Reply
 
Thread Tools Display Modes
  #1   Report Post  
Ric Oliva
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Ok, so I understand that 44.1k is 44,100 samples per second and 48k is
48,000 samples per second. Obviously 48,000 is better. I'm not exactly
sure what bit rate is though? CDs are 16 bit, DVDs are 24. What exactly
does that mean though?

Another question - if I'm recording a project to audio CD, is it better to
just record at 16/44 since that's what the CD will be anyway, and I can save
system resources? or should I do 24/48 and then dither it down, essentially
changing what I originally heard? I read in the ProTools book by Berklee
Press that its best to record on LE using 24/44 since you won't hear much
difference between the 48k and 44.1k. Any insights into this?


---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.541 / Virus Database: 335 - Release Date: 11/14/2003


  #3   Report Post  
Denny F
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Ric Oliva" wrote in message
...
Ok, so I understand that 44.1k is 44,100 samples per second and 48k is
48,000 samples per second. Obviously 48,000 is better. I'm not exactly
sure what bit rate is though? CDs are 16 bit, DVDs are 24. What exactly
does that mean though?


The term bit "rate" does contribute to the confusion. I think bit "depth"
would be better.

The quick answer is sampling rate (rate makes sense here) is directly
related to frequency response. Bit depth is directly related to dynamic
range.

According to the theory, your sampling rate needs to be twice the highest
frequency you want to record. So theoretically, 44.1 gives you 22kHz
response, which is beyond human hearing. In practice, the actual top end
limit will be somewhat lower due to analog filtering required to keep the
clock noise out of the audio. But still, anything over 44.1 is probably
superfluous rather than "better."

Each sample has to reflect the amplitude of the signal at that sample. That
value is stored in a digital "word." We're talking about storing the value
in either a 16-bit or 24-bit word. The more bits, the better the
resolution, which in audio is refered to as "dynamic range."

Another question - if I'm recording a project to audio CD, is it better to
just record at 16/44 since that's what the CD will be anyway, and I can

save
system resources? or should I do 24/48 and then dither it down,

essentially
changing what I originally heard? I read in the ProTools book by Berklee
Press that its best to record on LE using 24/44 since you won't hear much
difference between the 48k and 44.1k. Any insights into this?


24/44.

While your finished product can sound just fine to the vast majority of ears
at 16-bit depth, 24 is still worthwhile for recording, applying effects
(transforms) and mastering. The reason is that the greater dynamic range of
the 24-bit depth manifests itself in a lower "noise floor." This extra
"room" at the bottom of your dynamic range is valuable because each time you
perform any kind of transform to your audio signal(s), you'll add a bit of
noise due to rounding errors. A greater bit depth makes these errors
smaller, and when you resample or dither your final, mastered recording to
16-bit, most of those rounding errors will hopefully live in those truncated
bits.

That's not to imply that you can't do a fair number of transforms on a
16-bit file without seriously degrading it. But there is at least a good
argument for using greater bit depths for recording/editing. Moreso than for
higher sampling rates, anyway.

--------------------------------------------------
Denny Fohringer
Itinerant guitarist
--------------------------------------------------
Lessons and music:
http://surf.to/dennyf
Bands:
http://bluepearlband.com http://doubletakeband.com
--------------------------------------------------


  #4   Report Post  
DJ
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

What he said.

;-)


"Denny F" wrote in message
...
"Ric Oliva" wrote in message
...
Ok, so I understand that 44.1k is 44,100 samples per second and 48k is
48,000 samples per second. Obviously 48,000 is better. I'm not exactly
sure what bit rate is though? CDs are 16 bit, DVDs are 24. What

exactly
does that mean though?


The term bit "rate" does contribute to the confusion. I think bit "depth"
would be better.

The quick answer is sampling rate (rate makes sense here) is directly
related to frequency response. Bit depth is directly related to dynamic
range.

According to the theory, your sampling rate needs to be twice the highest
frequency you want to record. So theoretically, 44.1 gives you 22kHz
response, which is beyond human hearing. In practice, the actual top end
limit will be somewhat lower due to analog filtering required to keep the
clock noise out of the audio. But still, anything over 44.1 is probably
superfluous rather than "better."

Each sample has to reflect the amplitude of the signal at that sample.

That
value is stored in a digital "word." We're talking about storing the value
in either a 16-bit or 24-bit word. The more bits, the better the
resolution, which in audio is refered to as "dynamic range."

Another question - if I'm recording a project to audio CD, is it better

to
just record at 16/44 since that's what the CD will be anyway, and I can

save
system resources? or should I do 24/48 and then dither it down,

essentially
changing what I originally heard? I read in the ProTools book by

Berklee
Press that its best to record on LE using 24/44 since you won't hear

much
difference between the 48k and 44.1k. Any insights into this?


24/44.

While your finished product can sound just fine to the vast majority of

ears
at 16-bit depth, 24 is still worthwhile for recording, applying effects
(transforms) and mastering. The reason is that the greater dynamic range

of
the 24-bit depth manifests itself in a lower "noise floor." This extra
"room" at the bottom of your dynamic range is valuable because each time

you
perform any kind of transform to your audio signal(s), you'll add a bit of
noise due to rounding errors. A greater bit depth makes these errors
smaller, and when you resample or dither your final, mastered recording to
16-bit, most of those rounding errors will hopefully live in those

truncated
bits.

That's not to imply that you can't do a fair number of transforms on a
16-bit file without seriously degrading it. But there is at least a good
argument for using greater bit depths for recording/editing. Moreso than

for
higher sampling rates, anyway.

--------------------------------------------------
Denny Fohringer
Itinerant guitarist
--------------------------------------------------
Lessons and music:
http://surf.to/dennyf
Bands:
http://bluepearlband.com http://doubletakeband.com
--------------------------------------------------




  #5   Report Post  
Arny Krueger
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Ric Oliva" wrote in message


Ok, so I understand that 44.1k is 44,100 samples per second and 48k is
48,000 samples per second. Obviously 48,000 is better.


Why is this obvious?

I'm not
exactly sure what bit rate is though? CDs are 16 bit, DVDs are 24.
What exactly does that mean though?


That's not bit rate. It's sample size.

Another question - if I'm recording a project to audio CD, is it
better to just record at 16/44 since that's what the CD will be
anyway, and I can save system resources?


If you are going to do very much processing, it is wise to record with 24
bit samples to preserve dynamic range as you process the tracks. After
you've mixed the channels you are going to distribute, dither them down to
16 bits.

or should I do 24/48 and then dither it down, essentially changing what I

originally heard?

Dithering down is a fast operation with most software. A proper job of
downsampling can involve quite a bit of processing time. If you're going to
throw away all audio 22.05 KHz in the end, why bother ever recording it?

I read in the ProTools book by Berklee Press that its best to record on
LE using 24/44 since you won't hear much difference between the 48k
and 44.1k. Any insights into this?


Yes, I just made a post entitled "Why 24/96 sampling isn't necessarily
better-sounding than 24/44 sampling" that addresses this question.

You can also investigate this issue yourself by downloading and listening to
files of the same musical sounds recorded in various sample formats, from
http://www.pcabx.com/technical/sample_rates/index.htm .






  #6   Report Post  
White Swan
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

If sample rate is viewed as resolution over time (horizontal axis),
bit rate is resolution of the amplitude (vertical axis). Each bit
doubles the resolution, or in other words, the smallest increment of
volume possible is hlaved with each additional bit.

The difference between 16 bits and 24 bits is 2 to eighth power, or
256. That means between each volume increment in a 16 bit recording
there are 256 intermediate steps added in a 24 bit recording. This
means that volume changes can be portrayed far more accurately and
smoothly. Also, when you manuipulate tracks with faders and plug-ins,
you are essentially doing mathematical operations, so with much higher
resolution the rounding errors are minimized. In practice, the result
is increased dynamic range, better stereo imaging, smoother less
grainy fades and reverb tails, and less worry about having to track
"hot". There is no reason not to track atr 24 bits if you can. The
only disadvantage is each sound file will be 150% bigger.

The difference between recording at 44.1 and 48k, on the other hand,
is pretty tiny. Many people (myself included) record at 44.1 so that
you don't have to worry about doing a sample rate conversion somewhere
down the line (to a 44.1 CD) which may do more harm than whatever tiny
gain you are getting from the slightly higher sample rate. If you are
concerned with using a higher sample rate, 88.2 seems to make more
sense. If you are going to do all your mixing on an analog board,
however, then you might as well use 48k.
  #7   Report Post  
Rick Powell
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Ric Oliva" wrote in message m...
Ok, so I understand that 44.1k is 44,100 samples per second and 48k is
48,000 samples per second. Obviously 48,000 is better. I'm not exactly
sure what bit rate is though? CDs are 16 bit, DVDs are 24. What exactly
does that mean though?

Another question - if I'm recording a project to audio CD, is it better to
just record at 16/44 since that's what the CD will be anyway, and I can save
system resources? or should I do 24/48 and then dither it down, essentially
changing what I originally heard? I read in the ProTools book by Berklee
Press that its best to record on LE using 24/44 since you won't hear much
difference between the 48k and 44.1k. Any insights into this?


---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.541 / Virus Database: 335 - Release Date: 11/14/2003


Basically speaking, the bit resolution determines the ability to
describe the amplitude of a signal. Having 24 bits available gives
you a safety cushion in digital recording, among other things. The
same input signal that you are slamming to 0 dbfs in 16 bit format
(not a good thing because of the possibility of "overs") can be
recorded in 24 bit format with the same or better resolution while
staying well below the red. In a MIX article years ago, Stephen St.
Croix stated that, sound improvement-wise, he'd rather have 17 bits
vs. 16 instead of 96 kHz sampling rate vs. 48 kHz, if he had to make a
choice.

Most pro's would rather work in the higher resolutions until the
absolute last bounce or mix to 16/44.1. This is partly because, with
digital processing (EQ, compression, etc.), the extra headroom yields
real sonic benefits when recording, editing, etc. There are some that
prefer to keep everything in 44.1 all the way through to avoid sample
rate conversion at the end, but there is almost universal use of
higher bit resolutions whenever possible.

RP
  #8   Report Post  
Jay - atldigi
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

In article , "Ric Oliva"
wrote:

Ok, so I understand that 44.1k is 44,100 samples per second and 48k is
48,000 samples per second. Obviously 48,000 is better. I'm not exactly
sure what bit rate is though? CDs are 16 bit, DVDs are 24. What exactly
does that mean though?


http://www.promastering.com/pages/techtalk.html

Article 1 and 2 deal with bit depth and dither, article 3 with sampling
rates. Recording at wordlegths higher than 16 bit is helpful. In
practice, 20 is almost always as good as 24 for recording since A to D
converters don't have the dynamic range to capture 24 bits and the lower
bits just contain the self noise of the box. For digital processing,
however, you want to use longer wordlengths like 48 bit, or at the very
least, 32 bit floating point.

Most simply stated, wordlength (or bit depth) is dynamic range. Bit rate
actually means something a little different, but we won't get into that
right now as you obviously are asking about bit depth. For every bit you
get about 6 dB (just over actually) of dynamic range. 16 bit CD has 96dB
while 24 bit has 144. Extra bits do not add headroom; they add footroom.
0 dB FS (full scale) represents the same value in both 16 bit and 24 bit
audio. The extra bits come into play at the bottom of the range. You are
able to record smaller events - sounds at a lower level.

In addition to dynamic range, it also means noise. in 16 bit, there is a
noise floor of -96dB while 24 bit has a noise floor of -144 dB. 24 bit
offers no additional accuracy in the top 96db of the dynamic range.
Actually, an 8 bit recording is just as accurate as a 24 bit recording
from 0dBFS to -48 dB. The -48 dB noise floor is quite obtrusive and the
8 bit recording certainly sounds worse, but those top 48 dB are just as
accurate as a 24 bit recording. If you took a 24 bit file and added 96
dB of noise, it would sound like an 8 bit file.

Invariably any discussion of bit depths must eventually include dither.
This, however, I'll leave to the tech talk articles I've pointed you to,
or to a google search for the many posts that have appeared here in
r.a.p. Be aware, however, that there are some common mistakes made
quite often when discussing these subjects, so avoid the myths.
Sometimes common sense tends to fail you until you understand how
digital audio truly works, so some things that seem to make intuitive
sense at first are actually technical rubbish.

--
Jay Frigoletto
Mastersuite
Los Angeles
promastering.com
  #9   Report Post  
Peter Gemmell
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Jay - atldigi wrote:
In article , "Ric Oliva"
wrote:


Ok, so I understand that 44.1k is 44,100 samples per second and 48k is
48,000 samples per second. Obviously 48,000 is better. I'm not exactly
sure what bit rate is though? CDs are 16 bit, DVDs are 24. What exactly
does that mean though?



http://www.promastering.com/pages/techtalk.html

Article 1 and 2 deal with bit depth and dither, article 3 with sampling
rates. Recording at wordlegths higher than 16 bit is helpful. In
practice, 20 is almost always as good as 24 for recording since A to D
converters don't have the dynamic range to capture 24 bits and the lower
bits just contain the self noise of the box. For digital processing,
however, you want to use longer wordlengths like 48 bit, or at the very
least, 32 bit floating point.

Most simply stated, wordlength (or bit depth) is dynamic range. Bit rate
actually means something a little different, but we won't get into that
right now as you obviously are asking about bit depth. For every bit you
get about 6 dB (just over actually) of dynamic range. 16 bit CD has 96dB
while 24 bit has 144. Extra bits do not add headroom; they add footroom.
0 dB FS (full scale) represents the same value in both 16 bit and 24 bit
audio. The extra bits come into play at the bottom of the range. You are
able to record smaller events - sounds at a lower level.

In addition to dynamic range, it also means noise. in 16 bit, there is a
noise floor of -96dB while 24 bit has a noise floor of -144 dB. 24 bit
offers no additional accuracy in the top 96db of the dynamic range.
Actually, an 8 bit recording is just as accurate as a 24 bit recording
from 0dBFS to -48 dB. The -48 dB noise floor is quite obtrusive and the
8 bit recording certainly sounds worse, but those top 48 dB are just as
accurate as a 24 bit recording. If you took a 24 bit file and added 96
dB of noise, it would sound like an 8 bit file.

Invariably any discussion of bit depths must eventually include dither.
This, however, I'll leave to the tech talk articles I've pointed you to,
or to a google search for the many posts that have appeared here in
r.a.p. Be aware, however, that there are some common mistakes made
quite often when discussing these subjects, so avoid the myths.
Sometimes common sense tends to fail you until you understand how
digital audio truly works, so some things that seem to make intuitive
sense at first are actually technical rubbish.


So, what answer is correct? Whiteswan, Rick Powell, and Jay have given
three answers that sound good but are mutually exclusive. I've been at
this a few years and I still don't know what is right. Does 24 bit give
greater resolution than 16 bit or does it merely give a larger dynamic
range without a finer resolution?

Peter

  #10   Report Post  
Bob Cain
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain



Peter Gemmell wrote:

So, what answer is correct? Whiteswan, Rick Powell, and Jay have given
three answers that sound good but are mutually exclusive. I've been at
this a few years and I still don't know what is right. Does 24 bit give
greater resolution than 16 bit or does it merely give a larger dynamic
range without a finer resolution?


The way that finer resolution manifests is as a larger
signal to noise ratio. The noise is due to quantization and
the wider the sample, the lower the noise is relative to the
maximum representable signal. The noise is an approximately
random error of +-1/2 the value of the low order bit. It is
inescapable. It is intimately related to the dynamic range
because it determines how small the signal can be before it
loses signifigance relative to that error noise limit. The
ratio of how large a signal that can be represented to how
small a signal can be represented is the dynamic range.

In practice, I don't think that yet any front end to a 24
bit ADC is itself nearly as quiet as that quantization noise
so that you will see specifications, if they are honest,
that are signifigantly lower than the theoretical 144 dB SNR
that can be achieved with 24 bits.


Bob
--

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

A. Einstein


  #11   Report Post  
Jay - atldigi
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

In article ,
wrote:

Peter Gemmell wrote:

So, what answer is correct? Whiteswan, Rick Powell, and Jay have given
three answers that sound good but are mutually exclusive. I've been at
this a few years and I still don't know what is right. Does 24 bit give
greater resolution than 16 bit or does it merely give a larger dynamic
range without a finer resolution?


The way that finer resolution manifests is as a larger
signal to noise ratio. The noise is due to quantization and
the wider the sample, the lower the noise is relative to the
maximum representable signal. The noise is an approximately
random error of +-1/2 the value of the low order bit. It is
inescapable. It is intimately related to the dynamic range
because it determines how small the signal can be before it
loses signifigance relative to that error noise limit. The
ratio of how large a signal that can be represented to how
small a signal can be represented is the dynamic range.

In practice, I don't think that yet any front end to a 24
bit ADC is itself nearly as quiet as that quantization noise
so that you will see specifications, if they are honest,
that are signifigantly lower than the theoretical 144 dB SNR
that can be achieved with 24 bits.


Bob has it. Read his post and my post and you'll see that they are not
mutually exclusive.

White Sawn's satement seems to indicate that the extra bits are within
the same dynamic range, thereby giving you greater detail within that
range. You can't into the trap of viewing digital audio like it's
digital imagery. Unfortunately, 24 bits leaves the top 96db range of 16
bit alone, but lowers the noise floor and allows the recording of audio
events that are even smaller, at a lower level, i.e. below -96dB.

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.

Some of these technical distinctions may not seem to matter much in
every day practice, but that's no reason to be uninformed. The lower
noise floor and extra dynamic range can really make a difference. Some
say it's "merely" dynamic range or "just" lower noise, like those things
are somehow unimportant and could hardly make a difference. It must be
something more esoteric that makes it sound better! Well, it's not.
Those things can be very important and it often does sound better. The
little details that were once buried below the noise floor may now be
audible, whether minute audio events or perhaps subtle overtones, and
the noise floor of the recording may be below your ability to hear it
anymore. That's not minced meat!

--
Jay Frigoletto
Mastersuite
Los Angeles
promastering.com
  #12   Report Post  
Arny Krueger
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Jay - atldigi" wrote in message


White Sawn's satement seems to indicate that the extra bits are within
the same dynamic range, thereby giving you greater detail within that
range. You can't into the trap of viewing digital audio like it's
digital imagery. Unfortunately, 24 bits leaves the top 96db range of
16 bit alone, but lowers the noise floor and allows the recording of
audio events that are even smaller, at a lower level, i.e. below
-96dB.


24 bits puts 16 extra levels between each pair of levels that exist with 16
bits. Thus, the resolution is increased at any level, not just the smallest
one.

The reduction of the noise floor due to 24 bits is a consequence of the
extra resolution 24 bit coding provides between any of the two levels in a
16 bit representation. The two go together hand-in-hand because the coding
is linear.

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.

In an exactly linear system, whether digital or analog, the noise floor and
resolution are exactly the same. In a nominally linear (i.e., real-world)
system, whether digital or analog, the noise floor and resolution are
nominally the same.




  #13   Report Post  
Mike Rivers
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain


In article writes:

Ok, so I understand that 44.1k is 44,100 samples per second and 48k is
48,000 samples per second. Obviously 48,000 is better.


Obviously, but do you understand why, and why it might not be better
sometimes?

I'm not exactly
sure what bit rate is though? CDs are 16 bit, DVDs are 24. What exactly
does that mean though?


That's actually the length of the digital word that represents the
voltage of each sample. The more bits, the greater the resolution, and
the greater the potential for accuracy. Of course the actual accuracy
is a function of how good the analog-to-digital and digital-to-analog
converters that use that digital word is.

Another question - if I'm recording a project to audio CD, is it better to
just record at 16/44 since that's what the CD will be anyway, and I can save
system resources?


You'll save some disk space. As to whether it's better, it depends on
what you're going to do between making the recording and making the
CD.

I read in the ProTools book by Berklee
Press that its best to record on LE using 24/44 since you won't hear much
difference between the 48k and 44.1k. Any insights into this?


I believe that most people won't. Every mathematical process that you
avoid (like changing the sample rate) means there's less chance to
change the sound. But as a beginner, your auditory perception isn't
likely to be acute enough to make these judgements yourself by
listening, so you might as well take someone's word for it.


--
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
and reach me he double-m-eleven-double-zero at yahoo
  #14   Report Post  
Tommi
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain


"Arny Krueger" wrote in message
...
"Jay - atldigi" wrote in message

White Sawn's satement seems to indicate that the extra bits are within
the same dynamic range, thereby giving you greater detail within that
range. You can't into the trap of viewing digital audio like it's
digital imagery. Unfortunately, 24 bits leaves the top 96db range of
16 bit alone, but lowers the noise floor and allows the recording of
audio events that are even smaller, at a lower level, i.e. below
-96dB.



24 bits puts 16 extra levels between each pair of levels that exist with

16
bits. Thus, the resolution is increased at any level, not just the

smallest
one.

The reduction of the noise floor due to 24 bits is a consequence of the
extra resolution 24 bit coding provides between any of the two levels in a
16 bit representation. The two go together hand-in-hand because the coding
is linear.

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?


  #15   Report Post  
Arny Krueger
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Tommi" wrote in message

"Arny Krueger" wrote in message
...
"Jay - atldigi" wrote in message

White Sawn's satement seems to indicate that the extra bits are
within the same dynamic range, thereby giving you greater detail
within that range. You can't into the trap of viewing digital audio
like it's digital imagery. Unfortunately, 24 bits leaves the top
96db range of 16 bit alone, but lowers the noise floor and allows
the recording of audio events that are even smaller, at a lower
level, i.e. below
-96dB.



24 bits puts 16 extra levels between each pair of levels that exist
with 16 bits. Thus, the resolution is increased at any level, not
just the smallest one.

The reduction of the noise floor due to 24 bits is a consequence of
the extra resolution 24 bit coding provides between any of the two
levels in a 16 bit representation. The two go together hand-in-hand
because the coding is linear.

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.




  #16   Report Post  
Mike Rivers
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain


In article writes:

So, what answer is correct? Whiteswan, Rick Powell, and Jay have given
three answers that sound good but are mutually exclusive. I've been at
this a few years and I still don't know what is right. Does 24 bit give
greater resolution than 16 bit or does it merely give a larger dynamic
range without a finer resolution?


All of the above. Dynamic range and resolution are inseparable, even
in the analog world. In the digital world it's easier to see both.

With more bits, the actual value recorded for each sample is
potentially more accurate, hence better resolution. The greater the
resolution, the more likely that you'll be able to tell that two
adjacent samples along a time-varying waveform actually have different
values. I say "potentially" since as has been pointed out already,
once you get much beyond 20 bits, the amplitude of the noise in the
input or output parts of the system (the A/D or D/A converters) is
greater than one bit's worth of change, so the last couple of bits
really don't contribute anything to accuracy. They only allow you to
record the low level system noise with reasonable accuracy. So that's
your limit to dynamic range.

The reason why it's important to allow for longer word lengths in
signal processing (including mixing) is that those are purely
mathematical, and theoretically noise-free processes. Since you aren't
adding noise, you can take advantage of the resolution of the longer
word length so that when you ultimately shorten it to accommodate the
output circuitry or final delivery medium, all the numbers to the
resolution of that final word length will be accurate.



--
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
and reach me he double-m-eleven-double-zero at yahoo
  #17   Report Post  
Carey Carlan
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"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.
  #18   Report Post  
White Swan
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Arny and I obviously agree about the resolution issue. I believe Arny
is cvorrect, but I make no claim to infallibility, so I'm very
interested in the opposing points of view as well.

Having no tech expertise, I'm going more on intuitve logic. Let's use
reductio ad absurdum (or whatever it's called!). Let's take a 1 bit
system. Now we have only two volume values: full volume, and full
silence. using the 6dB per bit formula, we have values of 0dB and 6
dB. (I'm not using full scale dB, obviously, for this purpose). Now
let's assume a 2 bit system. Now we have inserted two intermediate
steps between the on and off. And we've also increased the dynamic
range by 6 dB. This would give us values of 0db, 4 dB, 8 dB, and 12
dB. (For the sake of argument, let's assume a linear system in terms
of dB values. For all I know, it might not be.)

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.

Now, my logic may well be flawed, so I'm most interested in finding
out where the flaw is. This is a great way to learn, and i thank
everyone who is teaching me!
  #20   Report Post  
Arny Krueger
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Arny Krueger" wrote in message

"Jay - atldigi" wrote in message


White Sawn's satement seems to indicate that the extra bits are
within the same dynamic range, thereby giving you greater detail
within that range. You can't into the trap of viewing digital audio
like it's digital imagery. Unfortunately, 24 bits leaves the top
96db range of 16 bit alone, but lowers the noise floor and allows
the recording of audio events that are even smaller, at a lower
level, i.e. below
-96dB.


24 bits puts 16 extra levels between each pair of levels that exist
with 16 bits. Thus, the resolution is increased at any level, not
just the smallest one.


Correction: 24 bits puts 256 (!!) additional levels between every pair of
levels that exist with 16 bits.




  #21   Report Post  
Arny Krueger
 
Posts: n/a
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.


  #23   Report Post  
Mike Rivers
 
Posts: n/a
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
and reach me he double-m-eleven-double-zero at yahoo
  #25   Report Post  
Jay - atldigi
 
Posts: n/a
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   Report Post  
Tommi
 
Posts: n/a
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   Report Post  
Jay - atldigi
 
Posts: n/a
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   Report Post  
Tommi
 
Posts: n/a
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   Report Post  
Rick Powell
 
Posts: n/a
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   Report Post  
Bob Cain
 
Posts: n/a
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


  #32   Report Post  
Justin Ulysses Morse
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Ric Oliva wrote:

Ok, so I understand that 44.1k is 44,100 samples per second and 48k is
48,000 samples per second. Obviously 48,000 is better. I'm not exactly
sure what bit rate is though? CDs are 16 bit, DVDs are 24. What exactly
does that mean though?

Another question - if I'm recording a project to audio CD, is it better to
just record at 16/44 since that's what the CD will be anyway, and I can save
system resources? or should I do 24/48 and then dither it down, essentially
changing what I originally heard? I read in the ProTools book by Berklee
Press that its best to record on LE using 24/44 since you won't hear much
difference between the 48k and 44.1k. Any insights into this?



The minimum necessary sample rate and bit depth is determined by the
sounds you want to reproduce.

The dynamic range (basically signal-to-noise ratio) of your material
determines the minimum necessary bit depth. In practice, you will
never record a source with a dynamic range greater than can be
represented in 16 bits.

The bandwidth (frequency range) determines the necessary sample rate.
The sample rate is required only to be more than double the highest
frequency you want to reproduce. In practice, almost nobody owns
reproduction equipment that is useful beyond 20kHz.

So in theory you can record at 44.1/16 and your digital audio will hold
all of the audio data necessary to reproduce anything you can put into
and pull out of any equipment.

The need for greater data rates comes when you plan to manipulate the
recording one or more times between the record and reproduce moments.
If you are going to re-sample your data (through sample rate conversion
or a D/A-A/D process), then you might benefit from an increased sample
rate. More importantly, if you plan to manipulate the VALUES of those
existing samples (by DSP processes such as gain changes, EQ, or
anything else), then you might benefit from an increased bit depth.

Both of these considerations serve to push the limitations of the
quantization processes beyond our ability to detect them. The idea is
that stacking these processes can compound their inherent errors and
eventually make them audible. Even this precaution represents a
judicious level of overkill in almost all cases, but data storage is
getting cheap so it doesn't hurt.

In summary: If you're recording live to 2-track and won't be doing any
processing at all, then 44.1/16 is more than adequate. If you will be
doing any processing, then start with 24 bits. If you will be doing
any resampling, then start with 96k. Whatever rate you choose for your
initial recording, you should maintain that rate until the final stage
of processing.
  #33   Report Post  
Arny Krueger
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Jay - atldigi" wrote in message

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.


Scott was clearly addressing the "massive gain" part of the comment, not the
"8 extra bits per sample". That there are 8 extra bits per sample is an
inarguable fact that we can all agree on.


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.


It means that, but it also means that a 24 bit word describes 255 additional
levels between every pair of levels that can be described by a 16 bit word.
It means both things. Again, this is inarguable and readily observable in
the real world.

The 16 bit number is already describing 96 dB of dynamic range
just fine.


Psychoacoustically 16 bits does do a fine job, however technically the 24
bit word codes 255 additional levels between each pair of levels described
by a 16 bit word.

If you want to carry the precision further and capture
signals that are lower, say to -144 dB, then 24 bits is your ticket.


24 bits also adds resolution in any region between -144 dB and full scale.

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.


That an accrete 24 bit representation of a signal has more resolution at any
level than a 16 bit representation of signal is readily observable in the
real world as soon as you have converters that are sufficiently accurate,
which we now have quite commonly.

Look at how this works with DC levels. In the following examples some
numbers may be off by 1 which is obviously practically irrelevant.

If you have a 16 bit converter with a 1 volt range, there are 65,535 levels
that can be uniquely described between 0 and 1 volt. 1.0000 volts is
represented by 65,535. 0.0000 volts is represented by 0. 0.50000 volts is
represented as 32,767 and there are 32,767 unique levels between 0 and
0.5000 volts and 32,767 more unique levels between 0.5000 volts and 1.000
volts. The smallest voltage 0 volts that can be coded is
3.0518509475997192297128208258309e-5 volts.

If you have a 24 bit converter with a 1 volt range, there are 16,776,960
levels that can be uniquely described between 0 and 1 volt. 1.0000 volts is
represented by 16,776,960. 0.0000 volts is represented by 0. 0.50000 volts
is represented as 8,388,480 and there are 8,388,480 unique levels between 0
and 0.5000 volts and 8388480 more unique levels between 0.5000 volts and
1.000 volts. The smallest voltage 0 volts that can be coded is
5.9605554283970397497520408941787e-8 volts.

Thus the 24 bit representation of voltages between 0 and 1 volt has both
more dynamic range and also more resolution than the 16 bit representation
of the same voltages. The same concept relates to audio signals.





  #34   Report Post  
Arny Krueger
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Tommi" wrote in message


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.


For the purpose of discussion, I'll stipulate that your facts are correct to
this point. I really don't know that, but it would help me make an important
point if we don't argue over that part of your comments.

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.


The myth here is that signals in a digital system can have interchannel
timing differences that are only integer numbers of sample periods. IOW this
myth as applied to 44,100 Hz sampling is that interchannel timing
differences can only be multiples of 22.675736961451247165532879818594
microseconds. I agree that this seems to be intuitively clear. But it is
also quite wrong.

The myth comes from the idea that two signals in different channels that are
displaced in time can only be expressed as the same set of sample values,
but time-shifted. This is not the case. Two signals in different channels
that are displaced in time can be expressed as different sample values.

For example, if two slowly-increasing (ramp) signals are displaced in time,
one signal might have a set of sample values that starts out 0, 10, 20,
30... This is a ramp that starts at t = 0. The time-delayed version of this
signal in another channel could have a set of values that is 0 at t = 0, but
is 5, 15, 25... for successive samples. If you looked at these two signals
over time, you'd say that the second signal is time-shifted by an amount of
time equal to half a sample period. And, that is how it would sound.

The correct time resolution of sampled signals is the sample period divided
by the number of distinct amplitude levels. In the case of 16/44 this would
be 5.1418904674492623958124444033093e-10 seconds or
514.18904674492623958124444033093 picoseconds. This is a tiny, tiny number.
In reality, it is lost in the noise.


  #35   Report Post  
Arny Krueger
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Bob Cain" wrote in message


Within the Nyquist criterion a signal can be produced with
any arbitrary phase or delay until you consider the
quantization of the samples.


Agreed.

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.)


Agreed. I just gave examples of this in another post.

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.


I estimated the minimum time delay that is quantizable in 16/44 in the other
post and found that it was
514.18904674492623958124444033093 picoseconds. Not only is this a very small
amount of time in the audio domain, but in reality it is lost in the noise,
due to Shannon's theories. Shannon and Nyquist really had their acts
together back in the late 1920s and early 1930s.





  #36   Report Post  
Arny Krueger
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Mike Rivers" wrote in message
news:znr1069010813k@trad
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.


This is clearly audible in a series of samples, originally recorded by Ethan
Winer, that people can download from
http://www.pcavtech.com/test_data His
article is good background reading and can be found at
http://www.ethanwiner.com/BitsTest.html .

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.


Sad but true. Only I'd raise the bar to about 10 bits. People can listen to
the files at http://www.pcavtech.com/test_data and reach their own
conclusions.



  #40   Report Post  
Jay - atldigi
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

In article , "Arny Krueger"
wrote:

That there are 8 extra bits per sample is an
inarguable fact that we can all agree on.


That we can agree on, but how audio behaves because of them we don't.

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.


It means that, but it also means that a 24 bit word describes 255
additional
levels between every pair of levels that can be described by a 16 bit
word.
It means both things. Again, this is inarguable and readily observable in
the real world.

The 16 bit number is already describing 96 dB of dynamic range
just fine.


Psychoacoustically 16 bits does do a fine job, however technically the 24
bit word codes 255 additional levels between each pair of levels
described by a 16 bit word.


Looking at the system as a whole, dither and all, you gain no advantage
in the top 96dB. You get and extra 48dB below them.

--
Jay Frigoletto
Mastersuite
Los Angeles
promastering.com
Reply
Thread Tools
Display Modes

Posting Rules

Smilies are On
[IMG] code is On
HTML code is Off


Similar Threads
Thread Thread Starter Forum Replies Last Post
Explain me this Schizoid Man Audio Opinions 6 April 11th 04 12:43 AM
TS/TRS balanced/unbalanced can someone explain TheKeith General 6 March 4th 04 07:56 PM
Can you explain this 50Hz hum?? me Pro Audio 18 October 28th 03 09:46 PM
Reverb & EQ and "damping" etc .. please explain .. Daniel Pro Audio 3 October 13th 03 09:09 AM


All times are GMT +1. The time now is 10:47 AM.

Powered by: vBulletin
Copyright ©2000 - 2024, Jelsoft Enterprises Ltd.
Copyright ©2004-2024 AudioBanter.com.
The comments are property of their posters.
 

About Us

"It's about Audio and hi-fi"