Can't control input level

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ezlxq1949
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Joined: Sun Nov 19, 2006 9:05 pm
Location: Canberra, Australia

Can't control input level

Post by ezlxq1949 »

This may have been answered elsewhere, but if it has, I can't find it.

I'm doing the popular thing, transferring some of my vinyl collection to digital, and am having trouble with clipping.

An LP (or more usually a 45) recorded at too high a level drives Amadeus into clipping when recording.

The cartridge on the turntable has fixed output level of course, and so does the phono stage. The ADC is the M-Audio 2496 USB, which also has fixed output level.

Problem: Amadeus is unable to alter the input level. From M-Audio's web site I read:
Q: On my Mac running OS X, it displays "The selected device has no output controls" when I select my M-Audio device for output from System Preferences>Sound; is this normal?

A: Yes. M-Audio devices bypass the Core Audio kernel mixer; depending on which M-Audio device you have, output level may be controlled from the M-Audio control panel, the device itself, or the software you are using.
No: there is no level control available anywhere, not even in Amadeus.

Does anybody know if level control is available? Is it possible to do anything about this in Amadeus? Martin, could you provide a solution?

Sure, I can declip either manually or using one of the few declippers out there, but this is an extra step which should be unnecessary. All I need is 1 or 2 dB of attenuation and everything will be fine.

Thanks in advance.
WTS

ezlxq1949
Posts: 9
Joined: Sun Nov 19, 2006 9:05 pm
Location: Canberra, Australia

Post by ezlxq1949 »

OK, after poring over the manual I discover that there's no way that my problem can be solved by Amadeus.

Drat.
WTS

Gerard Bik
Posts: 353
Joined: Fri Nov 10, 2006 9:04 pm

Can't control input level

Post by Gerard Bik »

One solution is to attenuate the analogue signal before it is converted.
You would need the increasingly rare skill of handling a soldering iron, for this.

good luck
gerard
OK, after poring over the manual I discover that there's no way that my problem can be solved by Amadeus.

Drat.

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Lou Kash
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Post by Lou Kash »

I'm not familiar with M-Audio USB interfaces (I used to have a PCI card and I still use FW410), is that this one?
http://www.m-audio.com/products/en_us/Transit-main.html

Their software for the FW interface does have a sort of virtual mixer. But then again, the FW interface also has two line/mic inputs with gain twisters - which is where I would connect the source signal. The USB model obviously doesn't.

Anyway...
Please let us know your actual setup (Mac, turntable, amplifier etc.), because usually a phono signal shouldn't be that "hot".

I'm digitalizing vinyl very often and there are several posible ways to do it right. Mostly I simply connect my PowerBook to the amplifier as if it were a tape deck, using the built-in audio which is good enough for this purpose. Then I record the signal from the amp's phono input. Important step here is to make sure to set the built-in audio to 24bit and to record at relatively low level into a 24bit file. Like that you can avoid clipping completely. After cleaning up the recording - especially loud pops - you can then safely normalize to the maximum peak level without any quality loss. (Recording at low level at 16bit and then normalizing can significantly affect the sound quality; think of it as if you scan a bad photo and then you try to increase the contrast - it will also look pretty unnatural in colors...)

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Martin Hairer
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Can't control input level

Post by Martin Hairer »

A: Yes. M-Audio devices bypass the Core Audio kernel mixer;
depending on which M-Audio device you have, output level may be
controlled from the M-Audio control panel, the device itself, or
the software you are using.
No: there is no level control available anywhere, not even in Amadeus.
Amadeus Pro allows to adjust the gain only if the hardware driver
allows for it. It would of course be trivial for me to allow adjusting
the gain of the data handed over by the driver, but it would also be
silly: attenuating a clipped signal won't prevent it from clipping...
I am afraid that the only solution is to get a weaker signal into the
device. Regards,

Martin

HairerSoft
http://www.hairersoft.com/


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CDJonah_alt
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Can't control input level

Post by CDJonah_alt »

Certainly the easiest route is to use either a line-level output or an
earphone output by plugging the turntable into a stereo amplifier. My
enthusiasm for soldering up a kludge is relatively negligible; one could
get some unfortunate frequency response characteristics with mismatched
impedences. Possibly you could find a cheap pre-amp to go to line level.

Chuck

Martin Hairer wrote:
A: Yes. M-Audio devices bypass the Core Audio kernel mixer;
depending on which M-Audio device you have, output level may be
controlled from the M-Audio control panel, the device itself, or the
software you are using.
No: there is no level control available anywhere, not even in Amadeus.
Amadeus Pro allows to adjust the gain only if the hardware driver
allows for it. It would of course be trivial for me to allow adjusting
the gain of the data handed over by the driver, but it would also be
silly: attenuating a clipped signal won't prevent it from clipping...
I am afraid that the only solution is to get a weaker signal into the
device. Regards,

Martin

HairerSoft
http://www.hairersoft.com/


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Sonic Purity
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Re: Can't control input level

Post by Sonic Purity »

ezlxq1949 wrote:Problem: Amadeus is unable to alter the input level. From M-Audio's web site I read:
Q: On my Mac running OS X, it displays "The selected device has no output controls" when I select my M-Audio device for output from System Preferences>Sound; is this normal?

A: Yes. M-Audio devices bypass the Core Audio kernel mixer; depending on which M-Audio device you have, output level may be controlled from the M-Audio control panel, the device itself, or the software you are using.
No: there is no level control available anywhere, not even in Amadeus.
I am still puzzled. Yes, true, the level will not be adjusted in Amadeus. But what about the M-Audio Control panel? That is how my friend does it. His setup, working backwards, is a PowerBook G4 titanium, Amadeus, M-Audio Transit USB interface with its M-Audio control panel, standard home receiver phono preamp, Technics SL 1200 MKII and Stanton 681 series cartridge/stylus. He adjusts the levels via the M-Audio control panel, and has no problems.

Aside: i am totally “synced” with Lou Kash’s way of doing things, esp. the “pretend the Mac is a tape deck” hookup philosophy. I only differ in working in 16 bits with the level of the desired audio just below 0 dBFS. So what if the clicks and pops attempt to go to +10 dB above full scale and square off severely?: they are undesired distortion already, and will be removed anyway (in my case, via some combination of ClickRepair and/or manually in Amadeus).
))Sonic((

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Lou Kash
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Re: Can't control input level

Post by Lou Kash »

Sonic Purity wrote:But what about the M-Audio Control panel?
If the signal is already too hot at the interface input, the control panel won't help either. That's why I record through the front line inputs on FW410 (if using this device - but that hasn't been necessary for vinyl recording since having the Powerbook with 24bit built-in audio) where I can comfortably adujst the input gain.
Sonic Purity wrote:His setup, working backwards, is a PowerBook G4 titanium
... which has only 16bit built-in audio...
Sonic Purity wrote:I only differ in working in 16 bits with the level of the desired audio just below 0 dBFS.
Well, usually I'm going even further, that's why I'm all for 24bit recording: I use my audio files partly for deejaying. Recording from vinyl produces much quieter audio than remastered tracks from CDs (not to speak of the overcompressed overlimited mastering that's common to most 2000s productions). So I'm even very lightly compressing and limiting the vinyl recordings just to "cut off" the very highest peaks to gain a few dB more, so the tracks are not all that quiet compared to the CD tracks. That's another step where having a 24bit file retains the original audio quality.
Sonic Purity wrote:So what if the clicks and pops attempt to go to +10 dB above full scale and square off severely?: they are undesired distortion already, and will be removed anyway (in my case, via some combination of ClickRepair and/or manually in Amadeus).
Exactly. If the pops are clipping, that doesn't matter. But since many records can have quite some unexpected peaks, you still should keep quite a lot of headroom.

Sonic Purity
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Re: Can't control input level

Post by Sonic Purity »

Lou Kash wrote:
Sonic Purity wrote:But what about the M-Audio Control panel?
If the signal is already too hot at the interface input, the control panel won't help either.
The OP indicated he needed only one or two dB lower. It would amaze me if the standard -10 dBV average output level of a standard phono preamp was too hot for the input stage of his M-Audio device, which i would think would be designed for good headroom above +4 dBu pro level. And i was under the impression that the M-Audio control panel was controlling hardware gain in the M-Audio device (though i do not know this for a fact).
Lou Kash wrote:But since many records can have quite some unexpected peaks, you still should keep quite a lot of headroom.
Well, i guess we have to agree to disagree here. I routinely run up to -0.3 dBFS, sometimes higher, when digitizing pre-recorded analog material. Sure, it takes time to get the correct level setting so the absolute highest peak will not exceed this level, yet i do not find it onerous to do so, and on my 16-bit systems i need to keep those levels as high as possible. As this is 100% pre-recorded, not live, the option exists to push things this much. (I am doing this on an OS 9 system using Coaster as the A to D software, as i vastly prefer its meters and have so far not been able to convince Martin to get that sort of metering into Amadeus.)

Yes, if i were under time pressure to digitize correctly in one pass, i would need to do more or less what you suggest if i wanted good results. Taking a little more time for what is usually 1 1/2 or 2 passes allows me to use my older 16-bit equipment, which is extremely appealing to my pocketbook.
))Sonic((

CDJonah_alt
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Can't control input level

Post by CDJonah_alt »

I just did a search on the web and according to the Audacity FAQ, the
average 33 1/3 lp bit depth is 9-10 bits.
http://www.audacityteam.org/wiki/index. ... #Vinyl_etc
To me, that means trying to get to within 1/2 dB of 0 dB a job, that if
you enjoy it, wirth doing but unlikely to make a different in sound.
For cassette tapes, the bit depth is even less. Minidisks have a bit
depth of 24 bits but real life probably will keep you from using the
full depth.

Chuck

Sonic Purity wrote:
Lou Kash wrote:
Sonic Purity wrote:
But what about the M-Audio Control panel?
If the signal is already too hot at the interface input, the control panel won't help either.
The OP indicated he needed only one or two dB lower. It would amaze me if the standard -10 dBV average output level of a standard phono preamp was too hot for the input stage of his M-Audio device, which i would think would be designed for good headroom above +4 dBu pro level. And i was under the impression that the M-Audio control panel was controlling hardware gain in the M-Audio device (though i do not know this for a fact).


Lou Kash wrote:
But since many records can have quite some unexpected peaks, you still should keep quite a lot of headroom.
Well, i guess we have to agree to disagree here. I routinely run up to -0.3 dBFS, sometimes higher, when digitizing pre-recorded analog material. Sure, it takes time to get the correct level setting so the absolute highest peak will not exceed this level, yet i do not find it onerous to do so, and on my 16-bit systems i need to keep those levels as high as possible. As this is 100% pre-recorded, not live, the option exists to push things this much. (I am doing this on an OS 9 system using Coaster as the A to D software, as i vastly prefer its meters and have so far not been able to convince Martin to get that sort of metering into Amadeus.)

Yes, if i were under time pressure to digitize correctly in one pass, i would need to do more or less what you suggest if i wanted good results. Taking a little more time for what is usually 1 1/2 or 2 passes allows me to use my older 16-bit equipment, which is extremely appealing to my pocketbook.

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))Sonic((




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Lou Kash
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Re: Can't control input level

Post by Lou Kash »

Never trust a wiki you didn't falsify yourself... ;)

I'm by no means an expert. But from what I read some time ago, the specs of audio interfaces (especially the built-in in Macs) are supposedly too "optimistic". If you record at 16bit, in fact you would get significantly less. If you record at 24 bit, you'll get less, too, but still more than 16bit. Hence it doesn't hurt anybody having some "overhead" here if you intend to perform any adjusting of the dynamics. If you have a 24bit interface, why not make use of it. You already paid for it, after all. :)

In the beginning of my vinyl digitalizing "career" I used to use the cheap Griffin iMic (16bit) to get the audio into the computer (a PowerMac G4 without a line-in or an iBook G3). These recordings sound significantly worse than anything I've recorded later at 24bit and downsampled to 16bit after editing.

Similar applies to displays, by the way. Laptop displays, by specs capable of 24bit depth (RGB à 8bit) obviously never displays all of those 16.7 mio colors they claim they would. Supposedly there's only 6bit per color, the rest is faked by dithering. (Not so good news for us professional graphic designers...)
(see e.g. http://blog.wired.com/gadgets/2007/05/a ... lawsu.html )

CDJonah_alt
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Can't control input level

Post by CDJonah_alt »

True -- also I have seen numbers between 10 and 12 bits but based on my
recordings and background noise, I think the numbers are probably
reasonable. The question was not on using 24 bits but trying to come
within a half dB of 0 on 16 bits.

If I had 24 bits I would use them, but I don't. (Of course, I don't
know how many bits of the 24 bits are real. At line level, we are down
to the region where contact voltage effets will be larger than the
resoltion) But I also have an external A/D converter. I would not want
to count on any internal converter to even 16 bits -- electrical noise
is probably too high. I once did test an external A/D and did find the
low order bits to be true. But for a cheap converter, I suspect there
are a lot of missing codes.

Chuck

Lou Kash wrote:
CDJonah_alt wrote:
Never trust a wiki you didn't falsify yourself... ;)

I'm by no means an expert. But from what I read some time ago, the specs of audio interfaces (especially the built-in in Macs) are supposedly too "optimistic". If you record at 16bit, in fact you would get significantly less. If you record at 24 bit, you'll get less, too, but still more than 16bit. Hence it doesn't hurt anybody having some "overhead" here if you intend to perform any adjusting of the dynamics. If you have a 24bit interface, why not make use of it. You already paid for it, after all. :)

In the beginning of my vinyl digitalizing "career" I used to use the cheap Griffin iMic (16bit) to get the audio into the computer (a PowerMac G4 without a line-in or an iBook G3). These recordings sound significantly worse than anything I've recorded later at 24bit and downsampled to 16bit after editing.

Similar applies to displays, by the way. Laptop displays, by specs capable of 24bit depth (RGB à 8bit) obviously never displays all of those 16.7 mio colors they claim they would. Supposedly there's only 6bit per color, the rest is faked by dithering. (Not so good news for us professional graphic designers...)
(see e.g. http://blog.wired.com/gadgets/2007/05/a ... lawsu.html )





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ezlxq1949
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Can't control input level

Post by ezlxq1949 »

Thanks everybody for the ideas and suggestions. I'm surprised this topic received so much attention.

Like you I'm surprised that devices which allegedly operate at line level have no headroom.

My phono stage (pre-amp) is the highly-regarded Graham Slee Jazz Club. I'm not about to abandon it for a cheap pre-amp!

M-Audio's Delta 2496 comes in two forms: PCI card and USB. have both, and both behave similarly. The USB driver provides settings for bit rate (8 to 96,000), sample depth (16 or 24) and latency. The PCI card provides the same bit rates and sample depth, and also output levels of 0db or -10dBV.

I have thought about making or buying some attenuators, but I don't want to introduce phase shifts and who knows what else into the signal.

It would seem that the only way ahead is to declip in software. Fortunately, the amount of clipping to deal with is very small. As I said, only a few records so far have produced clipping. A rummage around the Internet has produced a few candidates, one of which might be capable of being turned into a plugin for Amadeus. Provided I can figure out how to do this, of course! It was possible in the days of Amadeus II for users to write plugins.
WTS

Sonic Purity
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Re: Can't control input level

Post by Sonic Purity »

ezlxq1949 wrote:Like you I'm surprised that devices which allegedly operate at line level have no headroom.

My phono stage (pre-amp) is the highly-regarded Graham Slee Jazz Club. I'm not about to abandon it for a cheap pre-amp!
Ah… i looked it up… nice. And not at all a standard preamp with a standard output (it runs a bit hot at 472 mV vs. the standard 300 mV), which contributes to the problem.
M-Audio's Delta 2496…
The Technical Specifications claim “Peak analog input signal: +2 dBV”… only 12 dB of headroom vs. the nominal -10 dBV, which strikes me as awfully wimpy for an allegedly pro-ish device! This is even more of the problem, esp. with the GSJC having a nominal of 472 mV≈ -6.5 dBV… only 8.5 dB of headroom! Yikes!
I have thought about making or buying some attenuators, but I don't want to introduce phase shifts and who knows what else into the signal.

It would seem that the only way ahead is to declip in software. Fortunately, the amount of clipping to deal with is very small. As I said, only a few records so far have produced clipping.
Well, you could do that (declip). Why not fix the problem? I’d be looking at a better audio card with more reasonable (higher) peak input level specification. I have trouble taking anything with less than +6 dBV peak input level seriously… that has been the standard highest analog output of mid-fi CD players since the beginning. (Note: i don’t take any of my A-D seriously: i am using Mac built-ins! So, i have no recommendations for another A-D card.)

If you don’t want to invest in a different card, i’d consider discussing the issue with the GSJC folks. They can probably recommend some slight gain changes in their device which could lower its peak output just enough for things to work.

Best Wishes,
))Sonic((

CDJonah_alt
Posts: 379
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Can't control input level

Post by CDJonah_alt »

Remember, just because something doesn't digitally clip, doesn't mean
there isn't high-level distortion. I got a CD that I attempted to edit
that had distorted/clipping but at a level well below the maximum value
for a CD. Really couldn't do much to minimize the problem.

The output level of the preamp may be hot because the ear seems to wir
so that in general one appears the higher/louder output.

Chuck

Sonic Purity wrote:
ezlxq1949 wrote:
Like you I'm surprised that devices which allegedly operate at line level have no headroom.

My phono stage (pre-amp) is the highly-regarded Graham Slee Jazz Club. I'm not about to abandon it for a cheap pre-amp!
Ah… i looked it up… nice. And not at all a standard preamp with a standard output (it runs a bit hot at 472 mV vs. the standard 300 mV), which contributes to the problem.


M-Audio's Delta 2496…
The Technical Specifications (http://www.m-audio.com/images/global/ma ... Manual.pdf) claim “Peak analog input signal: +2 dBV”… only 12 dB of headroom vs. the nominal -10 dBV, which strikes me as awfully wimpy for an allegedly pro-ish device! This is even more of the problem, esp. with the GSJC having a nominal of 472 mV≈ -6.5 dBV… only 8.5 dB of headroom! Yikes!


I have thought about making or buying some attenuators, but I don't want to introduce phase shifts and who knows what else into the signal.

It would seem that the only way ahead is to declip in software. Fortunately, the amount of clipping to deal with is very small. As I said, only a few records so far have produced clipping.
Well, you could do that (declip). Why not fix the problem? I’d be looking at a better audio card with more reasonable (higher) peak input level specification. I have trouble taking anything with less than +6 dBV peak input level seriously… that has been the standard highest analog output of mid-fi CD players since the beginning. (Note: i don’t take any of my A-D seriously: i am using Mac built-ins! So, i have no recommendations for another A-D card.)

If you don’t want to invest in a different card, i’d consider discussing the issue with the GSJC folks. They can probably recommend some slight gain changes in their device which could lower its peak output just enough for things to work.

Best Wishes,

------------------------
))Sonic((





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