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Recording line in with Soundforge, and Audiophile 24/96
evereux
post Aug 10 2003, 10:34
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I'm having a problem recording the output from my CD player through the line in of my soundcard, the Midiman Audiophile (Window's XP).

The problem is distortion. I can't seem to find a way to attenuate the line level input in soundforge and this is causing clipping.

I'm beginning to wonder if this can be removed since during normal playback (through PC) I also get audible clipping, even when the line in is attenuated using Midiman's control panel. I suspect the Midiman doesn't like the output levels of my CD player?

I'm quite sure the input signal itself isn't a problem since playback direct through my amplifier reveals no clipping.

Any ideas?


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Pio2001
post Aug 10 2003, 18:01
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The line in level is not in Soundforge, it must be somewhere in the midiman control panel.
Reduce it to a very low level and record a bit of audio.
If it is reduced, but is still clipped, like this :



but the wav directly ripped from the CD is not :



Then the soundcard input can't bear the signal outputted from the CD player. If the rip from the CD is clipped too



then the recording is correct, and the mastering of the CD is faulty.

I've got the same problem from my vinyl turntable. Using a DJ cartridge from Stanton, the line out level from the phono section of the ampli is insanely high, and the Marian Marc soundcard can't handle it, though it can record from a CD Player without problem.

If this is your problem, see if you have a variable output on the CD player, that can play quieter than the fixed one (monitored by the phone output level). If not, use the headphone output, if it has a level control. If not, try to use a recorder (cassette recorder, CD Burner...) between the CD Player and the soundcard, in record mode, so as to decrease the level with its recording level controls.

Anyway, if indeed the line in clips regardless of the line in level setting, you should complain about it at the technical support. This is a very annoying defect.
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evereux
post Aug 10 2003, 19:25
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QUOTE
If it is reduced, but is still clipped


It is definitely still clipped, it's quite audible. Turning down H/W in 1/2 slider on the control panel reduces the playback volume but doesn't affect the soundforge input volume.

QUOTE
but the wav directly ripped from the CD is not


The ripped wav is not clipped, at least it's not audible although there are other problems with it. The reason I'm trying to rip the audio in this manner is since it's copy protected, I've posted about it here.

Here are two samples:

sample-ana.flac
sample-rip.flac

Interestingly I've tried another couple of CDs and they playback fine, although they may not of been good tests since the volume wasn't that high on them. I'll try and see if I can find some other's that do clip.

QUOTE
If this is your problem, see if you have a variable output on the CD player, that can play quieter than the fixed one (monitored by the phone output level). If not, use the headphone output, if it has a level control.


Yeah, this was my next plan of action. Was hoping I didn't have to do that though.


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Pio2001
post Aug 10 2003, 21:40
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I think the solution is to find the line in recording level. It must be a different slider than the line in playback level. On old Sb cards, there is an option to switch between "playback" mixer and "record" mixer. They look the same, but only the later affects SoundForge recording level. The former just ajusts the monitoring volume into the line out.

There is no problem until you actually see clipping below 0 db, like in the first picture I posted.
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ancl
post Aug 11 2003, 00:01
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I have an Audiophile 24/96 myself, and as far as i now there is no way of changing the line in level. There are no volume controls built in the card itself - all adjustments have to be done in software.
The only solution to your problem is to somehow lower the signal before it reaches the card.
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Pio2001
post Aug 11 2003, 11:40
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¿ WTF ?!

I'm not going to get rid of my DAT deck anytime soon (good external ADC )
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Brajanath
post Nov 16 2003, 05:38
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Hi,

I'm new here (just started yesterday), and am still finging my way around on how to post/quote a message. I'm getting confused...

The reason why I joined this forum is because of a problem I had recently with my turntable/vinyl recording setup. I was searching the web for any link on how to get around with a problem with too much gain from my amplifier output that that feeds into my computer line-in, and I got right smuck into this message.
I changed to a more expensive turmtable and I got stuck with so much goodness!?

My windows mixer recording volume control is all the way down, but still I get too much signal that saturates my recordings. I had to connect in series an old tape recorder (as an interface) to control the signal. The Tape recorder is set at stanby record mode, then I use the recorder volume control to set the ideal output, which really works. But is there better way? I'm using a cheap Soundcard, that has no line-in, input controls like the Soundblaster cards. Maybe a third party software that acts as an intermidiary between Windows and the soundcard???

Before I proceed, lemme first know if I'm going out with this message.

-Dan
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fewtch
post Nov 16 2003, 08:39
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QUOTE (Brajanath @ Nov 15 2003, 09:38 PM)
My windows mixer recording volume control is all the way down, but still I get too much signal that saturates my recordings.  I had to connect in series an old tape recorder (as an interface) to control the signal. The Tape recorder is set at stanby record mode, then I use the recorder volume control to set the ideal output, which really works.  But is there better way?  I'm using a cheap Soundcard, that has no line-in, input controls like the Soundblaster cards.  Maybe a third party software that acts as an intermidiary between Windows and the soundcard???

Not really -- if the problem is a too-loud analog signal (too much voltage) overloading the card, then you need an attenuator of some sort in the analog domain. A couple fixed-value resistors per channel would do it, or a passive variable resistor (potentiometer).

Maybe a cheap mixing board from Radio Shack?

This post has been edited by fewtch: Nov 16 2003, 08:44


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Pio2001
post Nov 16 2003, 16:22
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QUOTE (Brajanath @ Nov 16 2003, 05:38 AM)
Before I proceed, lemme first know if I'm going out with this message.

Your message is perfectly OK.
Thank you for letting us know that you've got the same problem as Evereux and I. People seem to find it normal, but I'm not aware of any line input with this problem outside a computer, and if I have not had the DAT deck solution, I would have returned the card to the shop.
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Brajanath
post Nov 16 2003, 16:56
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[quote=Pio2001,Nov 16 2003, 07:22 AM] [/QUOTE]
Your message is perfectly OK.
I have not had the DAT deck solution, I would have returned the card to the shop.

[/quote]

The setup really works, and I've been using it during my analog days in recording cassette tapes. Which would you prefer, using the earphone output of your amplifier (this way you control the output signal w/ the amplifier Volume control) or the cassette recorder interface? Thanks for your reply.

-Dan
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Pio2001
post Nov 16 2003, 17:10
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I don't know...
The problem with the cassette deck is that it might have much background noise (EDIT oh, by the way, no problem if the source is a vinyl).
If the ampli headphone output is a 6.5 mm TRS jack, I'd go with it. But not if it is a small 3.5 mm one.

This post has been edited by Pio2001: Nov 16 2003, 17:11
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Brajanath
post Nov 16 2003, 17:20
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QUOTE (fewtch @ Nov 15 2003, 11:39 PM)
Not really -- if the problem is a too-loud analog signal (too much voltage) overloading the card, then you need an attenuator of some sort in the analog domain.  A couple fixed-value resistors per channel would do it, or a passive variable resistor (potentiometer).

Maybe a cheap mixing board from Radio Shack?

I've heard of Soundcards with provision for adjusting the line-in signal. If this is true, do you think it is a better approach to control the signal from within the soundcard (digital) than attenuate the strong signal from the analog source. We took all the trouble of amplifying the minute signal from our record player cartridge, only to attenuate and dump some of it...

I couldn't believe that the Windows mixer can't handle volume control problems like this. Thanks for your reply...

-Dan
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fewtch
post Nov 16 2003, 17:53
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QUOTE (Brajanath @ Nov 16 2003, 09:20 AM)
QUOTE (fewtch @ Nov 15 2003, 11:39 PM)
Not really -- if the problem is a too-loud analog signal (too much voltage) overloading the card, then you need an attenuator of some sort in the analog domain.  A couple fixed-value resistors per channel would do it, or a passive variable resistor (potentiometer).

Maybe a cheap mixing board from Radio Shack?

I've heard of Soundcards with provision for adjusting the line-in signal. If this is true, do you think it is a better approach to control the signal from within the soundcard (digital) than attenuate the strong signal from the analog source. We took all the trouble of amplifying the minute signal from our record player cartridge, only to attenuate and dump some of it...

I couldn't believe that the Windows mixer can't handle volume control problems like this. Thanks for your reply...

-Dan

If the card is being overloaded, nothing you do in the digital domain will fix the problem. The Windows Mixer is designed only to deal with input signals that are within spec for the soundcard. I don't know why your phono stage would be putting out a high enough level signal to cause problems (mine doesn't) but probably your best bet would be to deal with it in the analog domain.

BTW, if in fact you could decrease the "record" volume with the Windows mixer, you're attenuating and dumping some of the signal that way as well (at least as I understand it). Whether digitally or analog, if it's too loud then quiet it...

This post has been edited by fewtch: Nov 16 2003, 17:56


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JeanLuc
post Nov 16 2003, 18:07
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Of course you can adjust input sensitivity with the Delta AP 2496 Control Panel (I just build one into my dad's pc) ... and check whether the input level is set to consumer (should be) or +4dB ...

A normal CD player should produce an output voltage of 1,2 (low) - 2 Volts (standard) ... that's what these line inputs are tuned to. They can be further attenuated for the use of high-output studio equipment


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fewtch
post Nov 16 2003, 18:16
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QUOTE (JeanLuc @ Nov 16 2003, 10:07 AM)
Of course you can adjust input sensitivity with the Delta AP 2496 Control Panel (I just build one into my dad's pc) ... and check whether the input level is set to consumer (should be) or +4dB ...

I couldn't find any such input level setting with my AP 2496 (other than the monitor mixer), only an output level setting (Consumer/-10dBV). Where is it in the control panel applet?

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JeanLuc
post Nov 16 2003, 18:25
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It is the HW mixer which is situated besides the "patchbay/routing" tab (at least in the latest driver/control panel software) ... there are 8 mono sliders that can be linked to stereo (2 Analog - I/O and 2 SPDIF - I/O) which will adjust input and output levels ...


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Pio2001
post Nov 16 2003, 18:50
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QUOTE (fewtch @ Nov 16 2003, 05:53 PM)
I don't know why your phono stage would be putting out a high enough level signal to cause problems (mine doesn't)

Because there has been a volume race between DJ cartridge manufacturers, that now output louder than a standard hi fi cartridge. Also, if you play a 45 RPM maxi, the output is noticeably louder than with a 33 RPM record (more space to modulate the groove ?), not to mention the volume race on records themselves, than are not digitally limited.
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ancl
post Nov 16 2003, 18:52
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QUOTE (JeanLuc @ Nov 16 2003, 07:25 PM)
It is the HW mixer which is situated besides the "patchbay/routing" tab (at least in the latest driver/control panel software) ... there are 8 mono sliders that can be linked to stereo (2 Analog - I/O and 2 SPDIF - I/O) which will adjust input and output levels ...

I guess we all now about these but they control the output and not the input, and only in the digital domain.
The problem (or feature) with the Audiophile is that it can't change the amplitude of the analog signal before going to the D/A converter.

If the input signal to the soundcard is to high, the signal won't "fit" in the 24-bit digital range and will be clipped. What you do with the digital data after this does not matter - what is lost is lost. The only way to solve this is to lower the amplitude of the signal before it reaches the soundcard.
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JeanLuc
post Nov 16 2003, 18:55
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QUOTE (ancl @ Nov 16 2003, 05:52 PM)
I guess we all now about these but they control the output and not the input, and only in the digital domain.
The problem (or feature) with the Audiophile is that it can't change the amplitude of the analog signal before going to the D/A converter.

If the input signal to the soundcard is to high, the signal won't "fit" in the 24-bit digital range and will be clipped. What you do with the digital data after this does not matter - what is lost is lost. The only way to solve this is to lower the amplitude of the signal before it reaches the soundcard.

I still wonder why I can attenuate any analog input signal ...


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ancl
post Nov 16 2003, 19:01
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QUOTE (JeanLuc @ Nov 16 2003, 07:55 PM)
QUOTE (ancl @ Nov 16 2003, 05:52 PM)
I guess we all now about these but they control the output and not the input, and only in the digital domain.
The problem (or feature) with the Audiophile is that it can't change the amplitude of the analog signal before going to the D/A converter.

If the input signal to the soundcard is to high, the signal won't "fit" in the 24-bit digital range and will be clipped. What you do with the digital data after this does not matter - what is lost is lost. The only way to solve this is to lower the amplitude of the signal before it reaches the soundcard.

I still wonder why I can attenuate any analog input signal ...

You can (in the digital domain), but you can't get back the part of the analog signal that was to loud (because this require attenuation in the analog domain).
At least not on my AP2496 card...

Edit: clarification

This post has been edited by ancl: Nov 16 2003, 19:03
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JeanLuc
post Nov 16 2003, 19:09
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QUOTE (ancl @ Nov 16 2003, 06:01 PM)
You can, but you can't get back the part of the analog signal that was to loud.
At least not on my AP2496 card...

Well, routing in my case is as follows (since there is no external mixer available yet, the soundcard itself acts as a mixer) ...

Midi Keyboard I/O <-> Delta AP Midi I/O

Midi Keyboard Line Out -> Delta AP HW In

Delta AP HW Out -> Amplification

If the Keyboard produces an output that is too high, you will hear noticeable clipping artifacts emerging from the Delta AP's ADC ... if I slide back input level (HW In) some 6 dB, everything is fine (Soundforge recordings do not show any clipping artifacts) ...

So I just can't imagine why an analog signal cannot be attenuated to a non-clipping ADC output in that case ... it is just a matter of voltage

EDIT:

My Terratec EWX 2496 allows to adjust the ADC's gain level separately to match your (in this case: low) source voltage without the need of further software normalization (maybe this is what you mean)

This post has been edited by JeanLuc: Nov 16 2003, 19:21


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Brajanath
post Nov 16 2003, 22:54
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QUOTE (Pio2001 @ Nov 16 2003, 09:50 AM)
QUOTE (fewtch @ Nov 16 2003, 05:53 PM)
I don't know why your phono stage would be putting out a high enough level signal to cause problems (mine doesn't)

Because there has been a volume race between DJ cartridge manufacturers, that now output louder than a standard hi fi cartridge. Also, if you play a 45 RPM maxi, the output is noticeably louder than with a 33 RPM record (more space to modulate the groove ?), not to mention the volume race on records themselves, than are not digitally limited.

Ideally, Windows mixer if set to zero (0), should have NO signal, irrespective of any frontend load. If not, then an adjustment to set the threshold should be made available so we could easily cope up with component changes. Since no known software can do it, our old method works!

If there was no message (thread) of this nature in this forum, I would ne completely in the dark.

My original turntable was a Technics, and I recently acquired (from a Thrift Shop), this Thorens high End Turntable with an Ortofon cartridge. I'm not using a DJ cartrige, but this Ortofon sells for about $150. I don't have an external pre-amp or a dedicated mixer, but instead I just use the pre-amp of my stereo receiver. My recordings with this new Turntable is exceptionallt brilliant... OH, I tell ya... there is a marked difference in the quLity of the sound!! My old David Rose, Percy Faith, Mantovani, beautiful instrumentals LP records are so nice to listen to...

Thank you all, for your input... It's so interesting to read messages from you Guy's. It's eduacational, informative and refreshing!!!

-Dan
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fewtch
post Nov 16 2003, 23:22
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QUOTE (Brajanath @ Nov 16 2003, 02:54 PM)
My original turntable was a Technics, and I recently acquired (from a Thrift Shop), this Thorens high End Turntable with an Ortofon cartridge.  I'm not using a DJ cartrige, but this Ortofon sells for about $150. I don't have an external pre-amp or a dedicated mixer, but instead I just use the pre-amp of my stereo receiver.  My recordings with this new Turntable is exceptionallt brilliant... OH, I tell ya... there is a marked difference in the quLity of the sound!!  My old David Rose, Percy Faith, Mantovani, beautiful instrumentals LP records are so nice to listen to...

No surprise there... some (many?) people seem to be fooled by the way digital scales (i.e. cheap player sounds almost as good as a $1000 player)... the analog SQ curve is much smoother/more linear and doesn't hit the point of diminishing returns until much higher in price and/or quality. There are many reasons for this, and most of them involve resonance and vibration control (not only motor vibrations but tonearm bearings, plinth solidity, platter bearing, etc). In other words, the same cartridge can sound crappy on a cheaper turntable, and excellent on a better quality turntable.

Glad you're enjoying... smile.gif. I've got some of that Percy Faith and Mantovani type stuff lying around too, I went on a big lounge/EZ kick awhile ago and am still into 'space age pop' and exotica to some extent.

This post has been edited by fewtch: Nov 16 2003, 23:30


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Pio2001
post Nov 17 2003, 01:46
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QUOTE (JeanLuc @ Nov 16 2003, 07:09 PM)
So I just can't imagine why an analog signal cannot be attenuated to a non-clipping ADC output in that case ... it is just a matter of voltage

Let's give an example :

Maximum recordable level of the line input : 1 Volt. It clips the 24 bits ADC
Mixer setting : at -12 dB, the full scale of the 24 bits is scaled to the full scale of the digital file output (gain = 1)

In you case : output of your keyboard = 0.5 Volts
ADC : Full scale minus 6 dB
Mixer setting : 0 dB, gain = +12 dB
Result : -6+12=+6 dB
Your file is clipped by 6 dB

Now you lower the mixer setting to -6dB : gain = +6 dB
Result : -6+6=0 dB
Your recording is OK

In our case :
Output of the phono preamp : 2 Volts
ADC : clipped by 6 dB
Mixer setting : 0 dB (gain = +12 dB)
Result : 6 dB clipping, then 12 more dB clipping.

We lower the mixer setting at -24 dB (gain = -12 dB)
Result : 6 dB clipping remapped to -12 dB, like in the first picture at the top of this thread.
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Kos
post Apr 21 2006, 09:56
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Thiis is probably very banale for you in here, but I found a way to reduce the way-too-high volume level coming from my turntable-receiver end to my pc when vinyl ripping: I simply switched from MC (moving coil cartridge) til MM (moving magnet cartridge) on the back of my NAD receiver and this reduces the volume to “normal”.

Regards,
Kristian
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