I can't read Gearslutz. It makes me genuinely depressed.

John
I think Goddard is confused about 1 bit audio recording and the rest of the audio recorders. 
 
This may clear up the confusion http://www.bhphotovideo.c...bit-better-24-bit.html
 
This also may help. http://en.wikipedia.org/wiki/Direct_Stream_Digital

I'd be wary of any source that claims DSD is superior.
 
And I think Goddard might just be pointing out that many converters use oversampling.

drewfx1

John
I think Goddard is confused about 1 bit audio recording and the rest of the audio recorders. 
 
This may clear up the confusion http://www.bhphotovideo.c...bit-better-24-bit.html
 
This also may help. http://en.wikipedia.org/wiki/Direct_Stream_Digital

I'd be wary of any source that claims DSD is superior.
 
And I think Goddard might just be pointing out that many converters use oversampling.

You're right but the one that is selling stuff is meant only to show the difference. Having never had occasion to hear or use DSD I have no opinion on its fidelity. 

I have to apologize for what this turned into! OMG, I "thought" I was asking a simple practical question that would be useful to myself (and others) and it turned into rehash of the last one. Ugh...
 
Just out of curiosity (and also because I didn't want to get caught up in it)... I pulled open a few of the 32BFP samples I had made and figured "if those extra 8 bits are empty (an entire byte/character), then the data portion of the 32BPF wav file should have repeating characters (every fourth character) even if opened with NotePad"... so I popped a few open, and basically scrolled to the middle, and found this to be true... the 4th character would repeat for a chunk of samples (no, I didn't count how many), and then shift to a new character that had the same pattern. I scrolled around a bit, then looked at couple other files, and saw the same thing. My "logic" was that even if those characters were not aligned properly to the samples themselves, the 4th character (8 bits) should repeat if "empty." (More layman logic, but was enough to confirm to me what John responded to my initial question... essentially that 32BFP files on disk were not buying me anything (other than 25% wasted space). Why Adobe even has a 64BFP file format available makes even less sense to me now.)
 
For me, this circles back to where it all started, and John's initial reply is still "my" answer, which was that anything beyond 24bit on disk is not gaining me anything, but for processing/rendering higher bit depths should be used (which is pretty standard in all DAWs anyway, so not an issue).

mettelus
Just out of curiosity (and also because I didn't want to get caught up in it)... I pulled open a few of the 32BFP samples I had made and figured "if those extra 8 bits are empty (an entire byte/character), then the data portion of the 32BPF wav file should have repeating characters (every fourth character) even if opened with NotePad"... so I popped a few open, and basically scrolled to the middle, and found this to be true... the 4th character would repeat for a chunk of samples (no, I didn't count how many), and then shift to a new character that had the same pattern. I scrolled around a bit, then looked at couple other files, and saw the same thing. My "logic" was that even if those characters were not aligned properly to the samples themselves, the 4th character (8 bits) should repeat if "empty." (More layman logic, but was enough to confirm to me what John responded to my initial question... essentially that 32BFP files on disk were not buying me anything (other than 25% wasted space). Why Adobe even has a 64BFP file format available makes even less sense to me now.)

It's not as simple as you assume - floating point consists of a sign bit, a series of exponent bits that scale the data up or down, and the actual number ("fraction") as described here (there is also an implied leading digit for non-subnormals that I mention mostly so the technically minded folks don't correct me):
 
http://en.wikipedia.org/wiki/Single_precision_floating-point_format
 
 
 
What ends up happening is your 24bit audio goes into the sign bit, the implied leading bit, and the fraction. The exponent then scales it up or down appropriately. IOW, the extra 8 bits are not just padded zeros because it's not just more bits; floating point means the bits represent things differently than in fixed point (aka integer).
 

For me, this circles back to where it all started, and John's initial reply is still "my" answer, which was that anything beyond 24bit on disk is not gaining me anything, but for processing/rendering higher bit depths should be used (which is pretty standard in all DAWs anyway, so not an issue).

If you are going to be processing further, 32bit floating point can be better because if you happen to have any samples that go over 0dBFS they will not be clipped, and low level precision is preserved if you subsequently raise the volume of the signal. I recommend it for interim formats (i.e. transferring to another programs for further processing or to external mp3 encoders) and for Render Bit Depth - even though might not really be necessary in most cases.

I guess the somewhat less technical explanation of the point I was making is that the 4th character isn't going to change in the way you might have been thinking because the translation between fixed and floating point is not straightforward. 

John
I think Goddard is confused about 1 bit audio recording and the rest of the audio recorders. 
 
This may clear up the confusion http://www.bhphotovideo.c...bit-better-24-bit.html
 
This also may help. http://en.wikipedia.org/wiki/Direct_Stream_Digital

You think I'm confused? Hah, that's rich.
 
Ar least you're getting warmer with your DSD/SACD links, although you still appear to be unaware of how oversampling multi-bit DSM ADCs and DACs (as equpped in almost all audio interfaces for many years now) operate, or of the effects/advantages of oversampling wrt the Nyquist frequency and filtering requirements/effectiveness (see previously linked skywired.net blog post).
 
FYI, there is a very simple reason (of which I strongly suspect that facetious scientist/music technology educator is completely unaware) why the lowly onboard sound chip in every PC/Mac for many years now (at least since Vista and Intel Macs) is capable of competently performing 24-bit (yeah, I wrote "24-bit" there, not "1-bit") @192kHz multi-channel recording and playback (and doing so simultaneously, i.e., at full duplex), and that reason comes down to Intel's "High Definition Audio" (HDA) specification (successor to the earlier "AC97") along with MS' stringent (as in, requiring use of an AP audio test rig) Windows logo cerification testing requirements, with which all those lowly little cheapo onboard sound chips must conform and comply before the millions of systems into which they are equpped ever reach market.
 
I suspect that you don't bother actually reading any of the material I've posted links for, as you've raised no objection or question in relation to anything I've linked-to, despite your earlier insistence that I cite autority for my critique of that facetious science blog, but I'll post some more links anyway just in case anyone else might genuinely be interested in this stuff:
 
http://www.intel.com/content/www/us/en/standards/high-definition-audio-specification.html
 
(I'll leave it for the curious to learn the reason for inclusion of 24-bit/192kHz in the HDA requirements)
 
http://msdn.microsoft.com...8866%28v=vs.85%29.aspx
 
http://msdn.microsoft.com/en-us/library/ff563343%28v=vs.85%29.aspx
 
http://msdn.microsoft.com/en-us/library/ff563349%28v=vs.85%29.aspx
 
http://www.realtek.com.tw/images/products/High_Definition_Audio_Codec_Selection_Guide_07182008.jpg
 
And of course, fwiw Sonar has supported 24/192 (and even 24/384) samples for some time now as well (since Sonar 4 iirc).
 
But, instead of focusing only on lowly onboard audio chips whose specs and performance are dictated by Intel and MS, let's see if I can't make more apparent the actual situation wrt more serious audio production gear which some folks hereabouts might likely be using (and let them judge for themselves whether the "problems" with 192k sampling raised in that facetious "Science of..." blog are a reality or merely non-existent pseudo-science baloney) by giving a real world example of a generally well-regarded 24-bit @192kHz-capable "audio interface". Say, one which was touted as featuring "mastering grade" converters just like those on the PTHD192 rigs used by real "pros", namely the E-MU 1212m:
 
http://www.creative.com/emu/products/product.aspx?pid=19169
 
on which, per the specs given, the following ADC and DAC chips are respectively employed:
 
http://www.akm.com/akm/en/product/datasheet1/?partno=AK5394AVS
 
(notice that AKM list "128x Oversampling" and "multi bit Architecture ADC")
 
http://www.cirrus.com/en/products/cs4398.html#
 
(note the "oversampled multibit Delta-Sigma modulator")
 
Now, just to avoid there arising any confusion, in case of any doubt as to how such oversampling multi-bit DSM converters actually operate when converting to and from 24-bit 192kHz samples, simply follow the links on the above pages to access the data sheets. And iirc, the Cirrus Logic/Crystal Semi developers published at the time about their clever stuff, so anyone interested in the real esoterica might search that out (possibly behind the IEEE's paywall though).
 
Not that any of this oversampling stuff is really all that new or even magical, even if it can be difficult to grasp at first. Drewfx alluded to it very briefly in a post early on this thread, although you may not have grasped what he was saying. Here's an article from the early DAW days ProRec webzine in which the principle of exchanging sample rate for bit-depth was given treatment (in the very last section):
 
http://web.archive.org/we...FA5EAA862566B20022F4CA
 
(sigh... I do sorely miss having Jose on this forum, those were some intelligent discussions, and CW benefitted too)
 
So, no, I'm not confused, John, because (unlike you and JohnT) I've actually known about and understood sampling and oversampling (and even floating-point math too) for a very long time, and very well know the difference between 1-bit DSD and 16/24-bit PCM (and even how to convert between the two).
 
And just to be clear, absolutely nobody has to take my word for anything which I've asserted, as I've posted links to independent references which explain and/or corroborate any technical points asserted. That is, if anyone ever actually reads what I've linked-to rather than just carrying on with hiking their post count in oblivious ignorance.
 
So by all means please don't take my word about oversampling DSM converters actually sampling in the MHz range, when you can instead read Dan Lavry's very own words saying so (to make it easier for those too lazy to follow a link, I'll just copy-in an excerpt of the relevant bit):
 
 

Dan Lavry
What rate and bits are enough for today music reproduction and recording?
 
Regarding the rate:

One has to make a distinction between the audio sample rate and the rate of a localized process:

The audio sample rate is the rate that carries the music data itself. Roughly speaking, the audio bandwidth itself is slightly less then half the sample rate. A 44.1KHz CD can contains music to about 20KHz.

At the same time, there are many cases when we use much higher “localized rates”. Such higher rates do not increase the musical content. The higher rates still offer the same original bandwidth of the sample rate. We up sample or down sample between localized rates for various technical reasons. For example, virtually all modern DA’s operate at 64-1024 times the sample rate speeds (in the many MHz range). Operating at such high rates simplifies the requirements of the anti imaging filter (an analog filter located after the DA conversion). The decision about the ideal localized rate depends on the technology and the task at hand. It is an engineering decision, not an ear based decision. As always a poor implementation may introduce Sonics, and it would be wise to refrain from the often encountered practice of far reaching false generalizations, so common in the audio community.

http://www.monoandstereo.com/2008/06/interview-with-dan-lavry-of-lavry.html
 
So, now, who exactly is confused (or uninformed/just plain ignorant) here?
 

John
It bothers me that I and others have been accused of giving out misinformation. This is something I have been very much fighting against ever since i have been on this forum. I have been wrong in the past. Not often, however.  When I am wrong I will broadcast that fact and try my best to correct the error. 
 
Goddard has accused me and just about all that have participated in this thread of being wrong. In fact it is he that is totally wrong. he has somehow confused 1 bit recording with 20/24 bit PCM recording. They are very different things and are interesting as a study in their differences but the technologies are very different. The one has no reason for being interjected in this thread. 
 
 

Well, it bothers me when people with "platimum" post counts who've been around this forum for years continue racking up their count by posting ignorant and uninformed drivel, when they really should know better or at least have learned some accurate info by now. Just like it bothers me when people get taken in by the baloney a facetious/pseudo scientist blogger posts (and especially when CW's CTO links to it!).
 
Are you still unable to grasp floating-point math, John? I mean, I can see from this old forum thread
 
http://forum.cakewalk.com/A-question-about-the-64bit-engine-m1724023.aspx
 
that you were uninformed 5 years ago, although I can see that being chastised then didn't matter or prevent you from posting more misinformation then so why should now be any different...
 
No need to thank me for taking time to explain 32-bit fp math to you-- even if you still can't grasp it hopefully someone else seeking knowledge may benefit.
 
Yeah, it can be so confusing, where those bits go, or how it is that a 32-bit DAW application can perform 64-bit floating-point math.
 
How does that old adage go? Better to keep silent and let folks guess whether one is a fool , than to open one's trap and dispel any doubts?
 
Perhaps a modern equivalent might be along the lines of better to lurk and learn than let fly with the keyboard. But then, can't work up a platinum count with years' worth of uninformed misinfo that way.
 
Anyway, folks are free to judge who is confused or misinformed or a fool.
 

John
OK before I let myself get caught up in putting some one down let it end here and now. Often threads like this can and often do result in conflict. That is not the way we should let things happen any more. Goddard was only posting what he thought was true. There is no sin in that. 
 
I like it when some one posts a correction when they believe it is needed. We need to let members feel this place will not jump on them just because of a disagreement. 
 
We need to keep this place free of intimidation or make people feel uncomfortable no matter what is posted.   

Oh, too late for that now.
 
That you or anyone else is unable to discern that I might actually know of what I write is not my fault, nor is it my responsibility to educate you/them.
 
Post misinfo if you must (or don't know any better). But be forewarned that other folks more knowledgeable than you may well undertake to point out the errors therein so that others seeking knowledge here might hopefully not become as misinformed (or remain as ignorant) as you clearly appear to be.
 
The level of knowledge and expertise around here has truly and sadly declined. Was a time when a lot more knowledgeable folks participated here and on the forerunner newsgroup which at one time was on the cutting edge in many ways concerning DAWs and digital audio. Maybe they just tired of the ignorance, er, noise level. Let's see if we can't do something about that.

bitflipper
Gee whilickers, it's been awhile since we've had a heated multi-page technical discussion! And one with a pretty decent signal-to-noise ratio, too. (Look up similar threads on Gearslutz to see just how uninformed and rude such conversations can get.)

 
Oh, sometimes the threads on GS aren't really all that uninformed (or even rude) at all...
 
http://www.gearslutz.com/board/geekslutz-forum/771247-oversampling-what.html
 

bitflipper 
If nothing else, this has prompted folks to seek additional self-education on the subject. Way to go, CW forum.

Have a good journey on your quest for self-enlightenment!

I'm confused as to what's being argued about and am really trying here... seems Goddard is focusing on converters and other hardware issues but how does this relate to the settings we use within our DAW?  My Focusrite Forte supposedly has great components and truly sounds magnificent... but if Sonar already uses higher rates/bit depth to deal with digital distortions and all the other yaya within the DAW's elaborate signal chains to output clean signals to the master buss feeding the interface, is anything gained by outputting high rates/bit depth to the interface as long as we don't overload that last buss? 
 
Goddard, I did peruse all your links and found them not really at odds with many of the bottom line thoughts conveyed here... as it relates to what we should be inputting/outputting....as modern daws deal with the summing/additive signals (or cumulative, all my dum dum words) internally to provide the headroom for boiling it all down nicely... if I understand it right... a few indicate 96/32float is beyond necessary but some may feel prudent
 
And at least to this layperson, the points Goddard makes seem to be validated by modern DAW designs which deal with these issues...internally.  If the DAWs do their job with this and we do our job at inputing/outputting reasonable signal levels, isn't it modern DAW design that reduces the need for higher rates/depth on the input/output side of things?  In that respect, both sides of this argument are valid! Peacepipe anyone???
 
I think the anger on both sides is unfortunate and don't really get it...when the issue boils down to what makes sense to create coherent and awesome sounding music...seems that's the goal most of us are striving towards...which really rests on mixing skills, listening skills, and developing the rigor and understanding of how to wield all the tools we have available.  
 
I must say, this is the first time I've seen egos go kinda weird around here.  On both sides.  
 
BTW, drewfx... really appreciated your tackling my spew trying to comprehend it all... nicely done.
 
 

I tend to agree, Alex at least after looking into all of this... never did before! I must say I have improved the quality of creations FAR more by upgrading to the Forte interface than anything... probably because of some of the technical issues being bandied about here and implemented in uber quality pres and dacs... I am truly blown away by the a/b comparison of the Forte and a couple other interfaces I have used and tried, leading up to my getting it... just night and day... way more significant than whatever's at stake here...
 

So to summarize, 96 kHz and 24 bit depth rate is all that anyone will ever need for making music, because human ears are pretty terrible.

And 44.1 KHz and 16 bit depth rate are still plenty enough. As for CDs, do people actually still buy those instead of using itunes or similar MP3 downloads anymore? Digital downloads (legal ones!) have totally destroyed CDs now, and its a good thing too because it eliminates CD printing and distribution costs. For video games at least, just about everyone uses digital downloads now, I though music was the same, its unbelievable convenience.

The internet is for porn, no wait, arguing!

robert_e_bone
This thread has gotten way out of hand.
 
Bob Bone

From bits to digits!