under Debian Linux, but I also tested it under Windows XP,
with same results. The input volume for digitilizing vinyl records is
too low, and I can't enhance it. On the contrary, with my old audio card
that was part of my ASUS motherboard, I was getting to high input
volumes and I had to lower the input to 7 to 13% of the maximum so as to
avoid saturating, such low settings were not in the green area for
recording.

Here is what I get with Alsamixer :


Use m-audio's driver software. You may prefer not to use the newest
version, the newest version does not let you record the output of the
monitor mixer as I recall this.


Ever since I wrote my first post, I have found this link, where my audio
card is being discussed by users. All comments are positive, except for
the first one on the list, and it just happens that the guy purchased that
card for the same purpose as mine, that is, digitilizing vinyl records. He
also found input levels too low, and digital 'burps' in his audio samples
(I don't know what that is, and I doubt if I have these). He also says
that that card did a poor sampling job, compared to his former audio card.
All other comments are positive, but no one else seems to have digitilized
vinyls with that card...

http://www.newegg.com/Product/Produc...e=1 00&Page=1

However, it is not sure whether this guy's problems would reproduce on my
system. At the end of this post, I will propose links for a few 30 seconds
samples that I obtained : maybe someone could then tell me if he or she
notices such digital burps or other defects besides scratches, and what to
think of the quality of sampling and of background noise.


It is generally preferable to use a complete hifi-preamp (or "pre-out"
from an integrated with a patch point beteeen preamp and poweramp) and
use the output to power-amplifier, that gives you a record volume
control. The next best thing may be to use the headphone output.


Great idea : I just tested it ! Whatever comes from this output has gone
through the amplifier, while what comes from Tape Out has only gone
through the RIAA adjustment. So, I can enhance the output level so that it
gets close to 0 dB for treatment and record. As you point out,
processing that way may not be the best strategy for quality, and the
"phones" output is obviously not meant for that purpose. I propose
samples below :

http://www.teaser.fr/~bdebreil/musiq...Mozart.wav.bz2

is a 30 sec sample of a vinyl record that has been digitilized using my
M-Audio 2496 audio card. No treatment was applied to the files, except an
export to 32bits float wav file (and then losslessly compressed using
bzip2)

http://www.teaser.fr/~bdebreil/musiq...t_ampl.wav.bz2

is the same music sample, but I have amplified it of 7.3 dB using audacity.

http://www.teaser.fr/~bdebreil/musiq...phones.wav.bz2

Is another digitilizing of the same music sample, processed from the
headphones output, amplification adjusted on the hardware amplifier so as
to approach 0 dB. No treatment was done afterwards.

Thanks in advance for your comments.

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Peter Larsen[_3_] Peter Larsen[_3_] is offline
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Default need help to set up a M-Audio Audiophile 2496

Bernard wrote:


Ever since I wrote my first post, I have found this link, where my
audio card is being discussed by users. All comments are positive,
except for the first one on the list, and it just happens that the
guy purchased that card for the same purpose as mine, that is,
digitilizing vinyl records. He also found input levels too low,


Give it more input then. Seriously: it is already too sensitive to reall
give you 24 bit AD conversion because of the analog noise floor.

and
digital 'burps' in his audio samples (I don't know what that is, and
I doubt if I have these).


That guy has something else broke in his setup, perhaps an on board graphics
card, perhaps something networking that needs to be disabled in that
specific box. Or his actual card is defective out of the box.

He also says that that card did a poor
sampling job, compared to his former audio card. All other comments
are positive, but no one else seems to have digitilized vinyls with
that card...


It is the card I use and I use it as I suggested that you should, ie. with a
japanese preamp from the age o the quality wars and via tape out, levels
match perfectly, ie. I can digitize the loudest scratch without clipping and
it is REQUIRED to be able to do just that if you want optimum results.

Thanks in advance for your comments.


I didn't read the rest ... busy morning, bear with me.

Kind regards

Peter Larsen




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Richard Crowley Richard Crowley is offline
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Default need help to set up a M-Audio Audiophile 2496

"Ken" wrote ...
"Richard Crowley" wrote:
Bernard wrote:
Could someone help me managing my new audio card ?

It operates under Debian Linux, but I also tested it under Windows XP,
with same results. The input volume for digitilizing vinyl records is
too low, and I can't enhance it.


You must use an RIAA phono preamp between the pickup
cartridge and the line input to the sound card. AFAIK, there
are NO computer audio cards which have inputs appropriate
for phono cartrdges (47K input impedance, high gain, and
the RIAA equalization curve).



RIAA computer audio cards
http://www.terratec.net/en/products/...inyl_2064.html
http://www.hama.de/portal/searchSele...bySearch*40515



Not exactly plug-in I/O cards, but thanks.
OTOH, the sound card cited does NOT have RIAA inputs and
no mention of a phono preamp was made in the OP's initial post.


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Richard Crowley Richard Crowley is offline
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Default need help to set up a M-Audio Audiophile 2496

"Bernard" wrote ...
Ever since I wrote my first post, I have found this link, where my audio
card is being discussed by users. All comments are positive, except for
the first one on the list, and it just happens that the guy purchased that
card for the same purpose as mine, that is, digitilizing vinyl records. He
also found input levels too low,


Unless your card is broken I can't see how "digitizing vinyl records"
has anything to do with "levels too low". If you are using an RIAA
phono preamp which is putting out the proper line level, it should
drive the M-Audio sound card perfectly. I am quite certain that
hundreds (or even thousands) of people are digitizing vinyl records
with this M-Audio card with no problems at all.

Repeating: Test both your RIAA phono preamp AND your M-Audio
card with other equipment to determine if the preamp is outputing the
nominal consumer line level signal. And try some other source (such as
a CD player, etc.) as the source into the M-Audio card to establish
whether that is the problem.

Nothing you have written causes me to believe that there is anything
intrinsicly problematic with the M-Audio card that causes it to be
unsuitable for digitizing vinyl records. You appear to have a level-
setting problem and so far you have not reported doing any of the
recommended tests to establish where the problem lies. Only you
can do this experiment, we cannot guess how your hardware is
working, especially at this distance.

and digital 'burps' in his audio samples
(I don't know what that is, and I doubt if I have these). He also says
that that card did a poor sampling job, compared to his former audio card.
All other comments are positive, but no one else seems to have digitilized
vinyls with that card...

http://www.newegg.com/Product/Produc...e=1 00&Page=1


Nothing about that comment makes me believe there is anything
wrong with the M-Audio card. It seems quite obvious that the
person has not set up his hardware and/or software properly.

...I propose samples below :

http://www.teaser.fr/~bdebreil/musiq...Mozart.wav.bz2


If you post files that only Linux users can easily decrypt, then you
can expect answers from only Linux users and there are very very
few (if any?) of them here. You might have better success taking
your question over to a Linux newsgroup. But PLEASE do not
cross-post anything from Linux-land to this newsgroup. The past
results are consistently disasterous.


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Peter Larsen[_3_] Peter Larsen[_3_] is offline
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Default need help to set up a M-Audio Audiophile 2496

Richard Crowley wrote:

Nothing you have written causes me to believe that there is anything
intrinsicly problematic with the M-Audio card that causes it to be
unsuitable for digitizing vinyl records.


it is excellent for the purpose and fits the tape-out level on my preamp
perfectly when using a high output MC. The record level may not be louder
than what allows a clean recording of the loudest possible click on a
record.

Kind regards

Peter Larsen



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Bernard Bernard is offline
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Default need help to set up a M-Audio Audiophile 2496

Le Wed, 15 Jul 2009 10:34:01 -0700, Richard Crowley a écrit*:

"Bernard" wrote ...
Ever since I wrote my first post, I have found this link, where my audio
card is being discussed by users. All comments are positive, except for
the first one on the list, and it just happens that the guy purchased
that card for the same purpose as mine, that is, digitilizing vinyl
records. He also found input levels too low,


Unless your card is broken I can't see how "digitizing vinyl records" has
anything to do with "levels too low". If you are using an RIAA phono
preamp which is putting out the proper line level, it should drive the
M-Audio sound card perfectly. I am quite certain that hundreds (or even
thousands) of people are digitizing vinyl records with this M-Audio card
with no problems at all.

Repeating: Test both your RIAA phono preamp AND your M-Audio card with
other equipment to determine if the preamp is outputing the nominal
consumer line level signal. And try some other source (such as a CD
player, etc.) as the source into the M-Audio card to establish whether
that is the problem.

Nothing you have written causes me to believe that there is anything
intrinsicly problematic with the M-Audio card that causes it to be
unsuitable for digitizing vinyl records. You appear to have a level-
setting problem and so far you have not reported doing any of the
recommended tests to establish where the problem lies. Only you can do
this experiment, we cannot guess how your hardware is working, especially
at this distance.

and digital 'burps' in his audio samples (I don't know what that is, and
I doubt if I have these). He also says that that card did a poor
sampling job, compared to his former audio card. All other comments are
positive, but no one else seems to have digitilized vinyls with that
card...

http://www.newegg.com/Product/Produc...e=1 00&Page=1


Nothing about that comment makes me believe there is anything wrong with
the M-Audio card. It seems quite obvious that the person has not set up
his hardware and/or software properly.

...I propose samples below :

http://www.teaser.fr/~bdebreil/musiq...Mozart.wav.bz2


If you post files that only Linux users can easily decrypt, then you can
expect answers from only Linux users and there are very very few (if any?)
of them here. You might have better success taking your question over to a
Linux newsgroup. But PLEASE do not cross-post anything from Linux-land to
this newsgroup. The past results are consistently disasterous.


I don't know anything about 'Linux-land', assuming that this is not a
generic word. The files which I have proposed links for downloading, are
fully workable under MSWIN systems, as I have tested myself. 'Audacity',
which is the software I am using for my purpose, is available for MSWIN
operating system ; I even tested it under Windows XP. The only thing I
have mentioned is the uncompressing of bz2 files. I made sure that
'bunzip' existed as MSWIN version, but I am quite aware that not everyone
may have this software on his/her system. For your information and
that of everyone else in this group, I have rightfully purchased the
Windows XP version that I have installed in a partition of my HDs, even
though I only boot on this system maybe once a month or so : this is my
choice, and I have no complain about this. I am in favour of GNU software,
still being fully aware that this would not exist if such commercial work
as that of Microsoft had not existed.

If you or anyone else are willing to test those links, I am quite prepared
to propose other files without compression, so as to avoid installing
weird compressing software.

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Richard Crowley Richard Crowley is offline
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"Bernard" wrote ...
If you or anyone else are willing to test those links, I am quite prepared
to propose other files without compression, so as to avoid installing
weird compressing software.


I think it would be a very good idea for you to use a more familiar
file type if you want help from people (most of whom are NOT using
Linux). I was unable to find anything that would decode BZ2 files
under MS Windows.




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Le Wed, 15 Jul 2009 15:09:41 -0700, Richard Crowley a écrit*:

"Bernard" wrote ...
If you or anyone else are willing to test those links, I am quite
prepared to propose other files without compression, so as to avoid
installing weird compressing software.


I think it would be a very good idea for you to use a more familiar file
type if you want help from people (most of whom are NOT using Linux). I
was unable to find anything that would decode BZ2 files under MS Windows.


I am very sorry for this. I had just checked that bzip2 and bunzip2 did
exist for Windows platforms, but I should have thought that it may have
not been so easy and quick to get it and install.

I have prepared new files : 2 are uncompressed ; you will find them
adequate if you have a fast DSL connexion. Two more are zip compressed:
'Winzip' will handle them as did 'pkzip' and 'pkunzip' in the old days of
MSDOS 6.2. The original uncompressed files are about 10 mB, the zip files
about 6 mB (the bz2 were about 4.4 mB). Other compressing
lossless utilities that I know don't handle that sort of files (FLAC only
compress 16 bit audio files, with a ratio that is about 2:1)

Here are the new links below :

http://www/teaser.fr/~bdebreil/musiq...ozart_ampl.wav

http://www/teaser.fr/~bdebreil/musiq...art_phones.wav

http://www/teaser.fr/~bdebreil/musiq...t_ampl.wav.zip

http://www/teaser.fr/~bdebreil/musiq...phones.wav.zip
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Le Thu, 16 Jul 2009 14:33:16 +0200, Bernard a écrit*:

Le Wed, 15 Jul 2009 15:09:41 -0700, Richard Crowley a écrit*:

"Bernard" wrote ...
If you or anyone else are willing to test those links, I am quite
prepared to propose other files without compression, so as to avoid
installing weird compressing software.


I think it would be a very good idea for you to use a more familiar file
type if you want help from people (most of whom are NOT using Linux). I
was unable to find anything that would decode BZ2 files under MS
Windows.


I am very sorry for this. I had just checked that bzip2 and bunzip2 did
exist for Windows platforms, but I should have thought that it may have
not been so easy and quick to get it and install.

I have prepared new files : 2 are uncompressed ; you will find them
adequate if you have a fast DSL connexion. Two more are zip compressed:
'Winzip' will handle them as did 'pkzip' and 'pkunzip' in the old days of
MSDOS 6.2. The original uncompressed files are about 10 mB, the zip files
about 6 mB (the bz2 were about 4.4 mB). Other compressing lossless
utilities that I know don't handle that sort of files (FLAC only compress
16 bit audio files, with a ratio that is about 2:1)

Here are the new links below :

http://www/teaser.fr/~bdebreil/musiq...ozart_ampl.wav

http://www/teaser.fr/~bdebreil/musiq...art_phones.wav

http://www/teaser.fr/~bdebreil/musiq...t_ampl.wav.zip

http://www/teaser.fr/~bdebreil/musiq...phones.wav.zip


I am even more sorry for these links , especially for the typing mistake.
Here are the valid links below :

http://www.teaser.fr/~bdebreil/musiq...ozart_ampl.wav

http://www.teaser.fr/~bdebreil/musiq...art_phones.wav

http://www.teaser.fr/~bdebreil/musiq...t_ampl.wav.zip

http://www.teaser.fr/~bdebreil/musiq...phones.wav.zip







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Peter Larsen[_3_] Peter Larsen[_3_] is offline
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Bernard wrote:

I am even more sorry for these links , especially for the typing
mistake. Here are the valid links below :


http://www.teaser.fr/~bdebreil/musiq...ozart_ampl.wav


http://www.teaser.fr/~bdebreil/musiq...art_phones.wav


Thank you, it appears to me that you do not have a problem, but if you get
the forte at offset 5 seconds at -6 or even -10 dB via tape out, then that
is how you should record a record. As I have said previously you need to
allow for making a clean recording of the loudest scratch or click because
fixing it requires it to be properly recorded. You will also very soon start
wanting to fix the channel imbalance, there is no such thing as a "no
processing" transfer ....

What I do with vinyl is to equalize the actual cartridge + riaa combination
to linear based on playback of a bruel and kjaer test-record I got at some
equipment demonstration way long time ago and then I do a manual "fix single
click" of the large clicks and nothing else. I have decided on this strategy
after detecting just how many clicks automated click fixing routines will
find and fix in pristine recordings causing them to sound plain boring. As
in the old days: if clicks and "needle noise" is a problem in playback, then
you should fix the playback system.

Note: you should NOT try to compensate for the treble roll-off innermost on
the record, leave it be. With some records it seems to me that the
"innermost bands" are cut at lower level so as to make room for more music,
that I compensate some of of the time, but not always fully. You need to
respect the artistic choices, the Droit Morale.

I always digitize to 16 bit 96 kHz sample rate and downsample, my preferred
audio software has very good downsampling. It would have merit to digitize
to cd-format with the target being CD's, but generally no grammophone
playback is acceptable without fixing frequency response errors and gross
channel imbalance, I usually normalize to -2.5 dB after fixing large clicks
manually with "fix single click functionality". Software that doesn't offer
that functionality is toy software, but - that said - it does happen that I
use some kind of automated repair, generally I prefer the clean sound of
doing less.

Kind regards

Peter Larsen




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"Bernard" wrote ...
Here are the new links below :

http://www/teaser.fr/~bdebreil/musiq...ozart_ampl.wav

http://www/teaser.fr/~bdebreil/musiq...art_phones.wav


I am not sure what these files are an example of, but they both appear
to be completely normal to me. There is some very slight clipping on
some peaks around 3.4 seconds into the "phones" file. But if these are
the levels you are getting for the original recording, you are better at
setting levels than I am. I would have been more conservative unless
I had identified the peaks on the vinyl and set the levels accordingly.
(which may be what you did???)

DO NOT compare levels from a vinyl disc with anything coming from
a modern CD, even the same performance of the same music. Note
that the dynamics on modern CDs are MUCH more processed than
vinyl discs ever were.


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In article ,
Richard Crowley wrote:

DO NOT compare levels from a vinyl disc with anything coming from
a modern CD, even the same performance of the same music. Note
that the dynamics on modern CDs are MUCH more processed than
vinyl discs ever were.


Often true, and it's a dreadful shame. The CD medium has quite a bit
more dynamic range capability than vinyl... and most CDs are mastered
so much level compression that most of the dynamic range is squandered.

--
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
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Le Thu, 16 Jul 2009 16:36:28 +0100, Peter Larsen a écrit*:

Bernard wrote:

I am even more sorry for these links , especially for the typing
mistake. Here are the valid links below :


http://www.teaser.fr/~bdebreil/musiq...ozart_ampl.wav


http://www.teaser.fr/~bdebreil/musiq...art_phones.wav


Thank you, it appears to me that you do not have a problem, but if you get
the forte at offset 5 seconds at -6 or even -10 dB via tape out,


This is what I got at first,but the file I sent
has been amplified after recording, using the audacity tool for that
purpose, so it was closer to 0 dB.

then that
is how you should record a record. As I have said previously you need to
allow for making a clean recording of the loudest scratch or click because
fixing it requires it to be properly recorded. You will also very soon
start wanting to fix the channel imbalance, there is no such thing as a
"no processing" transfer ....

What I do with vinyl is to equalize the actual cartridge + riaa
combination to linear based on playback of a bruel and kjaer test-record I
got at some equipment demonstration way long time ago and then I do a
manual "fix single click" of the large clicks and nothing else. I have
decided on this strategy after detecting just how many clicks automated
click fixing routines will find and fix in pristine recordings causing
them to sound plain boring. As in the old days: if clicks and "needle
noise" is a problem in playback, then you should fix the playback system.

Note: you should NOT try to compensate for the treble roll-off innermost
on the record, leave it be. With some records it seems to me that the
"innermost bands" are cut at lower level so as to make room for more
music, that I compensate some of of the time, but not always fully. You
need to respect the artistic choices, the Droit Morale.

I always digitize to 16 bit 96 kHz sample rate and downsample,


So far, I have digitized at 32 bit float and 44.1 kHz, which were default
values on my software (audacity), (other possible values are 24 bits and
16 bits). Then, at 'export', I do a downgrading to 16 bits so as to comply
with CD format. The sample I sent was not downgraded, so it was 32 bit
float and 44.1 kHz. Do you think that I would be better digitizing to 16
bits at start, so as to avoid a resampling process ? I thought that
digitizing at 32 bits would allow a better sampling afterward, the job
being carried with a greater number of data... also, the help file of
audacity says : "it is recommended that you use 32-bit float unless you
have a slow computer or are running out of disk space". In any case, my
audio card is 24 bit. As for the sampling rate, this was not until a
few minutes ago that I tried changing the default value (44.1 kHz) to 96
kHz, which is the maximum value allowed with my audio card. I felt that
I also had to change the master clock using the audio card monitor, and
set it from 44100 int to 96000 int. The audacity help file says that the
sampling rate can be set to anything, and that tracks are automatically
resampled to the project sample rate.


my
preferred audio software has very good downsampling.


How do I know if 'audacity' does a good downsampling job ? If I uploaded
a 96000 Hz file, and a downsampled version of it at 44100 Hz, would you
have a mean to say if it went through a good or average or poor
downsampling job ? In 'audacity', you have a choice of two processes
for downsampling :

Real-time sample rate converter : set by default to 'fast interpolation'

High-quality sample rate converter : set by default to 'high quality
interpolation'.

Besides this, it has dithering facilities, which, on my audacity, are set
by default to :

Real-time dithering is set to 'none'. Other options are 'triangle',
'rectangular' and 'wave'

High quality dithering is set to 'Triangle'. Other options are
'rectangular', 'wave' and 'none'


It would have merit
to digitize to cd-format with the target being CD's, but generally no
grammophone playback is acceptable without fixing frequency response
errors and gross channel imbalance,


my "normalize" function "allows you to correct for DC offset (a vertical
displacement of the track) and/or to amplify such that...". Is this what
you call 'fixing the channel imbalance' ?

I usually normalize to -2.5 dB after
fixing large clicks manually with "fix single click functionality".
Software that doesn't offer that functionality is toy software, but - that
said - it does happen that I use some kind of automated repair, generally
I prefer the clean sound of doing less.


Clicks don't bother me that much either. I don't know whether audacity has
a 'fix single click' fonction. I think so, even though no function bears
that name. In order to manually remove a click that I can see on the
graph, I operate the 'zoom in' tool till I get, say, 0.05 sec per inch on
the screen. Then I set 'snap off', which de-activate the 'snap on' mode
which constrain the selection to the nearest interval, by default the
second. In Snap off mode, I can select very narrow ranges such as one
thousandth of a second or so, and cut that much, no more. Besides, there
is another facility that I have not tested yet. It is called 'find zero
crossings': 'modify the selection slightly so that both left and right
edge of the selection appear on a positive slope zero crossing. This make
it easier to cut and paste audio without resulting in a large audible
clicking sound'.

Thanks again for your very relevant help which I appreciate much

Kind regards

Bernard Debreil

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Le Thu, 16 Jul 2009 09:58:50 -0700, Richard Crowley a écrit*:

"Bernard" wrote ...
Here are the new links below :

http://www/teaser.fr/~bdebreil/musiq...ozart_ampl.wav

http://www/teaser.fr/~bdebreil/musiq...art_phones.wav


I am not sure what these files are an example of, but they both appear to
be completely normal to me. There is some very slight clipping on some
peaks around 3.4 seconds into the "phones" file. But if these are the
levels you are getting for the original recording,


But no... the original levels were much lower (about -8 dB at the peaks).
That file that I sent as example of has been processed through the
'audacity' amplifier (+7 dB if I recall well), this is why I used the word
'ampl' in the filename. Do you wish to see the original unprocessed file ?

http://www/teaser.fr/~bdebreil/musiq...art_noampl.wav

As for the other file (test_Mozart_phones.wav), the audio card was fed
from what came out of the "phones" (headphones out), therefore it went
through the amplifier... but this is supposedly not the right way to do if
you expect a clean record



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Bernard wrote:
But no... the original levels were much lower (about -8 dB at the
peaks). That file that I sent as example of has been processed
through the 'audacity' amplifier (+7 dB if I recall well), this is
why I used the word 'ampl' in the filename. Do you wish to see the
original unprocessed file ?

http://www/teaser.fr/~bdebreil/musiq...art_noampl.wav


That looks exactly right. I don't see anything wrong with that recording.

I sometimes visually scan a disc to see where the highest levels are
likely recorded, and then I set the recording level given that likely
peak. That sometimes lets me get a big higher level recording without
fear of clipping. But your example seems perfectly acceptable. I don't
know what you are complaining about? Again, DO NOT compare the
levels you get from a vinyl disc from anything recorded on a modern CD.


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Bernard wrote:

But no... the original levels were much lower (about -8 dB at the
peaks). That file that I sent as example of has been processed
through the 'audacity' amplifier (+7 dB if I recall well), this is
why I used the word 'ampl' in the filename. Do you wish to see the
original unprocessed file ?


http://www/teaser.fr/~bdebreil/musiq...art_noampl.wav


An excellent example of the preferred record level for set and forget
transfer of grammophone records. It is in my opinion malpractice to adjust
record level on a pr. record basis, there is so much that needs fixing even
on a high quality record that it is better to have a standard start level
for processing that allows you ample headroom.

As for the other file (test_Mozart_phones.wav), the audio card was fed
from what came out of the "phones" (headphones out), therefore it went
through the amplifier... but this is supposedly not the right way to
do if you expect a clean record


I assume the example is from tape out, and it is recorded perfectly for
processing, and there is headroom enough to allow a clean recording of the
loudest possible transient. Again, you DO need to record the clicks and
scratches cleanly, otherwise they can get more difficult to fix. Some people
go to agonizing lenghts to get perfection, in my opinion that is
misunderstood, vinyl never was silent anyway and I much prefer good violins
and small record scratches to plastic violins and fake digital silence.

Kind regards

Peter Larsen


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Steven Sullivan Steven Sullivan is offline
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Bernard wrote:
Le Tue, 14 Jul 2009 07:07:00 +0100, Peter Larsen a ?crit?:
the first one on the list, and it just happens that the guy purchased that
card for the same purpose as mine, that is, digitilizing vinyl records. He
also found input levels too low, and digital 'burps' in his audio samples
(I don't know what that is, and I doubt if I have these). He also says
that that card did a poor sampling job, compared to his former audio card.
All other comments are positive, but no one else seems to have digitilized
vinyls with that card...


I have. You need a preamp with adjustable gain inline, if you want to
manually raise input level. If you record from a D/A source,
like a CDP or DVDP, you may actually find you have to use a preamp to
LOWER the input gain, to avoid clipping.

Digital 'burps' only occurred for me when there was an IRC conflict with
my PC hardware. In my case it turned out to be a conflict with my
aftermarket wireless network card, which I now disable whenever I need to
record with the 2496. Other causes of digital pops and clicks are noted in
the Troubleshooting faq of the user;s manual. Inadequate DMA buffer size
is one.

The claim that the sampling (I assume he meant A/D conversion) is
inadequate is nonsense. The M-audio 2496 is an excellent card when it and
your system are configured correctly, as Rightmark testing can
demonstrate.

http://ixbtlabs.com/articles/maudioaudiophile/

It is not the most user-friendly card by any means, though. You do need
to read the manual. Such as this:

Variable Signal Levels. Here you can choose two volume levels of the
output signal: +2 dBV (Consumer) and -4 dBV (-10 dBV). The level of the
input signal is +2 dBV and can't be changed. It is obvious that for
recording from the linear-in it's necessary to use audio equipment with a
standard linear-out or a recording level controller of the external device
(for example, faders of the audio mixer). H/W 1/2 In in the Monitor Mixer
means that we can monitor a signal and adjust volume of the playback
level, but it doesn't affect a recording level.


--
-S
We have it in our power to begin the world over again - Thomas Paine


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My older posts dealt with too low levels for inputs in audacity... I now
got a better RIAA preamp with higher output, and all seemed OK... except
that I now have a vinyl record that gives too high output, and I have
problem recording/digitizing it.

I can't monitor the analog signal input using Audacity. The input comes from a
Cambridge Audio 640 P RIAA preamp, which is fed from a vinyl turntable
Thorens TD 90. The preamp does not have an output setting knob. In audacity,
there is a cursor that allows
to decrease the input... but it does not really work like I would expect,
since the decrease shows only on the spectrogram level, not on the meter
toolbar, which keeps showing same high ratings with some over zero, even
when output setting is set to 0 and the spectrogram is flat. Am I right
to think that, even if the recording level is lowered, the input is still
distorted because of too high levels ?

How am I to lower the volume of the signal fed into Audacity ? I have
achieved nothing trying to play with the sound card control
(envy24control under Linux), and I don't know really how to use it.
Alsamixer has done no better, and, most times what you lower in alsamixer
also gets lowered in envy24control. Here are the settings that I have in
envy24control :

Monitor Inputs panel :

H/W In 1 Left = 58 Right = Mute

H/W In 2 Left = Mute Right = 58

SPDIF In L Mute

SPDIF In R Mute

The H/W In 1 and 2 boxes show green bars that go up and down while the music
plays, sometimes in the red. Moving the cursors up and down to change the
figures does not change anything in the level of green bars in each box,
but it shows changes in the left panel called "digital mixer". The lower
the figure is for each channel, the higher the same channel goes in the
"digital mixer" box. At first I had thought that I should set each
channel to a figure that would allow the green bar in the digital mixer
to go as high as possible without entering the red zone... but I soon
figured out that this had no effect on the input level in audacity nor in
the output of my earphones. I can mute both H/W In 1 and H/W In 2
monitors, and the recording input in Audacity is still at maximum, as
well as output in earphones.

Monitor PCMs panel :

PCM Out 1 is "2" on Left channel, Mute on right channel,

PCM Out 2 is Mute on Left channel, 2 on right

PCM Out 3 to 8 are Mute


Patchbay Router panel :

H/W Out 1 (L) : the check is on H/W In 1

H/W Out 2 (R) : the check is on H/W In 2

SPDIF Out L : the check is on H/W In 1

SPDIF Out R : the check is on H/W In 2


Hardware settings panel :

Master Clock : Int 96000

Rate State : Locked and Reset both checked

Volume Change : Rate 170


Analog Volume panel :

DAC 0 = 87
DAC 1 = 90
ADC 0 = 96
ADC 1 = 96


Thanks in advance for helping me with envy24control settings.

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Bernard wrote:

My older posts dealt with too low levels for inputs in audacity...


Again: your levels then were perfect.

I
now
got a better RIAA preamp with higher output, and all seemed OK...
except
that I now have a vinyl record that gives too high output, and I have
problem recording/digitizing it.


You then have to reduce the signal level before the signal enters the
soundcard. Otherwise the sound card will clip. A simple 10 dB attenuator
will do, say 10 kOhm in series and 5 kOhm to ground in a small plastic box.

I can't monitor the analog signal input using Audacity. The input
comes from a Cambridge Audio 640 P RIAA preamp, which is fed from a
vinyl turntable Thorens TD 90. The preamp does not have an output
setting knob.


With grammophone cartridges having their output signals within a 10 dB
window it seems like folly to me not to have an output level adjustment
option, a -5 and -10 dB switch would have been wise design from the
manufacturer ...

Thanks in advance for helping me with envy24control settings.


Forget everything about adjusting the record level after digitizing, it
needs to be adjusted before enters the sound card or the sound card will
clip. Welcome to the simplicity of line level contraptions. Your problem is
partly that the AP2496 is adapted to home audio line levels and partly that
Cambrigde appear to have put extra gain in that RIAA to match the +8 dB
(ref. 0.775 volt) output from a CD-player.

If you need to vent your frustration, then write to Cambridge Audio and ask
them how come they forgot a -5 and -10 dB option on the output of the RIAA
box. It is well warranted to have that since cartridges for 47 kOhm loading
have their out spread within approximately 12 dB.

Kind regards

Peter Larsen




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David Nebenzahl David Nebenzahl is offline
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On 7/22/2009 12:13 PM Peter Larsen spake thus:

Bernard wrote:

I now got a better RIAA preamp with higher output, and all seemed
OK... except that I now have a vinyl record that gives too high
output, and I have problem recording/digitizing it.


You then have to reduce the signal level before the signal enters the
soundcard. Otherwise the sound card will clip. A simple 10 dB
attenuator will do, say 10 kOhm in series and 5 kOhm to ground in a
small plastic box.


I'd say better yet would be an adjustable attenuator, which is just a
potentiometer across the output of the preamp with the wiper connected
to the input of the sound card. This is what I use between my sound card
and my amplifier (Technics) which produces a slightly too-high signal. I
use a 50K pot.

Mine fits in a small metal box (like an Altoids tin).


--
Found--the gene that causes belief in genetic determinism
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David Nebenzahl wrote:

On 7/22/2009 12:13 PM Peter Larsen spake thus:


Bernard wrote:


I now got a better RIAA preamp with higher output, and all seemed
OK... except that I now have a vinyl record that gives too high
output, and I have problem recording/digitizing it.


You then have to reduce the signal level before the signal enters the
soundcard. Otherwise the sound card will clip. A simple 10 dB
attenuator will do, say 10 kOhm in series and 5 kOhm to ground in a
small plastic box.


I'd say better yet would be an adjustable attenuator, which is just a
potentiometer across the output of the preamp with the wiper connected
to the input of the sound card. This is what I use between my sound
card and my amplifier (Technics) which produces a slightly too-high
signal. I use a 50K pot.


There are many ways to do things, but you really should not go higher than
10 kOhm ... otherwise the cable capacity between box and sound card begins
to matter too much. The disadvantages of an adjustable record level in this
context are - with my work process - a lot larger than the advantages.

Kind regards

Peter Larsen



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Arny Krueger Arny Krueger is offline
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"Peter Larsen" wrote in message
k
David Nebenzahl wrote:

On 7/22/2009 12:13 PM Peter Larsen spake thus:


Bernard wrote:


I now got a better RIAA preamp with higher output, and
all seemed OK... except that I now have a vinyl record
that gives too high output, and I have problem
recording/digitizing it.


You then have to reduce the signal level before the
signal enters the soundcard. Otherwise the sound card
will clip. A simple 10 dB attenuator will do, say 10
kOhm in series and 5 kOhm to ground in a small plastic
box.


I'd say better yet would be an adjustable attenuator,
which is just a potentiometer across the output of the
preamp with the wiper connected to the input of the
sound card. This is what I use between my sound card and
my amplifier (Technics) which produces a slightly
too-high signal. I use a 50K pot.


There are many ways to do things, but you really should
not go higher than 10 kOhm ... otherwise the cable
capacity between box and sound card begins to matter too
much. The disadvantages of an adjustable record level in
this context are - with my work process - a lot larger
than the advantages.


Yes indeed. My standard line-level attenuation pot is a 5K 2 watt RN50 type.




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Default need help to set up a M-Audio Audiophile 2496

Peter Larsen wrote:

There are many ways to do things, but you really should not go higher
than 10 kOhm ... otherwise the cable capacity between box and sound
card begins to matter too much.


Not to quibble, but "capacitance"

;-)

geoff


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Peter Larsen[_3_] Peter Larsen[_3_] is offline
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Default need help to set up a M-Audio Audiophile 2496

geoff wrote:

Peter Larsen wrote:


There are many ways to do things, but you really should not go higher
than 10 kOhm ... otherwise the cable capacity between box and sound
card begins to matter too much.


Not to quibble, but "capacitance"


Thanks, will try to remember.

;-)


geoff


Kind regards

Peter Larsen



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Geoff Geoff is offline
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Default need help to set up a M-Audio Audiophile 2496

Peter Larsen wrote:
geoff wrote:

Peter Larsen wrote:


There are many ways to do things, but you really should not go
higher than 10 kOhm ... otherwise the cable capacity between box
and sound card begins to matter too much.


Not to quibble, but "capacitance"


Thanks, will try to remember.



Tha8t's fine - even some cable manufacturers don't seem to get the subtle
difference, and print it on cable jackets !

;-)

geoff


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