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. 

The kicker is when recording in Audition 4.0 (from CS 5.5), I can choose up to 32 bit FP... but then in the save dialog the "sample type" is limited to 32 bit FP, but the "Format Settings" (same dialog box) goes up to 64 bit FP. If the processing is limited to 32 bit FP, I do not understand what would be saved in a 64 bit FP file. It is about impossible to find information on this from Adobe's website.

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.
 

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.
 
 

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
 
 

Helpful article - especially for my needs which is just a basic understanding. For now, recording at 24bit/48k as my default. However, I'm thinking of going back to what I used for years - 41k
 
 
 

Goddard

John
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
 

Care to point out in what way?

John, I don't think you will find satisfaction here unless a circular argument is your cup of tea. It's really not worth the trouble to argue with some guys.

Well, I'm not actually arguing, or at least not at the moment. I'm just finding Goddard strangely elliptical and a bit unconvincing. He certainly doesn't seem interested in explaining or discussing anything, but he does seem awfully keen for us all to think he's really clever. I'm trying to give him the opportunity to impress us.

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.

 
You've misunderstood. A converter is actually sampling its input at a far higher frequency (in the MHz "low radio frequency" band) than "ultrasonic". Yet remarkably, no reports yet of heterodyning artifacts showing up in recorded digital audio.
 
And "super-sonic" usually refers to the speed of sound (or an NBA team), not sampling frequency.
 
Btw, 44.1kHz is "ultrasonic" territory too. If a converter's anti-aliasing LPF isn't designed or working properly, well then... Fido will inform.
 

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.

No, he's saying a current laptop's converter is better (at CD audio sampling rate and bit-depth) than earlier generation pro converters, without grasping that the very same laptop about which he's talking is, thanks to its cheap little codec chip, well and truly capable of handling multi-channel 24-bit/192kHz digital audio (e.g. when playing a Bluray disc) without suffering any of the "problems" to which he later points for justifying not sampling at 192kHz. If there's a gap in his knowledge (and there appear to be many) it's that he can't recognize that the example he posits (the capability of that laptop's measly little codec chip's converter) actually disproves a lot of the baloney he later foists as being problematic of 192k sampling.

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.