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Offline szumsteg

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Analog to 24bit question
« on: December 07, 2007, 07:04:08 PM »
How much am I going to gain taking old analog tapes, master caseettes or 1st gens whatnot, and putting them to 24bit vs 16bit. Is an analog tape going to suddenly get better at 24bits, or is it just the fact its a better replication of the analog tape. I don't notice that when I convert an analog tape to 16 bits it sounds worse than if I just play it through the tape deck into my amplifier than the CD I make. It seems to sound exactly the same. It would just seem to me that the same would hold if I took the master tape and played it through my amp direct vs DVD audio, it cant get better, all it can do in the end if all goes well is sound exaclty the same,correct?

I totally understand that its probably better to have more 0's and 1's digitally to replicate my analog master, but what does that mean to the ears. Or is the classic example of it feels better to have it in the largest size file possible...Is everyone going to do the same thing all over again when we get to the next level of bit recording.

Granted for future audio recordings, 24bit is the way to go, because the machine is capturing more to start with, but old recordings on analog tape, I guess the real question is what bit rate were they being captured at...

Offline JasonSobel

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Re: Analog to 24bit question
« Reply #1 on: December 07, 2007, 07:31:31 PM »
it's not going to sound better than the original tape.
24 bits is good because, as you say, "its a better replication of the analog tape."
However, I think the real benefit for 24 bit is if you do any processing on your computer after you've transferred.  maybe some slight hiss reduction, or something along those lines.  I think you'll find that you have much better results starting with a 24 bit capture rather than a 16 bit capture.

Offline boojum

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Re: Analog to 24bit question
« Reply #2 on: December 08, 2007, 09:48:20 PM »
16 bit gives you about a 100 dB range; 24 gives you about 145 - 150 dB range.  24 bit allows more manipulation with less loss for mastering.  Better sounding?  Is Ford better than Chevy?  8)
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Offline Petrus

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Re: Analog to 24bit question
« Reply #3 on: December 09, 2007, 09:20:51 AM »
Old analog tapes propably have the noise floor at about -55 to -60 dB levels below peaks, even with 16 bits the lowest 30 dB at least will be hiss only. Using 24 bits will not give you any benefits, only almost 80 dB of hiss...

Offline boojum

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Re: Analog to 24bit question
« Reply #4 on: December 10, 2007, 12:30:16 AM »
Old analog tapes propably have the noise floor at about -55 to -60 dB levels below peaks, even with 16 bits the lowest 30 dB at least will be hiss only. Using 24 bits will not give you any benefits, only almost 80 dB of hiss...

Are you saying that by allowing a greater dynamic range the hiss level will be higher?  How does that happen?
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Offline Petrus

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Re: Analog to 24bit question
« Reply #5 on: December 10, 2007, 12:52:33 AM »
No. Just that with 24 and 16 bit systems you will have exactly the same usable signal (top part of the dynamic range) and the rest is just noise.

Analog tape has something like 50 to 70 dB of DR (depending on Dolby sytems et.)
16 bit digital has over 90 dB (96 dB in theory)
24 bit digital has about 115 db at best (144 dB in theory)

Using 24 bits does not help, does not hurt.

Offline F.O.Bean

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Re: Analog to 24bit question
« Reply #6 on: December 10, 2007, 01:04:41 AM »
if you do record in 16-bits, make sure your DAW's temp file is done at 24-bit processing tho.
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Offline willndmb

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Re: Analog to 24bit question
« Reply #7 on: December 10, 2007, 10:02:28 AM »
Is Ford better than Chevy?  8)
my grandfather says yes  ;D
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Offline Tim

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Re: Analog to 24bit question
« Reply #8 on: December 10, 2007, 10:07:10 AM »
why wouldn't you want the best possible representation of the analog tape?
I’ve had a few weird experiences and a few close brushes with total weirdness of one sort or another, but nothing that’s really freaked me out or made me feel too awful about it. - Jerry Garcia

Offline Petrus

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Re: Analog to 24bit question
« Reply #9 on: December 10, 2007, 12:17:39 PM »
Because in this case 24 bits is not any better than 16 bits. 24 bits adds nothing to the quality, bacause there is nothing on the analog tape that 16 bit system can not catch, hold and reproduce. Like driving a car alone and always pulling a trailer just because car (16 bits) and trailer (8 bits more) is "better" than having just a car... Or if your analog tape was a pint of milk, a gallon jar (16 bits) is plenty big enough for it, there is no point in getting a two gallon (24/96) or a four gallon (24/192) bottle for it.

But like I said, 24 bits only makes files larger, no hurt there with modern systems.
« Last Edit: December 11, 2007, 08:16:24 AM by Petrus »

Offline dmccabe

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Re: Analog to 24bit question
« Reply #10 on: December 10, 2007, 12:35:33 PM »
Because in this case 24 bits is not any better than 16 bits. 24 bits adds nothing to the quality, because there is nothing on the analog tape that 16 bit system can not catch, hold and reproduce.

We run a Dead project called MOTB. We transfer older analog recordings. We release every show in 2 versions (from the same analog transfer), one in 24-bit and one in 16-bit to burn to Cds. You can HEAR the difference. As mentioned, any post-mastering is one reason for 24-bit... but even WITHOUT any post transfer changes, you can still hear the difference between a 24-bit transfer and a 16-bit transfer -- using the same master recording and the same A/D but at different sample rates. Why? Frequency response detail. Well, some people can't hear it -- but that is another discussion.

I can write pages here on many topics that effect the quality of sound -- especially when taking an old master cassette to digital.

But let's look at how major studios release CDs. How many older analog commercial albums have been remastered to CD in the last 10 years. Did they go to 24-bit from the master analog... or did they just go directly to 16-bit? duh! If there is no audible difference, why would they ALL go to 24-bit...

Petrus' statement is related to dynamic range, not fidelity... by trying to associate dynamic range to the "quality" of recording (frequency response) is misunderstanding what human ear's hear. You can't change the dynamic range of an original analog cassette, but you can change the amount of frequency response. The higher, the more natural sounding.

The frequency response of audio CD is sufficiently wide to cover the entire audible range, which roughly extends from 20 Hz to 20 kHz. Analog audio is unrestricted in its possible frequency response, but the limitations of the particular analog format will provide a cap. High-quality metal-particle cassettes may have a response extending up to 14 kHz at full (0 dB) recording level.

Why do early 16-bit digital converters sound different than today's digital converters at 16-bit? So all 16-bit digital is not the same? If they are the same bit rate, how can they sound different? How about 24-bit?

When I record to 1-bit DSD... is that lower quality or higher "quality" than 16-bit PCM? Does the same analog transfer "sound" different going to 1-bit DSD than 24/96 PCM?

Don't confuse bits with sound quality... to help determine what bit rate to use when converting an older analog cassette is to check the frequency response of the microphones used in the recording. But on that same note, frequency response does not guarantee a specific fidelity either.
« Last Edit: December 10, 2007, 01:19:42 PM by dmccabe »

Offline boojum

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Re: Analog to 24bit question
« Reply #11 on: December 10, 2007, 04:50:35 PM »
How a copy can sound better than the original eludes me.  How can that be?
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Offline Gutbucket

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Re: Analog to 24bit question
« Reply #12 on: December 10, 2007, 04:58:36 PM »
'Sound better' could mean EQ, noise reduction, dynamics or other 'mastering' work, or something more subtle - maybe sweetened through a nice sounding preamp or other playback equipment during the transfer.  That doesn't make it more 'accurate' to the original tape though.
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Offline F.O.Bean

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Re: Analog to 24bit question
« Reply #13 on: December 10, 2007, 07:46:47 PM »
I would at least record in 24/44 for any processing that may need done, but thats just me and the purist in me :)
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Offline boojum

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Re: Analog to 24bit question
« Reply #14 on: December 10, 2007, 09:19:58 PM »
'Sound better' could mean EQ, noise reduction, dynamics or other 'mastering' work, or something more subtle - maybe sweetened through a nice sounding preamp or other playback equipment during the transfer.  That doesn't make it more 'accurate' to the original tape though.

This is true.  But the part that got my interest is this: " . . . but even WITHOUT any post transfer changes, you can still hear the difference between a 24-bit transfer and a 16-bit transfer --"
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Offline Gutbucket

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Re: Analog to 24bit question
« Reply #15 on: December 10, 2007, 09:40:56 PM »
This is true.  But the part that got my interest is this: " . . . but even WITHOUT any post transfer changes, you can still hear the difference between a 24-bit transfer and a 16-bit transfer --"

Yeah, that caught my eye too, and if so then that's the bottom line.

I would at least record in 24/44 for any processing that may need done, but thats just me and the purist in me :)

I don't get this argument, Bean. You can process the file in 24bit, 32bit, 64bit, or whatever your software can handle regardless of the bit depth of the recording.  In other words, the processing need not be tied to the resolution of the file you're working on.  I think the argument is capturing something you may miss with a 16 bit transfer, even if that isn't an extended dynamic range.
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Offline Petrus

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Re: Analog to 24bit question
« Reply #16 on: December 11, 2007, 02:21:44 AM »
Don't confuse bits with sound quality... to help determine what bit rate to use when converting an older analog cassette is to check the frequency response of the microphones used in the recording. But on that same note, frequency response does not guarantee a specific fidelity either.

Bit depth has nothing to do with frequency range. Bit depth governs the (theoretical) maximum dynamic range, simplified it is 6 dB for each bit. Sample frequency determines the maximum audio frequency, max frequency is sample rate/2 (Nyquist theorem). This is so basic that I find it funny to write about it here.

Not all converters are created equal, some are more accurate and have better analog stages. Also in the beginning times of digital audio the art of dithering was not as highly developed as it is now. 16 bit systems are just about perfect now.

We should also remember that even the best 24 bit systems are seldom if ever better than about 19bits in real dynamic range (115 db or so) thanks to less than perfect analog stages and power supplies.

Offline rich

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Re: Analog to 24bit question
« Reply #17 on: December 11, 2007, 05:40:06 AM »
Cool...I have been taping for years, some good tapes some not so good tapes.  Either way, many thanks for the info Petrus.  I would be interested to hear your thoughts on DSD and 1 bit converters.  I used to run the Panasonic SV - 255 which I thought sounded amazing (1 bit PCM).  Thanks.

Rich
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Offline Petrus

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Re: Analog to 24bit question
« Reply #18 on: December 11, 2007, 06:08:53 AM »
It is not possible to compare one bit converters and "normal" PCM converters directly, the working principals are so different.

With PCM converters the signal is basically sent thru a gate, which flips when voltage is half of maximum. If it is, the most significant bit is set to 1, next to 1/4 gate, then 1/8 gate and so one. This happens once for each sample, typically 44100 times a second. We have relativelly few samples (but enough to describe the signal), but good measurement of the amplitude.

With one bit systems we have extremely high sample rates, but the lone sample just records the change, is the next sample bigger (higher voltage) than the previous. If yes we get 1, if smaller, 0. If we have enough samples this system also can record the waveform well enough. To describe a signal with accuracy of 16 bit PCM we need 2^16 as many samples per second or 44100*65536=2,889,037,600, almost 3 billion samples per sec. To compete with 24, even more. This gets unpractical and normal PCM is now prefererd. The high sample rate of one bit systems is nothing to write home about, it is not any advantage, just the way that system operates.

Offline Gutbucket

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Re: Analog to 24bit question
« Reply #19 on: December 11, 2007, 09:47:40 AM »
... Good explanation by Nika Aldrich (who has a good book on digital audio) here:

http://www.users.qwest.net/%7Evolt42/cadenzarecording/DitherExplained.pdf
...

Excellent explanation.  One interesting thing he mentions that I was not aware of is that 'noise shaped' types of dither (which he refers to as 'colored' dither such a UV22, etc), should only be used at the very last wordlength reduction or could cause noise artifacts if dithered again.  Triangle Probability dither should instead be used if any additional processing might follow.
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Offline dmccabe

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Re: Analog to 24bit question
« Reply #20 on: December 11, 2007, 10:14:00 AM »
Bit depth has nothing to do with frequency range. Bit depth governs the (theoretical) maximum dynamic range, simplified it is 6 dB for each bit. Sample frequency determines the maximum audio frequency, max frequency is sample rate/2 (Nyquist theorem). This is so basic that I find it funny to write about it here.

You can't have sound without bit rate (dynamic range) AND frequency response. This is SO BASIC, I am still amazing you are continuing to post like you actually have experience in transferring analog cassettes to digital. Dynamic range IS DIFFERENT than frequency response, but they are directly related to each other. The fidelity of frequency response is also directly related to the quality of the bit converter. A sine curve at 24/96 has more detail than a sine curve at 16/44.1 -- even when the dynamic range is the same!

The human ear can't just listen to the dynamic range of sound... there needs to be frequency response at the same time.
Without going into great detail, the process in a pro A/D converter is not the same at different bit rates. At 16-bit, the A/D processes conversion DIFFERENTLY than at 24-bit... and you can hear that difference, even with the same source that has a limited dynamic range.

There are also playback D/A considerations that affect your perception of that sound. Bottom line, one should ALWAYS transfer in 24-bit to achieve the best fidelity when transferring analog recordings. To make a statement that is doesn't matter is misinformed. I can show you to many examples where an analog source was transferred to 24-bit/96 and 16/44.1... and the raw unaltered transfer will always sound better on the 24-bit version. I have never... ever seen an audio engineer recommend that an analog transfer be done in 16-bit... because there is "no benefit" to go up to 24-bit.

When you bring up the Nyquist theorem, it shows that you are trying to make your argument based on the mathematical limitations of dynamic range. Are you also saying that all A/D's running at the same bit rate and frequency sound the same?
Are you saying that any A/D sounds the same at 16-bit and 24-bit?

We use Mytek, Benchmark, Grace Design and Apogee A/D/A converters... they all sound different. Do you need to listen to some audio samples? Do you have the correct hardware to listen to a 24-bit/16-bit comparison? What is the monitoring path?

Old analog tapes propably have the noise floor at about -55 to -60 dB levels below peaks, even with 16 bits the lowest 30 dB at least will be hiss only. Using 24 bits will not give you any benefits, only almost 80 dB of hiss...

No benefits? To be blunt... are you sure you have any idea what you are talking about?


Offline Gutbucket

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Re: Analog to 24bit question
« Reply #21 on: December 11, 2007, 10:59:19 AM »
...A sine curve at 24/96 has more detail than a sine curve at 16/44.1 -- even when the dynamic range is the same!

Isn't a sine curve a sine curve?  Doesn't the 16/44.1 version contain all the information to reconstruct the sine perfectly as long as it is within the frequency and dynamic limits of 16/44.1? Isn't the additional time domain / frequency data redundant in this simplified case? Of course this is closer to the mathematical theory side than a complex musical signal through real world converters.

Quote

...Without going into great detail, the process in a pro A/D converter is not the same at different bit rates. At 16-bit, the A/D processes conversion DIFFERENTLY than at 24-bit... and you can hear that difference, even with the same source that has a limited dynamic range.

There are also playback D/A considerations that affect your perception of that sound. Bottom line, one should ALWAYS transfer in 24-bit to achieve the best fidelity when transferring analog recordings. To make a statement that is doesn't matter is misinformed. I can show you to many examples where an analog source was transferred to 24-bit/96 and 16/44.1... and the raw unaltered transfer will always sound better on the 24-bit version...


^^^
I think this is the heart of the matter.   A 24bit tape transfer can sound better than a 16bit one.  At least in part because different circuits just sound different, but maybe for additional theoretical reasons too? 

Now here's the next question: After the initial Tape > A/D transfer at 24bits, assuming the musical portion of the signal is optimized in that available bit depth with plenty of range to spare, would a later wordlength reduction in the digital ream to 16 bits sound inferior?  Is it the A/D transfer stage that is the critical part here? or the depth of the eventual storage format?

Thanks to those of you with a deeper understanding and hands on-experience with this.
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Re: Analog to 24bit question
« Reply #22 on: December 11, 2007, 12:49:16 PM »
Sorry to interrupt - just curious here....

Which would you prefer for 16 bit listening?

A) 24 bit master > dithered to 16 bits

B) Native 16 bit samples

I would tend to favor "B" - theoretically...

Offline boojum

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Re: Analog to 24bit question
« Reply #23 on: December 11, 2007, 01:07:29 PM »
With the little knowledge that I have I would prefer 24 > 16 with dither.  I say this because the capture and any post processing would be done in 24 bit where there is less loss from this work, I believe.  I always record 24/48 and do post in 24/48 for this reason.  When I am done I mix down and dither to 16/44.1.

If it is just captured at 24 and then dithered down to 16 I do not see what benefit there can be.  It can never sound better than the original.  As has been posted earlier, it can sound different, but only with tweaking.  If I am wrong someone please tell me how the copy can sound better than the original.  Can a 24 bit copy sound better than a 16 bit copy?  I do not know.  Some think so.  But I would not put a lot of credence in that myself until it had been tested many times in double-blind tests.  As usual, YMMV.   8)

Happy Trails.
« Last Edit: December 11, 2007, 05:24:34 PM by boojum »
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Offline Tim

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Re: Analog to 24bit question
« Reply #24 on: December 11, 2007, 01:28:05 PM »
Sorry to interrupt - just curious here....

Which would you prefer for 16 bit listening?

A) 24 bit master > dithered to 16 bits

B) Native 16 bit samples

I would tend to favor "B" - theoretically...

depends on the dither but most of the time I would prefer option "A"

why did we all love those ad500 and ad1000's? It was their onboard dithering that gave them that famous Apogee sound
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Offline Brian Skalinder

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Re: Analog to 24bit question
« Reply #25 on: December 11, 2007, 01:29:36 PM »
Old analog tapes propably have the noise floor at about -55 to -60 dB levels below peaks, even with 16 bits the lowest 30 dB at least will be hiss only. Using 24 bits will not give you any benefits, only almost 80 dB of hiss...
Even with tape's limited SNR spec, it's still possible to hear signal below the noise, so it probably is still helpful to run reasonably close to peak on transfer to 16 bit.

These two statements get to the crux of the issue for me.  This topic perpetually gives me brain cramps, but I'll dive in anyway, if only with a couple questions.  Both questions assume the same sample rate, for simplicity's sake:

If an analog cassette provides 60 dB of dynamic range, and below -60 dB is only noise...

  • am I correct that we should achieve the same results recording at 10-bit, much less at 16- or 24-bit?  (forgetting for the moment that none of us likely have a 10-bit ADC)
  • does a 24-bit transfer capture lower level detail about the noise than 16-bit (i.e. noise below -96 dB) , and therefore provide greater accuracy in performing noise reduction (if one planned on doing so now, or in the future)?  (I suppose an implied question here:  if the noise on an analog cassette starts at -60 dB...how low does it go?  Below -96 dB?  If so, how far?)

If an analog cassette provides 60 dB of dynamic range, and below -60 dB is shared by both noise and signal...

  • aren't there benefits to recording at 16- or 24-bit to capture the signal sharing those least significant bits with the noise?

Just found this in the rec.audio.pro FAQ suggesting that we hear signal below the noise floor:

Quote from: http://stason.org/TULARC/entertainment/audio/pro/5-14-How-can-a-16-bit-word-length-be-enough-to-record-all.html
5.14 - How can a 16-bit word length be enough to record all the detailin music? Doesn't that mean that the sound below -96 dB gets lost in thenoise? Since it is commonly understood that humans can perceive audiothat IS below the noise floor, aren't we losing something in digitalthat we don't lose in analog?

You're correct in saying that human hearing is capable of perceiving
audio that is well below the noise floor (we won't say what kind of
noise floor just yet). The reason it can do this is through a process
the ear and brain employ called averaging.

If we look at a single sample in a digital system or an instantaneous
shapshot in an analog system, the resulting value that we measure will
consist of some part signal and some part ambiguity. Regardless of the
real value of the signal, the presence of noise in the analog system
or quantization in the digital system sets a limit on the accuracy to
which we can unambiguously know what the original signal value was. So
on an individual sample or instantaneous snapshot, there is no way
that either ear or measurement instrument can detect signals that are
buried below either the noise or the quantization level (when properly
dithered).

However, if we look at (or listen to) much more than a single sample,
through the process of averaging, both instruments and the ear are
capable of detecting real signals below the noise floor.
Let's look at
the simple case of a constant voltage that is 1/10th the value of the
noise floor. At the instantaneous or sample point, the noise value
overwhelms the signal completely. But, as we collect more consecutive
snapshots or samples, an interesting thing begins to happen. The noise
(or dither) is random and its long term average is, in fact, 0. But the
signal has a definite value, 1/10. Average the signal long enough, and the
average value due to the noise approaches 0, but the average value of
the signal remains constant at 1/10.

A somewhat analogous process happens with high frequency tones. In
this case the averaging effect is that of a narrow-band filter. The
spectrum of the noise (or simple dither) is broadband, but the
spectrum of the tone is very narrow band. Place a filter centered on
the tone and while we make the filter narrower and narrower, the
contribution of the noise gets less and less, but the contribution of
the signal remains the same.

Both the ear and measurement instruments are capable of averaging
and filtering, and together are capable of pulling real signals from
deep down within the noise, as long as the signals have one of two
properties: either a period that is long compared to the inherent
sampling period of the signal in a digital system or long compared to
the reciprocal of the bandwidth in an analog system, or a periodic
signal that remains periodic for a comparably long time.

Special measurement instrument were developed decades ago that were
capable of easily detecting real signals that were 60 dB below the
broadband noise floor. And these devices are equally capable of
detecting signals under similar conditions in properly dithered
digital systems as well.

How much the ear is capable of detecting is dependent upon many
conditions, such as the frequency and relative strength of the tone,
as well as individual factors such as aging, hearing damage and the
like.

But the same rules apply to both analog systems with noise and digital
systems with decorrelated quantization noise.
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Re: Analog to 24bit question
« Reply #26 on: December 11, 2007, 01:52:13 PM »
Another interjection:

If I follow the dmccabe logic - shouldnt we all be remastering our old 16 bit DATs at 24 bits - just like the cassette process? If 24 bits makes cassettes sound better, why not 16 bit DATs???

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Re: Analog to 24bit question
« Reply #27 on: December 11, 2007, 01:56:12 PM »
Another interjection:

If I follow the dmccabe logic - shouldnt we all be remastering our old 16 bit DATs at 24 bits - just like the cassette process? If 24 bits makes cassettes sound better, why not 16 bit DATs???

I think many people are indeed doing just this

I know that's what I plan on doing

edit: Maybe I am just confused :P
« Last Edit: December 11, 2007, 02:06:58 PM by Tim »
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Re: Analog to 24bit question
« Reply #28 on: December 11, 2007, 02:05:32 PM »
If I follow the dmccabe logic - shouldnt we all be remastering our old 16 bit DATs at 24 bits - just like the cassette process? If 24 bits makes cassettes sound better, why not 16 bit DATs???

If making edits to 16-bit masters after transferring to computer, then yes - you should do so with software that uses internal precision of 24-bit, or better yet 32bfp.  In this case, we're talking about the precision used in manipulation of the digital signal within the digital realm, not the transfer of an analog signal into the digital realm.  Different beast entirely.

There's no point in transferring a 16-bit DAT to PC at 24-bit, since it's all within the digital realm (DAT > bit-transparent S/PDIF > PC).  The least significant bits would simply be padded with zeroes.  I suppose one could transfer into the analog realm and then back (DAT > DAC > ADC > PC), but...not sure what the point would be, unless one wanted to take advantage of the particular sonic signature of the intermediate analog gear.
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Re: Analog to 24bit question
« Reply #29 on: December 11, 2007, 02:06:01 PM »
Another interjection:

If I follow the dmccabe logic - shouldnt we all be remastering our old 16 bit DATs at 24 bits - just like the cassette process? If 24 bits makes cassettes sound better, why not 16 bit DATs???

no, because the DAT 16-bit was already dithered... and adding another dither would be a destructive edit and hurt the overall quality. But we have found that if you run a DAT out through a high-quality DAC back to "the best analog you can get out of the DAT"... then go back in 24/96 to do any mastering... that sounds way better than upsampling a 16/44.1 to 24/96 for editing or remastering.

As noted in previous post... "noise"... especially analog tape noise is an integral part of the overall fidelity. Dismissing the noise as just wasted bits is not correct... it affects the overall sound. Forgetting about broadband noise reduction in 24-bit as a main beneift... as I stated: We have done many tests converting high quality analog reels through the same A/D in 16/44.1 - 24/44.1 and 24/96. Listening to those raw transfers, you can hear the difference between all three. Maybe alot of that has to do with playing the 24-bit back through a 24-bit DAC... but there you go... you can still HEAR the difference.

As for going to 16-bit from a 24-bit source...

Most top mastering engineers (I am not going to drop a name here, but think the entire Hendrix remastered catalog) take their 24/192 digital masters and send them back out to analog... then back into 16-bit/44.1 A/D of their choice for Cd master... rather than dither them in software. Anyone who states that once it is "digital it should stay digital"... is not doing what the top mastering engineers are doing in LA today.
« Last Edit: December 11, 2007, 02:12:23 PM by dmccabe »

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Re: Analog to 24bit question
« Reply #30 on: December 11, 2007, 02:06:16 PM »
Another interjection:

If I follow the dmccabe logic - shouldnt we all be remastering our old 16 bit DATs at 24 bits - just like the cassette process? If 24 bits makes cassettes sound better, why not 16 bit DATs???

I think many people are indeed doing just this

I know that's what I plan on doing

When it comes to preservation and archiving - shouldn't you be trying to keep the exact digital info as it is on the DAT?

Once that is done - everything else (including any sound enhancing tricks)- is infinitely repeatable...

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Re: Analog to 24bit question
« Reply #31 on: December 11, 2007, 02:10:23 PM »
no, because the DAT 16-bit was already dithered...

Doesn't this assume all 16-bit ADCs (whether internal to the DAT recorder, or external) employ on-board dither?  Maybe they do, I dunno.  If not, then seems the answer should really be "it depends on whether the 16-bit ADC employed on-board dither".
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Re: Analog to 24bit question
« Reply #32 on: December 11, 2007, 02:14:24 PM »
If I follow the dmccabe logic - shouldnt we all be remastering our old 16 bit DATs at 24 bits - just like the cassette process? If 24 bits makes cassettes sound better, why not 16 bit DATs???

If making edits to 16-bit masters after transferring to computer, then yes - you should do so with software that uses internal precision of 24-bit, or better yet 32bfp.  In this case, we're talking about the precision used in manipulation of the digital signal within the digital realm, not the transfer of an analog signal into the digital realm.  Different beast entirely.

There's no point in transferring a 16-bit DAT to PC at 24-bit, since it's all within the digital realm (DAT > bit-transparent S/PDIF > PC).  The least significant bits would simply be padded with zeroes. 

Understood.

I suppose one could transfer into the analog realm and then back (DAT > DAC > ADC > PC), but...not sure what the point would be,

Thats more what I am talking about - the fact the analog signal is from a 16 bit DAT seems irrelevant following the mccabe logic...be it from a cassette or a DAT, its still an analog signal...

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Re: Analog to 24bit question
« Reply #33 on: December 11, 2007, 02:23:16 PM »


When it comes to preservation and archiving - shouldn't you be trying to keep the exact digital info as it is on the DAT?

Once that is done - everything else (including any sound enhancing tricks)- is infinitely repeatable...

It all depends, there is no definite rule for every recording... is it a DAT master... or an analog recording transferred to DAT - big difference. looking back now, DAT has turned out to be a shitty format. It compressed all the dynamic range of the mic info down to 16-bits, and you can't change that. I'll take an analog master over a DAT master any day. The only benefit we all got from DAT was less tape hiss and the ability to make a digital copy.

BTW, Micheal Grace is a contributing member of MOTB and has donated quite a few pieces of hardware to our project. We also have custom modded V3's done by Mike for using the analog inputs for line in rather than mic in (impedance matching mods). We use V2s and V3s as our preamp front end out of our Technics reels.

If you have a DAT master that is a good show you want to edit... you want to do all those edits in 24-bit. So the best way to get it there is out through a good DAC, like the Benchmark DAC-1. If you use a shitty DAC, then you might as well upsample the 16/48 DAT for a better result. You need a good DAC to get it back to analog.

If anyone needs some audio files transferred for a listening test... we have lots of hardware and can post the samples.

Personally I really love the Korg DSD stuff more than the higher end A/Ds...

Also I have quite a few live shows where I used KM184 > V3 > digital out to 24/96 Marantz 671 and at the same time out of the V3 analog out > Korg MR-1000 DSD. Then you can compare the Audiogate 24/96 to the V3 24/96. I like the Korg much better.

« Last Edit: December 11, 2007, 02:27:55 PM by dmccabe »

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Re: Analog to 24bit question
« Reply #34 on: December 11, 2007, 02:25:17 PM »
Lots of posts wile I was typing.. here it is anyway..

Sorry to interrupt - just curious here....

Which would you prefer for 16 bit listening?

A) 24 bit master > dithered to 16 bits

B) Native 16 bit samples

I would tend to favor "B" - theoretically...
With the little knowledge that I have I would prefer 24 > 16 with dither.  I say this because the capture and any post processing[/] would be done in 24 bit where there is less loss from this work, I believe.  I always record 24/48 and do post in 24/48 for this reason.  When I am done I mix down and dither to 16/44.1.

If it is just captured at 24 and then dithered down to 16 I do not see what benefit there can be.  It can never sound better than the original.  As has been posted earlier, it can sound different, but only with tweaking.  If I am wrong someone please tell me how the copy can sound better than the original.  Can a 24 bit copy sound better than a 16 bit copy?  I do not know.  Some think so.  But I would not put a lot of credence in that myself until it had been tested many times in double-blind tests.  As usual, YMMV.   8)

Happy Trails.

No, you both are missing the point.  The dithered 16bit signal doesn't sound better than the 24bit one.  Information is lost durring that transformation of course. But the resulting 16bit signal, after being dithered from 24bits is better than one originally recording at 16bits.  There is more information there because of the dithering process.

Keep in mind that processing can be done at a higher bit rate than the origional file, so future processing is not the concern, it's capturing extra information at the inital A>D stage in hte original recording.  

Tim is spot on with the comment on the A/D's that sample at 20 or 24bits and imediately dither down to a 16bits for recording to DAT or any other 16bit format.  It 'crams' more information in the 16bit file by sampling at 24bits and dithering than it would get by sampling at 16bits.

Another interjection:

If I follow the dmccabe logic - shouldnt we all be remastering our old 16 bit DATs at 24 bits - just like the cassette process? If 24 bits makes cassettes sound better, why not 16 bit DATs???

Good question.  I think the answer is that the 16bit tapes have already gone through the inital A/D stage when they were recorded, so any information beyond 16 bits is already lost.  Analog tapes have not been sampled yet and so can benefit from a better inital A/D conversion. Just a guess.

Here's another good article on dither from the Rane website. (Here's the same as a PDF). One thing mentioned inb it that I found interesting and I think applies to this topic (bold is my emphasis)-

Quote
Here is what is gained by using 20-bits:

  • 24 dB more dynamic range
  • 24 dB less residual noise
  • 16:1 reduction in quantization error
  • Improved jitter (timing stability) performance

And if it is 24-bits, add another 24 dB to each of the above and make it a 256:1 reduction in quantizing error, with essentially zero jitter!

As stated in the beginning of this note, with today's technology, analog-to-digital-to-analog conversion is the element defining the sound of a piece of equipment, and if it's not done perfectly then everything that follows is compromised.

With 20-bit high-resolution conversion, low signal-level detail is preserved. The improvement in fine detail shows up most noticeably by reducing the quantization errors of low-level signals. Under certain conditions, these course data steps can create audio passband harmonics not related to the input signal. Audibility of this quantizing noise is much higher than in normal analog distortion, and is also known as granulation noise. 20-bits virtually eliminates granulation noise. Commonly heard examples are musical fades, like reverb tails and cymbal decay. With only 16-bits to work with, they don't so much fade, as collapse in noisy chunks.

Where it really matters most is in measuring very small things. It doesn't make much difference when measuring big things. If your ruler measures in whole inch increments and you are measuring something 10 feet long, the most you can be off is 1/2 inch. Not a big deal. However, if what you're measuring is less than an inch, and your error can be as much as 1/2 inch, well, now you've got an accuracy problem. This is exactly the problem in digitizing small audio signals. Graduating our audio digital ruler finer and finer means we can accurately resolve smaller and smaller signal levels, allowing us to capture the musical details. Getting the exact right answer does result in better reproduction of music.


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Re: Analog to 24bit question
« Reply #35 on: December 11, 2007, 02:29:54 PM »
I think the answer is that the 16bit tapes have already gone through the inital A/D stage when they were recorded, so any information beyond 16 bits is already lost.  Analog tapes have not been sampled yet and so can benefit from a better inital A/D conversion. Just a guess.

Exactly.

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Re: Analog to 24bit question
« Reply #36 on: December 11, 2007, 02:37:42 PM »
You can't have sound without bit rate (dynamic range) AND frequency response. This is SO BASIC, I am still amazing you are continuing to post like you actually have experience in transferring analog cassettes to digital. Dynamic range IS DIFFERENT than frequency response, but they are directly related to each other. The fidelity of frequency response is also directly related to the quality of the bit converter. A sine curve at 24/96 has more detail than a sine curve at 16/44.1 -- even when the dynamic range is the same!

-----------------

When you bring up the Nyquist theorem, it shows that you are trying to make your argument based on the mathematical limitations of dynamic range. Are you also saying that all A/D's running at the same bit rate and frequency sound the same?
Are you saying that any A/D sounds the same at 16-bit and 24-bit?

If the sine wave is within the frequency range that the digital system can handle (less than half of sample rate), then it is perfect on both 16 and 24 bit systems. Basic fact of digital recording. That's why we have low-pass filters after the D/A converter to smooth the signal...

Nyquist theorem has nothing to do with dynamic range, only with maximum frequency range of a given digital system. Basically the sample rate has to be twice the highest frequency recorded.

Dynamic range and frequency range are not connected in any way. Bit rate dictates the DR, sample rate only the maximum recordable frequency.

You have your bitrate/samplereate wires crossed somewhere, dmccabe, sorry.

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Re: Analog to 24bit question
« Reply #37 on: December 11, 2007, 02:39:30 PM »
...
As for going to 16-bit from a 24-bit source.

Most top mastering engineers (I am not going to drop a name here, but think the entire Hendrix remastered catalog) take their 24/192 digital masters and send them back out to analog... then back into 16-bit/44.1 A/D of their choice for Cd master... rather than dither them in software
....
If you have a DAT master that is a good show you want to edit... you want to do all those edits in 24-bit. So the best way to get it there is out through a good DAC, like the Benchmark DAC-1. If you use a shitty DAC, then you might as well upsample the 16/48 DAT for a better result. You need a good DAC to get it back to analog.
...

Interesting, thanks.  I suppose conversion is an area where the digital schemes can still use plenty of improvement to compete with quality hardware when going either direction.

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Re: Analog to 24bit question
« Reply #38 on: December 11, 2007, 02:44:03 PM »
I think the answer is that the 16bit tapes have already gone through the inital A/D stage when they were recorded, so any information beyond 16 bits is already lost.  Analog tapes have not been sampled yet and so can benefit from a better inital A/D conversion. Just a guess.

Exactly.

Both are analog signals - so because one was derived from a DAT, it cant benefit from your process?

It would seem to me that both should benefit...

furthermore - the above seems in conflict with:

Quote
you want to do all those edits in 24-bit. So the best way to get it there is out through a good DAC, like the Benchmark DAC-1. If you use a shitty DAC, then you might as well upsample the 16/48 DAT for a better result. You need a good DAC to get it back to analog.

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Re: Analog to 24bit question
« Reply #39 on: December 11, 2007, 02:50:17 PM »
And one step futher, but much more contemporary, is the Grace ANSR.
Were you running it, or no?
Do you want to further change that?

Interesting thought.  I ran ANSR, quite a bit.  I generally didn't edit those recordings in post, but I did on a few occasions.  For some reason I thought ANSR is different from traditional dither, though I couldn't tell you why.  ???

I think the answer is that the 16bit tapes have already gone through the inital A/D stage when they were recorded, so any information beyond 16 bits is already lost.  Analog tapes have not been sampled yet and so can benefit from a better inital A/D conversion. Just a guess.

Exactly.

I'm struggling to reconcile the above with...

If you have a DAT master that is a good show you want to edit... you want to do all those edits in 24-bit. So the best way to get it there is out through a good DAC...

So if information beyond the original 16-bits is already lost, why transfer 16-bit DAT > DAC > ADC > 24-bit?  If the additional information is already lost, converting to analog and then back to digital doesn't re-gain any of the lost information.

And if the point of converting back to analog (as has been suggested) is to avoid dithering twice (dither once when capturing the original 16-bit capture, then dithered again after editing)...isn't the dither noise in the 16-bit recording simply passing into the analog realm, then back into digital, i.e. doesn't the dither noise remain in the signal, so you're experiencing the effects of double-dithering regardless?

Additionally, if one's editing software uses 32bfp internal precision, then wouldn't going 16-bit > DAC > ADC > 24-bit still require two dither stages?  Once to capture the 24-bit recording, and again dithering from 32bfp to one's target bit depth?  So really, if we assume converting into the analog realm and then back to digital, is, in fact, better, shouldn't we go 16-bit > DAC > ADC > 32 bfp (or at whatever bit-depth our editing software uses internally)?

If the answer to going back into the analog realm is to leverage the sonic characteristics of the intermediate analog gear, that's one thing.  But if it's to avoid double-dither, or some other issue, I'm still not grasping why it's better.
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Re: Analog to 24bit question
« Reply #40 on: December 11, 2007, 03:05:39 PM »
If the sine wave is within the frequency range that the digital system can handle (less than half of sample rate), then it is perfect on both 16 and 24 bit systems. Basic fact of digital recording.

There is no such thing as a perfect digital reproduction from an analog signal!

You have your bitrate/samplereate wires crossed somewhere, dmccabe, sorry.

The topic is analog to digital...
The example was an analog "sine wave" converted it to digital... Mr. Nyquest.
It's always the guys who don't actually master audio that bring up the Nyquest Theory that they just Google'd...

Do you agree that different A/Ds have different sounds? Some are more transparent than others, some have a warmer coloration. OK... but they are both at 16/44.1, how do you account for the fact that they sound different.

Do you have the hardware to play back a 24-bit file? There are plenty of MOTB releases done at 24-bit and 16-bit -- both from the same analog source. Go see if you can hear the difference?
« Last Edit: December 11, 2007, 03:19:35 PM by dmccabe »

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Re: Analog to 24bit question
« Reply #41 on: December 11, 2007, 03:10:09 PM »
You have your bitrate/samplereate wires crossed somewhere, dmccabe, sorry.

The topic is analog to digital...
The example was an analog "sine wave" converted it to digital... Mr. Nyquest.
It's always the guys who don't actually master audio that bring up the Nyquest Theory that they just Google'd...

Do you have the hardware to play back a 24-bit file? There are plenty of MOTB releases done at 24-bit and 16-bit -- both from the same analog source. Go see if you can hear the difference?

Sorry for the topic sway - just trying to see if we arrive at the same conclusion from a different angle.

Im sure there are differences - however we dont seem sure what to attribute those differences to. You are pointing to a factor that we all dont agree on...

And..lighten up a little dude - you sound FAR from authoritative.

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Re: Analog to 24bit question
« Reply #42 on: December 11, 2007, 03:13:53 PM »


I'm struggling to reconcile the above with...

If you have a DAT master that is a good show you want to edit... you want to do all those edits in 24-bit. So the best way to get it there is out through a good DAC...

So if information beyond the original 16-bits is already lost, why transfer 16-bit DAT > DAC > ADC > 24-bit?  If the additional information is already lost, converting to analog and then back to digital doesn't re-gain any of the lost information.

no additional gain of info, the transfer to 24-bit from DAT is assuming you have additional edits to do... pitch change, cross fades, patches. You would not want to make any of those type of changes to a 16/48 file, you would hear digital artifacts from those types of changes. You would "hear them less" when done in 24/96 -- them sampled back down to 16/44.1

It's all about doing the least amount of destructive artifacts. You don't "gain" any quality. You just don't gain as much artifacts.
« Last Edit: December 11, 2007, 03:16:29 PM by dmccabe »

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Re: Analog to 24bit question
« Reply #43 on: December 11, 2007, 03:26:02 PM »

And..lighten up a little dude - you sound FAR from authoritative.

I don't pretend to be  the authoratative voice on the subject... but when some one makes a statement like:
"But like I said, 24 bits only makes files larger", then bullshit needs to be clarified.

If you have any audio samples you would like to post to show your point of view, feel free to include links.
There are over 50 shows with links to find them as examples over at www.motb.org.

Transferring analog sources is what the MOTB project is all about... and since we have some actual PRO AUDIO ENGINEERS in the group, maybe you should take some time listening.

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Re: Analog to 24bit question
« Reply #44 on: December 11, 2007, 03:30:14 PM »
Care to refer us to any sources?

What should we listen for?

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Re: Analog to 24bit question
« Reply #45 on: December 11, 2007, 03:35:25 PM »
On the Official Releases page, there are links to the trackers that have all those shows seeded. EVERY show is seeded in 24-bit and 16-bit. Go download 1 song from any of those shows, get the 16-bit version and the 24-bit version of the same song. Compare them.

This is an example of where transferring a cassette to 24-bit sounds different than 16-bit.

You might want to pick an older show to get the most noticeable difference, like one of the Garcia Legion of Mary shows from 1975. Mics were placed on stage at the Keystone in San Francisco taped by Bob Menke. Very good sounding master with huge dynamic range. Listen especially to the horns.

Same master tape... for both 16-bit and 24-bit versions. We seed the 16-bit for CD burns. Not everyone has access to burn DVD-A.
« Last Edit: December 11, 2007, 03:37:57 PM by dmccabe »

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Re: Analog to 24bit question
« Reply #46 on: December 11, 2007, 03:43:41 PM »
As for going to 16-bit from a 24-bit source...

Most top mastering engineers (I am not going to drop a name here, but think the entire Hendrix remastered catalog) take their 24/192 digital masters and send them back out to analog... then back into 16-bit/44.1 A/D of their choice for Cd master... rather than dither them in software. Anyone who states that once it is "digital it should stay digital"... is not doing what the top mastering engineers are doing in LA today.

I concur with all the major points dmccabe has made, especially pertaining to recordings originally captured as analog masters.

Furthermore, even with 24 bit files that were originally recorded as a 24bit digital signal, most top mastering engineers will take the final mix through a D>A>signal processing>A>D signal chain for ultimate mastering.  The reason: very few sound engineers would maintain that any digital signal processing - whether on a DAW or with digital external hardware - is as accurate or 'sounds as good' as signal processing through top quality (and generally vintage) analog gear.  Point me to a plug-in that can EQ as well as an Orban 622B ParaEQ (for one example), or a comressor/limiter that sounds as good on a master bus as a pair of the vintage DBX 165a/160, etc (for another example).  


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Re: Analog to 24bit question
« Reply #47 on: December 11, 2007, 03:54:43 PM »
There is also a trend going away from digital mixing... a company called Dangerous makes an analog summing buss.

You take your individual 24/192 digital tracks that you recorded with the musicians in the studio... and send each track out to a very-high quality DAC... then you MIX all the tracks in ANALOG. Engineers swear this sounds better than digital mixing. More natural in the way the tracks lay. Then the final mix goes BACK to final digital in many modes... 16/44.1 for the CD master and other rates for other releases but all from that same analog mix.

BTW, an 16x2 channel Dangerous Summing Box costs: List $2,999.00
How many channels does your band need?

here is more info on the subject: http://www.studioreviews.com/summing-box-shootout.htm

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Re: Analog to 24bit question
« Reply #48 on: December 11, 2007, 03:59:52 PM »
And one step futher, but much more contemporary, is the Grace ANSR.
Were you running it, or no?
Do you want to further change that?

Interesting thought.  I ran ANSR, quite a bit.  I generally didn't edit those recordings in post, but I did on a few occasions.  For some reason I thought ANSR is different from traditional dither, though I couldn't tell you why.  ???.

I haven't used that on my V3, but I suppose it's a 'noise shaped' or 'colored' dither like UV22 and others that attempt to put more of the dither noise in regions that are less audible because of the non-linear low level sensitivity of the ear (out of the midrange).

Quote
...
If the answer to going back into the analog realm is to leverage the sonic characteristics of the intermediate analog gear, that's one thing.  But if it's to avoid double-dither, or some other issue, I'm still not grasping why it's better.


I don't think it's the double-dither, double-dipper.  There's actually more dithering going on with the analog route since there is another A>D step in there.  

I can imagine that increasing the sample rate of an existing digital file by doing a D>A>D conversion with quality gear may sound better than using a digital sample rate conversion algorithm because of rounding errors in the complex math involved when doing it digitally.  I'd think increasing the bit depth digitally to edit the file with more precision would not be a problem though since you're just increasing the word length with zeros.. no complex math.

If so that would mean a 16/44.1 > 24/44.1 (or 32/44.1 or 64/44.1) digital conversion is just as good as going the analog route (excluding any nice 'coloration' the analog stuff my add of course) as long as you don't change the sample rate.

^^
That applies to adding zeros to the bit depth only, not converting sample frequency up or down converting bit depth or sample rate.  It also doesn't address any processing done later in the rarified air of the mastering suite, as easyjim noted.

Does that jibe with your observations, dmccabe?
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Offline dmccabe

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Re: Analog to 24bit question
« Reply #49 on: December 11, 2007, 04:10:42 PM »
The ANSR is the name of the Grace Design dither... for 16-bit only. It is off in 24-bit.
It sounds great for going to 16-bit.

So if you had a DAT that needs some patches.
You could transfer the DAT to analog (Benchmark DAC-1) then back into your V3 to 24/96.
Then do your patches or any other edits.
Then take that edited digital 24/96 and go back out again to analog...
into the line in*** of the V3 and use the V3 as the A/D to 16-bit using the ANSR dithering.

That would result in one dithering.

Don't underestimate what a good DAC can do for your sound.
They also double as a high-end headphone monitoring device.

When you edit your digital 24/96 files, what are you listening to? Are you sure?
If your headphone or speakers are playing out of your computer... through what?
What DAC is going to your headphones or speakers?

If you are using a $100 audio card... you are not really listening to the 24/96 file...


***the "line in" on a stock V3 is really set for the impedance of a mic signal... not a line signal.
You can still feed a line in source to a V3, but to be "perfect" and match the impedance, you need the mod.
Micheal Grace did a special mod for our gear to allow mic or line in (you have to open the case and change jumpers to go back and forth). You can have him mod your V3 if you want to use you V3 as a A/D from line sources.
« Last Edit: December 11, 2007, 04:21:49 PM by dmccabe »

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Re: Analog to 24bit question
« Reply #50 on: December 11, 2007, 04:19:15 PM »
...You take your individual 24/192 digital tracks that you recorded with the musicians in the studio... and send each track out to a very-high quality DAC... then you MIX all the tracks in ANALOG. Engineers swear this sounds better than digital mixing. More natural in the way the tracks lay. Then the final mix goes BACK to final digital in many modes... 16/44.1 for the CD master and other rates for other releases but all from that same analog mix...

So funny how going forward sometimes mean looking back.  So the next to uber quality step is to sum in the air? Play each track through it's own dedicated monitor and sum it all in the air to a fresh stereo pair sampled at every rate you'll ever need.  Let's bust out the Yamaha player pianos, the robotic drummers, re-invent the old theater organs but replace the old paper rolls with CNC controllers, all so the actual sound of the instrument is plucked on each playback, just like in the parlors of robber barons of the 1920's.  F%ck MTV, I want my full animatronic GratefulDead that I can arrange in my living room, and Bear/Healy can program the piano roll.

Apologies for the swerve, my brain needed a diversion.
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Offline Brian Skalinder

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Re: Analog to 24bit question
« Reply #51 on: December 11, 2007, 04:23:55 PM »
So if you had a DAT that needs some patches.
You could transfer the DAT to analog to 24/96.
Then do your patches or any other edits.
Then take that edited digital 24/96 and go back out again to analog...
into the line in*** of the V3 and use the V3 as the A/D to 16-bit using the ANSR dithering.

That would result in one dithering.

I don't understand.  The dither noise from the first dither (V3 w/ ANSR on > DAT) is still in the signal, even if you convert it to analog and then back to digital.  So aren't you still dithering twice?  There just happens to be an analog generation in between (which may matter, but if it does, I'm not clear why from your statements).
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Re: Analog to 24bit question
« Reply #52 on: December 11, 2007, 04:44:11 PM »
When you use a good DAC, it uses its internal SRC to add 8 bits of dither to the incoming 16-bit digital signal. This increases the apparent data depth to 24 bits. The added dither pushes the quantization residue without adding to the recording’s noise floor. So your new analog signal has 24-bits instead of 16. That's where the difference is instead of just using software to upsample a 16-bit to 24-bit. The hardware version is more "accurate" -- sounds better.

So, yes you are adding a dither in that stage, but it's an "up" dither... the ones that are destructive are when you are going from higher to lower. That's when the word length gets reduced... so that's where you only want the one "down" dither.
« Last Edit: December 11, 2007, 04:50:00 PM by dmccabe »

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Re: Analog to 24bit question
« Reply #53 on: December 11, 2007, 04:48:51 PM »
When you use a good DAC, it uses its internal SRC to adds 8 bits of dither to the incoming 16-bit digital signal. This increases the apparent data depth to 24 bits. The added dither pushes the quantization residue without adding to the recording’s noise floor.

So, yes you are adding a dither in that stage, but it's an "up" dither... the ones that are destructive are when you are going from higher to lower. That's when the word length gets reduced... so that's where you only want the one "down" dither.

SRC?  Only SRC I know is Sample Rate Conversion, which is a different animal than bit-depth and dither.

Interesting.  But even though you're adding "up" dither, the original "down" dither noise is still in the signal.  So you really have 1 down dither + 1 up dither + 1 down dither (the final down dither) once back in the digital realm.  So there are still 2 down dithers.  Does the DAC "up" dither somehow negate the previous "down" dither?
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Re: Analog to 24bit question
« Reply #54 on: December 11, 2007, 04:54:07 PM »
They don't think they ever negate each other... and yes, the less dithers the better... but the last dither is always the most critical. If you keep going back and forth too many times between analog and digital, I am sure you are going to start introducing audible artifacts. You have to plan ahead. The lesser of two evils. If you edit a 16-bit in software with a 32-bit plugin... you are technically dithering there too...
« Last Edit: December 11, 2007, 04:57:03 PM by dmccabe »

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Re: Analog to 24bit question
« Reply #55 on: December 11, 2007, 04:59:52 PM »
I think you want to dither anytime you convert bit depth downwards or anytime you move between the analog and digital worlds, in either direction. No?
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Re: Analog to 24bit question
« Reply #56 on: December 11, 2007, 05:08:07 PM »
Question about MOTB approach for analog transfers:

Do you add gain during the transfer or just take the tape to the AD at a unity setting?

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Re: Analog to 24bit question
« Reply #57 on: December 11, 2007, 05:09:13 PM »
They don't think they ever negate each other... and yes, the less the better... but the last dither is always the most critical.

I understand less dither is better.  My point is that regardless of whether one takes the original recording into the analog realm or not, the final audio still contains noise from two "down" dithers:

<A>  (dither 1) 16-bit master > DAC > ADC (24-bit or 32bfp) > edit at 24-bit or 32bfp > dither to target output (dither 2)
<B>  (dither 1) 16-bit master > edit at 24-bit or 32bfp > dither to target output (dither 2)

(As I hope the above two examples illustrate), both still have dither noise from two different "down" dithers.  So I still don't understand how/why one is better than the other.  I guess I'll poke around online for references / documentation (if you have any to share, I'd like to check them out) to help me better understand why one is better than the other.

On a related note:  does the software MOTB uses for edits use 24-bit or 32bfp internal precision?  And if the latter, wouldn't it make sense to convert 16-bit > DAC > ADC > 32bfp?
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Re: Analog to 24bit question
« Reply #58 on: December 11, 2007, 05:17:29 PM »

On a related note:  does the software MOTB uses for edits use 24-bit or 32bfp internal precision?  And if the latter, wouldn't it make sense to convert 16-bit > DAC > ADC > 32bfp?

Taking that notion one step further - why not start with a 32bfp fileset on the analog transfers also? If 24 is better, why not 32?

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Re: Analog to 24bit question
« Reply #59 on: December 11, 2007, 05:25:03 PM »
Are we talking about increasing the bit depth and leaving the sample rate unchanged or is the sample rate being increased as well?
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Re: Analog to 24bit question
« Reply #60 on: December 11, 2007, 05:30:28 PM »
Are we talking about increasing the bit depth and leaving the sample rate unchanged or is the sample rate being increased as well?

Both bit-depth and sample rate are in play within this discussion.  Personally, I'm trying to understand the issues involving bit-depth first, before moving on to sample rate, hence my focus strictly on bit-depth in the last few posts.
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Re: Analog to 24bit question
« Reply #61 on: December 11, 2007, 05:39:38 PM »
I think I misspoke when stating all DACs "upsample" before the analog stage... some DACs do not. (I got slapped for that statement by someone who knows a lot more than me :)
If you look at different D/As, they use different methods, so there is no one way in the D/A stage.
It all depends on the actual chips in the particular hardware. Some DACs use 1-bit chips, so there is no dither at all in the D/A... but the point is not to get all hung up on the "engineering" of the specs.

The bottom line is "more bits" the better the signal to noise ratio.
The more frequency response, the better the detail on the transients.

They go together... you want both. That is why if you transfer a cassette at 24/96 rather than 16/44.1 it will sound more "open"... more analog.

And yes, "generally" the more dithers, the more chance for unwanted artifacts.

When you get into individual manufacturers of hardware you can then throw in a coloration equation.
Each piece of gear can make the sound different. So then it comes down to which sound you like best.

I don't design or make the hardware, I just use it :)
« Last Edit: December 11, 2007, 05:51:49 PM by dmccabe »

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Re: Analog to 24bit question
« Reply #62 on: December 11, 2007, 05:50:48 PM »
So I still don't understand how/why one is better than the other.

I am not stating one method is better than another... you need to compare specific hardware and/or software.

If you read that article I posted about the analog summing, the writer starts the entire comparison by stating he is not trying to say which one is "better".
For each type of music, there might be one method that sounds better. The key is to know all your options.

For some people, once they are in digital, they stay all digital.
Others like to use analog in their mixing or mastering stages.

But to get back to the very first original post, "does it make any difference to go to 24-bit when transferring a cassette: -- I'll answer YES.
Your mileage may vary.

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Re: Analog to 24bit question
« Reply #63 on: December 11, 2007, 06:02:44 PM »
The bottom line is "more bits" the better the signal to noise ratio.
The more frequency response, the better the detail on the transients.

And...we're back to page 1.  Oh, well.

I don't think anyone disputes this, in general.  The question from the very beginning of the thread, batted about in various forms since, is:  if the SNR and frequency response are limited in the analog master (for example, 60 dB SNR and 16 kHz), how does transferring at higher bit-depth and sample rate improve <a> SNR, and <b> frequency response / transient detail?  Nothing I've seen here suggests that it does, but I do suspect that the closer approximation of the analog waveform - while not necessarily improving SNR or frequency response - provides better precision for future editing in the digital realm.

Time to poke around on Google to dig into the above, and subsequent, unanswered questions (why the "best" way to get from a 16-bit digital master to 24-bit is through DAC > ADC, how introducing an analog stage somehow helps with the negative effects of double dither, etc.).  Nothing personal, it just seems we're not going to get to the answers here.
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Re: Analog to 24bit question
« Reply #64 on: December 11, 2007, 06:02:53 PM »
On a related note:  does the software MOTB uses for edits use 24-bit or 32bfp internal precision?  And if the latter, wouldn't it make sense to convert 16-bit > DAC > ADC > 32bfp?

There are many different editors, each uses their own DAW gear... so it's not all the same for every release. If you look at the lineage on each show, it is well documented.
And not all plug-ins are the same quality as well. Everyone uses what they can afford. We do get donated gear... and are very thankful to our sponsers, but we do this all for free, we don't get paid or anything.  But I think I can say generally most of us are going for 24/96 for our own personal files. Those files are big enough. Going to 32bfp is sooo much bigger... there is a point of diminishing return.

And now we are doing DSD stuff... each raw transfer ends up around 10megs per show. People have to ftp files... etc. 24/96 seems high enough for now. As cpu and internet bandwidth get faster...
who knows what we will all be working with in a few years. Last we knew 16-bit was the "best"!
« Last Edit: December 11, 2007, 06:07:08 PM by dmccabe »

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Re: Analog to 24bit question
« Reply #65 on: December 11, 2007, 06:05:57 PM »

Time to poke around on Google to dig into the above, and subsequent, unanswered questions (why the "best" way to get from a 16-bit digital master to 24-bit is through DAC > ADC, how introducing an analog stage somehow helps with the negative effects of double dither, etc.).

You are probably not going to find any studios who get paid for transferring DAT masters -- what bands used 2 track DAT for studio work? -- maybe ADATs but..., thus very little online discussion about the best methods of editing ADATs. The best way is to test yourself. Do you have a high-end DAC? If not, send me a DAT, I can transfer it through a Benchmark DAC-1, then back to digital via Mytek at 24/96. Then ftp you the file, you can do your own tests of that file compared to a raw 16/48 transfer.

Then also edit both files and do another comparison.


Nothing personal, it just seems we're not going to get to the answers here.

Oh how I agree... there is no manual for sound engineers that says "this is how it is supposed to be done".
And if there was, no one would pay the pros to do it!

I just consider myself an addicted audiophile with an expensive hobby. Others in our group make a living mastering audio.
« Last Edit: December 11, 2007, 06:16:19 PM by dmccabe »

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Re: Analog to 24bit question
« Reply #66 on: December 11, 2007, 07:18:30 PM »
If you come across anything you find enlightening Brian, please post here.

I still suspect that increasing bit depth only in the DAW is inconsequential as it just pads the word with zeros. In contrast to that, I suspect changing the sample rate either up or down (other than by a simple multiple, such as 44.1>88.2 or 48>96) is where the digital conversions cannot (yet?) do as nice a job as converting to analog and back to the target rate like MOTB & the mastering world are doing.

The reason I suspect that's the case is because SRC is a more complex mathematical operation (multiplication) not unlike digital summing (addition) that the guys going to analog summing that dmccabe posted the article about are avoiding.

Good discussion. dmccabe, thanks for sharing your on-hand experience with this stuff and especially for your MOTB work.
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Re: Analog to 24bit question
« Reply #67 on: December 11, 2007, 08:30:35 PM »
^^^^  Thank you!
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An engineer's take
« Reply #68 on: December 11, 2007, 08:38:53 PM »
Warning: huge post -- I'm just thinking out loud.

Here's my perspective as an image processing engineer... The same issues of bit depth and sampling rate (resolution) obviously come up in image processing. (I have also built a lot of audio processing circuits using pure data)

First, the difference between 44, 48 and 96k sampling rates. This is analogous to image resolution. In image processing, fine details are referred to as "high frequency," and this corresponds to high frequency audio. The better the sampling rate, the better you're able to reproduce high frequency detail. 44k has 0.92 times the resolution of 48k, and 96k has twice the resolution of 48k.

Here's 44k:


And 48k:


And 96k:


You can see how small the difference is between 44k and 48k. Both have enough resolution after Nyquist to cover the range of normal human hearing, which tops out at ~20k. There are some fine details you can see better in the 48k image, but you're squinting.

The 96k image is obviously larger, and there are more details, but... uh, unless you pitch shift that audio, it's just resolving details you can't hear. It would be like using a 20 megapixel camera when your lens could only resolve 10 MP of information. Just my opinion.

Bit depth is another issue. Bit depth is not "resolution." It simply defines how many steps of amplitude there are between 0 and 1. (1 usually being 0db). Here's the 24bit color image (8 bit per color channel, commonly but confusingly called "8 bit"):


Here's the same image resampled to 4 bit (16 amplitude values), so you can see the steps clearly:

Obviously this is why no one records at 4 bit. :o

However, here's the same image in 4 bit, but dithered using the algorithm generally accepted as "best" -- it's called error diffusion, or just "diffusion" for short:

Clearly, that is much, much better.

Here's a less impressive dithering technique ("pattern dither").

This demonstrates that the dithering style does make a difference -- pattern dither introduces distracting artifacts. IMO, for best results, use gear/software with error diffusion dithering.

Some audio gear actually just takes the bottom 16 bits of a 24bit signal (someone said Quicktime does this?). This is disastrous, since anything that reaches above 2/3 of the way to 0db will be unceremoniously clipped off. That would look like this:


24 bit vs 16bit
Okay, so how big of a difference is there between 16 and 24 bit? Well, in order to make the difference more clear, I'm going to filter out the high frequency information by blurring the image. Here's a 24bit image:


And here's the 16bit version (resampled to 24bit jpg for viewing on the web):


Wow, okay, the 24 bit version looks loads better! Well, that 16bit conversion didn't use any dithering. If we were converting from 24 > 16bit and used diffusion dithering, you would be hard pressed to spot the difference.

There's actually another factor here related to dithering. Error diffusion works by introducing minute errors (noise) to the signal. But your signal already has noise if it came from a mic, went through a pre, and passed through A/D. So even if you're not working with gear that dithers, odds are that your signal is doing the dithering for you.

Here's the same 24bit image with a small amount of noise added to emulate the amount of noise in a nice, clean recording:


And the signal with the same amount of noise at 16 bit:

It looks very close to the 24bit image. The 24bit is maybe a tiny bit nicer, but you have to be looking for it.

This is why Sound Devices says that at full signal level, 24 and 16 bit sound "largely identical." They should know.

Headroom
Of course, as SD points out, the world is not perfect, and you can't always be kissing 0db. So how does 24bit stack up against 16bit if you record with your levels down, and normalize (multiply) them later? Lets say you leave enough headroom that most of your audio peaks at -18db. You want to be sure that you won't clip if the dude next to you yells. That means you're only using 1/8th of the available levels -- in 16bit, 8192 levels; in 24bit, 2,097,152 levels. In other words, now your 16bit audio is really 13bit, and your 24bit is really 21bit.

The real issue, however, isn't the loss of amplitude fidelity. It's the fact that we're amplifying the ADC noise. Let's imagine that our ADC introduces an amount of noise that equals about 2 amplitude levels. So out of the 8192 levels used in our 16bit file, 2 of those are noise. And of the 2,097,152 levels used in our 24bit file, 2 of those are noise. See the issue? In the 16bit file, 2/8192 = 0.024% of the signal is noise, versus 0.0001% noise for the 24 bit file. Relative to the signal, the 16bit file has exactly 256 times more digital noise than the 24bit version.

That said, ADCs are all different, and will have different "characters," so if you do plan on running with a lot of overhead, it's probably best to try a few out and see what sounds good to you.

Dynamic Range
So, what about dynamic range? Dynamic range is one of the most misunderstood terms in both digital photography and digital audio. That's partially because DR figures are almost always given in logarithmic scale (stops in photography, dB in audio), and people always get tripped up with log numbers. It's also because people aren't sure what affects DR.

There are only two things that can affect DR: noise and bit depth. Noise is simple: your mic has a self-noise, your pre adds some noise, and your ADC adds some noise. Whatever you have left between the noise floor and 0db is your raw dynamic range. Bit depth affects DR, because audio is typically encoded linearly. So in 16bit, you have 65536 levels. Half of those levels (32768) cover 0db to -6db, half of the remaining levels (16384) cover -6 to -12, half of that (8192) cover -12 to -18, etc. By the time you get to the range between -66db and -72db, there are only 16 levels of amplitude to describe the waveform -- pretty gritty. By the time you get to the range between -84db and -90db, there are only two levels -- a square wave, either on or off. Of course, noise takes over long before we get to that point.

So the theoretical dynamic range of 16bit is 90db, but the last 30db or so are pretty rough. This is why some people think very soft sounds start to sound bad in 16bit -- for example, the oft-cited "end of the decay of a cymbal." There are plenty of microphones that have over 66db of dynamic range, so they can expose the limits of 16bit.

With 24bit, you start off with more levels. There are over 8 million levels to describe the amplitude between 0 and -6db!!! Obviously massive overkill. But the result is that between -66db and -72db, you have 4096 levels available vs 16 levels in 16bit. It takes 24bit a bit longer to get clipped to 16 levels -- you have to get down to the range between -114db and -120db. The theoretical limit to the DR in 24bit is 138db, because the range between -132 and -138db gives us only two levels.

Luckily, no microphone is capable of capturing that. Even microphone/pre/ADC combos capable of reaching 80db of DR will still have 1024 levels with which to describe their noise floor in 24bit. :)
« Last Edit: December 11, 2007, 09:18:01 PM by bensyverson »

Offline bensyverson

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Re: Analog to 24bit question
« Reply #69 on: December 11, 2007, 08:43:52 PM »
LOL

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Re: Analog to 24bit question
« Reply #70 on: December 11, 2007, 09:11:15 PM »
I just wanted to add something... for those people concerned about dithering "adding noise" to their recordings. First, dithering only occurs when going from a higher bit depth to a lower one. So you want it when going 24 > 16, but not 16 > 24. When going from a lower bit depth to a higher one, all your hardware will do is "add zeros" to the end of the values. Actually, you shift the bits, but "adding zeros" is a good way to think about it. The point is, going UP the bit depth chain in the digital realm is absolutely lossless. You literally lose nothing from the original. You also gain nothing unless you're doing processing -- that is, there's no reason to bump 16bit DAT materials to 24bit unless you want to apply effects in 24bit space. You can't improve your 16bit recordings by resampling them to 24bit.

Okay, back to the noise added by dithering. The noise added is always less than one value in the destination bit depth. So if you're going from 24 > 16, the amount of noise added is less than one value out of 65,536. Nothing to worry about.

The noise is basically added to round numbers -- say you have an amplitude sample that's roughly 33.33% of the max. In 24bit, maybe the value is 5,592,550. If we divide this by the max value (16,777,215) and then multiply it by the max value of 16bit (65535), we get 21845.566. Without dithering, we'd just chop it off at the decimal place and call it 21845. Error diffusion will add a random value between -1 and 1 before the rounding happens. So for example, it might add 0.6 to 21845.566, to get 21846.166. Once we round the number, now we have 21846. Or we might get 21844. That way, we get a smoother gradation between values.

The point is: I wouldn't sweat the dithering "noise."
« Last Edit: December 11, 2007, 09:14:39 PM by bensyverson »

Offline dmccabe

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Re: An engineer's take
« Reply #71 on: December 11, 2007, 09:32:12 PM »

Here's my perspective as an image processing engineer... The same issues of bit depth and sampling rate (resolution) obviously come up in image processing.

You can see how small the difference is between 44k and 48k. Both have enough resolution after Nyquist to cover the range of normal human hearing, which tops out at ~20k. There are some fine details you can see better in the 48k image, but you're squinting.

The 96k image is obviously larger, and there are more details, but... uh, unless you pitch shift that audio, it's just resolving details you can't hear.

Very good analogy... but as indicated in red, "unless you" -- meaning you are going to make an edit... so, if you were to make a retouching edit to an image, the 96K image would certainly give the best quality image for the edit - not on the smaller one, even if you then had to res it down to a smaller size later... but still needed the bigger size -- for your DVD-A burn...

so if you need to do a...
- pitch shift
- eq and tonal adjustments
- cross fade from patches
- broadband noise reduction
... what bit rate and frequency would you use on a cassette master?

--- what bit rate and frequency would you use to make the same edits on a DAT master?


You can't improve your 16bit recordings by resampling them to 24bit.

I don't think anyone is claiming resampling is "improving" anything. We are talking about destructive editing. And the initial thread started with an analog master.
The topics seem to shift around with people pulling out their slide rulers to demonstrate how brand X sounds so much better than brand Y.
« Last Edit: December 11, 2007, 10:06:20 PM by dmccabe »

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Re: An engineer's take
« Reply #72 on: December 11, 2007, 10:38:44 PM »
Warning: huge post -- I'm just thinking out loud.

Here's my perspective as an image processing engineer...

Wow thanks for this post! I was just reading the Wikipedia entry for optical resolution - as I thought there had to be an analogy there. Thanks for spelling it all out.

Offline bensyverson

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Re: An engineer's take
« Reply #73 on: December 11, 2007, 11:02:33 PM »
Very good analogy... but as indicated in red, "unless you" -- meaning you are going to make an edit... so, if you were to make a retouching edit to an image, the 96K image would certainly give the best quality image for the edit - not on the smaller one, even if you then had to res it down to a smaller size later... but still needed the bigger size -- for your DVD-A burn...

so if you need to do a...
- pitch shift
- eq and tonal adjustments
- cross fade from patches
- broadband noise reduction
... what bit rate and frequency would you use on a cassette master?

--- what bit rate and frequency would you use to make the same edits on a DAT master?
Very good point, and there is something that I missed that gives 96k an edge right off the bat. Your ADC will introduce errors in the form of digital noise -- at 48k, that noise is happening at (duh) 48k. If you record in 96k and output to 48k, it means you're essentially oversampling that noise by 2X. That helps smooth out errors in the ADC.

To return to my analogy, as I said, 96k is like a 20 megapixel camera, even though your lens (in this analogy, your ears) can only resolve 10 MP of information. But downsampled, the 20MP camera will make a less noisy 10MP image than a 10MP camera with the same S/N ratio. So for every one pixel in the 10MP image, there are two pixels in the 20MP image, which helps average out the noise.

Oversampling is a common method to improve your effective S/N ratio, so I have to admit, 96k has some real appeal.

How much of a difference will this make, considering 48k is already oversampling (ie, above Nyquist) the limits of human hearing? Who knows, but if you know you're going to be manipulating a signal, it always helps to start with more of it! :) If you think of 48k as roughly 2X oversampling human hearing, then 96k would be a 4X oversample, which is standard in image processing to avoid aliasing. I would think that 96k would make even a slightly noisy A/D effectively transparent compared to the same A/D running at 48k.

I don't think anyone is claiming resampling is "improving" anything. We are talking about destructive editing. And the initial thread started with an analog master.
The topics seem to shift around with people pulling out their slide rulers to demonstrate how brand X sounds so much better than brand Y.
I think that could easily be true (brand X is better than brand Y). What you'll need to faithfully reproduce an analog master will depend on the format (2" RTR? Mini cassette?), but it comes down to figuring out the noise floor of the analog master, and then determining exactly how well you want to reproduce that noise. If you basically want a digital clone of a high-quality analog tape (ie, 60db dynamic range or above), 24bit is necessary, and you'd want to sample at either 48k or 96k, depending on the quality of your ADC.
« Last Edit: December 11, 2007, 11:04:09 PM by bensyverson »

Offline Brian Skalinder

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Re: An engineer's take
« Reply #74 on: December 11, 2007, 11:34:22 PM »
... what bit rate and frequency would you use on a cassette master?

--- what bit rate and frequency would you use to make the same edits on a DAT master?

Higher the better, as most have agreed.  But most of the (well, my, anyway) questions and discussion revolved around your suggestion that it's better to convert from a lower bit-depth (or sampling rate) to a higher via a DAC / ADC analog stage, and why (which was never really addressed).  Anyway...gonna let that one go now...

The topics seem to shift around with people pulling out their slide rulers to demonstrate how brand X sounds so much better than brand Y.

Not sure where that came from - you tossed out more brand X, brand Y, engineer X, and engineer Y comments than anyone else.  :shrugs:

:coolguy:  Thanks bensyverson for your lengthy posts - they made a lot of sense to me.  I still need to review another time or two to digest fully, though.
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Re: An engineer's take
« Reply #75 on: December 12, 2007, 12:31:38 AM »
:coolguy:  Thanks bensyverson for your lengthy posts - they made a lot of sense to me.  I still need to review another time or two to digest fully, though.
Hey, thanks! It's a lot of information, but hopefully people can use it as reference too. It always helps me to have pictures to visualize tricky concepts like these...

Offline DSatz

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Re: Analog to 24bit question
« Reply #76 on: December 12, 2007, 12:32:16 AM »
Focusing on the original question: The only aspect of audio quality affected by the choice of bit depth (or word length, or whatever else you want to call it--maybe "moo shu") in a properly made digital recording is dynamic range. The noise floor will be lower when more bits are used to quantize the signal, "all other things being equal" as they say. There are no inherent differences in frequency response, distortion, imaging, warmth, jitter, toenail fungus or any other aspect of audio quality beside dynamic range where the number of bits per sample is concerned (in linear PCM at a fixed sampling rate, anyway).

As a result, the original question doesn't really have anything to do with digital! The same question would occur even if digital recording had never been invented (and yes, I do realize that for certain people that is a fond wish ...).

What matters is this: If the medium that you're copying to has a distinctly (say 10+ dB) lower noise floor than the noise floor of the source material, and you use the available range fully, then the copying process will add only negligible noise. What certain people seem not to realize (or to want to admit) is this: Once you've reached that point, any meaningful further reduction in the noise level of the copy (i.e. by choosing a quieter medium to copy to) is simply not possible. Copying from an analog source can never add zero noise, nor can it add "negative noise". And once you're adding, say, only 1/2 dB of noise, you couldn't notice the difference if that could be knocked down to, say, 1/3 dB instead, even with extremely critical listening.

But that's the only real issue in the choice between a 16- and a 24-bit transfer. The noise floor of a 16-bit digital recording is WAY the frigging fark below the noise floor of the cassette recording; heck, a 16-bit recording has distinctly wider dynamic range than a 15 ips half-track Dolby "A" master tape (which I should know--having made many, many such recordings back in the day). Thus it will not matter at all whether 24 bits are used instead of 16 for this application, as long as the copy is made with reasonable care on reasonably good equipment. Nothing audible will be gained or lost as the result of 16 vs 24 bit quantization of a cassette playback--so I say, the person should feel free to use either one! If 24 bits feels more comfortable, use it. If 16 bits feels more economical, use it. Flip a coin, use Tarot cards, spin the bottle, whatever.

Let me also just say that for live concert recording, I use 24 bits whenever I can--but that's because live performances have a far wider dynamic range than cassettes.

--best regards
« Last Edit: December 12, 2007, 12:51:28 AM by DSatz »
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Offline bensyverson

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Re: Analog to 24bit question
« Reply #77 on: December 12, 2007, 01:16:37 AM »
The noise floor of a 16-bit digital recording is WAY the frigging fark below the noise floor of the cassette recording; heck, a 16-bit recording has distinctly wider dynamic range than a 15 ips half-track Dolby "A" master tape (which I should know--having made many, many such recordings back in the day).

It depends on what you accept as "dynamic range." 16 bit maxes out at 90db of theoretical DR, but the last 30db or so are pretty nasty. So if you have a tape with 60db of DR, and you really want to faithfully reproduce the quietest parts of a recording, 24bit becomes not a luxury but a necessity.

Of course, popular music doesn't really utilize that much DR -- it's mastered so that you don't have to keep fiddling with the volume when you're in the car. It might only be an issue with classical or experimental stuff...

Offline szumsteg

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Re: Analog to 24bit question
« Reply #78 on: December 12, 2007, 01:19:59 AM »
First, thanks to all of you for the good arguments on both sides. Lots of great info here, and it seems you all have taken this as far as possible which is good discussion. I knew this could be a rough discussion, and you all brought up great points. What am I personally going to do, well I will still keep taking my analog cassettes in at 16bit mainly because part of preserving the tapes I do is that I do not eq them, edit them or enhance them which many of you agreed the best part of being 24bit is. I am simply capturing what is there, for better or worse, splitting it in CDWAV and making backup flacs and making CDs for myself. Most of what I am preserving as well is simply bass, elec keyboard, vocals, electric guitar and drums, no stringed instruments or natural sound, just big rock concerts coming out of large PA's. Another part brought up is that what is being argued as "better" becomes so incremental at the end, the master taper just by being in a better spot, or using a better tape, or whatever could make 100x more difference than some of the final benefits of being at 24bit. Funny thing is many of these tapes are from 1980s and from europe, where they all used normal bias cassettes, so I bet the freq range you were speaking of was for high bias casettes. Like someone said in one of the posts, it still boils down to good equipment, good cables, a nice high performance computer or everything at the end is not consequential.

When I am doing masters from here out, its at 24bit because we can do it with the equipment today, so I am not against 24bit at all. Just for purposes of getting old items archived, ill keep doing what I was doing. Its not as if people have had any complaints as of yet

Reminds me of just how many problems are solved with tapes by the three L's of taping. Location, Location, Location...nail that and you don't need to spend hours and hours post editing.
« Last Edit: December 12, 2007, 01:26:38 AM by szumsteg »

Offline DSatz

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Re: Analog to 24bit question
« Reply #79 on: December 13, 2007, 12:14:47 AM »
bensyverson wrote:

> It depends on what you accept as "dynamic range." 16 bit maxes out at 90db of theoretical DR, but the last 30db or so are pretty nasty. So if you have a tape with 60db of DR, and you really want to faithfully reproduce the quietest parts of a recording, 24bit becomes not a luxury but a necessity.

If the term "dynamic range" is too vague for you, we can use signal-to-noise ratio instead. 16-bit linear PCM has 93 - 94 dB of it (unweighted) depending on how you like your dither, while Dolby cassettes have maybe 65 dB on a good day, and I'm being pretty generous with that number; you should write and thank me. 60 dB would be more like it. You can't tape an LP onto a cassette and not hear any tape hiss during the silences; that should tell you something important about how much dynamic range a cassette doesn't have.

Meanwhile your remark about the "last 30 dB" is something that nearly everyone on this board could find out for him- or herself within ten minutes, which I think would be very nice for all concerned if they (and you) did.

--best regards
« Last Edit: December 13, 2007, 12:54:50 AM by DSatz »
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Offline Petrus

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Re: Analog to 24bit question
« Reply #80 on: December 13, 2007, 06:26:40 AM »
Using the digital photography analogy with digital audio is sometimes usefull. There is one danger, though: in photogaphy more resolution is better because we routinelly want to enlarge the pictures or zoom in into them. In audio the corresponding thing would be making the waves larger (wider) which correspods to SLOWING DOWN the audio. In normal life this is never done. In photography it is always possible to reach the limit of the resolution by routine enlargement, in audio there is a certain limit, 20 kHz, past which human audio resolution does not reach and for this reason those frequences are not needed in normal life.

In photography better true resolution is better, in audio after certain level more resolution does not improve the signal anymore. Resolution = frequency range in this context.

Unnormal life would be slowing down audio for effects etc. unnatural manipulations.

Offline DSatz

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Re: Analog to 24bit question
« Reply #81 on: December 13, 2007, 08:55:54 AM »
Analogies with photography seem unfortunately to be misleading when it comes to the specific things that we are talking about here. To see why that is, we would need to talk about how the statistical distribution of particle size and shape is controlled during film manufacture--it would get very specific, which would turn people off. Plus I don't actually know anything about film manufacture, so let me pay respect to the very wonderful field of photography by not making analogies with it.

But that's just the thing: Analogies make people comfortable drawing conclusions about "X" by applying what they know (or imagine to be true) about "Y," without ever having looked at how "X" actually works. Thus analogies can persuade people that they "understand" things which may not be factual! This is why demagogues lurve them so much. When you pursue an analogy, how do you know when you've exceeded the limits of its validity? You don't--unless you methodically test your beliefs against the real thing. And that takes time and effort, which are the very things people try to avoid. Instead you end up comparing one concept that you have to another concept that you have, and saying, "Yeah, that feels right."

That was the whole problem with medieval thinking, and why science was such a big breakthrough for our species. That's also a big problem with so many speculative discussions on the Internet--because all discussions about things that you're not actually doing are free from the constraint of reality. It is perfectly feasible on an Internet forum to discuss the theoretical basis of digital recording and to make it as clear as day in all essentials. Unfortunately it seems equally possible for people to make interesting, intelligent guesses which are plausible, sincere, self-consistent and very persuasive--but which lack just that one tiny element of being what happens in reality. And people literally don't know which is which. The true and the false assertions weigh the same, they smell the same, so people judge based on whatever they feel like believing.

Where digital audio is concerned, that pretty much already happened a long time ago for some people. I was an analog engineer for years while digital was still in the laboratory stages. Some of the earliest digital recording systems didn't use dither, and consequently had severe defects in their performance at low signal levels. The lower you went in level, the higher went the distortion because of fewer and fewer bits being used. It got uglier and uglier as you descended, and at the very bottom there was an absolute chasm--reverberation tails simply disappeared. As sounds went bye-bye over the cliff, you could hear this odd punctuated noise that was kind of like cheesecloth gently ripping and then fading out. This was usually called "granular noise" (although it's technically a form of distortion), and most people could identify it reliably after one hearing, if it was pointed out to them.

As a result of complaints over this problem (and due in large part to the work of two AES stalwarts, Stan Lipshitz and John Vanderkooy), within two to three years most audio manufacturers saw the light and started using dither appropriately. This subtracted a couple of dB from the spec sheet signal to noise ratio of the recording system, but it cured once and for all (notice that I didn't say "covered up" or "concealed"--it fundamentally cured) those problems with "digital deafness," granular noise and the ratty sound in the lower 30 (I would say 40) dB of the dynamic range. It became very clear that they had never been an inherent part of digital audio, but were an artifact of bad design in particular systems.

And yet you see people today, a quarter century later, arguing as if all that had never happened. They not only didn't learn anything--in many cases I would say that they consciously chose not to let the information in, for whatever reason. And while it's one thing to think whatever you want to think about whatever you want to think it about--that's just being human and ornery, and it can keep you alive--it's another thing to spread ignorance and to devise more and more clever ways of covering up the fact that it is ignorance.

I want to be persuasive, not combative, but sometimes I feel like telling people to shut up and do their homework before spreading further misinformation. On the other hand, at least misinformation in the audio field only bruises some bits and some ears, rather than, say, breaking up a marriage where a child is involved (which some friends of ours are going through right now) or starting a whole war on false pretenses (which my country did and is still doing). So that helps me to stay calm about the audio issues, though I have to say I'm pretty upset about the other stuff.

--best regards
« Last Edit: December 13, 2007, 09:02:19 AM by DSatz »
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Offline Gutbucket

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Re: Analog to 24bit question
« Reply #82 on: December 13, 2007, 10:29:46 AM »
^^^
Well that's a rational, informed and well articulated perspective.  At the same time I believe dmccabe when he says he hears a difference in the tape transfers (I haven't had a chance to find some MOTB recordings and listen myself, though I plan to), and those differences must be attributable to something.  Perhaps it's the fingerprint of the particular A/D circuitry that sounds different and not limitations of the file formats themselves.

...Meanwhile your remark about the "last 30 dB" is something that nearly everyone on this board could find out for him- or herself within ten minutes, which I think would be very nice for all concerned if they (and you) did...

How do you suggest we do this, crank up the volume and seriously listen to the quietest sections of our recordings? or is there some other method?

I'll bring this in from another thread since it pertains here, it's come up several times in various threads and we never really reached a satisfactory answer to this question:

I don't see the point of doing 24 bit on this recorder [a discussion on the Edirol R-09].  The specs of the chip inside say the max SNR is 92dB.  That means, the noise floor can be no lower than 16 bits (-96dB).

I verified this.  I put a 1k resistor load on each channel of a miniplug, plugged into line in, and recorded in 24 bit.  If this was a perfect recorder, there should be zero bits of noise.  In fact, the lower 9 bits were noise.  So, go ahead and record in 24 bit, but you'll just be recording 8 bits of extra noise.

...

Considering that dithering a signal works by adding a form of low level noise before truncating the bottom 8 bits (essentially allowing us to hear details beneath the noise floor that dissolve into the dither noise instead of into quantization artifacts), would the noise you measured in the bottom 9 bits of the R-09's 24bit recorded signal act as a form of dither allowing better than 16 bit performance?

Until this question is answered conclusively, I choose record in 24/48 on the R-09 because I'd rather err on the side of potential quality, recognizing that quality difference may be illusional, and also because the additional storage space required is not a big problem for me.

..but that's really a cost/benefit justification and rationalization I make to myself. I'd prefer a more scientific analysis of the matter, if only for my own education.  Realistically I'll probably go on recording 24/48 on the R-09 regardless until I have the opportunity to do some comparative testing and listening, because I love understanding why, but in the end I always trust my ears.
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Offline aegert

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Re: Analog to 24bit question
« Reply #83 on: December 13, 2007, 12:44:49 PM »
I am an electrical engineer. I do Digital signal processing... I could argue text book reasoning around all this including Nyquist Criteria... But!

We have taken cassettes played them through killer analog electronics and transfered them in parallel using bro a/d's to 16 and 24 bit....


The results are clear!

The 24 bit and more importantly the higher sample rate make them sound better. Sample rate is the best determination factor for your signal to noise ratio and your transient response Period... Its my ears that tell me that..


Here is a snap shot that shows it (SR that is):




Now I respect any opinions on this matter and if any one asks me and they do every day what resolution and SR to use I say the more the better!

Now if you are only ever going to release 16 bit cd style stuff then 24/88.2 is a clear choice but if tou want to make 24 bit releases that people will burn to lets say dvd-A go 24/96...

Now all that being said the argument of 16 bit dats that is not the discussion that was originally started and is a whole other thread..

I will sit back now and watch the sparks but for real we have done qualitative tests to prove this with our ears...

Use your own to see if we are right but remember your DAC is critical in the listening part. I use a benchmark DAC-1



« Last Edit: December 13, 2007, 12:54:45 PM by aegert »
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Offline Shawn

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Re: Analog to 24bit question
« Reply #84 on: December 13, 2007, 12:51:53 PM »
Sample rate is the best determination factor for your signal to noise ratio and your transient response Period...

I'm just trying to make sure I understand what you are saying.... so are you saying that signal to noise ratio is most directly affected by the sample rate?

Offline aegert

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Re: Analog to 24bit question
« Reply #85 on: December 13, 2007, 01:48:30 PM »
Sample rate is the best determination factor for your signal to noise ratio and your transient response Period...

I'm just trying to make sure I understand what you are saying.... so are you saying that signal to noise ratio is most directly affected by the sample rate?

In the dynamic range pertaining to cassette transfer. The ability to manage the transient response of analog signals, yes sample rate has the biggest effect....

To describe this with what I hear and for me all the tech stuff falls away the higher sample rates and bit depths for that matter are more open... If you want me to define in techincal terms open forget about it LOL

But you hear it it is not hiss but air int he recording that exsists when you listen to the cassettes or LP's or reels that you here that does not trasnlate to the cd as well.. In the higher res/rate formats it starts to come back.. But there is no perfect digital copy of the analog signals just aproximations... In the end what are you doing with this stuff...

Are the cassettes adn other analog media dying... Yes

Do I want to best preserve the master in as close to original sound as I can with digital... Yes

Do I want to edit these transfers for listening? Yes


Well then for me there is only one clear choice.

I have sat countless doubters down with the dac-1 and a pair of sennheiser 595's ultrasone 750's and proven the point...

Let your ears be the judge... Without a killer dac thought you will not hear it.. As well one 'audiophile' fought me tooth and nail on this and he was listening on his shitty $20 computer speakers... I told him you got to work with me here ROFL...

A
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Offline bensyverson

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Re: Analog to 24bit question
« Reply #86 on: December 13, 2007, 02:20:41 PM »
I'm just trying to make sure I understand what you are saying.... so are you saying that signal to noise ratio is most directly affected by the sample rate?

If that is what he's saying, it's misleading. Changing the sample rate alone can not change your actual S/N ratio. But by choosing higher sample rates, you're oversampling that noise. The more you oversample the signal, the lower the ADC noise will seem to be. Really, the ADC noise will be the same amplitude, but you'll just have to zoom in further to see it. It's the same with dithering. Sure, it does help, but it can not give you more than 16 levels between -66db and -72db. It can make those 16 levels sound smoother, but if you decide to boost the levels, that range will still not hold up very well. How big of an issue is it for cassettes that already have overwhelming tape noise at -66db? Like I said before, it all comes down to how well you want to reproduce that noise.  :P  If you don't care about preserving exactly the hiss from the tape, 16/44.1 is more than enough.

I do think the picture aegert posted is a little unfair. Yes, if you really want to totally faithfully reproduce that tiny electrical "pop" with the best fidelity, you need an insanely high sample rate. (Care to tell us the duration of the pop?) But most sounds are smoother waves, and are thus a lot easier to digitize. If you run the same test with a real sample from music or voice, it will not be so clear cut. And if you say you can hear the difference in transient response between 96k SR and 192k, so be it, but I highly doubt it's physically possible. Your ears are just not designed for that.

I also want to say this about my digital image (not photo) analogy: it's fine if you don't like it -- don't use it. But it is valid. I made it very clear in my text that image detail frequency = audio waveform frequency, so I don't think that's misleading. Nor do I think it's medieval demagogy to try to make these abstract concepts visual. I'm not asking anyone to draw any conclusions based on the images, I'm just using them as illustrations to help people get to the "a ha" moment when it all clicks together how SR and bit depth are related.

All of that said, the engineer in me agrees that "the higher the better." Higher sample rates and better bit depths can only help you. The question is, when do you reach the point of diminishing returns? That's a subjective call that you'll need your ears for, not an engineer. For digitizing a cassette tape "clean," without the intention of applying effects, I don't believe I could hear a "worthy" difference between 16/44.1 and 24/96. For a violin in a studio, it's obviously another story...
« Last Edit: December 13, 2007, 02:23:56 PM by bensyverson »

Offline Petrus

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Re: Analog to 24bit question
« Reply #87 on: December 13, 2007, 02:32:27 PM »
aegert, if I understod your graph correctly that "analog impulse" is 3 microseconds long, which means it represents a half wave of a 166666 Hz signal. What is the point of testing these systems with signals that are about 8 times outside the frequency range they are meant and designed to record or what we can hear? Besides all systems filter out all frequency content above the Nyquist frequency of the sampling rate before letting the signal to the A/D converter. This is so BASIC! For our normal 44.1 and 48 kHz everything above 21 kHz is filtered out, that kind of 166.666 kHz signal would never get past the low pass filters of even a 196 kHz sample rate recorder.

Must be from a broshure of a DSD recorder... For even that this is totally irrelevant.

Misinformation at worst.
--------------------------------

There is one analogy connecting dither to pre-exposure of printing paper (or even film) in the old days of film. It is possible to get about one half f-stop's worth of extra latitude to the highlights by exposing the printing paper to a weak even light before exposure proper. The pre-exposure must be so weak, that it itself does not have an effect on the paper, but combined with the weakest highlight signals (shadows in the neg, remember) cumulativelly causes a weak exposure. A neat trick, in digital audio we add weak noise to help the weakest signals to raise above the lowest digitizing level.
« Last Edit: December 13, 2007, 03:20:03 PM by Petrus »

Offline Petrus

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Re: Analog to 24bit question
« Reply #88 on: December 13, 2007, 03:30:20 PM »
I hope the comment above not directed at me... :-)

All of that said, the engineer in me agrees that "the higher the better." Higher sample rates and better bit depths can only help you. The question is, when do you reach the point of diminishing returns? That's a subjective call that you'll need your ears for, not an engineer.

There are no valid scientific tests that I know of (or any of my AES engineer friends know of) proving that people can even hear a difference between 16/44.1 and 24/96. I am talking about a controlled double blind test of real world audio, not one where you listen to test signals and/or know what you are listening to. People tend to hear what they expect or want to hear, not what they actually do hear.

I do know about tests where they did NOT hear a difference between original analog live signal and 16/44.1, between 16/44.1 and 24/96, and where 24/96 signal was low pass filtered at 20 kHz.

That makes me a sceptic...
« Last Edit: December 13, 2007, 03:33:55 PM by Petrus »

Offline boojum

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Re: Analog to 24bit question
« Reply #89 on: December 13, 2007, 03:52:11 PM »
I think it would be good for us to agree if we are chasing the real or the theoretical here.  A lot of the theory discussed is interesting,  sort of, but in the real world, what can we hear and what can we differentiate?  The Lavry paper, if I remember correctly, argues against insanely high sampling rates as unnecessary and wasted.  With those really high sampling rate and deep bit depths we are in the realm of the medieval argument of "how many angels can dance on the head of a pin?"  Well, some say an infinite number and some say none at all: the difference between the theoretical and the real.

For me and for all my practical purposes I will stick with 24/48; maybe a 24/88.2 sometime, but I doubt it.  I wonder how many of those who can hear a difference on analog xferred at 16/44.1 and 24/96 and 24/48 can hear those same differences in a double-blind test?  I like double-blind as it assures neutrality; and I like seeing others replicate it, just as in the scientific world of facts.

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Offline datbrad

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Re: Analog to 24bit question
« Reply #90 on: December 14, 2007, 11:01:35 AM »
I have been reading this thread and all the points being made have really made me think, which is really the point with honest discourse anyway. I backed away from the deep detail of the finer points of sampling theory and word length qualities and came up with what I think is good answer to the original posters real overall question.

A 16 bit recording with noise shaped dither sounds basically no different to a human ear than a straight 24 bit quantitized recording with the same gain level unless the dynamic range of the source exceeds about 114db. A 24 bit signal converted to 16 bit using a noise shaped dither such as UV22 will produce an effective dynamic range of 19 bits, since the quantitization noise is taken out of the 24 bit data before it is converted to 16 bit. A straight 16 bit master leaves the quantitization noise intact and is more audible than the noise left intact on a straight 24 bit recording. This is because the quantitization noise for 24 bit is below the analog noise floor of real world equipment, but not always so for straight 16 bit.

Here is where we get to the meat of the original poster's question. It's not about the final playback of the converted cassettes, it's about the recording of them to digital. Unless he is recording the cassettes using an A/D with real time noise shaped dither, like an Apogee, to master at 16 bit, he must use 24 bit to master and then apply the filter in the post conversion, using UV22 in Wavelab.

Then, it would up to him to retain and store the original 24 bit master file, or just the final noise shape dithered 16 bit file, since they would both sound basically identical with a cassette as the source.

Hope this helps!



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Re: Analog to 24bit question
« Reply #91 on: December 14, 2007, 02:49:49 PM »
For me and for all my practical purposes I will stick with 24/48; maybe a 24/88.2 sometime, but I doubt it. 

I have come to the same conclusion as far as 24 bit recording goes...that is unless I switch to DSD :turnevil: if/once it catches on more than now.

Offline F.O.Bean

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Re: Analog to 24bit question
« Reply #92 on: December 14, 2007, 07:59:35 PM »
Sorry Petrus, not directed at you.  I seem to live in a disabled world on this board where reply to doesn't work and quoting has to be done manually, so I skip it.

I can state what I can hear as well as what I can measure (D/A conversion is not required, just some analysis with a good SRC.  But a D/A/D analysis is not troublesome for a reasonable converter either).  I can hear the difference between 44.1 and 48; I expect that anyone with enough remaining high-frequency hearing* who knows what to listen for could hear it.  But it's rather subtle.

I can't hear the difference between 48 and any higher rate, but I can measure a difference in the audio spectrum between 48 and 64, but not any higher rate than that.  So Lavry's theory (also supported by other EEs who design converters; see Bruno Putzeys' board on PSW for examples) can be quickly shown to be measurable even without resorting to listening tests.

But the statement that higher rates (both sample and bit depth) can always only be good is false and unhelpful at a time when the thread had seemed to come near a conclusion.




* The other day for fun I tested my daughters' (7 and 10) hearing at 19.5kHz.  I'm sorry I didn't measure the volume accurately, but it would have been between 70 and 80dBSPL.  There were no lower-frequency distortion products anywhere near that SPL level.  I could not detect it at all (I frequently test my ears, and I'm good to 17kHz in one ear, and 12kHz in the other at 70dBSPL--ear infection damage and tinnitus in the one ear).  Both girls passed the test easily, without even hesitating when the tone started.

Perhaps I should give them the 48 vs 96 sample rate test . . . I can test up to about 22kHz accurately through my playback system . . .

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Offline Keyd

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Re: Analog to 24bit question
« Reply #93 on: December 15, 2007, 04:29:47 PM »
Interesting info in this thread.

Thanks folks.

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Re: Analog to 24bit question
« Reply #94 on: December 21, 2007, 03:35:44 PM »

You are probably not going to find any studios who get paid for transferring DAT masters -- what bands used 2 track DAT for studio work? -- maybe ADATs but..., thus very little online discussion about the best methods of editing ADATs.

I do :)   :P 

sorry to jump in late on this one.  seems like it got rather heated...  an interesting read nonetheless. 
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Offline DSatz

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Re: Analog to 24bit question
« Reply #95 on: December 30, 2007, 06:34:02 PM »
DATBRAD, the thing is, for a transfer from cassette, 16-bit with garden-variety neutral dither gives you way more than enough dynamic range. There's no need for UV 22, or 24-bit, or whatever. I mean for God's sake, 16-bit linear PCM with garden-variety neutral dither gives you more than enough dynamic range to handle 15 ips half-track Dolby "A" open reel master tapes already.

What is the problem here? Why do people imagine that you'd need to strap yourself into a frikkin' Atlas booster rocket just to cross the street in your own little neighborhood? It's a cassette, people. Take any that you have, with the widest dynamic range recorded on it, transfer it to 16-bit so that the peaks are at a nice, comfortable -3 dBFS, and the noise floor of the cassette will be 30+ dB above the noise floor of your transfer.

Just to make sure, I just now did what I described, and I'm seeing levels of around -60 dBFS on the blank part of the tape with Dolby "B" on and 70 microsecond playback EQ selected. I mean, 14 bits would be more than enough for that; 12 bits would be enough.

I do not, do not, do not get this whole "princess and the pea" attitude. (goes offstage muttering to self ...)

--best regards
« Last Edit: December 30, 2007, 10:55:37 PM by DSatz »
music > microphones > a recorder of some sort

Offline George2

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Re: Analog to 24bit question
« Reply #96 on: December 31, 2007, 12:48:18 AM »
I hope that puts an end to this drivel. Well said.
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