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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain



 
 
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  #21  
Old November 16th 03, 07:15 PM
Arny Krueger
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Carey Carlan" > wrote in message
. 205
> "Tommi" > wrote in
> :
>
>> So, if you're recording, say, someone's vocals at both 16 and 24
>> bits, and the peaks are at -6dB to 0dB FS, does the 24 bit recording
>> represent more accurately the signal in that region than the 16-bit
>> version?

>
> The extra 8 bits give you 48 db more dynamic range between EVERY
> sample. Between sample value = 0 and sample value = 1 they give you
> an extra 48 db on the bottom end.
>
> On the loud end, 16 bit max value is 32767 (0x7FFF), second value is
> 32766 (0x7FFE). That equates to 24 bit values 8388352 (0x7FFF00) and
> 8388096 (0x7FFE00), a difference of 256 values, the equivalent of 48
> dB dynamic range.


Agreed. I figured that out a few hours after I posted, but I was nowhere
near a computer with internet access. I then started wondering who would be
the first to catch my error.


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  #23  
Old November 16th 03, 09:48 PM
Mike Rivers
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain


In article > writes:

> So what has happened? Yes, we have increased out dynamic range by 6 dB
> between the loudest and softest signals the system can represent. But
> we have increased the resolution throughout the system: from a 6dB
> increment to a 4 dB increment.
>
> Continuing to a 3 bit system: dynamic range is 18 dB. Values are 0dB,
> 2.25 dB, 4.5 dB, etc. to 18 dB. Wth each additional bit the dynamic
> range is increasing, but ALSO the resolution is increasing everywhere
> in the system.


So your next question should be: "What do I HEAR that's different?"

A good way to answer that question is to listen to some very low bit
rate recordings. I know this sounds like blasphemy, but once you get
up to about 8 bits, you don't get the sense that you're increasing
resolution, you get the sense that you're reducing the background
noise level into which the signal disappears. So yes, your ears are
able to RESOLVE a lower level signal in the presence of noise because
the resolution down there is better.

However, on the practical side, since most of the music we listen to
today has a dynamic range of less than 10 dB and is played back well
above the system and ambient noise floor, you don't get much of a
chance to take advantage of the added resolution. Of course it doesn't
hurt to have it there (for the occasions where you actually can use
it) but I'll bet you could sell 8-bit pop music CDs today and nobody
would complain about the sound quality.




--
I'm really Mike Rivers - )
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
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  #25  
Old November 16th 03, 10:57 PM
Jay - atldigi
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

In article >, "Arny Krueger"
> wrote:

> "Tommi" > wrote in message
>
> > "Arny Krueger" > wrote in message


> >> The idea that adding bits does not increase resolution is yet another
> >> popular urban myth about digital. It's similar to the urban myth
> >> that analog has resolution below the noise floor.

>
> > So, if you're recording, say, someone's vocals at both 16 and 24
> > bits, and the peaks are at -6dB to 0dB FS, does the 24 bit recording
> > represent more accurately the signal in that region than the 16-bit
> > version?

>
> The 24 bit recording has the capability to represent the signal much more
> accurately in *any* range from zero to max, than the 16 bit recording.


I think you're suffering the myth, Arny. Let me quote from another
thread where Scott Dorsey is trying to explain the same thing that I am,
and I'll and try to explain it yet another way:


In article >, (Scott
Dorsey) wrote:

> > A 16 bit number is significantly
> > smaller and therefore less precise than a 24 bit number.

>
> Right.
>
> >So, in a nutshell. Moving from 16 bit to 24 bit, we have 8 extra bits
> >per sample to represent the analog wave which is a massive gain.

>
> Not really. It gives you more dynamic range, which is often wasted
> anyway. 96 dB is an awful lot.




The 24 bit number is more precise than the 16 bit. True enough. What
that means in audio, however, is that the 24 bit word can describe
smaller values than the 16 bit word, thus signals that are lower in
level. The 16 bit number is already describing 96 dB of dynamic range
just fine. If you want to carry the precision further and capture
signals that are lower, say to -144 dB, then 24 bits is your ticket.

The myth is the dynamic equivalent to the argument that 4 samples on a
20kHz sine wave will render it more accurately than 2, and 8 samples
even more so. That's not true either.

--
Jay Frigoletto
Mastersuite
Los Angeles
promastering.com
  #26  
Old November 16th 03, 11:40 PM
Tommi
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain


"Jay - atldigi" > wrote in message
...

> The 24 bit number is more precise than the 16 bit. True enough. What
> that means in audio, however, is that the 24 bit word can describe
> smaller values than the 16 bit word, thus signals that are lower in
> level. The 16 bit number is already describing 96 dB of dynamic range
> just fine. If you want to carry the precision further and capture
> signals that are lower, say to -144 dB, then 24 bits is your ticket.
>
> The myth is the dynamic equivalent to the argument that 4 samples on a
> 20kHz sine wave will render it more accurately than 2, and 8 samples
> even more so. That's not true either.



I may well be suffering the myth, but my understanding is that it matters
whether you sample a sine wave 2 or 8 times. Tests have been made where
subjects had to determine which sound came first from their headphones. The
same signal was fed to both L and R channels, only the other one was delayed
by 5-15 _micro_seconds.
Some of the people were able to "localize" the sound source even when it was
delayed only by 5 microseconds. This implies that a sampling rate of
192kHz(which results in 5.2 microsecond's sample intervals), for example, is
not only pushing the nyquist rate to the ultrasonic range, but also presents
better channel separation on multichannel systems.
So, it doesn't necessarily matter if you sample a sine wave 2 or 8 times on
a mono system, but on a multichannel system higher sample rates result in
better localization.



  #27  
Old November 17th 03, 12:10 AM
Jay - atldigi
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

In article >, "Tommi"
> wrote:


> I may well be suffering the myth, but my understanding is that it matters
> whether you sample a sine wave 2 or 8 times. Tests have been made where
> subjects had to determine which sound came first from their headphones.
> The same signal was fed to both L and R channels, only the other one was
> delayed by 5-15 _micro_seconds.
> Some of the people were able to "localize" the sound source even when it
> was delayed only by 5 microseconds. This implies that a sampling rate of
> 192kHz(which results in 5.2 microsecond's sample intervals), for example,
> is not only pushing the nyquist rate to the ultrasonic range, but also
> presents better channel separation on multichannel systems.
> So, it doesn't necessarily matter if you sample a sine wave 2 or 8 times
> on a mono system, but on a multichannel system higher sample rates result in
> better localization.



You have to take it one step at a time and separate the issues. 2
samples is enough to reconstruct the wave plain and simple. Bob Stuart
and Tom Holman have talked about the possibility of better time axis
resolution as it pertains to differences between two or more channels,
not to be confused with time axis resolution meaning a more detailed
representation of the waveform, and we're not just talking about single
sine waves here. These are two different issues.

It may well be that imaging improves with higher smaple rates, unless of
course you dither properly at the lower ones. It's a little known fact
that dither can also have an effect in the time domain. Other things
including filter issues certainly can make higher sample rates sound
better. However, this has nothing to do with the waveform being
reproduced more accurately within the bandwidth of the system (i.e below
Nyquist for the particular sample rate).

--
Jay Frigoletto
Mastersuite
Los Angeles
promastering.com
  #28  
Old November 17th 03, 12:47 AM
Tommi
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain


"Jay - atldigi" > wrote in message
...

> You have to take it one step at a time and separate the issues. 2
> samples is enough to reconstruct the wave plain and simple. Bob Stuart
> and Tom Holman have talked about the possibility of better time axis
> resolution as it pertains to differences between two or more channels,
> not to be confused with time axis resolution meaning a more detailed
> representation of the waveform, and we're not just talking about single
> sine waves here. These are two different issues.
>
> It may well be that imaging improves with higher smaple rates, unless of
> course you dither properly at the lower ones. It's a little known fact
> that dither can also have an effect in the time domain. Other things
> including filter issues certainly can make higher sample rates sound
> better. However, this has nothing to do with the waveform being
> reproduced more accurately within the bandwidth of the system (i.e below
> Nyquist for the particular sample rate).


I absolutely agree, multichannel imaging is a different matter and that
wasn't the topic here.





  #29  
Old November 17th 03, 02:05 AM
Rick Powell
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Jay - atldigi > wrote in message >...

> Rick understands that bit depth relates to amplitude and that DSP is
> better with longer wordlengths. A small clarification is in order,
> however. He seems to consider there to be extra headroom while
> technically there is not, unless you change the zero reference. In other
> words, increase the voltage that zero is referenced to. Nevermind
> working at -10 or +4, you'll be using a new, nonstandard reference
> voltage, and what about the analog electronics that probably can't
> handle that voltage? You're asking for trouble for that reason and
> several others (increasing the noise floor of the analog gear,
> compatability, and more). Unless you want to do that, you really are
> gaining what should be thought of as "footroom" more than headroom.
>
> In practice some feel that you need to push a digital recording right up
> to 0dB FS to "use all the bits". This really isn't as big an issue as
> some would have you believe, as long as you use good gain stageing and
> reasonable recording levels, especially with todays converters which
> perform far better than much or the early crappy digital stuff. It
> doesn't hurt to assumne that 24 bits gives you a little room to play
> with, but unless you are recording a program with greater than average
> dynamic range in a very quiet environment with excellent equipment and
> minimal processing, you really aren't going to be able to take advantage
> of those extra bits. Then again, they certainly don't hurt, and they
> could help, so there's no reason not to. Still, it helps to understand
> technically what's going on and when extra effort will pay off and when
> if won't.


Jay, I'm not suggesting changing the zero reference. Correct the
following if I'm wrong, but as a mastering engineer, you would rather
take in a 2-track digital mix that peaked at -2db than one that peaks
at 0 dbfs and has a few "flat tops". Using a 24 bit format to record
or mix down to allows less artifacts towards the noise floor, given 2
"identical" sources (one recorded at 16 bit and one at 24 bit) peaking
at, say, -2db. And reduces the need (perceived or real) to "slam" the
recording all the way to 0 dbfs to take "full advantage" of the bit
depth.

If this is "footroom' instead of "headroom", isn't it still a margin
nonetheless?

RP
  #30  
Old November 17th 03, 03:39 AM
Bob Cain
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain



Tommi wrote:
>
> "Jay - atldigi" > wrote in message
> ...
>
> > You have to take it one step at a time and separate the issues. 2
> > samples is enough to reconstruct the wave plain and simple. Bob Stuart
> > and Tom Holman have talked about the possibility of better time axis
> > resolution as it pertains to differences between two or more channels,
> > not to be confused with time axis resolution meaning a more detailed
> > representation of the waveform, and we're not just talking about single
> > sine waves here. These are two different issues.
> >
> > It may well be that imaging improves with higher smaple rates, unless of
> > course you dither properly at the lower ones. It's a little known fact
> > that dither can also have an effect in the time domain. Other things
> > including filter issues certainly can make higher sample rates sound
> > better. However, this has nothing to do with the waveform being
> > reproduced more accurately within the bandwidth of the system (i.e below
> > Nyquist for the particular sample rate).

>
> I absolutely agree, multichannel imaging is a different matter and that
> wasn't the topic here.


Within the Nyquist criterion a signal can be produced with
any arbitary phase or delay until you consider the
quantization of the samples. Then the achievable delays
become quantized as well and wider sample widths will have a
positive effect on the delay/phase resolution (which
controls the imaging resolution.) I don't know for sure but
I rather doubt that the ear is sensitive to the resolution
constraint imposed by even 16 bit samples.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein
 




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