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Topic: Recording line in with Soundforge (Read 38894 times) previous topic - next topic
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Recording line in with Soundforge

Reply #25
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.

Recording line in with Soundforge

Reply #26
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 ), 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.

Recording line in with Soundforge

Reply #27
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.

Recording line in with Soundforge

Reply #28
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 

Recording line in with Soundforge

Reply #29
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.