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?
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 :
(http://perso.numericable.fr/laguill2/pictures/clip1.gif)
but the wav directly ripped from the CD is not :
(http://perso.numericable.fr/laguill2/pictures/clip2.gif)
Then the soundcard input can't bear the signal outputted from the CD player. If the rip from the CD is clipped too
(http://perso.numericable.fr/laguill2/pictures/clip3.gif)
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.
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.
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 (http://ikonboard.napigator.com/ikonboard.cgi?s=3f3684bc0f4affff;act=ST;f=1;t=4697).
Here are two samples:
sample-ana.flac (http://homepage.ntlworld.com/pj.ntl/flac/sample-ana.flac)
sample-rip.flac (http://homepage.ntlworld.com/pj.ntl/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.
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.
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.
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.
¿ WTF ?! (http://perso.numericable.fr/laguill2/smileys/megaeek.gif)
I'm not going to get rid of my DAT deck anytime soon (good external ADC (http://perso.numericable.fr/laguill2/smileys/biggrin.gif) )
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
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?
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.
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
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.
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
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...
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
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?
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 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.
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.
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 ...
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
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)
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
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... . 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.
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.
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
I have to admit I have not read this very closely, but it seems there are some ideas that, in some sense the source equipment, or the soundcards, are not quite proper. I just want to point out that adjusting the input level is not a soundcard job and adjusting the output level is not normally a source device job.
I can't speak to what those gaming-multimedia cards might or might not do, but any kind of professional or semi-pro line level input expects a controlled line level input -- it is supposed to already be correct when it reaches the soundcard. Microphone preamps have a gain control, and that can be sufficient for microphone input, but phono preamps, CD player and tape deck line-outs, or any other such sources, do not normally have this control potential. Certainly it can exist as an add-on but it is not part of the normal ‘definition' of the source.
Analogue mixers are the standard professional method of controlling the signal level. Microphone preamps are also most commonly run throug mixers on their way to wherever they will be used or recorded. Mixers can often do quite a few different things. The part that is relevant here is the line level preamplifier. Any line level preamp can be used in exactly the same way but HiFi line level preamps are generally far more expensive than basic mixers and, for this kind of use, provide nothing significant for all that extra money. Of course, full featured, many channeled professional mixers can be very expensive, but ones that give full control, and very good results, with two channels are widely available and not terribly expensive.
I just want to point out that adjusting the input level is not a soundcard job
Yes it is. And nearly all soundcards do it, with the volume control of their "record mixer". In the same way, all tape decks can adjust the recording level. Some of them manually, with a volume control, and some of them automatically, with a built-in dynamics compressor.
The problem is the headroom available. According to this discussion ( http://www.audioasylum.com/forums/pro/messages/9971.html (http://www.audioasylum.com/forums/pro/messages/9971.html) ), the usual headroom in line inputs is around 20 or 25 dB. It means that they clip around 3 or 6 Volts. Soundcard seem to clip lower. This is just an indication of a poor performance, but it can completely prevent their use in some cases.
No that is incorrect. Some of those that use the Windows mixer may, but genuine audio cards do not have input level adjustments (I don't think any quality audio soundcard does use the Windows mixer). Their control panels only operate in the digital domain, after the signal is digitized. Maybe some of those Creative cards, and their clones, have an analogue adjustment preceding the ADC, but those are hardly considered serious recording devices. Only the most awful of the Creative cards have that nasty compressor/limiter.
No doubt there are one or two professional cards that have some adjustable analogue front-end, to be "the exception that makes the rule", but they are certainly a small minority. Of course this is sort-of different for the microphone inputs on professional soundcards that have microphone preamps in the same box; those operate entirely in the analogue domain and always include a gain control (not an input level control). The same applies to those with included instrument amplifiers. The line level inputs on those same cards are static, not adjustable.
Mixers are the norm in any professional recording setup. I doubt that any professinal or semi-pro card has too little headroom for any normal recording task.
I don't follow you.
You are making a distinction between analog and digital volume controls. Does it mean that analog volume control are passive ? If this is the case, then OK, they cannot suffer from clipping. But otherwise, then same problem should arise, shouldn't it ?
I doubt that any professinal or semi-pro card has too little headroom for any normal recording task.
This precisely what this thread was about. Semi-pro soundcards (Midiman Audiophile, Marian Marc 2) not having enough headroom in order to record the line output of a CD player or a phono preamp fed by a DJ cartridge respectively... Or maybe I'm not following you at all
Most analogue volume controls are passive. Potentiometers are the quintessential, and most common, examples. There are also a variety of resistor ladder designs but those are essentially identical in function to the potentiometer: a passive voltage divider.
There are some active analogue devices such as the voltage control amplifier, a special kind of opamp. Unlike almost every other kind of volume control, a VCA can also amplify the signal, thus this device can be an entire line level preamp. It uses a potentiometer to control the output but the signal does not pass through the pot as it does in a ‘normal' volume control.
A digital volume control operates on the data. It changes the numerical value of samples, thus it loses resolution that cannot be gained back by later amplification. In theory this is not true for analogue volume controls although the distinction is less than absolute.
I'm not sure what your question about the potential difference in clipping between analogue and digital volume controls is about, exactly. My earlier comment about soundcard level controls being digital, such as those in the Audiophile's mixer panel, means that the signal has already clipped (if it is going to) before it gets to the control. Clipping occurs in the ADC. The level controls are in the digital mixing chip, downstream from the ADC. They have no potential to effect clipping on the input; they are only useful when mixing multiple signals in the digital mode.
I don't think ‘headroom' rightfully applies to an ADC. Except for those weird gaming cards, there is always some specific input voltage that produces 0dBfs. Any higher voltage level will always clip; the result will be the same if the input signal is higher by 1mV or by 1V, there is never any non-clipping room above that voltage.
If you look at the operating characteristics curves for transistors or vacuum tubes, most have basically the same shape, only with different numbers defining the particular device. There is generally a significant stretch where the relationship between input and output forms a straight line. At lower or higher levels the relationship becomes radically un-linear.
Reasonably enough, most audio circuits are built around the linear portion of the ‘curve.' If the input signal exceeds circuit specifications (i.e. the linear operating range), the output tends to become rather nonlinear, thus distorted. Somewhere still higher it clips.
It is possible to add some automatic limiting circuitry to prevent clipping, but this is not normal on HiFi because it introduces non-linearities and distortion of its own; actual limiting is left to the fool with his hand on the volume control. Without limiting you get higher quality at the expense of potential danger (clipping, speaker damage, amplifier destruction).
An ADC does have some analogue front-end circuitry. This is more in the nature of impedance matching and buffering; it may have some small + or - amplification but that is not variable. Any considerations towards audio quality will strongly suggests that we want the ADC analogue front-end circuitry always operating on the linear portion of its curve. When the output reaches 0dBfs, and then begins clipping, the input should still be operating completely linearly. The input level that produces clipping in the converter should be lower than the voltage that forces the analogue front-end into non-linear operation, thus well below the clipping level of the analogue front-end. That way there is never any analogue distortion passed on to the digital data.
What an ADC has instead of headroom is dynamic range. For some ways down from 0dBfs its operation is linear and clean. For the Audiophile that is specified as slightly more than 100dB. This is not an exceptional figure, but it is far more than adequate for LPs, cassettes, and quite a few other sources. The peak input can be a fair amount below clipping without ever losing any source resolution. Thus your concern is not "how much headroom does the soundcard have?" but rather "where is 0dBfs relative to my source device, some place convenient or some place inconvenient?"
The dB scale is a relative one. It has no particular values unless it is referenced to something external. For soundcards, as for many other audio devices, there are two main standards. Nominal levels for these two are +4dBu and -10dBV. The "u" and "V" designate some particular meanings, which I believe are u: 1.23V and V: 0.316V. The higher level +4dBu is considered ‘professional' and the other is ‘consumer.'
The real difficulty is that many ‘consumer' devices are constructed without close consideration of the standards. The Audiophile's specs say that its peak analouge input is +2dBV, which is in accordance with the -10dBV standard, I believe. Changing its maximum to a higher voltage could only be done by shifting the entire scale, bring its minimum useable signal level higher also. Then it would not match with other standard devices.
Having read discussions in a number of places over the past four years, I have seen quite a few complaints about inconvenient input levels, when using the Audiophile and a number of other soundcards. A few of these complaints have been the same as in this thread, "my levels are too high for the soundcard," but the great majority have been on the other side " how do I get an adequate input level for good recordings?"
These later are almost always from people who mistakenly believe their peaks should be pushing the red line when in fact they are really doing very well with peaks in the range of -12dBfs to -6dBfs. Regardless, all of these complaints, of which ever stripe, arise because people are attempting to use professional type devices without the full professional equipment regalia. This might be possible if standards were strictly adhered to but I suspect most circuit designers consider themselves artists of some sort and take what ever ‘artistic liberties' they feel inclined towards.
These same discrepancies might well exist with professional equipment -- the actual output levels of source devices may be higher or lower than the standards -- but since most professional setups will include the adequate amenities (mainly mixers), the differences will be take care of in the normal course of set-up for recording, without triggering a second thought.