Goddard

John T

Goddard

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.

Uh, a Mac laptop's codec chip can record (sample) @24/192 too:
 
http://www.realtek.com.tw/products/productsView.aspx?Langid=1&PFid=28&Level=5&Conn=4&ProdID=138
 

I'm sure it can, but it's not relevant to the point the guy is making. He's simply saying that a bog standard consumer grade converter running at 44.1/16 is actually very good these days. If you want to get stuck on the fact that he's used the throwaway description "laptop mac converter", then go for your life. It's got nothing to do with the matter at hand.

Ah, so your earlier criticism of the guy for some historical inaccuracy about videophones was somehow relevant to some point the guy was making, but my pointing out that the guy's own "Mac laptop converter" example undercut all his hooey about 192kHz sampling being problematic isn't relevant and has nothing to do with the matter at hand?

It wasnt a criticism, it was a mainly humorous observation following up on bitflipper's post. I hope I'm being clear about this. I think the guy makes very good arguments. He seems to have gaps in his knowledge. But you know, so do we all. They matter when said gaps are fatal to the arguments we're making. In this case, hey are clearly not.

Goddard: let me ask you this. What's your point here? What are you batting at?

John T

Goddard
 
You've misunderstood. A converter is actually sampling its input at a far higher frequency (in the MHz "low radio frequency" band) than "ultrasonic".
 

This is flat out wrong. You're conflating the device's means of operation with its function. A converter is sampling its input at its functioning sample rate. There's all kinds of jiggery pokery going on with how it does that, some of which yer man touches on in his article.

No, it's not wrong, at least not as concerns Delta-Sigma converters employing oversampling, which for many years have been the prevalent type of ADCs and DACs used for audio (although thinking back, Lavry's and perhaps also some others' early "mastering" converters may have used conversion bits having a different archtecture (multi-pass flash?) way back when).
 
Your "conflating the device's means of operation with its function" statement is nonsensical and naiive.  Perhaps you should learn a bit about the "jiggery pokery" how of things before posting any further arguments. This might be informative:
 
http://skywired.net/blog/2011/07/how-delta-sigma-works-anti-aliasing-advantage/
 
And this might lend a bit more perspective than that facetious "Science of..." blog:
 
http://www.benchmarkmedia.com/discuss/feedback/newsletter/2010/08/1/unique-evils-digital-audio-and-how-defeat-them
 

We appear to be at an impasse. I think you're entirely wrong, but you seem to be unwilling to actually argue your case. I'm not sure what to do with that. Of course, you're free to post all the links you want to.

Again, I wonder: what is your point?

John T

Goddard

John T
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.

I really don't care whether you or anyone else is impressed, only that misinformation isn't perpetuated unchallenged, or taken in by the naiive and gullible who don't know any better, when those who should know better praise it.
 
You and others criticized that blogger for historical inaccuracy earlier, and I've merely done the same wrt obvious technical fallacies in that article.

In principle, this is a fair point but it's undermined by the fact that you've been profoundly incorrect in most of your claims. For example, your thing about MHz sampling. That was just nonsense.

 
MHz sampling (oversampling) is a reality in DSM audio ADCs and DACs, even if you are unable to comprehend that it occurs. You're confusing the actual sampling rate of a DSM converter (the frequency at which its input is sampled) with the converter's output sample rate.
 

John T
If you've a counterpoint to that, or anything else in this conversation, I'm honestly eager to hear it. If you're not willing or able to to counterpoint anything, though, well... I dunno what to do with that.

 
I've already responded as much as I care to. Done playing your time wasting games. Don't really care if you're persuaded. I'll leave it to others and posterity to assess any technical merit in what I've posted.

Well!  
 
Found this an interesting skim read... I've migrated to 96/24 recently to keep Latency reasonable on my new Focusrite Forte, which is exquisite in all other respects...and while I haven't had the time to delve into Goddard's references, I have bookmarked this thread as these issues intrigue the "infomaniac" in me; most of the big boys have a Tesla for a reason... if I might use an analogy... 
 
Fortunately, my DAW seems to handle this just fine and what am I giving up until fuller comprehension unfolds in my mind? Nothing!  And said links and references will at least be read by me, stumbling around and all as I learn the ropes with this magnificant tech we are so fortunate to be able to wield these days
 
Goddard, one point interests me in particular, as it relates to increasing RECORD BIT DEPTH  to 32, which I am going to try after reading this:
 

Goddard
 
 
A simplified analogy often given is to "scientifc notation" of decimal (base 10) numbers, where the decimal point position is represented in an exponent (power of 10).
 
In a 32-bit floating-point binary (i.e., single-precision) representation, the 1-bit "sign bit" plus 23-bit "signficand" (sometimes called the "mantissa" as in logarithms) portions together carry the basic 24-bit integer binary data (actually 25 bits' worth thanks to the "implied" leading "1" bit) while the 8-bit "exponent" portion carries the data indicating the position of the binary decimal point. The use of a floating point representation enables, by virtue of the exponent, a vastly wider range of possible values than can be represented by a 24-bit integer binary number, which is why a 32-bit foating-point audio/mix engine doesn't suffer risk of clipping (whenever the value would exceed the significand's maxium possible value, the exponent value can be increased).
 

As I am just beginning to tackle the art and science of recording/mixing my creations I would be very much interested in 32 bit float if it reduces the risk/digital distortion associated with level spikes or accumulations thru combined signals... especially if it doesn't interfere with overall DAW performance too much as I still have margins available as long as I make love to DIVA sparingly!
 
Speaking of "love"...I love ambience...stereo imaging, deep effects pushing and pulling at our spirits...I've acquired a lot of them and I plan on using them...so,  lot's of data to be massaged...and will 32bit float handle what I see as magical overload better??? My gut says: YES But I'd love to know what the math says.... say, 10 instances of Timeless2,  a few amp sims, 3 or 4 instances of "Spaces" and 10 other manglers, with AVOX, chorX, various channel tools...essentially loads of imaging, processing and complex signal chains... Does 32bit float play better in such an environment, where the sum of all the parts is WAY bigger than any one particular piece of the puzzle???
 
Please, gurus.... answer me that...with real world data or mathamatical extrapolations... as climate scientists are trying to do every frackin' moment we have left...
 
I LOVE open, clean yet emotionally powerful music and while I recognize most of that is achieved via mixing/mastering skills/tools... why not also employ this "edge" or "hedge" as I'm quite sure HZ and every other major producer does... and probably for a reason...
 
better safe than sorry??? What's the cost?  Doesn't it make sense till available cpu overhead is exhausted? Hell, by then I'll have moved to an 8 core haswell! What joys befall a lowely gardener doing eco/save the world music these days!
 
Such things should be measurable... and I am a deep believer in subtle nuances of nature and reality all around us which we can not necessarily SEE but are profoundly influential... 
 
Sadly, most childeren, including my own youngling will die of truths being distorted in an emotionally laden fashion...
 
which brings me back to mathamatics/measurements and not rhetoric... neither side has debated/reflected/responded as I would have had to argue during many years in GAO/OTA technical analysis' of very complex and charged issues...
 
Interesting thread... and I couldn't resist....
 
and anyone... Does 32bit float even touch on what I've postulated above?  Links, please if u got um...
 
 

Very little of what you wrote has anything to do with bit depth, I would say.
However, music is an emotional thing too, so if you are feeling better using 32 bit fp, knock yourself out.

Sander...Just trying to understand the implied additional headroom/processing implications... If "I would say..." is based on a real understanding of what God was sayin' (couldn't resist!) then thats good to know although "very little" tells me "very little"... I for one don't comprehend the full breadth and credabilities of the arguments yet but my curiosity was piqued by Goddard's comment...
 
It doesn't make me feel better or worse... information/data does that for me... I admit to being stupid or ignorant about a zillion things... but try to understand everything...just haven't delved and practice suspending judgement/positions till I understand...and never get stuck there, as new info always unfolds...
 
One thing I think I learned this week is to always record stereo to retain track bin imaging opportunities...all my mono tracks are stuck there... thus, smart to record stereo just in case...BitFlipper's eloquent solilaque on mono/stereo really helped me decide what works best for me.... Seems the same for 32fp, lacking reasons not to as I can always reduce it if the issues are clarified...but never truly realize any advantages by going there after the fact. I was hoping for some real data on the above...it helps wade through the human tendencies of denialism/false beliefs on either side of many issues which stem from ignorance or lack of compelling/comprehensive data/understanding.
 
 

Sander...Just trying to understand the implied additional headroom/processing implications... If "I would say..." is based on a real understanding of what God was sayin' (couldn't resist!) then thats good to know although "very little" tells me "very little"... I for one don't comprehend the full breadth and credabilities of the arguments yet but my curiosity was piqued by Goddard's comment...
 
It doesn't make me feel better or worse... information/data does that for me... I admit to being stupid or ignorant about a zillion things... but try to understand everything...just haven't delved and practice suspending judgement/positions till I understand...and never get stuck there, as new info always unfolds...
 
One thing I think I learned this week is to always record stereo to retain track bin imaging opportunities...all my mono tracks are stuck there... thus, smart to record stereo just in case...BitFlipper's eloquent solilaque on mono/stereo really helped me decide what works best for me.... Seems the same for 32fp, lacking reasons not to as I can always reduce it if the issues are clarified...but never truly realize any advantages by going there after the fact. I was hoping for some real data on the above...it helps wade through the human tendencies of denialism/false beliefs on either side of many issues which stem from ignorance or lack of compelling/comprehensive data/understanding.
 
 

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