John T

Goddard
Sigh... wotta buncha hooey. Facetious scientist indeed.
 
Surprise! ADCs actually sample at frequencies in the MHz even if they only output PCM streams at 44.1/48kHz. And DACs oversample 44.1kHz audio streams (in MHz) too! That's not ultrasonic, it's radio frequency (and relates to why the use of a CD player is prohibited at times on airliners). Not to worry though, decimation and lowpass (and often highpass) filtering fortunately keeps the out-of-band nasties from getting through (at least, its supposed to if things are designed and working properly).
 
Another surprise: the cheapo onboard 'high definition audio" codec chip inside the typical PC/Mac can handle 192kHz digital audio (such as one might find as a primary audio stream on a Bluray disc) just fine (by design).
 

None of that seems to address anything in the article. I'm not sure what point you're making here.

But be careful of designers who go for super-sonic sampling rates and set their filters too high. If you include too much super-sonic information in the signal it becomes likely that you will introduce super-high frequency “intermodulation distortion” on playback.

Now in 2013, the 16/44.1 converter of a Mac laptop can have better specs and real sound quality than most professional converters from a generation ago, not to mention a cassette deck or a consumer turntable. There’s always room for improvement, but the question now is where and how much?

Goddard

John
"This is probably a good place to ask this question. I did my first-ever sampling last weekend, and ended up using Audition to do the task. I recorded them 44.1KHz mono (drums), but Audition defaulted to a 32-bit float on saves so I just used that. Is that 32-bit buying anything at all, or just wasting space on me?"
 
 
For recording yes its wasting space. Keep in mind that your converters are incapable of recording anything above 24 bits. So the file that was created is 24 bits plus a lot of padding. This adds nothing useful to the recording at all.  
 
Now for processing it a very different story. However, I am of the opinion that the file on disk doesn't need to be greater than 24 bits even after processing.   I am sure I am alone in this view.  

No, the 32-bit floating-point file data saved by Audition isn't padded, it's just being represented in floating-point format (and using all of those 32 bits for its data) rather than in the fixed-point/integer 24-bit format as was output by the ADC. If it were being saved as 32-bit integer data instead, then yes, it would include padding.
 
Perhaps Noel (if he's still around) would care to confirm whether/when Sonar converts to integer format for disk storage (as WAV files iirc).

Goddard
My "Surprise!" statement was in response to what that blogger wrote here (about super-high frequency IM):
 

But be careful of designers who go for super-sonic sampling rates and set their filters too high. If you include too much super-sonic information in the signal it becomes likely that you will introduce super-high frequency “intermodulation distortion” on playback.

 
(Lavry's papers do refer to such high converter "modulator" sampling frequencies but I suspect the facestious scientist has no idea how converters actually operate)  

Goddard
And my "Another surprise" above was in reply to what the blogger wrote here:
 

Now in 2013, the 16/44.1 converter of a Mac laptop can have better specs and real sound quality than most professional converters from a generation ago, not to mention a cassette deck or a consumer turntable. There’s always room for improvement, but the question now is where and how much?

 
(last time I looked inside a Macbook (several years ago), it used a 192kHz-capable Realtek  chip in its "Intel HDA" componentry, but I suspect the facetious scientist really has no clue of what "the 16/44.1 converter of a Mac laptop" is actually capable)

John T

Goddard
My "Surprise!" statement was in response to what that blogger wrote here (about super-high frequency IM):
 

But be careful of designers who go for super-sonic sampling rates and set their filters too high. If you include too much super-sonic information in the signal it becomes likely that you will introduce super-high frequency “intermodulation distortion” on playback.

 
(Lavry's papers do refer to such high converter "modulator" sampling frequencies but I suspect the facestious scientist has no idea how converters actually operate)  

 
So if I'm reading you right, you seem to concede that he at least is reporting the information he has correctly, but because you suspect his knowledge doesn't extend to point X, you're dismissing him. But not because you think he's wrong. That's kind of odd.

John T 

Goddard
And my "Another surprise" above was in reply to what the blogger wrote here:
 

Now in 2013, the 16/44.1 converter of a Mac laptop can have better specs and real sound quality than most professional converters from a generation ago, not to mention a cassette deck or a consumer turntable. There’s always room for improvement, but the question now is where and how much?

 
(last time I looked inside a Macbook (several years ago), it used a 192kHz-capable Realtek  chip in its "Intel HDA" componentry, but I suspect the facetious scientist really has no clue of what "the 16/44.1 converter of a Mac laptop" is actually capable)

Same thing, really. His point stands regardless. He's talking about it being better than tape decks or turntables even operating at fairly low rates. That's correct. You're dismissing him because he doesn't seem aware that it can run at higher rates. That's a gap in his knowledge, sure, but not one that undermines his point.

Goddard

John

Goddard

John
"This is probably a good place to ask this question. I did my first-ever sampling last weekend, and ended up using Audition to do the task. I recorded them 44.1KHz mono (drums), but Audition defaulted to a 32-bit float on saves so I just used that. Is that 32-bit buying anything at all, or just wasting space on me?"
 
 
For recording yes its wasting space. Keep in mind that your converters are incapable of recording anything above 24 bits. So the file that was created is 24 bits plus a lot of padding. This adds nothing useful to the recording at all.  
 
Now for processing it a very different story. However, I am of the opinion that the file on disk doesn't need to be greater than 24 bits even after processing.   I am sure I am alone in this view.  

No, the 32-bit floating-point file data saved by Audition isn't padded, it's just being represented in floating-point format (and using all of those 32 bits for its data) rather than in the fixed-point/integer 24-bit format as was output by the ADC. If it were being saved as 32-bit integer data instead, then yes, it would include padding.
 
Perhaps Noel (if he's still around) would care to confirm whether/when Sonar converts to integer format for disk storage (as WAV files iirc).

Where is the audio that is filling those extra bits coming from? My understanding is that floating point will reduce rounding errors if the file has been processed and with a 32 bit FP audio engine used. Otherwise it will be exactly what the converter created. 

Your understanding is incorrect/incomplete:
 
http://en.wikipedia.org/wiki/Single-precision_floating-point_format
 
 

dubdisciple
Anyway, this thread has left me more confused then when i started it.

John T
Sampling and playback are not the same, and don't exhibit the same problems. That the mac can playback 192khz audio has nothing to do, for good or ill, with what he discusses in the article.