It seems to me like you're saying that 24 bit recording allows you to capture more dynamic range than 16 bit recording.

I'd have to agree with that.

well, yes technically, but...
I am also saying that the benefit of this is the clarity at or around the zero-crossings, not having the ability of juxtaposing "quieter and louder sounds" which is how most people think of dynamic range.

For example take a 24 bit file, and the same file carefully converted to 16 bit. They will have the same "dynamic range", or RMS to Peak. They will sound the same loudness everywhere. They will generate roughly the same "statistics" so to speak. Two average listeners would probably not notice any difference...Where they would differ is that, in fact, the 16bit file is more "grainy" mathematically speaking. Less resolution. Whether it is 16 bit or 24 bit, it does not actually change the musical dynamics. It captures more detail in the voltage change through the time domain.

When you convert to 16 bit, you lose some of the detail, but you retain the dynamics of the composition perceptually. Its like the same picture, but with less pixels per inch. Red is still red, but that spot in the picture where the red shades into some other color...not as smooth if you were to zoom in.

So my point is, the extra dynamic range is a de facto benefit of 24 bit recordings, but is manifested at the micro level (zero-crossings) not the macro level (eg - having a trumpet at -120dB and a snare a 0dB). In this case, you would turn down to hear the snare at reasonable level, and the trumpet would disappear, completely inaudible. So the dynamic range is not really helpful or relevant in that sense. But, the snare hit, as it rips back and forth from positive to negative, will have more detail encoded about those crossing points in 24 bit than it would in 16 bit. THIS increase in resolution to the original sound is why higher bit depths are good. This is the advantage a higher bit depth gives you.

Peace,
Nate

CrackBuddha wrote: well, yes technically, but...
I am also saying that the benefit of this is the clarity at or around the zero-crossings ...

which is what dynamic range is all about.

$ .02

well well... seems i've stumbled into something here... personally i've always found the resolution arguement interesting.... and i tend to agree with it... perhaps with a little different view.... electronically we start at an analog level .. and not withstanding increases in rail voltages over the years we can assume that the max/min that we display haven't changed... the noisefloor of the thing is still at ?? the max swing is still +/- 18V... so in that sense we dont have anymore dynamic range... at the point of conversion we stop time for in all intents and spit out a #which represents the nearist we can come to incrementally... so the resolution arguement then boils down to how many slices do we want to deal with in the verticle plane (amplitude)... the same arguement holds true for sample rate...on the horizontal if you will...

Excellent topic Nate!!

I couldn't agree with you more.

True, the signal to noise ratio is increased significantly, and this is a HUGE benefit during mixing and while applying effects. The true benefit comes from the nature of 24 bit being able to represent a higher quantity of voltage levels resulting in lower quantization errors. It's kind of like Mozart dynamics versus Mahler dynamics - both have PPs and FFs, but for Mozart to have gotten from a P to an F, he would have scored for more instruments whereas Mahler was able to (thanks mostly to musical instrument advances over the course of the 135 years) create P with a full orchestra as well as every dynamic in between it and FFFFFF...

Mozart was stuck with something resembling terraced dynamics and Mahler was free to write for an infinite level of dynamics.

Of course, that doesn't invalidate the beauty that is Mozart (and by analogous extension - CD), it simply shows the more simplistic nature.

In general, it isn't the lower noise floor and greater dynamic range of 24 bit that people hear as more pleasing, it is the greater clarity of dynamics represented within the scope of that range. So, not *DYNAMIC RANGE* per se, but dynamic accuracy.

J.

Re: $ .02

dementedchord wrote: the max swing is still +/- 18V...

now that's something I'd like to see - a delta-sigma A/D converter with a 36 v reference. That would add another 3 bits or so of resolution to the current 24 bits. ('course, voltages like that would probably punch right through the silicon)

One of the problems designers face is the ever-lowering maximum voltage levels on silicon. Years ago we'd have 15 v references on A/Ds, then 10, now 5, some probably down to 3.3. The problem is that the system noise floors aren't really changing.

24 bit is way beyond what we need from a dynamic range perspective but the resolution brings its benefits.

$ .02

understood but that's out of context in no way did i talk about +/- 18 v as the reference voltage for theconverter... what i was observing was more along the lines of general levels that we come from and eventially must return to ... incedently the general discussion was not specific to sig-delta type 1 bit converters but pcm...

The digital geeks like Nika and Bob.K and others ... including the Digi boys themselves ( Dave LB )
have often said that resolution in the way you have used it to describe PCM encoding is incorrect.

they say that 24 bit is not more resolute than 16 bit
it does provide for more dynamic range if required
if the signal is of low dynamic range then the detail of the sound at the given sample rate will remain the same.

I don't want to get into another deep discussion as we have done this so many times in the past.
Try some searches here and look for some of the very, very, old threads from the im and distant past with some og the guys from PSW and Gearslutz when they were mods he at RO.
... and some of the old stuff at Sweetwater Forum

Read some of the Bob K and Lavry notes and white papers and some of the Chip manufacturer white papers etc

Ultimately we will all use a new sytem and DSD could be it ... not yet though.

$ .02

yeah i'm aware of katz and lavery and aldridge to alesser degree ... just not sure why rexamining it from a slightly different perspective is such a bad thing... i always have thought anything worth defending should be worth rexamining...

if the signal is of low dynamic range then the detail of the sound at the given sample rate will remain the same.

am i to understand you to say that as an example a micro volt change at two different levels are not represented in the same way??? seems counterintuitive...

Yep, and I've gone on record several times disagreeing with these folks. My feeling is, the burden of proof is on them. They are trying to suggest that something is not how science and basic math (and some not so basic quantum physics) state that it is. Therefore, if they state otherwise, it is on them to back up their statements. This is something they have NOT done. They (both Dan and Nika) never get around to explaining WHY they suggest that higher resolution doesn't mean less errors (rounding/quantization) rather they go and explain why it means better noise floor/SNR.

Sorry, but just because Dan and Nika say it's so doesn't make it so. There's a lot of really simple math involved here. It's not as complicated as either of these gentlemen would have you believe.

In 24 bit or 16 bit 0=0 and granted there are 8 more bits on the "high" end of the binary spectrum of 24 bit, but that certainly does not mean that all of these 1s and 0s only affect the maximum or minimum amplitude of any given signal.

One good analogy is the bit range on digital photography. In 8 bits per color channel (24 bit), there are a certain amount of colors available. In 12 bits per channel (36 bit), there are significantly more colors available. This doesn't make blue any less blue, but if you take a picture in 24 bit of an image which is darn-near blue, you'll probably just get blue. However, in 32 bit, you will get "darn-near blue."

The fact is, with digital ANYTHING, audio, imaging, etc., the more values (in digits) you have, the more resolution between primary values you have.

What they're suggesting is that the ONLY difference between 4 bit audio (think Atari) and 16 bit audio is the SNR. Is this not absurd?

Are they suggesting that you reach a critical mass at some point and that point is clearly 16 bits? So, with 1 bit, 2 bit, 4 bit, and 8 bit audio, each incremental increase gains resolution, but once you hit that magical 16 bit threshold, you no longer gain resolution but only SNR?

No offense to either Dan or Nika, as I respect them greatly, but their arguments border on the insane. To me, it's apparent that they are attempting to put science before logic. Neither can exist without the other.

Sorry...Rant over.

Yea? Well... my DAW goes to 25 bit. That's more resolution than 24.

corrupted wrote: Yea? Well... my DAW goes to 25 bit. That's more resolution than 24.

Oh yeah?!?! Well I record everything at 2 bit. It saves hard drive space!

Cucco wrote: [quote=corrupted]Yea? Well... my DAW goes to 25 bit. That's more resolution than 24.

Oh yeah?!?! Well I record everything at 2 bit. It saves hard drive space!DIZZZAMN! Foo, you workin' fo ATARI up in this piece?

(Sorry to crap up your thread, I had a Spinal Tap moment and I apologize...)

corrupted wrote: [quote=Cucco][quote=corrupted]Yea? Well... my DAW goes to 25 bit. That's more resolution than 24.

Oh yeah?!?! Well I record everything at 2 bit. It saves hard drive space!DIZZZAMN! Foo, you workin' fo ATARI up in this piece?

(Sorry to crap up your thread, I had a Spinal Tap moment and I apologize...)

Fo Shizzle DAWG!

Besides, it ain't my thread you're "crapping up" (?) - I'm just here to bitch-slap conventional wisdom.

8)

I just started posting so my cred may not be high, but I have read ALL the posts here and probably anywhere pertaining to the subject. Again Cucco and I speak in concert; I disagree with anyone who says 24 bit is "not more resolute". This is why I started this thread. Total misunderstanding on this topic.
Again, anyone who says 24 bit is for capturing a wider MUSICAL dynamic range is missing the boat. It IS for capturing more detail at the lower voltages, which happen hundreds of times a second, no matter what the perceived musical dynamics are.
So literally hundreds of times a second, 24 is capturing and encoding information that 16 bit is incapable of.

The digital geeks like Nika and Bob.K and others ... including the Digi boys themselves ( Dave LB )
have often said that resolution in the way you have used it to describe PCM encoding is incorrect.

they say that 24 bit is not more resolute than 16 bit
it does provide for more dynamic range if required
if the signal is of low dynamic range then the detail of the sound at the given sample rate will remain the same

- Kev

Bob Katz (Bob K?) does NOT say this Kev :o You are misunderstanding bro. If you found someone who did, they are misunderstanding.

Like Cucco said, the burden of proof is on "them" :lol:
Dont believe what people say, believe what you can prove. The Earth is ROUND peeps!!! Represent!!!

Peace,
Nate

I'm no one to talk, but I can tell from a measly 2 years involved in recording, that my 24 bit converter sounds miles better than my old 16 bit converter. Mainly happens on things like cymbals (the high end sounds smoother or "nicer"..it was harsh in 16 bit), and vocals (more detail, more "reality" to the sounds). SO! I don't know...just felt like I had to stick my nose in and relate my limited experience to the topic I guess! ANDY

Cucco wrote: Yep, and I've gone on record several times disagreeing with these folks. My feeling is, the burden of proof is on them.

No offense to either Dan or Nika, as I respect them greatly, but their arguments border on the insane. To me, it's apparent that they are attempting to put science before logic. Neither can exist without the other.

Sorry...Rant over.

yep
and I know you have been in many of those and over a good portion of the time I suggested above.

The occasional rant is fine ... 8-)

I know what you maen about the nature of their argument at times.
This stuff is harder to explain and it does require the heavy maths.

I've been think about this again ... over night
I promised myself I wouldn't and as I said before I don't want to start yet another endless thread
BUT
I had a thought ... :shock: ... and even though it is not technically right it might help to bring a direction or point of perspective ... devils advocate if you like.

Vision / Video
1 bit video gives the two state situation. BLACK or WHITE
8 bit gives the 256 grey scale
but that scale falls between the same two extremes of the BLACK and the WHITE.
There is more information within the range.
... then comes the 16 bit and true colour stuff ... but still falls into the between BLACK and WHITE.
BUT cameras with CCD blocks can see into the Infra Red ... beyond the above range
???

PCM Audio
both 16 and 24 bit have the upper limit of 0dBFS ( Full Scale )
the largest 16 bit word send you down to the 16 bit noise floor
and the 24 bit work can send you LOWER.

so part of the expanded 16 to 24 bit is to expand the area we are working with.

wait for it

Does the expanded 16 to 24 bit ALSO provide for more detail within the OLD 16 bit area.

:roll:
get my drift ?

as I said above , this analogy is not strickly correct so don't all go juming on me ... just trying to present it in a deferent sort of way.

if that does sit ok then I can present the idea that a given signal captured into a 16 bit system can be identical to the same signal captured into a 24 bit system.

CrackBuddha wrote: Bob Katz (Bob K?) does NOT say this Kev :o You are misunderstanding bro. If you found someone who did, they are misunderstanding.

nup
I have a very clear understanding and Bob has at times used and chosen not to used the word Resolution
Currently he does use the phrase more resolved
http://www.tnt-audio.com/intervis/digidoe.html

In fact, with a little less lazy engineering, 44.1K/24 can be extremely good. (16 bit is out of the running; 20 and 24 bit at either sample rate is always superior, wider, warmer, deeper, more resolved). It's a lot easier (with the relaxed filter requirements) to make a 96/24 (or 88.2K/24) recording sound good.

is the filtering the dominant factor in what is being heard ? - a Kev comment only
... BUT in the past in some thread here ( I think) and at PSW has has tended to agree with Nika in that Resolution was less than a correct descriptor.

This is a very old subject and one must go back to the original PCM maths and we should use the words in the original context of PCM theory.

I think Lavry's white paper is a good presentation
Link removed
but it tends to focus on rates
... it would be nice to see an equally depth look at Bit depth and amplitude

Like Cucco said, the burden of proof is on "them"

and I totally agree
and feel that to date they have not done so well enough.
I fear that it will end up in a detailed look at converts and a chart full of mV values and digital words.

and meaningless to those that say ...

but I can tell from a measly 2 years involved in recording, that my 24 bit converter sounds miles better than my old 16 bit converter.

too often the white papers on bit depth end up being detailed on noisefloor and then introduce Dithering Theory.
It would be nice to see some detail on signals recorded in both 16 and 24 well above noise floor and below FS and see how the same signal compares in the two formats.

Buddha,
if you go back through some of those ancient threads you will see on many occasions I have banged heads with Nika and even though I do respect him greatly ... I often did not agree with many of Nika's presentations and interpretations. You may even find I have presented the exact agruments that you may chose to present here. I bring only part of an alternate point of view.

I think that when either side of a argument takes a point to an extreme, it can lose some credability unless it is based in an exact and restricted frame of reference.
Hence the use of Math and exact definitions that often seem to have no bearing on the types of signal we want to record.

The bottom line and beyond much of the math and theory is the fact that people can and reliably hear differences. Norrowing this down to just BIT depth as against the influances of Dithering and Error Correction and/or other algorithins contained within the hardware and chipsets is so very hard.

$ .02

curious that you bring up dither at this point as IMO thats what makes the argument for the resolution camp... all that little detail in the lsb end of things once truncated needs to be synthesized to make it sound right again... why would that be except that through poor gain staging brought on by the erronious belief that some perceived increase in available dynamic is going to save you from overs... when in fact you've simply been lulled into not useing the detail it affords you...

Re: $ .02

dementedchord wrote: curious that you bring up dither at this point ...

and bringing up dither is exactly what I don't want to do but nearly every white paper on the subject or 16 and 24 bit PCM audio , that I can find at the moment, does end up looking at noise floor and dithering

as I said above, it would be nice to see a detailed presentation of a limited dynamic range signal recorded in both 16 and 24 to see how they differ in accuracy to the original signal
and the same signal recorded at diferent levels with respect to FS in both formats and see which one is closer to the original and why/how

it can be difficult to present someone else's argument when you tend not to agree with their premise

I tend to like the way Roger Nichols presents a concept.
Even though this paper doesn't talk about the act of sampling and is more about subsequent processing and mixing,
it is worth a read
http://www.rogernichols.com/EQ/EQ_2001_03.html

If you get your input level up above -144 you will start to record some information. With no noise shaping or dithering your recording will be very distorted, but you can easily tell what is being recorded. At that level you are basically recording with a one-bit converter.

then it should follow that if you record at the other end and near to FS the detail should be at it's best
... I think this is dementedchord's line of thinking and to be honest I find it hard to side with Nika.

One of Roger's articles ends with the line

Now you will have to excuse me, I have to make room on my desk for my new 48bit/ 192kHz Digital Audio Workstation.

and it's a cool point
why don't we just make the leap now and be done with it or will it be a case of diminishing returns
and those golden ears will just continue to complain that ... "it's still not quite right"

roll on DSD ?

$ .02

boy i know what you mean about taking the otherside... i only hope that you (like i) find that it often helps us reevaluate our own beliefs... walk amile in my shoes kinda thing... i have always held rather strongly that any position worth defnding HAS to be worth reexaming... i appreciate you brother...

Re: $ .02

dementedchord wrote: ... i have always held rather strongly that any position worth defnding HAS to be worth reexaming... i appreciate you brother...

thanks mate

not easy and I've racked my brain over the last couple of days to find a new way to present the idea but I just can find a way that doesn't end up falling ito the same lsb and hard math sort of thing.
AND that math is not my strong point and although can follow I'm not the one to try to drive a point of view.

it is actually a harder subject to investigate than the more simple sample rate
even then the high sample rates 0f 96 and 192 has it's detractors
but
it is a much easier concept to see that 192 has more sample points in each second and therefore there is more data as the sampling takes place.
more = more resolution

The vertical or amplitude axis is clouded by the log scale and noise floor ... and dither ... and algorithms

An A to D for the sole purpose of tracking say temperature between a highest and a lowest defined point if much easier than tracking an audio signal.

I have a feeling that a presentation of PCM should be based around the zero crossing points
rather than the usual peak to peak presentations as found at the horrible ol' Wikipedia . :wink:
http://en.wikipedia.org/wiki/Pulse-code_modulation

Perhaps it is just easier to hope and accept that next years ADDA is cheaper and better sounding than the one I am currently using and in turn that was better than my old one.

I admit that I have been living that way since I traded my 888 for the newer 888/24 .... waaay back at around PT4.3
and now on HD with 96I/O's and yet I still record at 44.1/24.

What will the future bring ?
will it be higher sample rate AND an increase in bit depth ... 32 or 48 with an even bigger increase for the mix engines
or
will we jump from PCM and find a new method (DSD) ??

Can anyone remember the first time they were presented with serious digital audio ?
For me it was 81 or 80 and was the 16/44.1 destined to become CDaudio and I still have fond memories of the first Fairlight 16 I ever saw and heard (it was a CMI I think) .

Re: $ .02

Kev wrote: it is a much easier concerpt to see that 192 has more sample points in each second and therefore there is more data as the sampling takes place.
more = more resolution

i've always found it curious why it can't be held similarly... 2 to the 16th power versus 2 tothe 24th (no calculator sorry) if seen as discrete slices of the maximum 24 has 256Xs the # of possible representations....

The vertical or amplitude axis is clouded by the log scale

and perhaps thats where it breaks down for me... while i found math easy in school i chose not to take much... had to practice... and it's not too hard to imagine that some of that 256 is eatten up pretty quickly by a decade or two...

What will the future bring ?
will it be higher sample rate AND an increase in bit depth ... 32 or 48 with an even bigger increase for the mix engine

it's already a lovely bit of masturbation... because we can ??? there's better things to research... i might fancy leaving this rock for instance... then there's oh don't get me started...

will we jump from PCM and find a new method (DSD) ??

doesn't that qualifiy as everything old is new kinda thing??? not sure here but as i understand it it's just based on what sony tried to do with those mix down decks built on video transports... hey there's an idea...lol but the sony decks were pretty serious nakamichi made one that i had for a while... predictive delta modulation i think they called it i think lexicon also played with it some if memory serves...

Can anyone remember the first time they were presented with serious digital audio ?

well i never got my hands on it but saw a very early denon maschine at CES in chicago once... whoa... also sold hi-fi for years and carried the first philips cd players...

I still have fond memories of the first Fairlight 16 I ever saw and heard (it was a CMI I think) .

i know what you mean somewhere around here i have a copy of the original Synclavier brochure....
BTW sorry if my comments show inside your's afraid i'm rather inept at this ...

see ya round the playground....

Kev wrote: Vision / Video
1 bit video gives the two state situation. BLACK or WHITE
8 bit gives the 256 grey scale
but that scale falls between the same two extremes of the BLACK and the WHITE.
There is more information within the range.
... then comes the 16 bit and true colour stuff ... but still falls into the between BLACK and WHITE.
BUT cameras with CCD blocks can see into the Infra Red ... beyond the above range
???

Well...a couple things here. First, 8 bit is not entirely relegated to B&W or Grayscale. In fact, you can have 8 bit color.

Also, Infrared is not solely available to digital cameras with CCDs. In fact, any camera can see InfraRed. However, bear in mind, this is not to be thought of as "beyond the range." In fact, much like digital audio, digital video/photo is broken into 2 types of incoming signal - in audio, it's frequency and amplitude, in video/photo, it's luminence and chrominence. Infrared qualifies as a luminence (or light) value, not a color value

Kev wrote:
PCM Audio
both 16 and 24 bit have the upper limit of 0dBFS ( Full Scale )
the largest 16 bit word send you down to the 16 bit noise floor
and the 24 bit work can send you LOWER.

so part of the expanded 16 to 24 bit is to expand the area we are working with.

wait for it

Does the expanded 16 to 24 bit ALSO provide for more detail within the OLD 16 bit area.

:roll:
get my drift ?

as I said above , this analogy is not strickly correct so don't all go juming on me ... just trying to present it in a deferent sort of way.

if that does sit ok then I can present the idea that a given signal captured into a 16 bit system can be identical to the same signal captured into a 24 bit system.

Well, clearly you're playing devil's advocate here... :wink: But, of course, we know that, as more values (voltage) are able to be represented on an exponential basis in binary, then obviously the increase from 2 to the 16th versus 2 to the 24th obviously represents a broader range than merely 1.25x increase in SNR. Clearly, we are getting more values within ALL ranges (as is analogous to color/imaging/video)

J

Cucco wrote: First, 8 bit is not entirely relegated to B&W or Grayscale. In fact, you can have 8 bit color.

yep
I was just trying to keep things narrow just to illustrate a point and as I said vision is not a great way to explain audio but it can help to highlight an error that may also be in audio with a given technique if we go looking for it.
Still there really is no parallel with audio and vision in sampling and compression techniques.

Cucco wrote: Also, Infrared is not solely available to digital cameras with CCDs. In fact, any camera ... Infrared qualifies as a luminence (or light) value, not a color value

yep
I only mentioned the CCD as it was something that many could relate to here as this type of camera is common place now.

I wonder if I could get an IR image out of one of our old tube cameras

Cucco wrote: ... we know that, as more values (voltage) are able to be represented on an exponential basis in binary, then obviously the increase from 2 to the 16th versus 2 to the 24th obviously represents a broader range than merely 1.25x increase in SNR. Clearly, we are getting more values within ALL ranges (as is analogous to color/imaging/video)

and here is I think the point at which the two camps seems to get stuck

I think if you rescale the peak of the signal being sampled to the maximum value then you do have more bits/values to describe the original.
Hence the original premise of the thread.

But if you stay at -14dBFS or -20dBFS = 0dBu ... for example
then there may be little difference between the two samples.

Even when this re-scaling is brought up the Digital Boffins still continue the argument that resolution is incorrect.
It is here where I fall in a heap trying to present their argument cos I can help but fall in with Cucco's

... we are getting more values within ALL ranges ...

as I said above ... the grey scale idea was the best I could come up with. The boundaries of white and black are fixed and the 254 variations in between.
In audio one boundary of FS is fixed and the other boundary is moved
... SN increases ... and the scale is logarithmic

I haven't got a pictorial/graphical way to present this and an Excel Spreadsheet of values is too big and hard to visualise.
:roll:
so I fail at being devils advocate.

sometimes these things are not worth worrying about and I tend leave a little margin of say -6 to a max of -12dBFS on any take and even if it falls outside of that but is a RIPPER then I keep the take and move on. The performance is probably more important than anything.
Once you have chosen your gear and the sample rate and bit depth ... and the Mic and Mic-pre then all this geek stuff should be pushed aside and just record.
:roll:

not to be an ignorant asshole but which is the best 24 bit, or 48? has it been ruled out that 16 is unfit for natural selection?

do you believe that 192kHz is useless? not only that but truly decimation?

i honestly wanted to believe that analog tape and albums and what not display beyond 20-20khz. but he does make a point that if it goes to cd whats the point.

where can you even find a 48/192?

mmm
... is someone stirring here ?

liquidstudios wrote: do you believe that 192kHz is useless? not only that but truly decimation?

not useless
one simple useful fact for the high sample rate is

for a given function that may take say, 10 samples
at 192 khz it takes 1/4 of the time to get that job done than at 48khz
so latency is shorter

that could be a good thing for mix engines and live work

... but he does make a point that if it goes to cd whats the point.

because the calculation should be at better acuracy than the result
... calculators go to many decimal places internaly and yet you may choose to display only 2 decimal places

most stuff is being distributed at low MP3 qualities
so
we should record at lower quality

I'm happy with 44.1 and 24 for the time being

where can you even find a 48/192?

:roll:
as a mix engine .. or as a recorded file ?
there is a few mix engines and I've hear of some prototype stuff but not for retail
and I don't think there is an AES or EBU standard yet
but then I'm not that interested and may have been asleep

liquidstudios wrote: not to be an ignorant asshole but which is the best 24 bit, or 48? has it been ruled out that 16 is unfit for natural selection?

do you believe that 192kHz is useless? not only that but truly decimation?

i honestly wanted to believe that analog tape and albums and what not display beyond 20-20khz. but he does make a point that if it goes to cd whats the point.

where can you even find a 48/192?

Nope, not coming across as an ignorant a-hole, just an inquiring mind.

48 bit is superior to 24 bit. However, there are no converters or engines capable of recoring at 48 bit currently. (Even those DAWs which claim to operate at 64 bit don't record anything at 64 bit, the converters simply don't put that many bits out.)

48 bit is usually reserved for processing on digital devices. For example, Dan Weiss's gear operates at 48 bit fixed point as does some Z-sys and Waves stuff. Then, these devices have to dither back down to 24 bit. The good news about this is that much of the advantage of the 48 bit processing is retained when dithering back down to 24 bit. (The advantages being a cleaner treatment of the processed audio due to the fact that any artifacts are well outside the audible range amplitude-wise).

Personally, I believe that 192 IS a benefit despite what Dan Lavry states. As Kev mentions, there is less overall latency, but also less in-band distortion.

As far as analog tapes/albums going past 20kHz - it's entirely possible, but the limitations are that the hardware (analog) capable of doing this is quite expensive. Normal Sony or Pioneer tape decks or JVC turntables just aint gonna do it.

J.

we should record at lower quality

I'm happy with 44.1 and 24 for the time being

That's high quality!

Once you have chosen your gear and the sample rate and bit depth ... and the Mic and Mic-pre then all this geek stuff should be pushed aside and just record.

Yes sir!

also - higher sample rates allow an alias filter much farther above the frequencies of interest, thus skewing them less. Then, when you are processing, you are processing a better representation of the original. Eventually the rate has to come down, but higher sample rates allow you to forgo that damage until you are done doing your creative damage!

Having said that, do I think it really matters? no! It still sounds killer, so long as the music sounds killer. Im with Kev, 24/44.1 is adequate. I have recorded at 192khz and the only real tangible difference i noticed - The file is four times bigger! Now, if you do a lot of processing, the auditory diff can become noticeable - or not so much, if you are judicious, careful.
Conclusion; unless you are investing 150 g's on one album and expect platinum sales, who the f**k cares :roll: With some ingenuity, you should be able to achieve killer results at 44.1, 24bit.

I'd like to think I take a backwards approach to recording as most do. To me, it seems that the mentality amongst many is:

"Well, it's just going to be heard on an Eye-pod anyway, who cares if I use the highest quality?"

Whereas, my mentality is:

"Some dude out there is going to be listening to this on their Wilson Watt/Puppys with an AudioNote amplifier and Theta Digital transport. I want this stuff to please him. Let the Eye-pod users hear what they can."

However, all that being said, I still agree that, for most projects 44.1 is where it's at. The technology keeps getting better and better and so does the sound. For SERIOUS classical literature or stuff that I'm sure will wind up on SACD or DVD-A, I go with higher SRs, but usually - 44.1 does it for me. My hard drive and my CPU thank me for that.

j

CrackBuddha wrote: also - higher sample rates allow an alias filter much farther above the frequencies of interest, thus skewing them less.
... Eventually the rate has to come down, but higher sample rates allow you to forgo that damage until you are done doing your creative damage!

yep
and part of me feels that there might be something still worth persuing here even though the Filters and converters and clocks are better these days.
I think it was the Fairlight that had the analog filters set lower than some of the competitors ... such that the specs showed a frequency resp that was only around 18.5k to 19k instead of 20k at the 44.1 sample rate
Apogee may also have done this in one of there units.
many believed that unit , at the time , to be one of the best. There could have been many reasons.
BUT
in the early days there were issues with filters and the wobble and bump you get at the point of interest.

The freq wobble also brings the phase wobble and the bump when not amplitude aligned properly can bring a clip right at the highest frequency for the converter.

anyway

perhaps one day I'll re-visit 88.2 and 24 ... or even 176.4 (if I get an interface that can).
I still feel that it could be easier and more gentle on the results to go from 88.2 to 44.1 for CD delivery.

I don't really have any evidence to back that up.
just a gut feeling

Drives are bigger and I have enough DSP grunt to do the same productions as I did in the mid 90's
so I have no excuse and should really give it all another try.

and so it would be nice to have the option to use a more gentle nyquist filter when recording at the higher rates
I don't think we need a sine wave at 44k more than we need one at say 40k or 38k. It is well above what we accept at the 44.1k sample rate and a little higher than some of the best analog tape decks which were well gone by 30k

The manufacturers don't give us this option and I can't be bothered modifying my interfaces.

:roll:
... it might be enough to just put a filter on the front of the existing gear
:shock:
perhaps I should market a product to the Golden Ears and charge Sqwilians of Dollars for it and make a fortune
8)

hi guys

http://www.wescottdesign.com/articles/Sampling/sampling.html

i ran across this over at PSW and found it an interesting read thought perhaps ya might like it as well... the math is minimal and while it's open to all sample systems and not just audio (in fact he kinda stays away from us...) his ideas on sampling frq seem to have an impact for the Lavrey/no 192Khz camp...anyway best and enjoy...

nice paper that centres on Nyquist and Filters
and band limiting to within specific bandwidth

the thread here was centred on amplitude and bit depth
and whether more bits gave more accuracy or just more dynamic range with less noise

at a first level less noise does mean more accuracy
but is Bit Depth and Nyquist and Filters all equaly important or does one have more effect on quality than the others ?

I've been following these discussions on quantizing on the various recording forums for a few years now, and one thing that strikes me immediately is just how far audio experts are behind the curve in this field. It also strikes me odd that the people who are insulted by this, simply genuinely do not even realize this because they are detached from the one academic field which actually studies this as a specialty.

The field of quantizing is almost the exclusive domain of the electrical engineers who study information theory, and I've noticed that the applied knowledge in this field is decades behind the actual theoretical research. Digital communicatuion engineers are about 20 years behind the theorists, and audio experts are even farther behind... I'd say about 10 years behind the digital communications experts... meaning that audio experts are basically 30 years out of date with what the theorists knew 30 years ago.

I don't know how many here are electrical engineers who specialize in information theory, but if you are, then you would almost have to be astounded at how superficial, or at least outdated, is the level of knowledge of audio experts who specialize in the design of sampling products.

If you want to take a gander at what real experts know, just read any of the thousands of articles in the IEEE Transactions on Information Theory which deal with that topc. The level of maths involved is so far beyond the audio field... that... its really kind of astounding.

I mean, I cant even believe we're still using scalar uniform quantization for gods sake. Non uniform quantization would be a fairly radical improvement for something which is inherently so simple and requires very little adaptation from our current technology. From there though, we could make the jump from scalar to to even more sophisticated vector quantization which would almost be a quantum leap, not to overstate the case.

Unfortunately, we have spent decades now, retarding our progress by relying on traditional PCM... no wonder we're not making much progress. Had audio engineers not been 30 years behind the times, and 10 years behind the digital communications field... we could be roughly achieving the same level of perceived clarity while utilizing a significantly lower bit depth (or "rate", as its called in information theory).

A perfect example of how audio engineers are so far out of it, is that virtually none of their literature ever deals with the stochastic processes. They spend all their time rehashing the same stuff over and over again... talking about noise floors, etc... but unlike real experts in quantization theory, they never dedicate anytime studying or researching the statistical distribution of the source and then matching that source to the quantizer, and changing the distribution of the bins to match the source so that the mean values between the source and the quantizers sweet spot are aligned, thus optimizing the rate-distortion and its converse distortion-rates characteristics.

The only real area where we get into this discussion of stiochastic process is when we start discussion noise shaping, but other than that, we have locked ourselves into the rediculous paradigm of traditional PCM which was partially abandoned in many other fields 30 - 40 years ago.

IT LIVES !!!!!

wow... resurected an oldy.... i'm guessing that lagging behind is'nt peculiar to our area of focus... in fact i suspect that we're still ahead of a lot of people.... have to admit some of what you're saying goes past me.... but IIRC some of what you're espousing is taken into consideration in sacd format for instance.... but as i understand it breaks down from a practical standpoint when you want to apply DSP/mixing etc...

yes resurected

I read this when it was posted and didn't want to start all over again
... information theory ... yeah and we could use mathematics too
Non uniform quantization ... not sure that's going to bring an instant improvement for analog audio capture

as with all capture .. there is a consideration for what is to be captured and how
also the method by which it will be presented later

capturing the data of an earth quake might be a continuous stream but it might also be a capture of just events above a given threshold

then the presentation might be in the form of a histogram or a chart of data
at no stage is the earth quake to be ... re-produced in the original form

yes, it is a good discussion point and I do see where Mises is coming from

however
we do have PCM recording and it has been around since the 70s and the AudioCD has been with us from the early 80s
yes it is old
yes there is wide knowledge of it out there
yes people can reproduce the signals in a form near enough to the originals
yes it can be use to mix multiples that were not captured at the same time or during the same event

one day we may leave PCM behind ... DSD perhaps
but not today

sorry one more thing
stochastic processes
good for the repeatable events, in the same way that an FFT can tell us more than a simple sound pressure meter
but again
not sure that it provides for an instant improvement for analog audio capture

so far I have never seen any statictical representation of the sound of a Mic or a Speaker
( I deliberately chose electro mechanical devices )
that is as acurate as an Ear and a Brain.

These are great analytical tools ... after the event.
but we want to capture an event and then manipulate
... then present that event ... or a totally new event.
The time domain is probably the most important consideration in reproduction of music and real event video