Here's a conversation between Patsoe and me started in private messages, about the analog line in of soundcards. It seems that it's not possible to record properly many analog sources, because the recording level is clipped no matter what.



Here's what I can say about the marian Marc 2 Soundcard :

Max input : +8dBu

I took a track that I've got on CD and LP (45 RPM). I played the CD (an MPC file) throught the digital output, bit exact, no DSP, to the Sony DTC 55ES :
Impedance : 470 Ohm, Rated output : -4dBs. Load : Arcam Diva ampli + Marc 2 analog input + External equalizer (2 tape outs + monitoring). This stands for an analog source that should be calibrated to a correct line out level.

I recorded it with the Marian Marc 2 analog input, line in level at 0 dB, then I also recorded the vinyl, without changing the recording level.

Here are the results :

Original :


Copy of the Sony Playback :


A "statistics" tool run on 14 seconds of audio shows an RMS level 1 dB lower than the original. Thus the output of the Sony is well tuned with the input of the Marian. The 1 dB loss might even just come from the triple load.

Copy of the vinyl (phono tape out) at the same recording level :


The same analysis (ran on a selection that is not clipped) shows a level 10.4 dB higher than the Sony recording (same load). Thus if the Sony has a rated output of -4 dBs, the Phono out is +6.4 dBs.
The cartridge is a Stanton Trackmaster EL (rated 5.2 mV at 5.5 cm/s), the preampli an Arcam Diva A85 (phono input sensitivity 2.5 mV). Thus the +10.4 dB comes from the cartridge output, that is 5.2 mV instead of 2.5 (+6.4 dB) and that plays the 45 RPM at about 55 cm/s (ten times faster than for the 5.2 mV measurment).

In addition, the peaks are completely clipped.

Lowering the recording level leads to this :


Completely unuseful, the clipping is still there, but lower. The recording level is just a digital process applied to the line in digital stream. We could record at full volume and adjust the level later in the wave editor, the result would be the same.

It seems that Midiman understood this, since they don't feature any recording level slider in the Audiophile 2496 soundcard.

The clipping occurs at -0.5 dB. Granted that the recording level resulted in a file 1 dB lower than the original, it means that the analog input of this card has no headroom at all. It can record a line out signal that is equal to the Sony one (-4 dBu), and it will clip exactly at the same level as the Sony playing a clipped file (0.5 dB higher, to be precise).
If a CD Player has a line output that is 1 dB higher than the Sony, it can't be recorded with this soundcard ! The input is overloaded.

The phono output level is 10 dB higher, and neither the Arcam Diva A85 inputs, nor the Cyrus One inputs, nor even the analog inputs of the Sony DAT are overloaded by it ! They handle it perfectly.

The solution seems simple : asking soundcards manufacturers to add an analog volume control for the line input before the ADC in their soundcards. Otherwise, the line input feature is quite unuseful.

My Philips soundcard and my old Turtle Beach Fiji had this. I recorded many LPs via the analog input and was able to carefully adjust the input volume to optimize the dynamic range without clipping.

Is this a driver issue or is the Marian Marc designed with a fixed gain for pro use? I haven't come across a card that didn't allow adjustable analog input gain - but I haven't used that many cards.

If you need to, you could get a volume control potentiometer and make a simple passive attenuation circuit before the analog input of the sound card.

How do you know that your line in level was analog ? I could record the Sony output on my Marian and adjust the gain without problem too.

Thank you for the suggestion, but I have the Sony for analog recording.

I wonder if this by any chance could be related to the issue several people had when using said line in configured as headphone out. This caused a really annoying static for several people (including myself) on various models of Soundcards.

It is somewhere mentioned on the side in this thread. There was another one that I don't seem to be able to dig up right now. (already went through six pages of search for this one .

Basically I had the problem that whenever I used the headphone out (Terratec Six Pack) there was a constant very prominent hiss in the backgroud. Now that I read this thread this actually really reminds me of the kind of hiss that a normal amp could produce when on full blast with no signal through it.

So basically my guess would be that the same that goes for the line in also goes for the headphone amp. Ie.: it is constantly at full blast and the signal is only lowered digitally at some point before.

When you place an attenuator in front of the line in ADC, it will degrade the noise performance of the ADC.
Then the best is to not have the attennuator on the soundcard, so when using "passive" sources you get no signal attenuation and the lowest noise added.
When you have an "active", amplified source then you can add a passive attennuator if the amplifier does not have one to lower the signal amlitude to avoid clipping. A 24bit ADC has a dynamic range sufficient not to degrade the signal from the amplified soure.

In e-mail contact with a Marian engineer, he said: "The driver does not touch the audio in any kind. Gaining/Volume is made by the AD/DA converter on the card. "

So I guess it's not a driver thing, but it seems the gain control pins on the converter work on the digital side of it.

Sorry, deaf, I didn't understand what you mean. Can you give an example based on a CD player line-out -> Soundcard line-in, for example ? I'm a bit confused.


How does an attenuator introduces noise ? Can you explain, please ?


How do avoid clipping without attenuating ?


What is an "active" and a "passive" source ?


Stupid ADC !

Pio2001,

I think you're being overly critical of these sound cards. After all, they're simply doing the same thing on the input that we praise them for doing on the output: nothing!

They avoid having variable analogue attenuators on the input and output because the quality of these devices can be dubious, and it's better to have a pure input and output, and let the user adjust the level outside the card. At least, that's what the manufacturers would probably say!


FWIW most digital devices seem to match a common analogue audio level: digital full scale = 2Vrms (just over 6Vpp). This is much more consistent than analogue devices (e.g. record players, tuners etc) which can use just about any analogue voltage level as an output.

If you're recording one of these "non standard" analogue devices with a card without input level adjustment, then you may need to provide your own external signal attenuation. In practice, this can be just one or two resistors, or a simple potentiometer. Either would give a simple and cheap solution to the problem. Or you could buy a passive pre-amp with a volume control that would do the same job, but cost more. Or an active device, such as a mixing desk. Probably overkill.


Maybe it's misleading that the GUI of some of these sound cards appears to show an input level control, when the hardware doesn't actually have one. Maybe this should be made more clear in the manual - but who reads those?

Cheers,
David.

I guess I agree with David. About the second part of our conversation then:



Any opinions?

Let me try to explain again. Basically, we want to maximize the signal to noise ratio or the dynamic range of the sound card while keeping distortion low. One can not have higher dynamic range coming out of a sound card than going in.

1) When the sound card has a line in amplifier and an attennuator before the ADC, the signal amplitude to the ADC is limited by the power supply voltage of the line in amplifier or the attennuator. Therefore the input signal amplitude can not be higher or clipping will happen.

2) When no signal is present at the line input of the soundcard, depending on the attennuator setting, either the noise of the line in amplifier, or if that is attenuated too much, the noise of the ADC will determine the noise in the digitized output. But the (uncorrelated) noise from the input signal can only add to the noise of the line in amplifier. We want the best situation, when the noise of the input signal is dominating the noise throughout the chain. An attennuator can prohibit that. Therefore best possible point to connect your source is right into the ADC! (Some ADCs accept input level values above the power supply voltage, that is another reason not to place anything in front of it)

So when can the input signal to be be too high for the ADC? Only when you have an amplified or "active" source, like a CD player/preamplifier. They must have a power supply voltage higher than the sound card (and they should, since higher power supply voltages can mean higher dynamic range and lower distortion, we can discuss it another time).
But in those situation placing a passive attenuator in between the output of the source and the input of the sound card should not degrade the dynamic range of the source, since the source rarely has a dynamic range larger that the ADC's. Meaning, that attennuating both the signal and the noise of the source, in such that the signal values attennuated just below the clipping level of the ADC, the attennuated noise from the source still will be higher than the ADC's. (Don't forget, how was that signal creatred? Possibly digitized with an ADC pretty similar in performance that your own sound card's?)

3) A good attennuator operating from the higher power supply rails, or being a passive one, can be designed in such, that the dynamic range is preserved.

4) A sound card should be designed to accept the standard 2V signal from a CD player. But just a 6dB passive attennuator placed in between the CD player output and the sound card input should not compromise the digitized signal, since the dynamic range of most recorded media rarely higher than 80dB (or even 70dB) and most sound card's are better than that!
Some microphones can produce a dynamic range higher than 120dB, which a good ADC can convert, but who wants to record the sound of a jet airplane taking off, while also recording the sound of the surrounding animal life? How will you play it back? What is the threshold of your hearing?
(I was at an airshow this weekend and saw the Blue Angels flying, but could not hear the people talking around me, when a jet was passing.)

P.

If the maximum output level of CD Players (and soundcards and DACs) is normalized, I'm a bit relieved.
However, Evereux reported a CD Player clipping the Audiophile 2496 line in : http://www.hydrogenaudio.org/forums/index....showtopic=12102

And with current cartridges for vinyl (up to 7 mV ouput !) there is no way to record any vinyl with these soundcards using a proper phono preamp.
I never had this problem with my cassette or DAT decks ! They can record any analog line level source, including vinyls.
I really don't know how I could record my LPs to CD if I had not kept my old DAT deck that works as an external ADC. I'm not going to buy or solder a passive or active attenuator in order to record a line in signal in the line input of a soundcard I bought and paid for. I'd rather return it to the store telling that its line input can't record the line output of my ampli.

Deaf, thank you for the explanation. Interesting point. Personally, I prefer to record my vinyls with an inaudible noise than not at all. Following your point of view, a bypassable active input preamp would be the optimal solution.

deaf,

your distinction between an active and passive source is totally arbitrary, and I'm not sure I understand it!

The only passive (i.e. unpowered, unamplified) source I can think of is a phono cartridge. a few mV isn't going to clip a sound card(!!!), and you wouldn't plug the thing straight into the sound card anyway.

Anything else is active. The output level is totally arbitrary. Go back 20 or 30 years, and will not find any idea of a standard signal voltage level in domestic audio equipment.


Pio2001,

You could argue that it's the pre-amps fault! Most 1960s phono pre-amps have a sensitivity selector, to compensate for the differeing sensitivities of different cartridges. e.g. I have valve amps with potentiometers, a transistor amp with a four position switch, and a transistor amp with a two position switch.

These are needed to match the level of the phono source roughly with that of the other sources (Replay Gain 1960s style!), and to ensure that the level is suitable to drive the power amplifier optimally.

Later equipment designs have enough headroom to cope with a 2mV or 10mV cartridge output, so these switches are not included, and you just have to use the volume control to make the level match other sources!


Maybe there's a market for a simple, cheap, high quality 6dB passive attenuator!

You can buy something over the top like:
http://www.creekaudio.com/products/obh/obh_10_12.asp
but that probably costs more than your sound card!

Cheers,
David.

I wonder if other soundcards behave the same.

2Bdecided,

You are almost correct. It is hard to find a passive source which matches the input of a sound card. I had a turntable once, with a piezo pick-up. But moving coil phono cartridges are the norm now. Those require equalization so you better to start with a preamp. (So is the case of a microphone.)

Pio2001,

It seems sound cards having the limitations because of their ADCs run off of a +5V power supply. But they do work with most older equipment. Accepting the output of a tape recorder and even your DAT works just fine regardless that the electronics in those equipment runs off of a +/- 12-15 V power supply. Because the line out voltages were raised to the 2V level only with the appearence of CD players, because of their DAC's put out such a high voltage. Moving coil preamps can not standardize in such a way, since their input level is not fixed.
Your (and my DTC Z5) DAT or any tape recorder accepts the signal level of a phono preamp, because the front stage is a low gain amplifier running off of a similar power supply rails and has a manually controlled potentiometer right after that. A versatile configuration. A sound card usually does not have such controls. Electronic attennuation still has the +5V limitation. So to offer the highest performance, the line input of a sound card should be unamplified, accepting 2V and no more. The mic input should be your amplified input.
But you can have a sound card with manually controllable input attenuator, a break out box of a SB-Pro or Extigy? Or just put that 6-10dB attennuator after your phono preamp, if your pick up is too sensitive and your DAT/tape breaks down.

I have a 4.5mV cartridge which I plug directly into the phono-in on my DMX 6fire.. That input has a 5mV sensitivity so I get perfect levels!

Am I dumb for doing this, I never saw the need for a pre-amp considering I had this card.. ?

Mac,

I (and the others, I think) are talking about the line-in, which expects about 1-2V input, rather than 2-10mV. Also, the line in isn't equalised; a real phono stage has RIAA equalisation. But we've alredy discussed that I think! The problem is that the line out of some separate phono stages can overload the line-in of these sound cards, and there's no analogue gain control in these sound cards to prevent this.

Cheers,
David.

Your card is designed for recording vinyls directly. That's a good thing.

Yay, I'm not stoopid

I've not noticed line problems with the DMX when recording from line level inputs, I'm not sure if the gain controls provided are before the ADC, but they only clip when I set them too high

OK - summarizing: for highest compatibility, the recommendation to people looking for a recording card should be to be sure that the soundcard supports standard 2V input levels. If you have any higher/lower levels to record, you should get an external step-up/down amp.
For convenience, Mac's tip is interesting: the DMX 6fire seems to have hardware gain, and even supports MM-cartridges (RIAA equalising is done digitally). Note that nothing has been said yet about the quality of this card's inputs. Convenient, however, it certainly is.


It seems nobody is interested in this question?

Well, since I'd really like to see some opinions, here's an additional question:
seeing digit-life's measurements of digital inputs (examples here and here), I get the feeling its very easy to cheat RMAA - using a similar card, I found results that were much better: voila - and I don't even have a Lynx2 for playback!

I don't know what to say. The cards used for playback were different, the level playback might be different, if the soundcards are not in the same case, the difference in ground voltage can introduce noise or hum, the CPU activity or the proximity of other cards can also change the performances. I don't pay very much attention to RMAA results when it comes to comparing the sound quality of line inputs/outputs. I wouldn't be surprised if a card A have more distortion than a card B in a computer and the opposite in another computer. After all, the differences reported by RMAA are often very, very small.

Had the same problems. Solved by using some trim pots on the line in cable (make sure to calibrate both L & R before recording) to reduce level or a preamp with output volume setting.

Indeed there's a lot of difference. Some tests show power supply noise, some don't. Also there's some difference between production samples. Etc...

But if not by these kind of measurements, how are we going to back our advice (for anyone not getting the discussion: we were trying to get consensus over recommended sound cards)? Listening tests may differ as much between systems as measurements do...

However, the differences between the examples I posted weren't small - 10 to 18dB noise level differences (unweighted)! So that should be significant enough to say that e.g. the Aureon is a lousy recorder, or?.... not so fast -
here's some totally different results on the Aureon. It performs 10dB better as to noise.

If recording measurement is so unreliable, can we even trust playback measurement? Do we have better fundaments to the advice we offer here than is offered in fora we consider 'subjectivist'?

For "sound quality", it is very difficult to give an objective advice. The soundcard challenge showed that the distortion introduced by soundcards were at the limit of audibility. In the soundcard challenge, I couldn't tell the difference between the reference file, and the same loopbacked five times in the audiophile 2496 card !
Yes, the mysterious challenger who took the test was me
Fortunately, Bedeox managed to ABX an analog recording of CD with a CD player versus a digital copy ( http://www.hydrogenaudio.org/forums/index....st=0#entry67546 ), proving that low end CD players are not perfect (but it was a model from 1991, multibit I think, DACs have improved since then).
The example of the SB64 is easier, since the frequency response is uneven. It has too much treble, and it can be heard.
Resampling may introduce distortion, but is it visible on RMAA's results ? Analysis of SB Live digital loopback, anyone ? And noise shaped dither can raise the noise level without causing any harm to the sound. Remember that at digital silence, +/- 1bit dither is already 6 db louder than 1-bit peak-to-peak dither, that can itself be any amount of db louder than silence.
ABX results of loopback recordings against originals can be a valid way of backuping statements about sound.

An advice about a soundcard must not be only based on sound quality. Gaming abilities, sample rates supported, ASIO, midi support, mixer abilities, connexions, etc must be taken into account too.

Sorry to bring this thread "back from the dead", but no where did anyone mention one problem with the audiophile 2496, which I've just measured...


I previously said that most digital devices use 2Vrms = digital full scale as a reference level these days. The problem with the audiophile 2496 is that it doesn't. (Or, at least, mine doesn't).

The audiophile 2496 analogue output, fed with a digital full scale signal, gives 1.27Vrms (3.68Vpp) for a 1kHz sine wave. (Both values approximate.)

The analogue input attenuates the signal slightly (about -1.16dB), which means that an analogue input of 1.45Vrms will give a sampled signal which peaks near digital full scale, and any higher level input will clip.

This is very unhelpful, since most CD and DVD players have a maximum output of 2Vrms. It means that most modern sources will clip the audiophile 2496.


Can anyone else with an oscilloscope or multimeter confirm my measurements?

Cheers,
David.

Technically they shouldn't call it line in and out, as you said 1.4rms is the standard for line in and out.

Sorry to ask again, but can anyone confirm the line-out voltage for digital full scale using the Audiophile 2496?

I'm wondering if the low output on mine is standard (for this card), or a fault.

Cheers,
David.

If you use a multimeter, test at 50 Hz, no more.

Once we used a high-end electronic voltmeter on a signal that was several kHz, and got strange results... We checked all the circuitry until the teacher comes and tells "You won't get any meaningful result from a voltmeter at such frequencies, when measuring alternative voltage, always use an oscilloscope. Alternative voltmeters are designed for 50/60 Hz voltage..."

David:

I don't remember the exact figure, but IIRC my Audiophile max. I/O level was similar to yours, it was quite below 2 Vrms. I'm not really sure this 2 Vrms is a true "standard", but IIRC my father's old Denon cd-player has a similar output voltage.

But, this test was a little bit "unfair" in the first 5-times looped back version, because frequency response differences between all samples had been reduced to minimum via digital processing (not other kind of differences, though). In the next and current version, there was no digital processing applied but dither to the samples, but the number of loopbacks applied was just 3. However, so far nobody has been able to reliability tell the samples from each other in a blind test.

Thanks for confirming this.

It's true that there isn't a "standard", but it seems that, somehow, a standard has emerged.

All the DVD and CD players I could find to test (I found 7), and all the data sheets for current products I could find on the net - every single one of them used or said 2V. The data sheets tend to say just "2V", but testing the output shows this is 2Vrms.

So, whilst the sound card does not break any official standard (because there is none!), it is very unhelpful. As Pio2001 said, it can't record the output of most modern equipment (without an external pre-amp/attenuator), and it can't match the output level of any modern CD or DVD player. This won't matter most of the time, but it's worth knowing.

Cheers,
David.

1.)The forum suggests the use of a passive attenuator(volume control) with the M-Audio 2496, which may not be such a good idea. The version I had used the Inverting Input of the IC...noise would be a function of the source impedance, hence the floor noise would vary with the position of the potentiometer. (This would also explain why input overload measurements on this card vary with the test setup.)

The better approach would be to place the balance control ahead of a low-noise preamp/buffer eg AD797,SSM2134

2.)Ripping CD music via the Analog outputs is adding more links in the chain then needed, why not use the Digital out or Optical out from the CD player, the 2496 supports this....No more clipping!

3.) In all cases one has to Zero the DC offset on every file.

I doubt using an adequate simple passive voltage divider in order to avoid clipping would degrade notabily input SNR, but I haven't tried. Have you verified this personally via measurements? The use of an active device as input stage makes things much more complex, and I doubt would give any practical benefit.



We are not talking about ripping here. I believe we are talking about recording and analyzing of the output of a device.



Most cards don't have DC offset problems at the input, since they are AC-coupled (this means that they block the DC at their inputs).

I went through the testing when I first built my Vinyl Ripper ... over 2yrs ago!
I used a Neutrik 3000 series audio test rack for signal and phase measurements; Tektronix 500 rack for Impulse, Tone burst and very low frequency(1-20hz) signals:
CRO Kikusui COR5501

I have 3 vinyl sources, so a switch box/preamp was needed anyway.From memory, the main issue with the M-Audio was anomalies with the subsonics. I am not knocking the product, just putting forward my observation.



I stand corrected.



The cards don't have DC offsets?

DC offset on modern HQ sound cards should be negligible, but still measureable ... the DC offset is introduced after the AC coupling stage mostly due to the pc environment (em radiation cannot always be completely suppressed and might induce DC that leaks over to the card's analog input - not the ideal environment for a high end sound controller)

That's why I use an external ADC to feed digital signals only

The Prodigy 7.1's CODEC features analogue gain control for its inputs, and analogue volume control for the outputs.