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Author Topic: What sample rate are you using?  (Read 4057 times)

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Offline noahbickart

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Re: What sample rate are you using?
« Reply #45 on: June 13, 2018, 09:16:55 AM »
Dsatz is one of the main reasons that Ts.com remains the best place on the internet.

Thanks for your continued presence here.
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Offline aaronji

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Re: What sample rate are you using?
« Reply #46 on: June 13, 2018, 10:21:06 AM »
--aaronji wrote:

> A major advantage, as I understand it, is that the extra bits greatly reduce the noise introduced from quantization error during analog to digital conversion (each extra bit approximately halves the noise).  So even if the recording peaks considerably below full-scale, you can amplify it without making the quantization noise audible...

Aaronji, that just isn't true as a rule. It all depends on the noise in the signal that's being quantized versus the converter's own noise floor. The quantization noise of the converter only needs to be a few dB lower than the noise in the incoming analog signal, however many bits that takes. Any further bits beyond that will (or should!) be random; therefore they don't contribute to the precision or accuracy of the recording in any way. The notion that computation (post-processing) will somehow work out more accurately with more bits is a total fallacy when those bits contain only noise.

I have to admit (and with all due respect) that I am not quite sure what you mean.  My comment had nothing to do with post-processing accuracy.  What I was saying is that, as I understand it, the major (only?) advantage of the extra bits is to increase the channel's dynamic range by reducing quantization noise.  That makes it easier to fit the dynamic range of the signal, even with very low peak levels, into that space with (to use Gutbucket's term) sufficient footroom.  So when amplified, the signal's noise will dominate the converter's noise (rendering it inaudible).  Am I missing something here?  There are even some examples available (such as on the Sound Device's website) that appear to show this and it also seems to be in line with both your post and Gutbucket's.

Granted, the real world, live music recordings scenarios in which this will actually help are likely rare, but stuff happens due to accident or circumstance...

Offline H₂O

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Re: What sample rate are you using?
« Reply #47 on: June 13, 2018, 10:50:19 AM »
1bit 2.8224Mhz for 2 channel
24bit 96Khz for 4-8 channel
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Offline rigpimp

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Re: What sample rate are you using?
« Reply #48 on: June 13, 2018, 11:26:08 AM »
Dsatz is one of the main reasons that Ts.com remains the best place on the internet.

Thanks for your continued presence here.

This^^.  If I am perusing and I see a post of his I take off my coat and read his post thoroughly.  I do the same with a few others like Morst and Gutbucket, for example.  The historical and technical contributions these folks make is so valuable. 

With that said, I am still going to run at 24/96.  :-p
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Offline wforwumbo

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Re: What sample rate are you using?
« Reply #49 on: June 13, 2018, 02:53:17 PM »
I agree with much of what’s said here, particularly coming from DSatz (whose praises I want to echo throughout). BUT... I DO want to iron out something here, and I’ve brought it up before. The “1 bit == 6 dB of dynamic range” statement.

That is a theoretical rule that mostly follows how bits work at lower bit depths. A 24-bit word length will always contain 24 bits of information. It’s not that the extra bits are wasted; they are still there, and they are accessible in memory and computation. Where we run into limitations is in the design of ICs (mostly in amplifier design and topology) and getting THOSE low-noise enough. That is a pretty complex and technical challenge.

Something else worth noting... the average dynamic range of a young, healthy human hearing system is about 110 dB of dynamic range. So as long as you are peaking below 0 dB-FS, the purpose of having more bits and a wider dynamic range is to make it such that your ears - not the technical limitations of your system - is the bottleneck for getting things to sound “as good as you can” so to speak.

And again, another advantage of larger word lengths, is reduced filter error. The more data you are pumping through a system doing truncated (discrete) calculations, the more accurate the end product will be. It’s why 32-bit floating point operations are used internally in DAWs, and we aren’t stuck to using just 16- or 24-bit systems. Post work does improve significantly in my (and many other studio pros’) experience and to my/our ears.

Again, this isn’t to say that practically most of what’s said here is right, and it’s not to say that you can get away with 16/44.1 without issue; in many cases, it will do the job just fine. Just want to clarify, from a technical perspective.
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Offline goodcooker

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Re: What sample rate are you using?
« Reply #50 on: June 13, 2018, 03:30:41 PM »
the average dynamic range of a young, healthy human hearing system

I hope you realize that a lot of us here are old, worn out, half deaf and barely human... :guitarist: :zombie02:
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Offline Gutbucket

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Re: What sample rate are you using?
« Reply #51 on: June 13, 2018, 07:35:36 PM »
I'm happy wforwumbo has joined TS.  As far as I can tell by way of a few personal conversations he is sharp, extremely well versed in matthmatics, not old, only partially deafened thus far (yet retains a very keen ear), and is fully human.  Keep an eye open for his future contributions on the board.

I do want to bring up something which came to mind while reading his post above, because it is something I find commonly misunderstood at TS and could be easily misconstrued from what he wrote.  I regularly see statements in these kinds of threads which say something to this effect- "I record 24 bit files because I plan to do post work on the recordings, and 24 bit files are better for that", but that doesn't actually correlate with what he wrote above. 

Performing mathematics with sufficient precision such that rounding/truncation errors are avoided makes for more accurate calculations, and that can translate to better sounding audio.  But the bit-length of the data being manipulated doesn't dictate the bit-length of the calculation space.  The calculation space within the DAW is larger (32-bit floating point) so calculation precision is preserved.. until one outputs it again.  As DSatz describes above-

Quote
..as long as the channel has sufficient dynamic range and the levels are set optimally--where the peaks of the program material come as close as possible to the channel's maximum level without actually reaching it--any additional extension of the channel's dynamic range can't audibly improve things, because the signal itself is the limiting factor at that point, not the channel.

The raw 16 or 24 bit recorded file is a channel containing the signal.  Mathematics are performed on that signal in the DAW in a 32-bit floating point space regardless of the signal's bit-length.  We then export the data from the DAW via another file (channel) of sufficient bit-length to contain the manipulated signal.  Because we have full control over signal levels when manipulating and exporting that signal from the DAW (unlike when we were originally making the raw recording) we can make sure the signal fits comfortably within an output channel of an optimum size, so we can actually output a 16 bit file for almost everything without compromising the signal. We can make sure that "the signal itself is the limiting factor at that point, not the channel".  Or we can output 24 bit files with some extra unrelated noise at the bottom, which is easy, and space is cheap.  But in most cases a 16 bit output file can be of equivalent quality because the full dynamic range of almost all music can be fully represented by less than 16 bits.

The portions quoted below should be understood as being relevant within the DAW (or possibly within the recorder prior to a file being written in some cases), and not as specifying what file length is necessary in file storage formats-

A 24-bit word length will always contain 24 bits of information. It’s not that the extra bits are wasted; they are still there, and they are accessible in memory and computation.

Quote
The more data you are pumping through a system doing truncated (discrete) calculations, the more accurate the end product will be. It’s why 32-bit floating point operations are used internally in DAWs, and we aren’t stuck to using just 16- or 24-bit systems. Post work does improve significantly in my (and many other studio pros’) experience and to my/our ears.



tldr- As long as the full dynamics of the raw recording fits within 16 bits, recording at 24 bits will not make "post processing more accurate or sound better", but doing the post processing in a larger bit-length calculation space can do so. 

Sample rate conversion is a different animal.
« Last Edit: June 13, 2018, 07:39:35 PM by Gutbucket »
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Offline checht

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Re: What sample rate are you using?
« Reply #52 on: June 13, 2018, 08:35:11 PM »
....
tldr- As long as the full dynamics of the raw recording fits within 16 bits, recording at 24 bits will not make "post processing more accurate or sound better", but doing the post processing in a larger bit-length calculation space can do so. 

For math-challenged folks, what's the tldr of the tldr?

I record trying to get close to but not exceed 0db. Will my recordings be better in 16 or 24 bit?

Thanks for simplifying way way down.
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Offline wforwumbo

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Re: What sample rate are you using?
« Reply #53 on: June 13, 2018, 11:46:50 PM »
....
tldr- As long as the full dynamics of the raw recording fits within 16 bits, recording at 24 bits will not make "post processing more accurate or sound better", but doing the post processing in a larger bit-length calculation space can do so. 

For math-challenged folks, what's the tldr of the tldr?

I record trying to get close to but not exceed 0db. Will my recordings be better in 16 or 24 bit?

Thanks for simplifying way way down.

Try this post I wrote:

http://taperssection.com/index.php?topic=184569.msg2251598#msg2251598

Still on the technical side, but a simple explanation of the overall concept.
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Offline checht

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Re: What sample rate are you using?
« Reply #54 on: June 14, 2018, 12:52:41 AM »
^ That clarifies a lot.

Thanks!
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Offline Gutbucket

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Re: What sample rate are you using?
« Reply #55 on: June 14, 2018, 10:50:43 AM »
Quote
I record trying to get close to but not exceed 0db. Will my recordings be better in 16 or 24 bit?

In practical terms it likely depends on what recorder you are using, the ADC chip and specific implementation of it in the machine.  If you don't have measurements, don't care to do critical listening tests, don't want to worry about it too much, and don't care about storage size, my take is that it makes sense to default to 24 bit recording in this day and age, if simply as a belt and suspenders approach more than a definitely better kind of thing. This is a practical take rather than an engineering answer.


Practicality rules.  In the end, I suspect the two largest practical determinants on this is what era one started recording in, and how many TB of raw recordings one has amassed and needs to deal with! Storage space may be growing ever cheaper, but managing and backing up large catalogs is a PITA.
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Offline DSatz

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Re: What sample rate are you using?
« Reply #56 on: June 14, 2018, 11:40:34 AM »
aaronji, howdy. As for why I mentioned post-processing in my reply to you, please look again at the earlier message of yours that I replied to. It quotes a message from ycoop as follows: "What about the extra 8 bits allows for more work in post without losing as much quality?"--to which you replied that essentially, more bits in a/d conversion = lower quantization error. To which, in turn, I replied that no, that's not a valid general statement; it all depends on the noise in the signal being converted.

If that noise isn't correlated with the desired signal, and is entirely above the noise floor of the converter, then more bits don't (can't!) improve the accuracy of the conversion. Hypothetically and simplistically, if the noise floor of a signal is at a 13-bit level, you could convert that signal with a 14-bit, 16-bit, 20-bit or 24-bit converter all of which would give equal audible results (all other things being equal), because the noisy signal was the limiting factor, not the converter.

However, if the noise IS correlated with the desired signal, then it is a form of non-linear distortion and you've got yourself a defective converter. Most discussion assumes tacitly that the residual noise of a converter will be uncorrelated with the signal, because for decades now, that has normally been the case. That is another way of saying that the non-linear distortion is very low.

Still, the older ones among us, or anyone who's ever used an inadequately dithered A/D converter, will know what that kind of distortion sounds like, because some of the earliest available digital audio recording equipment didn't dither properly (ahem Sony, including their professional PCM-1600 which a lot of early CDs were recorded with). Some people describe it as "noise breathing" or "noise pumping" because it rises and falls with the signal levels, particularly on very soft sounds such as reverberation decay (or imagine taking a 400 Hz tone and fading it down slowly to nothingness with a smooth, analog fader). An accompanying phenomenon is (or was) "digital deafness"--signals below the scope of the lowest-order bit are simply blanked out.

Unfortunately, the way a lot of people think about digital audio is still based on the incredible flood of bullshit that was unleashed when digital audio was introduced around 1980, and which (when it was about anything real) harped greatly on shortcomings of conversion that were really due to not having proper dither. When an a/d converter is properly dithered, there's no increase in distortion at lower levels, no "stair-step levels" to be minimized by striving for more and more bits, nor is there "digital deafness" at the bottom of the range; it acts just like an analog channel as far as dynamic range is concerned. Since actual distortion in modern a/d converters is so low, I tend to shift the discussion away from the concept of "precision" in conversion (which would be a more appropriate concept if a signal were utterly noiseless, such as a computationally generated test signal) to dynamic range, which is much more pertinent to live, recorded audio.

--best regards
« Last Edit: June 14, 2018, 11:53:32 AM by DSatz »
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Offline wforwumbo

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Re: What sample rate are you using?
« Reply #57 on: June 14, 2018, 02:30:36 PM »
aaronji, howdy. As for why I mentioned post-processing in my reply to you, please look again at the earlier message of yours that I replied to. It quotes a message from ycoop as follows: "What about the extra 8 bits allows for more work in post without losing as much quality?"--to which you replied that essentially, more bits in a/d conversion = lower quantization error. To which, in turn, I replied that no, that's not a valid general statement; it all depends on the noise in the signal being converted.

If that noise isn't correlated with the desired signal, and is entirely above the noise floor of the converter, then more bits don't (can't!) improve the accuracy of the conversion. Hypothetically and simplistically, if the noise floor of a signal is at a 13-bit level, you could convert that signal with a 14-bit, 16-bit, 20-bit or 24-bit converter all of which would give equal audible results (all other things being equal), because the noisy signal was the limiting factor, not the converter.

However, if the noise IS correlated with the desired signal, then it is a form of non-linear distortion and you've got yourself a defective converter. Most discussion assumes tacitly that the residual noise of a converter will be uncorrelated with the signal, because for decades now, that has normally been the case. That is another way of saying that the non-linear distortion is very low.

Still, the older ones among us, or anyone who's ever used an inadequately dithered A/D converter, will know what that kind of distortion sounds like, because some of the earliest available digital audio recording equipment didn't dither properly (ahem Sony, including their professional PCM-1600 which a lot of early CDs were recorded with). Some people describe it as "noise breathing" or "noise pumping" because it rises and falls with the signal levels, particularly on very soft sounds such as reverberation decay (or imagine taking a 400 Hz tone and fading it down slowly to nothingness with a smooth, analog fader). An accompanying phenomenon is (or was) "digital deafness"--signals below the scope of the lowest-order bit are simply blanked out.

Unfortunately, the way a lot of people think about digital audio is still based on the incredible flood of bullshit that was unleashed when digital audio was introduced around 1980, and which (when it was about anything real) harped greatly on shortcomings of conversion that were really due to not having proper dither. When an a/d converter is properly dithered, there's no increase in distortion at lower levels, no "stair-step levels" to be minimized by striving for more and more bits, nor is there "digital deafness" at the bottom of the range; it acts just like an analog channel as far as dynamic range is concerned. Since actual distortion in modern a/d converters is so low, I tend to shift the discussion away from the concept of "precision" in conversion (which would be a more appropriate concept if a signal were utterly noiseless, such as a computationally generated test signal) to dynamic range, which is much more pertinent to live, recorded audio.

--best regards

Well-stated, eloquently and accurately. I 100% agree with every aspect of this post.

And while the digital “crunch” you mention from poor converters is considered charming or nostalgic by some, it definitely isn’t pleasant to listen to. I’ve heard my fair share of bad converters and understand why digital gets such a bad reputation. 

DSatz I hope I’m blessed enough to tape with you at some point and learn a thing or two from your experience.
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Offline nak700s

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Re: What sample rate are you using?
« Reply #58 on: June 14, 2018, 02:44:47 PM »
Wow, a big thank you to DSatz, and as always Gutbucket as well as others which made this stuff so much easier to understand.  Luckily, I've been doing the right thing, more or less, possibly, as pointed out above, because I'm an old timer that started out with analogue. Reading these specs put everything in perspective and helped me to understand why I do what I do, and that I may even be able to improve my craft a little to boot.  Can't wait to experiment...Thank you!
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Offline robeti

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Re: What sample rate are you using?
« Reply #59 on: June 14, 2018, 04:01:17 PM »
I tape, archive and listen at 24/48

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