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Old 12th May 2010, 07:55 PM   #11
M-chen is offline M-chen  Germany
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Join Date: Nov 2004
Location: the middle of Germany
Originally Posted by tritosine View Post
I have a better idea, load this track up into a wave editor, run a declipper, or draw the samples away by hand.

Well, to be true - I never thought about this.
Sometimes life is easy. This is a good option of course, I will try that (for some very odd CDs).



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Old 13th May 2010, 01:46 AM   #12
benb is offline benb  United States
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Join Date: Apr 2010
Originally Posted by guido View Post
Mmm, just had some beers after work, maybe that's why i don't get it.

If a signal clips, the digital stream contains a number of samples after each other with max value. Output of the dac is max current/voltage at one end; a DC signal. After that the analog stuff which handles this in some kind of (bad) way.

So now attenuate all with 2dB. Wouldn't that be a number of samples after each other with max value minues 2dB. So still a DC signal out of the dac, just a bit less current/voltage.

I could only understand this if the signal would be going through a big DSP, which would recontruct the part of the signal which was removed by the clipping. Which is not just a minus 2dB and what is not described in the wolfson datasheet.

The reconstruction filter does just that, and thesedays it may be done in the digital domain by the DAC. I'm not sure whether sigma-delta DACs would clip due to this, but they ARE designed so there's very little analog filtering needed on the output.
Originally Posted by tritosine View Post
By the way this intersample over thing is a few years old now, there are even plugins designed to show if it occurs , can't imagnie they press CD's without checking for it.
It's easy enough to show - I've got Cool Edit 2000 which draws the actual waveshape (AFTER the reconstruction filter) when zoomed in. I just generated a square wave a few dB below zero and zoomed in to show a few sample points, the rising and falling edges clearly go above the highest and below the lowest sample points. How often this happens with "real" music material and how audible it is when it DOES happen is up for debate, though ideally we don't want it to happen.

Here's a page on what happens to a band-limited square wave:
Gibbs phenomenon - Wikipedia, the free encyclopedia
Originally Posted by tritosine View Post
Well , now Im actually thinking about this, the builtin chip attenuation wont work if its not done on the 16bit input directly. Hm . Maybe this feature has something to it afterall. I seem to remember pmd200 filterchip has 1dB attenuation engaged all the time, not to clip incoming squarewave.

Btw Once we are speaking about squarewaves , does it matter?

I have a better idea, load this track up into a wave editor, run a declipper, or draw the samples away by hand.
Solid State Logic | Music
PSW Recording Forums: Brad Blackwood => SSL Intersample Peak Meter freeware

You can run this one simultaneously.
If you're going to load the CD into audio editor software (to "draw the samples away by hand"), you might as well just lower the amplitude by a dB or two (even Audacity does gain changes in floating point and presumably 'correctly' dithers back to 16 or 24 bits), and play that.

ETA: Any digital recording that has a significant number of intersample overs is surely compressed, limited and clipped to within an inch of its life anyway, and I'm doubtful anyone can tell the difference if these overs are also clipped.

Last edited by benb; 13th May 2010 at 01:50 AM.
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