[Chaz Cotton]

Hi


Is there a recommended order to process a digital audio signal using
Audacity or similar so that optimum results are obtained? Eg, is it
better to remove hiss, clicks, etc before Normalizing? I have a
number of old cassettes I need to copy to CDs and I want to get the
best results possible.

thanks

Chaz Cotton

[Scott Dorsey]

Chaz Cotton wrote:

Is there a recommended order to process a digital audio signal using
Audacity or similar so that optimum results are obtained? Eg, is it
better to remove hiss, clicks, etc before Normalizing? I have a
number of old cassettes I need to copy to CDs and I want to get the
best results possible.

Spend 90% of your time getting the azimuth on the tape playback right
and getting the Dolby levels so the damn thing pumps as little as possible.

Any digital processing is gravy. I would not trust the noise removal on
Audacity, but your goal is to deal with that before it ever goes into the
computer.
--scott


--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[John Williamson]

On 17/07/2012 16:35, Chaz Cotton wrote:
Hi


Is there a recommended order to process a digital audio signal using
Audacity or similar so that optimum results are obtained? Eg, is it
better to remove hiss, clicks, etc before Normalizing? I have a
number of old cassettes I need to copy to CDs and I want to get the
best results possible.

I'd say remove clicks, then noise, and use multiple passes to partially
remove them each time until you're happy with the sound rather than one
heavy handed pass. Then normalise.

Just be aware that no matter how you do it, you will lose some of the
musicality of the performance, which you may find more annoying than the
noise.

A quick and dirty method to clean up cassettes and tapes is to use a
free Winamp plugin called Tape Restorer Live!, currently at version
1.20. It has settings to allow for incorrectly adjusted Dolby B & C,
presets for removal of 19KHz pilot tone and hum from recordings and even
lets you time align the channels on tapes where the head gaps aren't in
the right place. Use the WAV writer output plugin to record the output
to HD.

http://www.winamp.com/plugin/tape-restore-live/154246

--
Tciao for Now!

John.

[Richard Webb[_3_]]

On Tue 2012-Jul-17 12:04, Scott Dorsey (1:3634/1000) wrote to All:

Chaz Cotton wrote:

Is there a recommended order to process a digital audio signal using
Audacity or similar so that optimum results are obtained? Eg, is it

Spend 90% of your time getting the azimuth on the tape playback
right and getting the Dolby levels so the damn thing pumps as little
as possible.

Right, but if he's asking about normalize and asking the
questions it's probably a given that he wouldn't know how to do the things you suggest.


Any digital processing is gravy. I would not trust the noise
removal on Audacity, but your goal is to deal with that before it
ever goes into the computer.

For this though he's going to need a good oscilloscope and
some other hardware.


Regards,
Richard
.... Remote audio in the southland: See www.gatasound.com
--
| Remove .my.foot for email
| via Waldo's Place USA Fidonet-Internet Gateway Site
| Standard disclaimer: The views of this user are strictly his own.

[Trevor]

"Chaz Cotton" wrote in message
...
Is there a recommended order to process a digital audio signal using
Audacity or similar so that optimum results are obtained? Eg, is it
better to remove hiss, clicks, etc before Normalizing? I have a
number of old cassettes I need to copy to CDs and I want to get the
best results possible.

The order is pretty well unimportant when copying cassette to CD even if you
used a Nakamichi Dragon for both the record and playback source. And if you
are using anything less, then you have FAR more problems to worry about, and
the digital processing order is NOT one of them these days.

Trevor.

[PStamler]

On Jul 17, 6:18*pm,
(Richard Webb) wrote:
On Tue 2012-Jul-17 12:04, Scott Dorsey (1:3634/1000) wrote to All:

Chaz Cotton wrote:
Is there a recommended order to process a digital audio signal using
Audacity or similar so that optimum results are obtained? Eg, is it
Spend 90% of your time getting the azimuth on the tape playback
right and getting the Dolby levels so the damn thing pumps as little
as possible.

Right, but if he's asking about normalize and asking the
questions it's probably a given that he wouldn't know how to do the things you suggest.

Any digital processing is gravy. *I would not trust the noise
removal on Audacity, but your goal is to deal with that before it
ever goes into the computer.

For this though he's going to need a good oscilloscope and
some other hardware.

A scope doesn't really help align azimuth unless the tape has high-
frequency test tones recorded on it. For aligning azimuth on program,
there's nothing better than a pair of ears listening for maximum
treble (and a monitoring system that can be switched to mono).

Peace,
Paul

[Scott Dorsey]

PStamler wrote:
(Richard Webb) wrote:

For this though he's going to need a good oscilloscope and
some other hardware.

A scope doesn't really help align azimuth unless the tape has high-
frequency test tones recorded on it. For aligning azimuth on program,
there's nothing better than a pair of ears listening for maximum
treble (and a monitoring system that can be switched to mono).

This is true, although once you have peaked for maximum treble it's good to
be able to make fine adjustments listening to how centered the stereo image
is.

And you're still going to need to scope to align the machine back to the
standard tape after you've futzed with it.
--scott


--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Chaz Cotton]

On Tue, 17 Jul 2012 17:06:24 +0100, John Williamson
wrote:

On 17/07/2012 16:35, Chaz Cotton wrote:
Hi


Is there a recommended order to process a digital audio signal using
Audacity or similar so that optimum results are obtained? Eg, is it
better to remove hiss, clicks, etc before Normalizing? I have a
number of old cassettes I need to copy to CDs and I want to get the
best results possible.

I'd say remove clicks, then noise, and use multiple passes to partially
remove them each time until you're happy with the sound rather than one
heavy handed pass. Then normalise.

Just be aware that no matter how you do it, you will lose some of the
musicality of the performance, which you may find more annoying than the
noise.

A quick and dirty method to clean up cassettes and tapes is to use a
free Winamp plugin called Tape Restorer Live!, currently at version
1.20. It has settings to allow for incorrectly adjusted Dolby B & C,
presets for removal of 19KHz pilot tone and hum from recordings and even
lets you time align the channels on tapes where the head gaps aren't in
the right place. Use the WAV writer output plugin to record the output
to HD.

http://www.winamp.com/plugin/tape-restore-live/154246


Thanks for this.

[Sean Conolly]

"Chaz Cotton" wrote in message
...
Hi


Is there a recommended order to process a digital audio signal using
Audacity or similar so that optimum results are obtained? Eg, is it
better to remove hiss, clicks, etc before Normalizing? I have a
number of old cassettes I need to copy to CDs and I want to get the
best results possible.

You shouldn't have clicks for a commercial cassette. Maybe if it was copied
from an LP.

You should do your final normalization (to set the level on CD) after you've
removed any clicks or applied any EQ, since these steps can alter the peak
levels in the recording.

FWIW, the order I normally do is:
* Capture (analog to digital), being fairly conservative about peak levels.
Saved as 24 bit or 32 bit files.
* Normalize to -3 dbfs for convenience when I'm working in the wave editor
* Apply whatever restorative (de-noise) or additive (EQ to taste) processing
* Normalize & hard limit (if needed) to the final levels I want for CD. I
hard limit sparingly, just enough to bring up the average level if there's a
few peaks that stand out.
* Convert to 16 bit for burning to CD

I'm usually putting a mix of songs on my CD's for listening, so I will do
some adjustments to the volume and EQ to roughly match the other songs in
the set.

Hope this helps,
Sean

[Dave Plowman (News)]

In article ,
Sean Conolly wrote:
* Normalize & hard limit (if needed) to the final levels I want for CD.
I hard limit sparingly, just enough to bring up the average level if
there's a few peaks that stand out.

Why on earth would you want to do that?

--
*Most people have more than the average number of legs*

Dave Plowman London SW
To e-mail, change noise into sound.

[Richard Webb[_3_]]

Paul writes:
Right, but if he's asking about normalize and asking the
questions it's probably a given that he wouldn't know how to do the
things you suggest.
For this though he's going to need a good oscilloscope and
some other hardware.

A scope doesn't really help align azimuth unless the tape has high-
frequency test tones recorded on it. For aligning azimuth on program,
there's nothing better than a pair of ears listening for maximum
treble (and a monitoring system that can be switched to mono).

Yep, was assuming the tape with the proper test tones of
course. Ears work if you know how to use 'em for most
purposes grin. Still, doing this setup before playing the tapes into the computer, whether with properly done test
tones or setting up the deck by ear will go a long way
toward eliminating the need for further band aid fixes.



Regards,
Richard
--
| Remove .my.foot for email
| via Waldo's Place USA Fidonet-Internet Gateway Site
| Standard disclaimer: The views of this user are strictly his own.

[hank alrich]

Sean Conolly wrote:

* Normalize & hard limit (if needed) to the final levels I want for CD. I
hard limit sparingly, just enough to bring up the average level if there's a
few peaks that stand out.
* Convert to 16 bit for burning to CD

Why in the world would you hard limit when moving from a medium with
maybe 50 or 60 dB of dynamic range to one of 96 dB of dynamic range?

Significant peaks probably didn't survive recording to cassette in the
first place.

--
shut up and play your guitar * http://hankalrich.com/
http://www.youtube.com/walkinaymusic
http://www.sonicbids.com/HankandShaidri

[Trevor]

"Richard Webb" wrote in
message ...
Paul writes:
Right, but if he's asking about normalize and asking the
questions it's probably a given that he wouldn't know how to do the
things you suggest.
For this though he's going to need a good oscilloscope and
some other hardware.

A scope doesn't really help align azimuth unless the tape has high-
frequency test tones recorded on it. For aligning azimuth on program,
there's nothing better than a pair of ears listening for maximum
treble (and a monitoring system that can be switched to mono).

Yep, was assuming the tape with the proper test tones of
course. Ears work if you know how to use 'em for most
purposes grin. Still, doing this setup before playing the tapes into
the computer, whether with properly done test
tones or setting up the deck by ear will go a long way
toward eliminating the need for further band aid fixes.


The point was to match the heads to the *particular* tape you are copying
for best response. Adjusting them to a test tape is what you do before you
record, and why would anybody want to do that on a cassette deck these days?
(unless of course you are trying to copy the test tape to CD, which would be
even more pointless!)
*Only* if the tape was recorded on the same machine with the heads adjusted
to a test tape before the recording you are copying will there be any point
in doing that again now, and for most cassettes decks, not even then!

Trevor.

[Jim Lesurf[_3_]]

In article , Sean Conolly
wrote:
"Chaz Cotton" wrote in message
...


* Normalize & hard limit (if needed) to the final levels I
want for CD. I hard limit sparingly, just enough to bring up the
average level if there's a few peaks that stand out.

* Convert to 16 bit for burning to CD

Like others, I'd suggest *avoiding* normalizing or limiting to anything
close to 0dBFS. if you normalise to within a dB or so of 0dBFS you'd need
to check any CD player you use can then handle intersample excursions
produced by the DAC/reconstruction filter that rise above 0dBFS.

And for the same reason I'd certainly be cautious about downsampling to 16
bit *after* normalizing to near 0dBFS. Again, that might lead to problems.

Unless you are *really* pushed for covering ultra-wide range material I'd
not 'normalise' to a level where any peak sample went above about -2 to
-3dBFS.

I appreciate that would not suit some in the 'biz' though, because of their
obsession with LOUDNESS and the assumption that the listener is incapable
of adjusting the volume.

Slainte,

Jim

--
Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm
Armstrong Audio http://www.audiomisc.co.uk/Armstrong/armstrong.html
Audio Misc http://www.audiomisc.co.uk/index.html

[Luxey]

�реда, 18. јул 2012. 08.12.56 UTC+2, PStamler је напи�ао/ла:

there's nothing better than a pair of ears listening for maximum
treble (and a monitoring system that can be switched to mono).

What is maximum treble? Where Hi mids and highs are strongest, or where the highest frequencies are best heard, although lower in level?

[Meindert Sprang]

"Jim Lesurf" wrote in message
...
Like others, I'd suggest *avoiding* normalizing or limiting to anything
close to 0dBFS. if you normalise to within a dB or so of 0dBFS you'd need
to check any CD player you use can then handle intersample excursions
produced by the DAC/reconstruction filter that rise above 0dBFS.

Is that *really* an existing problem ur an urban myth? The Red Book
specifies 16 bit data. Why would they do that if a player would not be able
to reconstruct full 16 bit data to an analog signal? to me this would really
be a severe design flaw in a CD player! If I were the designer of a DAC and
I would expect that an interpolation between two samples would rise above
the maximum dynamic range, I'd add the necessary bits to take care of that
or scale the input of that process down. In my DSP work I always checked the
input range and output range of calculations to see if it would fit in the
'width' of calculations and scale accordingly. This is standard design
practice.

Meindert

[Jim Lesurf[_3_]]

In article , Meindert
Sprang wrote:
"Jim Lesurf" wrote in message
...
Like others, I'd suggest *avoiding* normalizing or limiting to
anything close to 0dBFS. if you normalise to within a dB or so of
0dBFS you'd need to check any CD player you use can then handle
intersample excursions produced by the DAC/reconstruction filter that
rise above 0dBFS.

Is that *really* an existing problem ur an urban myth?

I can't say for all players. But when I was investigating this area for Hi
Fi News a few years ago one of the people there I was discussing it with
did some checks on some then-current players that he had. The measured
results showed that some of them did alter/distort waveform excursions
above 0dBFS due to intersample peaks.


The Red Book specifies 16 bit data. Why would they do that if a player
would not be able to reconstruct full 16 bit data to an analog signal?

The snag here is the usual one.

In theory, theory and practice agree. But in practice they may not... at
least some of the time.

to me this would really be a severe design flaw in a CD player! If I
were the designer of a DAC and I would expect that an interpolation
between two samples would rise above the maximum dynamic range, I'd add
the necessary bits to take care of that or scale the input of that
process down. In my DSP work I always checked the input range and output
range of calculations to see if it would fit in the 'width' of
calculations and scale accordingly. This is standard design practice.

For me, also. But any finite state system with finite value representations
will have a limit, And the evidence is that some designers haven't catered
for this.

For a CD player you should only need one more bit as the max possible
overshoot is of the order of 3dB so far as I was able to determine. Not a
very demanding requirement, but still needs to be implimented into the
design.

So a *good* designer making a *good* machine will allow for intersample
peaks. But are all designers and machines "good"?...

And given how clipped some pop/rock CDs are, how much difference would it
make given what has been done to the music before the player reads it from
the disc?!

I'll see if I can find the data and I'll ask the person who gave it to me
if he minds it being made public. At the time it was just sent to me as
part of our discussions about the topic.

One of the related discussions I've had with others is the speculation that
'NOS' DACs and players may be liked by some people because they avoid this
by having no digital values generated in between input samples, and can
have a following analogue filter for reconstruction that can cope with the
peaks.

Slainte,

Jim

--
Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm
Armstrong Audio http://www.audiomisc.co.uk/Armstrong/armstrong.html
Audio Misc http://www.audiomisc.co.uk/index.html

[Scott Dorsey]

In article ,
Luxey wrote:
=D1=81=D1=80=D0=B5=D0=B4=D0=B0, 18. =D1=98=D1=83=D0=BB 2012. 08.12.56 UTC+2=
, PStamler =D1=98=D0=B5 =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=D0=B0=D0=BE/=D0=BB=
=D0=B0:

there's nothing better than a pair of ears listening for maximum
treble (and a monitoring system that can be switched to mono).

What is maximum treble? Where Hi mids and highs are strongest, or where the=
highest frequencies are best heard, although lower in level?

You'll know it when you hear it. You're basically moving a comb filter
forward and back.... you want to be listening for the highest frequencies
you can, and peaking them. How easy this is to do and how accurate you can
be depends somewhat on the source material.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Scott Dorsey]

In article ,
Meindert Sprang wrote:
"Jim Lesurf" wrote in message
...
Like others, I'd suggest *avoiding* normalizing or limiting to anything
close to 0dBFS. if you normalise to within a dB or so of 0dBFS you'd need
to check any CD player you use can then handle intersample excursions
produced by the DAC/reconstruction filter that rise above 0dBFS.

Is that *really* an existing problem ur an urban myth?

It does exist on a few model players.

The Red Book
specifies 16 bit data. Why would they do that if a player would not be able
to reconstruct full 16 bit data to an analog signal?

Because the Red Book was written before any of the players existed.

to me this would really
be a severe design flaw in a CD player! If I were the designer of a DAC and
I would expect that an interpolation between two samples would rise above
the maximum dynamic range, I'd add the necessary bits to take care of that
or scale the input of that process down. In my DSP work I always checked the
input range and output range of calculations to see if it would fit in the
'width' of calculations and scale accordingly. This is standard design
practice.

Consumer equipment is designed to be as cheap as absolutely possible.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Jim Lesurf[_3_]]

In article , Scott Dorsey
wrote:

Consumer equipment is designed to be as cheap as absolutely possible.
--scott

My take on that is slightly different. That consumer equipment is made to
be *sold* (at a profit), not to be *used*. Some designers/makers may be
more concerned for the end-user (rather than customer) than others. But I'm
not sure price is always a good indicator...

Slainte,

Jim

--
Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm
Armstrong Audio http://www.audiomisc.co.uk/Armstrong/armstrong.html
Audio Misc http://www.audiomisc.co.uk/index.html

[Richard Webb[_3_]]

On Thu 2012-Jul-19 01:37, Trevor writes:
Yep, was assuming the tape with the proper test tones of
course. Ears work if you know how to use 'em for most
purposes grin. Still, doing this setup before playing the tapes into
the computer, whether with properly done test
tones or setting up the deck by ear will go a long way
toward eliminating the need for further band aid fixes.


The point was to match the heads to the *particular* tape you are
copying for best response. Adjusting them to a test tape is what you
do before you record, and why would anybody want to do that on a
cassette deck these days? (unless of course you are trying to copy
the test tape to CD, which would be even more pointless!)
*Only* if the tape was recorded on the same machine with the heads
adjusted to a test tape before the recording you are copying will
there be any point in doing that again now, and for most cassettes
decks, not even then!

Good point! That's what we get for assumptions. I'm
assuming that the recording was made on a properly set up
deck in the first place grin. Tain't necessarily so huge grin.



Regards,
Richard
--
| Remove .my.foot for email
| via Waldo's Place USA Fidonet-Internet Gateway Site
| Standard disclaimer: The views of this user are strictly his own.

[Sean Conolly]

"hank alrich" wrote in message
...
Sean Conolly wrote:

* Normalize & hard limit (if needed) to the final levels I want for CD. I
hard limit sparingly, just enough to bring up the average level if
there's a
few peaks that stand out.
* Convert to 16 bit for burning to CD

Why in the world would you hard limit when moving from a medium with
maybe 50 or 60 dB of dynamic range to one of 96 dB of dynamic range?

Simply to somewhat match the levels of the other material on the CD, so I'm
not adjusting the levels per song when I'm listening. Just my own
preference.

Trimming a few DB off a few transients is inaudible. Shaving the entire
track like a lawnmower is not the idea, but as you know is a common (and
bad) practice.


Significant peaks probably didn't survive recording to cassette in the
first place.

Agreed.

Sean

[Sean Conolly]

"Meindert Sprang" wrote in message
...
"Jim Lesurf" wrote in message
...
Like others, I'd suggest *avoiding* normalizing or limiting to anything
close to 0dBFS. if you normalise to within a dB or so of 0dBFS you'd need
to check any CD player you use can then handle intersample excursions
produced by the DAC/reconstruction filter that rise above 0dBFS.

Is that *really* an existing problem ur an urban myth? The Red Book
specifies 16 bit data. Why would they do that if a player would not be
able
to reconstruct full 16 bit data to an analog signal? to me this would
really
be a severe design flaw in a CD player! If I were the designer of a DAC
and
I would expect that an interpolation between two samples would rise above
the maximum dynamic range, I'd add the necessary bits to take care of that
or scale the input of that process down. In my DSP work I always checked
the
input range and output range of calculations to see if it would fit in the
'width' of calculations and scale accordingly. This is standard design
practice.

No, it's a legitmate problem with some gear, and I'm inclined to agree with
Jim on why.

For demos that I'm going to distribute I set the peaks to -1 db, which is
fine with all the players I've encountered.

Sean

[geoff]

Scott Dorsey wrote:

Consumer equipment is designed to be as cheap as absolutely possible.

Some is, and some is designed to be as expensive as absolutely possible, on
the hope that there will be a fool who pays the money. And then there is
us, in a range across the middle, with slightly varying criteria.

geoff

[Trevor]

"geoff" wrote in message
...
Scott Dorsey wrote:
Consumer equipment is designed to be as cheap as absolutely possible.

Some is, and some is designed to be as expensive as absolutely possible,
on the hope that there will be a fool who pays the money.

Nope, that is still designed to be as cheap as possible *for the
manufacturer* (obviously NOT for the purchaser) so the manufacturer can
spend money on advertising a very low volume item whilst still making a very
large profit.

Trevor.

[geoff]

Trevor wrote:
"geoff" wrote in message
...
Scott Dorsey wrote:
Consumer equipment is designed to be as cheap as absolutely
possible.

Some is, and some is designed to be as expensive as absolutely
possible, on the hope that there will be a fool who pays the money.

Nope, that is still designed to be as cheap as possible *for the
manufacturer* (obviously NOT for the purchaser) so the manufacturer
can spend money on advertising a very low volume item whilst still
making a very large profit.

Trevor.

And you can buy a high-priced Loewe that is actually LG inside a pretty box
!

geoff

[Jim Lesurf[_3_]]

In article , Jim Lesurf
wrote:

[snip discussion of players distorting inter-sample peaks]

I'll see if I can find the data and I'll ask the person who gave it to
me if he minds it being made public. At the time it was just sent to me
as part of our discussions about the topic.

He has just replied and said this is OK. So I'll publish his results on my
website when I get a 'round tuit'. :-)

Slainte,

Jim

--
Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm
Armstrong Audio http://www.audiomisc.co.uk/Armstrong/armstrong.html
Audio Misc http://www.audiomisc.co.uk/index.html

[Jim Lesurf[_3_]]

On 23 Jul, wrote:
In article , Jim Lesurf
wrote:

[snip discussion of players distorting inter-sample peaks]

I'll see if I can find the data and I'll ask the person who gave it to
me if he minds it being made public. At the time it was just sent to
me as part of our discussions about the topic.

He has just replied and said this is OK. So I'll publish his results on
my website when I get a 'round tuit'. :-)

I've now done this and the results can be seen at

http://www.audiomisc.co.uk/HFN/OTTre...d/results.html

My thanks to Keith Howard for kindly agreeing to let me publish his
measured results. Note they were made in 2007 so only covered the players
and DACs he had to hand at that time. But - rather depressingly - all but
one of the nine he tried showed they had problems coping with the waveform
that has peaks at +3dBFS.

FWIW personally I'd love to see all reviews on DACs or player use the
'waveform from hell' I devised for the original article (a link to that is
on the above page) as a test of how they cope - or not! :-) I have the
uncomfortable feeling that whilst reviews continue to overlook this area,
problems will continue to afflict some new designs without anyone knowing.

Slainte,

Jim

--
Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm
Armstrong Audio http://www.audiomisc.co.uk/Armstrong/armstrong.html
Audio Misc http://www.audiomisc.co.uk/index.html

[Meindert Sprang]

"Jim Lesurf" wrote in message
...
I've now done this and the results can be seen at

http://www.audiomisc.co.uk/HFN/OTTre...d/results.html

My thanks to Keith Howard for kindly agreeing to let me publish his
measured results. Note they were made in 2007 so only covered the players
and DACs he had to hand at that time. But - rather depressingly - all but
one of the nine he tried showed they had problems coping with the waveform
that has peaks at +3dBFS.

FWIW personally I'd love to see all reviews on DACs or player use the
'waveform from hell' I devised for the original article (a link to that is
on the above page) as a test of how they cope - or not! :-) I have the
uncomfortable feeling that whilst reviews continue to overlook this area,
problems will continue to afflict some new designs without anyone knowing.

This test is flawed. The sample points on the right represent a sinewave of
+3dBFS. If you would sample a sinewave on the tops, you would indeed get the
picture as shown on the left. If you move the sample points to the positions
as shown in the picture on the right, the sample points would be 3dB down
and they would still represent a sinewave of 0dBFS.

Following this reasoning, one could say that if a DAW normalizes by simply
measuring the highest sample and scaling all others accordingly, that DAW is
flawed too. It should at least "reconstruct" the whole waveform to be able
to determine the *real* maximum amplitude.

So the question is: do we know how a DAW measures the maximum amplitude? Is
it documented in the manual/specs? If not, using -3dBFS is always safe
because this is the worst case we could encounter as seen in the test.

In reality, a piece of audio with thousands of samples would probably have a
few that are nearly on the top of a loudest sinewave, statistically
speaking, so -1dBFS would *probably* do as well as someone else already
mentioned.

Interesting stuff when you think about it....

Meindert

[Jim Lesurf[_3_]]

In article , Meindert
Sprang wrote:
"Jim Lesurf" wrote in message
...
I've now done this and the results can be seen at

http://www.audiomisc.co.uk/HFN/OTTre...d/results.html

My thanks to Keith Howard for kindly agreeing to let me publish his
measured results. Note they were made in 2007 so only covered the
players and DACs he had to hand at that time. But - rather
depressingly - all but one of the nine he tried showed they had
problems coping with the waveform that has peaks at +3dBFS.

FWIW personally I'd love to see all reviews on DACs or player use the
'waveform from hell' I devised for the original article (a link to
that is on the above page) as a test of how they cope - or not! :-)
I have the uncomfortable feeling that whilst reviews continue to
overlook this area, problems will continue to afflict some new designs
without anyone knowing.

This test is flawed.

Depends what you mean by "flawed". cf below...

The sample points on the right represent a sinewave of +3dBFS.

Correct. That is what sampling theory, etc, indicate the samples should
mean when reconstructed if we assume sampling/filtering at ADC was done
using standard-theory time-symmetric sinc-like filters.

If you would sample a sinewave on the tops, you would indeed
get the picture as shown on the left. If you move the sample points to
the positions as shown in the picture on the right, the sample points
would be 3dB down and they would still represent a sinewave of 0dBFS.

Following this reasoning, one could say that if a DAW normalizes by
simply measuring the highest sample and scaling all others accordingly,
that DAW is flawed too.

I tend to agree. However this once again depends on how you choose to
define "flawed". Information theory is quite happy to accept the sample
values as representing a waveform that peaks at 3dBFS. And the critical
point is that some CDs, etc, do carry sample values that will require
excursions 0dBFS.

So I don't think the *test* is "flawed" given that players may often be
presented with series of samples that require peaks above 0dBFS to
reconstruct as part of the music. Saying the *discs* are "flawed" might be
a better argument. But in theory they should be fine *provided* the
players/DAC designer realised the implications of sampling theory, etc, and
made a player/DAC that copes as theory indicates.

It should at least "reconstruct" the whole waveform to be able to
determine the *real* maximum amplitude.

I agree. However the reality is that some Audio CDs, etc, *do* give series
of sample values that reconstruct to have excursions above 0dBFS. Hence the
test exposes that some players may have a problem reconstructing such
waveforms.

The problem them becomes a real one for those who buy CDs/files. Should
their player/DAC cope with such material or not? My view is that it should.
This is for two reasons.

1) Such material exists, so a player/DAC should reconstruct it in an
orderly way in accord with the sampling theorem, etc.

2) It is simple enough these days for the designer to arrange (1) if they
have a clue. So why should they refuse give the existence of such material?

The snag being, how to tell if no-one checks using such a test or examines
the CDs to see if any require this? The test exposes a problem area.

In an ideal world, though, I'd certainly *much* prefer those who release
CDs and files (and stream) to avoid any intersample peaks reaching 0dBFS or
more. Alas, that world isn't the one we live in. :-/

So the question is:
do we know how a DAW measures the maximum amplitude?

Nielsen and Lund have looked at such issues IIRC in AES papers. However I
assume that it would depend on the details of the specific cases and
equipment. The actual max possible peak intersample values will depend on
the bandwidth/shape/etc of the source material being processed as well as
the sample rates, etc. So the complication is that a precise figure will
depend on the material as well as the details of the system used.

Overall, though, I'd expect allowing around 3dB to be generally 'safe' cf
below.

One problem, alas, is the obsession some CD/file 'mastering' people have
with LOUDNESS. To deal with avoiding intersample peak problems that would
have to be tackled, I think. TBH I wonder if some of those 'mastering' have
a clue about any of these questions, or give a hoot... :-/

FWIW if you look at my website and many other places you will find plenty
of examples of commercial CDs 'mastered' with plenty of samples in the
range above -1dBFS.

My experience is that 'classical' types of music avoid these problems
because those who make the CDs take more care and aren't obsessed with
loudness, and perhaps because those involved take care. But pop/rock
material is often LOUD and the assumption seems to be "that is what sells".

Is it documented in the manual/specs? If not, using -3dBFS is always
safe because this is the worst case we could encounter as seen in the
test.

The peak excursions can potentially be much higher than 3dB. e.g. if you
look at

http://www.audiomisc.co.uk/HFN/OverTheTop/OTT.html

you can see where I got over +5dBFS for a deliberately chosen 'waveform
from hell'. 8-]

So if you wish to be *really* careful you'd need to avoid samples going
above -5dBFS. But I would agree that such extreme examples would be
unlikely in real practice with acoustic music. (Not so sure for
'electronic' creations for pop/rock!)

In reality, a piece of audio with thousands of samples would probably
have a few that are nearly on the top of a loudest sinewave,
statistically speaking, so -1dBFS would *probably* do as well as someone
else already mentioned.

FWIW my own opinion is that for material at rates like 48k or 44.1k it is
wisest to ensure no samples go above about -3dBFS. Although more like -1 or
-2 may often be OK - depending on the material. That was why I raised this
issue when people were discussing 'normalisation'. The bottom line is that
simply normalising to the biggest sample being at 0dBFS is unwise.

IIRC If you look at BBC Radio 3 material, they generally try to keep peak
samples below about -4dBFS. (This is even more important for methods that
encode using methods like AAC which are 'lossy' so may alter the size of
peaks.) So BBC R3 caution puts them in between the 3dB value from the
simple sinewave example Keith used, and my 'waveform from hell'.

BTW if anyone wants to experiment with the 'waveform from hell' it is still
available as a zipped wave file from

http://jcgl.orpheusweb.co.uk/temp/WaveFromHell.zip

(Actually a pair of wave files with different sign relationships
left/right.)

Use with care though, not sure how happy some amps or speakers would be
with that waveform. :-)

Slainte,

Jim

--
Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm
Armstrong Audio http://www.audiomisc.co.uk/Armstrong/armstrong.html
Audio Misc http://www.audiomisc.co.uk/index.html

[Johny B Good[_2_]]

On Thu, 19 Jul 2012 15:13:45 +0100, Jim Lesurf
wrote:

In article , Scott Dorsey
wrote:

Consumer equipment is designed to be as cheap as absolutely possible.
--scott

My take on that is slightly different. That consumer equipment is made to
be *sold* (at a profit), not to be *used*. Some designers/makers may be
more concerned for the end-user (rather than customer) than others. But I'm
not sure price is always a good indicator...

Apologies for the lateness of my response but I've only just
subscribed to this NG and I feel that I have a useful contribution to
make...

Almost certainly cost will be a major factor (even, as in the case of
Akai's flagship GX630DBm at that time, it is only a matter of shaving
a few pence off the cost of a piece of kit priced close to 500 quid).

In this case, looking at the plots, it seems more likely (especially
so for the Pioneer DV-939A) that the analogue output stage of the DAC
or the following stages were starved of sufficient DC bias voltage to
encompass the full peak to peak swing, i.e. they were clipping the
output waveform.

The extra 5db required for the 'waveform from Hell' is pretty close
to the 6db mark (which I suspect would only just exceed that of the
_ulimate_ 'waveform from Hell') which suggests a doubling of bias
voltage over and above that required to prevent clipping at 0db.

If the 0db voltage level is, as is commonly the case, based on the
0dbm 600R reference level (775mV rms sine), The peak to peak voltage
swing requirement at +6db becomes 4.3834v. Theoretically possible to
achieve with a single ended buffer amp powered from a 5 volt rail
(quite possibly just achievable with a FET based amplifier stage).

In this case, I suspect the real culprit isn't so much the 'extra
cost of doubling the output buffer amp rail voltage' so much as a
'lack of attention to detail' since the internal comparitor reference
voltage setting can be arbitarily adjusted[1] to avoid such
interpolated excursions exceeding whatever the peak to peak clipping
level limit happens to be in the analogue output stage(s)[2].

Getting back to the order of processing of digitised vinyl recordings
(whether direct or from tape dubs), I would suggest you look at the
whole waveform and home in on suspicious peaks manually and apply
clip/pop filtering to any that are obviously pops or clicks rather
than musical transients.

Once you've cleaned up such blatent pops and clicks, scan the
waveform to determine whether or not you now have headroom for
normalisng the tracks and assess whether the gain values look
reasonable for how the level appears by eye in the waveform editor
display (it's still possible that some strong clicks may remain
without showing in the edit display).

Once you've done this, you can apply normalisation and verify that,
in bulk, the result still looks (and sounds) ok. You could home in on
any peaks to verify whether they represent musical transients or
previously missed clicks or pops.

I tend not to apply declicking to the whole recording since it can
result in distorting effects on certain musical instruments (trumpets
can be very badly mangled by this process). I find such processing is
best concentrated on the quieter parts (typically intros and outros).

Aside from very loud clicks or pops, the music in the louder parts
will mask modest scratch noise and clicks that would otherwise become
objectionable in the very quiet parts of the recording.

I don't think there are any such cleanup filters that can be trusted
to automatically de-click / de-pop / de-scratch a complete vinyl
recording in one go without risk of objectional distortions arising
from the processing but I could well be out of touch with the latest
developments in such DSP technology[3]

If you are going to attempt to apply an all-in-one cleanup process,
audition the result thoroughly before deleting the original
un-processed wav file. Talking of which, deleting such files after
producing a cleaned up version was quite common back in the day some
ten to fifteen years ago, but with the much larger disk drive
capacities now availble, there's no longer any need for such space
savings. As long as you archive the orginal wav files, you can be as
cavalier as you like with the post digitsing processing.

[1] Of course, all bets are off if the DAC chip employed offers no
such adjustment.

[2] Assuming the manufacturer isn't totally obsessed with the 0dbm
600R reference or else is prepared to adjust to this reference by
adding the extra gain required in a seperate buffer amp that is fed
from a voltage rail that will allow a peak to peak swing without
clipping.

[3] After re- researching the subject of DSP software to bypass the
Dolby decoder stage in the tape deck itself and apply the Dolby B
decoding function in software, I don't hold out much hope for any
improvements in such de-clicking software.
--
Regards, J B Good