There are, in fairness, good engineering reasons for using higher rated converters at the recording stage. Simply put, it's a lot easier to build a good, non-artefacting gently sloped cutoff filter than a steep one. If your converter can work at a higher rate, it's entirely possible that you can get better recordings at the initial conversion stage.
 
Some points to note are:
- after that initial step, there's no value in the higher rate.
- manufacturers don't tend to publish very clear or reliable information about how their converters are operating
- a 44.1 converter may well be already dealing with this via an initial oversampling step
- there's a definite law of diminishing returns here. Once the filter isn't causing artefacts, it isn't causing artefacts. You rapidly reach a point where increasing the sample rate gains nothing.
 
It is all a lot less like rocket science than it's been presented throughout this thread.

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.

Over the years I've become convinced that the clock plays a pretty big role in the sound quality of a given DAC or ADC process. I've personally heard e.g. the old Digi 002 improve dramatically with a different clock. My friend who studied sound engineering told me they run a rack of Behringer converters at his school that they clock from an Apogee, also greatly improving the sound.
 
 Many soundcards that list similar specs sound very different, and what a given chips maximum sample rate is doesn't really seem reflect on what its sound quality is like. It seems akin to the photography thing where they pretend that using more megapixels gives you a better picture. I'm completely convinced you can get a good sound at 44.1K, 24 bit, as long as you have a decent converter and a good clock. Additionally, I don't think you can entirely "solve" a jittery clock problem by doubling or quadrupling the sample rate.

Yeah, clocks matter. Again, though, this is one of those things that is, in general, a lot better than it was ten years ago. It's not that easy any more to buy an audio interface with a truly sucky clock.
 
I like your photography analogy. You're quite right; the overall issue is how all the components are working together as a system. There's no simple "More X is better" "More Y is better" here.

I managed to fix my settings as advised earlier in the thread. The maximum that my soundcard is specified for recording is 96 / 24, but I could set the bit rate up to 64 bit in the preferences, but havnt done any audio recording yet so don't know if that will work, I'm just focusing on getting the midi backing tracks done for my ideas atm.

I'm of the view that you can agonise too much about this stuff. If you're not an electronics engineer specialising in digital audio, then it's sensible - to some extent, I'm not arguing for foolish credulity - to assume the equipment designers have done their job properly, and get on with yours, which is creating and recording music.

John

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
 
 

Sorry I'm not following you. If the recorded data/audio is only 24 bits how can converting it into 32 bits add further data? It will not become audio with any more detail or precision. It is limited to the 24 bits that the A/D converter produced. If it is processed it is possible that the resulting audio will have greater precision due to an internal 32 bit FP audio engine but it will be inaudible.
 
It really should be similar to recording at 16 bits and then converting the 16 bit file to 24 bits. You gain nothing except a patted file with the original 16 and a bunch of zeros filling out the rest of the file. Its like having a container that you fill up. No interpolation occurs simply because the file is bigger.  
 

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).
 
If you stilll can't comprehend from the preceding and from that wiki page how all the data of a 24-bit integer PCM sample (plus 1 extra bit's worth!) is represented by all the bits of a 32-bit floating point binary number representation then I'd suggest searching for another online explanation of floating point binary math which is hopefully easier to grasp.

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.

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?

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.

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.

As a matter of record, I didn't criticise yer man for historical accuracy at all. And even if I had, it would be irrelevant to what we're discussing. Your fondness for ducking the point at hand is not unnoticed. I give you notice that it's not likely to fly.