unclipping

On Mon, 05 Mar 2012 20:43:10 +0000, glen herrmannsfeldt wrote:

> Tim Wescott <tim@seemywebsite.com> wrote:
> 
> (snip, I wrote)
>>> Well, I could try listening even more carefully.
> 
>> Well, next time use a floating-point ADC, for heaven's sake!
> 
>> (Slightly) more seriously, in theory you could use an ADC with
>> sufficient dynamic range to encompass the dynamic range of your analog
>> system.  Assuming you have lots of $$.
> 
> The DR-1 has, and I used, 24 bit WAV, and usually not so close to
> clipping.
> 
> This one I did with a different microphone position, which might have
> made the level a little higher, but I didn't think that much higher.
> Either this symphony is a lot louder, or the microphone position makes
> more difference than I though. (or both.)
> 
> I record them 24 bit, check the RMS and peak amplitude for each track,
> and then choose a shift value from those. (That is, changes in 6dB
> increments.)
> 
> Since the rest of the analog signal chain might not be good enough for
> 24 bits, I try not to get too low, but if I use 18 to 20 bits it
> shouldn't be so far off.

You should have plenty of overhead if you set your levels such that the 
preamp into the ADC clips just before the ADC does -- do that, record 
some silence, and I rather suspect that you'll have way more than 1LSB of 
amplifier noise in your recording.

Then, when you get clipping you'll know that your ADC was doing it's 
best, at least.

-- 
My liberal friends think I'm a conservative kook.
My conservative friends think I'm a liberal kook.
Why am I not happy that they have found common ground?

Tim Wescott, Communications, Control, Circuits & Software
http://www.wescottdesign.com

On Mon, 05 Mar 2012 01:12:03 +0000, glen herrmannsfeldt wrote:

> I have a digitized WAV file about 71 minutes long (actually, 10 files
> totaling 71 minutes) with three samples clipped, I believe not all
> consecutive.
> 
> I can't actually hear them, as I usually can with much more clipping,
> but I do know that they are there. I was, then, wondering about possibly
> reducing the effect by substituting (after rescaling) a more appropriate
> value. Most likely it wouldn't work well for longer clipped regions, but
> maybe for such short ones.
> 
> One that I have thought about so far is to fit a quadratic to the two
> points before and after and use its value at the point (or two) in
> question. I don't remember ever hearing about anyone else doing
> something like this.

It's worth a try, though.  I'm not sure if a quadratic or some higher-
order polynomial is best.

Whatever you do, I agree with you that it won't do much good over more 
than a few samples.

-- 
My liberal friends think I'm a conservative kook.
My conservative friends think I'm a liberal kook.
Why am I not happy that they have found common ground?

Tim Wescott, Communications, Control, Circuits & Software
http://www.wescottdesign.com

Tim Wescott <tim@seemywebsite.com> wrote:

(snip, I wrote)
>> The DR-1 has, and I used, 24 bit WAV, and usually not so close to
>> clipping.
 
(snip) 
>> I record them 24 bit, check the RMS and peak amplitude for each track,
>> and then choose a shift value from those. (That is, changes in 6dB
>> increments.)
 
>> Since the rest of the analog signal chain might not be good enough for
>> 24 bits, I try not to get too low, but if I use 18 to 20 bits it
>> shouldn't be so far off.

> You should have plenty of overhead if you set your levels such that the 
> preamp into the ADC clips just before the ADC does -- do that, record 
> some silence, and I rather suspect that you'll have way more than 1LSB 
> of amplifier noise in your recording.

I have an Audio-Technica AT-822, which is a nice, but not too
expensive stereo microphone. Interestingly, it uses the usual XLR
connector and cable, but not as a balanced line but for stereo.

The spec. sheet says maximum 125dB SPL, dynamic range 101dB, 1kHz
at max SPL. S/N 70dB 1kHz at 1Pa. 

The spec. sheet even says that positive acoustic pressure gives a 
positive signal on pins 2 and 3. (pin 1 is the ground/shield.)
Assuming that the DR-1 records positive voltage as positive values,
that means that my peaks were at negative acoustic pressure.


> Then, when you get clipping you'll know that your ADC was doing it's 
> best, at least.

I can't measure between the amp and the ADC, but I could between
the microphone output and DR-1 input. 

I should at least find out if the microphone built-in pre-amp or
the DR-1 is the source of the noise floor.

-- glen

On 05.03.2012 03:47, glen herrmannsfeldt wrote:
> robert bristow-johnson<rbj@audioimagination.com>  wrote:
>> On 3/4/12 8:12 PM, glen herrmannsfeldt wrote:
>>> I have a digitized WAV file about 71 minutes long (actually, 10
>>> files totaling 71 minutes) with three samples clipped, I believe
>>> not all consecutive.
>
>> only three isolated samples?  they're at 0x7FFF or 0x8000 ?
>
> All three at X'8000'. The peak light blinked at least twice,
> so two might be together.

Just a thought, if 3 single samples just a bit overloaded are audible, 
could it be that some software or hardware component, maybe during 
playback, does not process this value 0x8000 correctly, and outputs, for 
example, 0x7fff?

Andre <lodwig@pathme.de> wrote:

(snip, I wrote)
>> All three at X'8000'. The peak light blinked at least twice,
>> so two might be together.

> Just a thought, if 3 single samples just a bit overloaded are audible, 
> could it be that some software or hardware component, maybe during 
> playback, does not process this value 0x8000 correctly, and outputs, 
> for example, 0x7fff?

Well, the usual thing about clipping is that it generates high
frequencies where they weren't before.

If you slowly increase the amplitude of a 20Hz sine wave until
clipping, I expect you to hear the new high frequency components
at a fairly low level.

-- glen
>