"Karl Uppiano" writes:
[...]
A well-designed PCM system (e.g., CD-Audio) uses phase-linear, oversampling
digital FIR filters to allow the use of very gentle, phase-linear analog
filtering. I think the original Sony/Philips engineers that knew what they
were doing when they designed the CD-Audio spec must have retired without
passing their expertise on to the next generation.

Right-on, brother. I've been watching (sometimes not so passively)
this "DVD-A/SACD" blindside on the consumers for a couple of years now
and my conclusion, as one whose career is in digital signal
processing, is that it's a complete and utter farce.

In the end, consumers will have emptied out their pocketbooks for new
players and, more importantly, new media which are indistinguishable
in sound quality from CD audio.

The only possible rational justification for a new format is the
inclusion of multiple ( 2) tracks. The sound quality arguments are
empty.
--
% Randy Yates % "Though you ride on the wheels of tomorrow,
%% Fuquay-Varina, NC % you still wander the fields of your
%%% 919-577-9882 % sorrow."
%%%% % '21st Century Man', *Time*, ELO
http://home.earthlink.net/~yatescr

"Randy Yates" wrote in message

"Karl Uppiano" writes:
[...]
A well-designed PCM system (e.g., CD-Audio) uses phase-linear,
oversampling digital FIR filters to allow the use of very gentle,
phase-linear analog filtering. I think the original Sony/Philips
engineers that knew what they were doing when they designed the
CD-Audio spec must have retired without passing their expertise on
to the next generation.

Right-on, brother. I've been watching (sometimes not so passively)
this "DVD-A/SACD" blindside on the consumers for a couple of years now
and my conclusion, as one whose career is in digital signal
processing, is that it's a complete and utter farce.

In the end, consumers will have emptied out their pocketbooks for new
players and, more importantly, new media which are indistinguishable
in sound quality from CD audio.

The only possible rational justification for a new format is the
inclusion of multiple ( 2) tracks. The sound quality arguments are
empty.

Looks to me like a failed marketing effort. The drive behind attempts to
obsolete the CD Audio format have to be money. Once the Chinese started
producing CD players with what most consumers found to be acceptable sound
quality, the Japanese were largely cut out of the picture unless they had
some unique technology to sell. SACD and DVD-A were that technology, but
insufficient product was sold before the new-technology players were
commoditized.

With something like 600,000 DVD-A discs and perhaps as many as a few times
that in SACD discs sold last year, it's quite clear that there just aren't a
lot of players out there that consumers are trying to "feed". With
SACD-DAV-A players selling for under $300, this is hardly a true high end,
nice market play. It's a failed attempt to sell consumers a line of 'bigger
numbers always sound better" BS.

Now, we're faced with retro-tech flacks like François who push vinyl out of
one side of their mouths, and high sample rates out of the other. Anybody
who has a memory long enough to remember the last time they talked out of
both sides of their mouth will not grant them sufficient credibility to
actually sell product.

"Randy Yates" wrote in message

"Karl Uppiano" writes:
[...]
A well-designed PCM system (e.g., CD-Audio) uses phase-linear,
oversampling digital FIR filters to allow the use of very gentle,
phase-linear analog filtering. I think the original Sony/Philips
engineers that knew what they were doing when they designed the
CD-Audio spec must have retired without passing their expertise on
to the next generation.

Right-on, brother. I've been watching (sometimes not so passively)
this "DVD-A/SACD" blindside on the consumers for a couple of years now
and my conclusion, as one whose career is in digital signal
processing, is that it's a complete and utter farce.

In the end, consumers will have emptied out their pocketbooks for new
players and, more importantly, new media which are indistinguishable
in sound quality from CD audio.

The only possible rational justification for a new format is the
inclusion of multiple ( 2) tracks. The sound quality arguments are
empty.

Looks to me like a failed marketing effort. The drive behind attempts to
obsolete the CD Audio format have to be money. Once the Chinese started
producing CD players with what most consumers found to be acceptable sound
quality, the Japanese were largely cut out of the picture unless they had
some unique technology to sell. SACD and DVD-A were that technology, but
insufficient product was sold before the new-technology players were
commoditized.

With something like 600,000 DVD-A discs and perhaps as many as a few times
that in SACD discs sold last year, it's quite clear that there just aren't a
lot of players out there that consumers are trying to "feed". With
SACD-DAV-A players selling for under $300, this is hardly a true high end,
nice market play. It's a failed attempt to sell consumers a line of 'bigger
numbers always sound better" BS.

Now, we're faced with retro-tech flacks like François who push vinyl out of
one side of their mouths, and high sample rates out of the other. Anybody
who has a memory long enough to remember the last time they talked out of
both sides of their mouth will not grant them sufficient credibility to
actually sell product.

"Randy Yates" wrote in message

"Karl Uppiano" writes:
[...]
A well-designed PCM system (e.g., CD-Audio) uses phase-linear,
oversampling digital FIR filters to allow the use of very gentle,
phase-linear analog filtering. I think the original Sony/Philips
engineers that knew what they were doing when they designed the
CD-Audio spec must have retired without passing their expertise on
to the next generation.

Right-on, brother. I've been watching (sometimes not so passively)
this "DVD-A/SACD" blindside on the consumers for a couple of years now
and my conclusion, as one whose career is in digital signal
processing, is that it's a complete and utter farce.

In the end, consumers will have emptied out their pocketbooks for new
players and, more importantly, new media which are indistinguishable
in sound quality from CD audio.

The only possible rational justification for a new format is the
inclusion of multiple ( 2) tracks. The sound quality arguments are
empty.

Looks to me like a failed marketing effort. The drive behind attempts to
obsolete the CD Audio format have to be money. Once the Chinese started
producing CD players with what most consumers found to be acceptable sound
quality, the Japanese were largely cut out of the picture unless they had
some unique technology to sell. SACD and DVD-A were that technology, but
insufficient product was sold before the new-technology players were
commoditized.

With something like 600,000 DVD-A discs and perhaps as many as a few times
that in SACD discs sold last year, it's quite clear that there just aren't a
lot of players out there that consumers are trying to "feed". With
SACD-DAV-A players selling for under $300, this is hardly a true high end,
nice market play. It's a failed attempt to sell consumers a line of 'bigger
numbers always sound better" BS.

Now, we're faced with retro-tech flacks like François who push vinyl out of
one side of their mouths, and high sample rates out of the other. Anybody
who has a memory long enough to remember the last time they talked out of
both sides of their mouth will not grant them sufficient credibility to
actually sell product.

"Randy Yates" wrote in message

"Karl Uppiano" writes:
[...]
A well-designed PCM system (e.g., CD-Audio) uses phase-linear,
oversampling digital FIR filters to allow the use of very gentle,
phase-linear analog filtering. I think the original Sony/Philips
engineers that knew what they were doing when they designed the
CD-Audio spec must have retired without passing their expertise on
to the next generation.

Right-on, brother. I've been watching (sometimes not so passively)
this "DVD-A/SACD" blindside on the consumers for a couple of years now
and my conclusion, as one whose career is in digital signal
processing, is that it's a complete and utter farce.

In the end, consumers will have emptied out their pocketbooks for new
players and, more importantly, new media which are indistinguishable
in sound quality from CD audio.

The only possible rational justification for a new format is the
inclusion of multiple ( 2) tracks. The sound quality arguments are
empty.

Looks to me like a failed marketing effort. The drive behind attempts to
obsolete the CD Audio format have to be money. Once the Chinese started
producing CD players with what most consumers found to be acceptable sound
quality, the Japanese were largely cut out of the picture unless they had
some unique technology to sell. SACD and DVD-A were that technology, but
insufficient product was sold before the new-technology players were
commoditized.

With something like 600,000 DVD-A discs and perhaps as many as a few times
that in SACD discs sold last year, it's quite clear that there just aren't a
lot of players out there that consumers are trying to "feed". With
SACD-DAV-A players selling for under $300, this is hardly a true high end,
nice market play. It's a failed attempt to sell consumers a line of 'bigger
numbers always sound better" BS.

Now, we're faced with retro-tech flacks like François who push vinyl out of
one side of their mouths, and high sample rates out of the other. Anybody
who has a memory long enough to remember the last time they talked out of
both sides of their mouth will not grant them sufficient credibility to
actually sell product.

"Arny Krueger" wrote in message
...
"Karl Uppiano" wrote in message

"François Yves Le Gal" wrote in message
...
On Sun, 01 Feb 2004 12:53:18 -0800, chung wrote:

One, DSD/SACD proponents
claim the much wider bandwidth over CD's, and filtering will reduce
significantly that claimed advantage.

Having a gentle low pass filter at 60 or 100 KHz doesn't
significantly reduce SACD's bandwith. You get more than PCM 96 or
192 in both cases!

I would start a 3 dB/octave rolloff at 20kHz or so.

Ironically, most if not all people can't hear the difference a brickwall
filter at 16 KHz makes, if the filter is well-designed.

Don't believe me?

Listen for yourself at http://www.pcabx.com/technical/low_pass/index.htm .

I believe you. I realize it isn't considered state of the art anymore, but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

"Arny Krueger" wrote in message
...
"Karl Uppiano" wrote in message

"François Yves Le Gal" wrote in message
...
On Sun, 01 Feb 2004 12:53:18 -0800, chung wrote:

One, DSD/SACD proponents
claim the much wider bandwidth over CD's, and filtering will reduce
significantly that claimed advantage.

Having a gentle low pass filter at 60 or 100 KHz doesn't
significantly reduce SACD's bandwith. You get more than PCM 96 or
192 in both cases!

I would start a 3 dB/octave rolloff at 20kHz or so.

Ironically, most if not all people can't hear the difference a brickwall
filter at 16 KHz makes, if the filter is well-designed.

Don't believe me?

Listen for yourself at http://www.pcabx.com/technical/low_pass/index.htm .

I believe you. I realize it isn't considered state of the art anymore, but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

"Arny Krueger" wrote in message
...
"Karl Uppiano" wrote in message

"François Yves Le Gal" wrote in message
...
On Sun, 01 Feb 2004 12:53:18 -0800, chung wrote:

One, DSD/SACD proponents
claim the much wider bandwidth over CD's, and filtering will reduce
significantly that claimed advantage.

Having a gentle low pass filter at 60 or 100 KHz doesn't
significantly reduce SACD's bandwith. You get more than PCM 96 or
192 in both cases!

I would start a 3 dB/octave rolloff at 20kHz or so.

Ironically, most if not all people can't hear the difference a brickwall
filter at 16 KHz makes, if the filter is well-designed.

Don't believe me?

Listen for yourself at http://www.pcabx.com/technical/low_pass/index.htm .

I believe you. I realize it isn't considered state of the art anymore, but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

"Arny Krueger" wrote in message
...
"Karl Uppiano" wrote in message

"François Yves Le Gal" wrote in message
...
On Sun, 01 Feb 2004 12:53:18 -0800, chung wrote:

One, DSD/SACD proponents
claim the much wider bandwidth over CD's, and filtering will reduce
significantly that claimed advantage.

Having a gentle low pass filter at 60 or 100 KHz doesn't
significantly reduce SACD's bandwith. You get more than PCM 96 or
192 in both cases!

I would start a 3 dB/octave rolloff at 20kHz or so.

Ironically, most if not all people can't hear the difference a brickwall
filter at 16 KHz makes, if the filter is well-designed.

Don't believe me?

Listen for yourself at http://www.pcabx.com/technical/low_pass/index.htm .

I believe you. I realize it isn't considered state of the art anymore, but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

"Arny Krueger" wrote in message ...
"Robert Stanton" wrote in message
m
chung wrote in message
...
Harry Lavo wrote:



Isn't it interesting that we "subjectivists" here are always
whipsawed by the "objectivists" for thinking that extended
frequency response is a benefit "because it can't be heard". Then,
in defense of DVD-A, the increased noise in the ultrasonic range is
bandied about as making DSD/SACD "inferior".


Actually it's interesting that subjectivists said that they need the
ultrasonic bandwidth to fully perceive music, and yet they like
SACD's despite the much higher (by orders of magnitude) ultrasonic
noise inherent in the SACD format.


I often see ultrasonic noise mentioned as a problem for SACD players.
If ultrasonic noise were really a problem, it could be easily
eliminated with a small, active lowpass filter.

Why wouldn't the manfactures of "high end" SACD players, just filter
it out?

Wouldn't the filter affect the overtones of the music just as much as it
affects the noise?

Yes, it will chop off all overtones above 30kHz. But, we humans can't
hear above 25kHz, so we won't hear the difference.

I took a standard Chebychev 0.1 dB ripple, 5th order, active filter
from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353
op-amps.

Here are the simulation results:

Freq. Gain Group Delay
Hz dB usec

1000 -0.09 20.9
5000 -0.15 20.7
10000 -0.18 21.3
20000 -0.21 25.6
30000 -3.01 43.5
40000 -19.7 10.4
50000 -31.7 4.7
60000 -40.7 2.8
70000 -48.1
80000 -54.4
90000 -60.0


The filter is flat from 20 to 20kHz, and the group delay is almost flat
from 20 to 20kHz.

Bob Stanton

"Arny Krueger" wrote in message ...
"Robert Stanton" wrote in message
m
chung wrote in message
...
Harry Lavo wrote:



Isn't it interesting that we "subjectivists" here are always
whipsawed by the "objectivists" for thinking that extended
frequency response is a benefit "because it can't be heard". Then,
in defense of DVD-A, the increased noise in the ultrasonic range is
bandied about as making DSD/SACD "inferior".


Actually it's interesting that subjectivists said that they need the
ultrasonic bandwidth to fully perceive music, and yet they like
SACD's despite the much higher (by orders of magnitude) ultrasonic
noise inherent in the SACD format.


I often see ultrasonic noise mentioned as a problem for SACD players.
If ultrasonic noise were really a problem, it could be easily
eliminated with a small, active lowpass filter.

Why wouldn't the manfactures of "high end" SACD players, just filter
it out?

Wouldn't the filter affect the overtones of the music just as much as it
affects the noise?

Yes, it will chop off all overtones above 30kHz. But, we humans can't
hear above 25kHz, so we won't hear the difference.

I took a standard Chebychev 0.1 dB ripple, 5th order, active filter
from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353
op-amps.

Here are the simulation results:

Freq. Gain Group Delay
Hz dB usec

1000 -0.09 20.9
5000 -0.15 20.7
10000 -0.18 21.3
20000 -0.21 25.6
30000 -3.01 43.5
40000 -19.7 10.4
50000 -31.7 4.7
60000 -40.7 2.8
70000 -48.1
80000 -54.4
90000 -60.0


The filter is flat from 20 to 20kHz, and the group delay is almost flat
from 20 to 20kHz.

Bob Stanton

"Arny Krueger" wrote in message ...
"Robert Stanton" wrote in message
m
chung wrote in message
...
Harry Lavo wrote:



Isn't it interesting that we "subjectivists" here are always
whipsawed by the "objectivists" for thinking that extended
frequency response is a benefit "because it can't be heard". Then,
in defense of DVD-A, the increased noise in the ultrasonic range is
bandied about as making DSD/SACD "inferior".


Actually it's interesting that subjectivists said that they need the
ultrasonic bandwidth to fully perceive music, and yet they like
SACD's despite the much higher (by orders of magnitude) ultrasonic
noise inherent in the SACD format.


I often see ultrasonic noise mentioned as a problem for SACD players.
If ultrasonic noise were really a problem, it could be easily
eliminated with a small, active lowpass filter.

Why wouldn't the manfactures of "high end" SACD players, just filter
it out?

Wouldn't the filter affect the overtones of the music just as much as it
affects the noise?

Yes, it will chop off all overtones above 30kHz. But, we humans can't
hear above 25kHz, so we won't hear the difference.

I took a standard Chebychev 0.1 dB ripple, 5th order, active filter
from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353
op-amps.

Here are the simulation results:

Freq. Gain Group Delay
Hz dB usec

1000 -0.09 20.9
5000 -0.15 20.7
10000 -0.18 21.3
20000 -0.21 25.6
30000 -3.01 43.5
40000 -19.7 10.4
50000 -31.7 4.7
60000 -40.7 2.8
70000 -48.1
80000 -54.4
90000 -60.0


The filter is flat from 20 to 20kHz, and the group delay is almost flat
from 20 to 20kHz.

Bob Stanton

"Arny Krueger" wrote in message ...
"Robert Stanton" wrote in message
m
chung wrote in message
...
Harry Lavo wrote:



Isn't it interesting that we "subjectivists" here are always
whipsawed by the "objectivists" for thinking that extended
frequency response is a benefit "because it can't be heard". Then,
in defense of DVD-A, the increased noise in the ultrasonic range is
bandied about as making DSD/SACD "inferior".


Actually it's interesting that subjectivists said that they need the
ultrasonic bandwidth to fully perceive music, and yet they like
SACD's despite the much higher (by orders of magnitude) ultrasonic
noise inherent in the SACD format.


I often see ultrasonic noise mentioned as a problem for SACD players.
If ultrasonic noise were really a problem, it could be easily
eliminated with a small, active lowpass filter.

Why wouldn't the manfactures of "high end" SACD players, just filter
it out?

Wouldn't the filter affect the overtones of the music just as much as it
affects the noise?

Yes, it will chop off all overtones above 30kHz. But, we humans can't
hear above 25kHz, so we won't hear the difference.

I took a standard Chebychev 0.1 dB ripple, 5th order, active filter
from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353
op-amps.

Here are the simulation results:

Freq. Gain Group Delay
Hz dB usec

1000 -0.09 20.9
5000 -0.15 20.7
10000 -0.18 21.3
20000 -0.21 25.6
30000 -3.01 43.5
40000 -19.7 10.4
50000 -31.7 4.7
60000 -40.7 2.8
70000 -48.1
80000 -54.4
90000 -60.0


The filter is flat from 20 to 20kHz, and the group delay is almost flat
from 20 to 20kHz.

Bob Stanton

Robert Stanton wrote:

"Arny Krueger" wrote in message ...
"Robert Stanton" wrote in message
m
chung wrote in message
...
Harry Lavo wrote:



Isn't it interesting that we "subjectivists" here are always
whipsawed by the "objectivists" for thinking that extended
frequency response is a benefit "because it can't be heard". Then,
in defense of DVD-A, the increased noise in the ultrasonic range is
bandied about as making DSD/SACD "inferior".


Actually it's interesting that subjectivists said that they need the
ultrasonic bandwidth to fully perceive music, and yet they like
SACD's despite the much higher (by orders of magnitude) ultrasonic
noise inherent in the SACD format.


I often see ultrasonic noise mentioned as a problem for SACD players.
If ultrasonic noise were really a problem, it could be easily
eliminated with a small, active lowpass filter.

Why wouldn't the manfactures of "high end" SACD players, just filter
it out?

Wouldn't the filter affect the overtones of the music just as much as it
affects the noise?

Yes, it will chop off all overtones above 30kHz. But, we humans can't
hear above 25kHz, so we won't hear the difference.

You and I may agree on that, but one of the touted features of SACD is
the much broader bandwidth, higher than 25 KHz. If you limit it to
25KHz, it could not compete against the other hi-rez formats, such as
24/96 or 24/192, which are flat up to close to half the sampling frequency.


I took a standard Chebychev 0.1 dB ripple, 5th order, active filter
from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353
op-amps.

Here are the simulation results:

Freq. Gain Group Delay
Hz dB usec

1000 -0.09 20.9
5000 -0.15 20.7
10000 -0.18 21.3
20000 -0.21 25.6
30000 -3.01 43.5
40000 -19.7 10.4
50000 -31.7 4.7
60000 -40.7 2.8
70000 -48.1
80000 -54.4
90000 -60.0


The filter is flat from 20 to 20kHz, and the group delay is almost flat
from 20 to 20kHz.

If you take into account tolerances of the 5 caps, you would have a
noticeable ripple in the passband, as well as mismatches between L/R.
Plus the cost of the 10 capacitors is not insignificant. And more if you
have 5 channels.

The much better way is simply apply digital filtering, or use better
dithering schemes, but then we are back to LPCM, and not DSD .


Bob Stanton

Robert Stanton wrote:

"Arny Krueger" wrote in message ...
"Robert Stanton" wrote in message
m
chung wrote in message
...
Harry Lavo wrote:



Isn't it interesting that we "subjectivists" here are always
whipsawed by the "objectivists" for thinking that extended
frequency response is a benefit "because it can't be heard". Then,
in defense of DVD-A, the increased noise in the ultrasonic range is
bandied about as making DSD/SACD "inferior".


Actually it's interesting that subjectivists said that they need the
ultrasonic bandwidth to fully perceive music, and yet they like
SACD's despite the much higher (by orders of magnitude) ultrasonic
noise inherent in the SACD format.


I often see ultrasonic noise mentioned as a problem for SACD players.
If ultrasonic noise were really a problem, it could be easily
eliminated with a small, active lowpass filter.

Why wouldn't the manfactures of "high end" SACD players, just filter
it out?

Wouldn't the filter affect the overtones of the music just as much as it
affects the noise?

Yes, it will chop off all overtones above 30kHz. But, we humans can't
hear above 25kHz, so we won't hear the difference.

You and I may agree on that, but one of the touted features of SACD is
the much broader bandwidth, higher than 25 KHz. If you limit it to
25KHz, it could not compete against the other hi-rez formats, such as
24/96 or 24/192, which are flat up to close to half the sampling frequency.


I took a standard Chebychev 0.1 dB ripple, 5th order, active filter
from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353
op-amps.

Here are the simulation results:

Freq. Gain Group Delay
Hz dB usec

1000 -0.09 20.9
5000 -0.15 20.7
10000 -0.18 21.3
20000 -0.21 25.6
30000 -3.01 43.5
40000 -19.7 10.4
50000 -31.7 4.7
60000 -40.7 2.8
70000 -48.1
80000 -54.4
90000 -60.0


The filter is flat from 20 to 20kHz, and the group delay is almost flat
from 20 to 20kHz.

If you take into account tolerances of the 5 caps, you would have a
noticeable ripple in the passband, as well as mismatches between L/R.
Plus the cost of the 10 capacitors is not insignificant. And more if you
have 5 channels.

The much better way is simply apply digital filtering, or use better
dithering schemes, but then we are back to LPCM, and not DSD .


Bob Stanton

Robert Stanton wrote:

"Arny Krueger" wrote in message ...
"Robert Stanton" wrote in message
m
chung wrote in message
...
Harry Lavo wrote:



Isn't it interesting that we "subjectivists" here are always
whipsawed by the "objectivists" for thinking that extended
frequency response is a benefit "because it can't be heard". Then,
in defense of DVD-A, the increased noise in the ultrasonic range is
bandied about as making DSD/SACD "inferior".


Actually it's interesting that subjectivists said that they need the
ultrasonic bandwidth to fully perceive music, and yet they like
SACD's despite the much higher (by orders of magnitude) ultrasonic
noise inherent in the SACD format.


I often see ultrasonic noise mentioned as a problem for SACD players.
If ultrasonic noise were really a problem, it could be easily
eliminated with a small, active lowpass filter.

Why wouldn't the manfactures of "high end" SACD players, just filter
it out?

Wouldn't the filter affect the overtones of the music just as much as it
affects the noise?

Yes, it will chop off all overtones above 30kHz. But, we humans can't
hear above 25kHz, so we won't hear the difference.

You and I may agree on that, but one of the touted features of SACD is
the much broader bandwidth, higher than 25 KHz. If you limit it to
25KHz, it could not compete against the other hi-rez formats, such as
24/96 or 24/192, which are flat up to close to half the sampling frequency.


I took a standard Chebychev 0.1 dB ripple, 5th order, active filter
from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353
op-amps.

Here are the simulation results:

Freq. Gain Group Delay
Hz dB usec

1000 -0.09 20.9
5000 -0.15 20.7
10000 -0.18 21.3
20000 -0.21 25.6
30000 -3.01 43.5
40000 -19.7 10.4
50000 -31.7 4.7
60000 -40.7 2.8
70000 -48.1
80000 -54.4
90000 -60.0


The filter is flat from 20 to 20kHz, and the group delay is almost flat
from 20 to 20kHz.

If you take into account tolerances of the 5 caps, you would have a
noticeable ripple in the passband, as well as mismatches between L/R.
Plus the cost of the 10 capacitors is not insignificant. And more if you
have 5 channels.

The much better way is simply apply digital filtering, or use better
dithering schemes, but then we are back to LPCM, and not DSD .


Bob Stanton

Robert Stanton wrote:

"Arny Krueger" wrote in message ...
"Robert Stanton" wrote in message
m
chung wrote in message
...
Harry Lavo wrote:



Isn't it interesting that we "subjectivists" here are always
whipsawed by the "objectivists" for thinking that extended
frequency response is a benefit "because it can't be heard". Then,
in defense of DVD-A, the increased noise in the ultrasonic range is
bandied about as making DSD/SACD "inferior".


Actually it's interesting that subjectivists said that they need the
ultrasonic bandwidth to fully perceive music, and yet they like
SACD's despite the much higher (by orders of magnitude) ultrasonic
noise inherent in the SACD format.


I often see ultrasonic noise mentioned as a problem for SACD players.
If ultrasonic noise were really a problem, it could be easily
eliminated with a small, active lowpass filter.

Why wouldn't the manfactures of "high end" SACD players, just filter
it out?

Wouldn't the filter affect the overtones of the music just as much as it
affects the noise?

Yes, it will chop off all overtones above 30kHz. But, we humans can't
hear above 25kHz, so we won't hear the difference.

You and I may agree on that, but one of the touted features of SACD is
the much broader bandwidth, higher than 25 KHz. If you limit it to
25KHz, it could not compete against the other hi-rez formats, such as
24/96 or 24/192, which are flat up to close to half the sampling frequency.


I took a standard Chebychev 0.1 dB ripple, 5th order, active filter
from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353
op-amps.

Here are the simulation results:

Freq. Gain Group Delay
Hz dB usec

1000 -0.09 20.9
5000 -0.15 20.7
10000 -0.18 21.3
20000 -0.21 25.6
30000 -3.01 43.5
40000 -19.7 10.4
50000 -31.7 4.7
60000 -40.7 2.8
70000 -48.1
80000 -54.4
90000 -60.0


The filter is flat from 20 to 20kHz, and the group delay is almost flat
from 20 to 20kHz.

If you take into account tolerances of the 5 caps, you would have a
noticeable ripple in the passband, as well as mismatches between L/R.
Plus the cost of the 10 capacitors is not insignificant. And more if you
have 5 channels.

The much better way is simply apply digital filtering, or use better
dithering schemes, but then we are back to LPCM, and not DSD .


Bob Stanton

"Karl Uppiano" writes:

"Arny Krueger" wrote in message
...
"Karl Uppiano" wrote in message

"François Yves Le Gal" wrote in message
...
On Sun, 01 Feb 2004 12:53:18 -0800, chung wrote:

One, DSD/SACD proponents
claim the much wider bandwidth over CD's, and filtering will reduce
significantly that claimed advantage.

Having a gentle low pass filter at 60 or 100 KHz doesn't
significantly reduce SACD's bandwith. You get more than PCM 96 or
192 in both cases!

I would start a 3 dB/octave rolloff at 20kHz or so.

Ironically, most if not all people can't hear the difference a brickwall
filter at 16 KHz makes, if the filter is well-designed.

Don't believe me?

Listen for yourself at http://www.pcabx.com/technical/low_pass/index.htm .

I believe you. I realize it isn't considered state of the art anymore, but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and
a 19 kHz pilot tone added. This, along with the L+R baseband signal, is
then FM modulated onto the carrier.
--
% Randy Yates % "And all that I can do
%% Fuquay-Varina, NC % is say I'm sorry,
%%% 919-577-9882 % that's the way it goes..."
%%%% % Getting To The Point', *Balance of Power*, ELO
http://home.earthlink.net/~yatescr

"Karl Uppiano" writes:

"Arny Krueger" wrote in message
...
"Karl Uppiano" wrote in message

"François Yves Le Gal" wrote in message
...
On Sun, 01 Feb 2004 12:53:18 -0800, chung wrote:

One, DSD/SACD proponents
claim the much wider bandwidth over CD's, and filtering will reduce
significantly that claimed advantage.

Having a gentle low pass filter at 60 or 100 KHz doesn't
significantly reduce SACD's bandwith. You get more than PCM 96 or
192 in both cases!

I would start a 3 dB/octave rolloff at 20kHz or so.

Ironically, most if not all people can't hear the difference a brickwall
filter at 16 KHz makes, if the filter is well-designed.

Don't believe me?

Listen for yourself at http://www.pcabx.com/technical/low_pass/index.htm .

I believe you. I realize it isn't considered state of the art anymore, but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and
a 19 kHz pilot tone added. This, along with the L+R baseband signal, is
then FM modulated onto the carrier.
--
% Randy Yates % "And all that I can do
%% Fuquay-Varina, NC % is say I'm sorry,
%%% 919-577-9882 % that's the way it goes..."
%%%% % Getting To The Point', *Balance of Power*, ELO
http://home.earthlink.net/~yatescr

"Karl Uppiano" writes:

"Arny Krueger" wrote in message
...
"Karl Uppiano" wrote in message

"François Yves Le Gal" wrote in message
...
On Sun, 01 Feb 2004 12:53:18 -0800, chung wrote:

One, DSD/SACD proponents
claim the much wider bandwidth over CD's, and filtering will reduce
significantly that claimed advantage.

Having a gentle low pass filter at 60 or 100 KHz doesn't
significantly reduce SACD's bandwith. You get more than PCM 96 or
192 in both cases!

I would start a 3 dB/octave rolloff at 20kHz or so.

Ironically, most if not all people can't hear the difference a brickwall
filter at 16 KHz makes, if the filter is well-designed.

Don't believe me?

Listen for yourself at http://www.pcabx.com/technical/low_pass/index.htm .

I believe you. I realize it isn't considered state of the art anymore, but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and
a 19 kHz pilot tone added. This, along with the L+R baseband signal, is
then FM modulated onto the carrier.
--
% Randy Yates % "And all that I can do
%% Fuquay-Varina, NC % is say I'm sorry,
%%% 919-577-9882 % that's the way it goes..."
%%%% % Getting To The Point', *Balance of Power*, ELO
http://home.earthlink.net/~yatescr

"Karl Uppiano" writes:

"Arny Krueger" wrote in message
...
"Karl Uppiano" wrote in message

"François Yves Le Gal" wrote in message
...
On Sun, 01 Feb 2004 12:53:18 -0800, chung wrote:

One, DSD/SACD proponents
claim the much wider bandwidth over CD's, and filtering will reduce
significantly that claimed advantage.

Having a gentle low pass filter at 60 or 100 KHz doesn't
significantly reduce SACD's bandwith. You get more than PCM 96 or
192 in both cases!

I would start a 3 dB/octave rolloff at 20kHz or so.

Ironically, most if not all people can't hear the difference a brickwall
filter at 16 KHz makes, if the filter is well-designed.

Don't believe me?

Listen for yourself at http://www.pcabx.com/technical/low_pass/index.htm .

I believe you. I realize it isn't considered state of the art anymore, but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and
a 19 kHz pilot tone added. This, along with the L+R baseband signal, is
then FM modulated onto the carrier.
--
% Randy Yates % "And all that I can do
%% Fuquay-Varina, NC % is say I'm sorry,
%%% 919-577-9882 % that's the way it goes..."
%%%% % Getting To The Point', *Balance of Power*, ELO
http://home.earthlink.net/~yatescr

chung writes:

François Yves Le Gal wrote:
On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote:

That does very little to the noise between 20KHz and 50KHz.
The noise is at much lower levels in these frequency bands.


The noise starts rising from 10KHz, and is significant between 20KHz
and 50KHz. Check out some player measurements:

http://www.stereophile.com/digitalso...06/index5.html

Noise is 30 dB higher than 24 bit LPCM at 20 KHz.

Oh piddle - that means the noise is only 114 dB below full-scale at
20 kHz. You know, we just might hear that...
--
% Randy Yates % "Remember the good old 1980's, when
%% Fuquay-Varina, NC % things were so uncomplicated?"
%%% 919-577-9882 % 'Ticket To The Moon'
%%%% % *Time*, Electric Light Orchestra
http://home.earthlink.net/~yatescr

chung writes:

François Yves Le Gal wrote:
On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote:

That does very little to the noise between 20KHz and 50KHz.
The noise is at much lower levels in these frequency bands.


The noise starts rising from 10KHz, and is significant between 20KHz
and 50KHz. Check out some player measurements:

http://www.stereophile.com/digitalso...06/index5.html

Noise is 30 dB higher than 24 bit LPCM at 20 KHz.

Oh piddle - that means the noise is only 114 dB below full-scale at
20 kHz. You know, we just might hear that...
--
% Randy Yates % "Remember the good old 1980's, when
%% Fuquay-Varina, NC % things were so uncomplicated?"
%%% 919-577-9882 % 'Ticket To The Moon'
%%%% % *Time*, Electric Light Orchestra
http://home.earthlink.net/~yatescr

chung writes:

François Yves Le Gal wrote:
On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote:

That does very little to the noise between 20KHz and 50KHz.
The noise is at much lower levels in these frequency bands.


The noise starts rising from 10KHz, and is significant between 20KHz
and 50KHz. Check out some player measurements:

http://www.stereophile.com/digitalso...06/index5.html

Noise is 30 dB higher than 24 bit LPCM at 20 KHz.

Oh piddle - that means the noise is only 114 dB below full-scale at
20 kHz. You know, we just might hear that...
--
% Randy Yates % "Remember the good old 1980's, when
%% Fuquay-Varina, NC % things were so uncomplicated?"
%%% 919-577-9882 % 'Ticket To The Moon'
%%%% % *Time*, Electric Light Orchestra
http://home.earthlink.net/~yatescr

chung writes:

François Yves Le Gal wrote:
On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote:

That does very little to the noise between 20KHz and 50KHz.
The noise is at much lower levels in these frequency bands.


The noise starts rising from 10KHz, and is significant between 20KHz
and 50KHz. Check out some player measurements:

http://www.stereophile.com/digitalso...06/index5.html

Noise is 30 dB higher than 24 bit LPCM at 20 KHz.

Oh piddle - that means the noise is only 114 dB below full-scale at
20 kHz. You know, we just might hear that...
--
% Randy Yates % "Remember the good old 1980's, when
%% Fuquay-Varina, NC % things were so uncomplicated?"
%%% 919-577-9882 % 'Ticket To The Moon'
%%%% % *Time*, Electric Light Orchestra
http://home.earthlink.net/~yatescr

Randy Yates wrote:
chung writes:
=20
Fran=E7ois Yves Le Gal wrote:
On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote:=


That does very little to the noise between 20KHz and 50KHz.
The noise is at much lower levels in these frequency bands.


The noise starts rising from 10KHz, and is significant between 20KHz
and 50KHz. Check out some player measurements:

http://www.stereophile.com/digitalso...06/index5.html

Noise is 30 dB higher than 24 bit LPCM at 20 KHz.
=20
Oh piddle - that means the noise is only 114 dB below full-scale at
20 kHz. You know, we just might hear that...

Actually the noise in a 1/3 octave BW centered at 20KHz is about 87 dB=20
below FS, according to that plot. Not that it is necessarily audible,=20
but the possibility exists that the sharply rising noise could cause=20
problems with some amps.

Randy Yates wrote:
chung writes:
=20
Fran=E7ois Yves Le Gal wrote:
On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote:=


That does very little to the noise between 20KHz and 50KHz.
The noise is at much lower levels in these frequency bands.


The noise starts rising from 10KHz, and is significant between 20KHz
and 50KHz. Check out some player measurements:

http://www.stereophile.com/digitalso...06/index5.html

Noise is 30 dB higher than 24 bit LPCM at 20 KHz.
=20
Oh piddle - that means the noise is only 114 dB below full-scale at
20 kHz. You know, we just might hear that...

Actually the noise in a 1/3 octave BW centered at 20KHz is about 87 dB=20
below FS, according to that plot. Not that it is necessarily audible,=20
but the possibility exists that the sharply rising noise could cause=20
problems with some amps.

Randy Yates wrote:
chung writes:
=20
Fran=E7ois Yves Le Gal wrote:
On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote:=


That does very little to the noise between 20KHz and 50KHz.
The noise is at much lower levels in these frequency bands.


The noise starts rising from 10KHz, and is significant between 20KHz
and 50KHz. Check out some player measurements:

http://www.stereophile.com/digitalso...06/index5.html

Noise is 30 dB higher than 24 bit LPCM at 20 KHz.
=20
Oh piddle - that means the noise is only 114 dB below full-scale at
20 kHz. You know, we just might hear that...

Actually the noise in a 1/3 octave BW centered at 20KHz is about 87 dB=20
below FS, according to that plot. Not that it is necessarily audible,=20
but the possibility exists that the sharply rising noise could cause=20
problems with some amps.

Randy Yates wrote:
chung writes:
=20
Fran=E7ois Yves Le Gal wrote:
On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote:=


That does very little to the noise between 20KHz and 50KHz.
The noise is at much lower levels in these frequency bands.


The noise starts rising from 10KHz, and is significant between 20KHz
and 50KHz. Check out some player measurements:

http://www.stereophile.com/digitalso...06/index5.html

Noise is 30 dB higher than 24 bit LPCM at 20 KHz.
=20
Oh piddle - that means the noise is only 114 dB below full-scale at
20 kHz. You know, we just might hear that...

Actually the noise in a 1/3 octave BW centered at 20KHz is about 87 dB=20
below FS, according to that plot. Not that it is necessarily audible,=20
but the possibility exists that the sharply rising noise could cause=20
problems with some amps.

"François Yves Le Gal" wrote in message
...
On Mon, 02 Feb 2004 16:52:57 GMT, "Karl Uppiano"

wrote:

I believe you. I realize it isn't considered state of the art anymore,
but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

Sample rate? FM is 100% analog, no smpale rate, but a carrier frequency
modulated in frequency.

Nope, FM stereo is a sampled system. It's analog, but it's definitely
sampled. Left and right sampled alternately at 38 kHz is the same as L+R and
L-R fed into a balanced modulator with a suppressed carrier. The balanced
modulator approach was used in FM stereo generators until the mid 1970's or
so, when the alternately sampled approach became possible with the advent of
TTL and FET analog switches. Both systems also inject a 19 kHz pilot tone at
a 9% modulation level to turn on the "stereo" light, and to give
phase-lock-loop stereo decoders (also sampled systems) something to lock
onto.

"François Yves Le Gal" wrote in message
...
On Mon, 02 Feb 2004 16:52:57 GMT, "Karl Uppiano"

wrote:

I believe you. I realize it isn't considered state of the art anymore,
but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

Sample rate? FM is 100% analog, no smpale rate, but a carrier frequency
modulated in frequency.

Nope, FM stereo is a sampled system. It's analog, but it's definitely
sampled. Left and right sampled alternately at 38 kHz is the same as L+R and
L-R fed into a balanced modulator with a suppressed carrier. The balanced
modulator approach was used in FM stereo generators until the mid 1970's or
so, when the alternately sampled approach became possible with the advent of
TTL and FET analog switches. Both systems also inject a 19 kHz pilot tone at
a 9% modulation level to turn on the "stereo" light, and to give
phase-lock-loop stereo decoders (also sampled systems) something to lock
onto.

"François Yves Le Gal" wrote in message
...
On Mon, 02 Feb 2004 16:52:57 GMT, "Karl Uppiano"

wrote:

I believe you. I realize it isn't considered state of the art anymore,
but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

Sample rate? FM is 100% analog, no smpale rate, but a carrier frequency
modulated in frequency.

Nope, FM stereo is a sampled system. It's analog, but it's definitely
sampled. Left and right sampled alternately at 38 kHz is the same as L+R and
L-R fed into a balanced modulator with a suppressed carrier. The balanced
modulator approach was used in FM stereo generators until the mid 1970's or
so, when the alternately sampled approach became possible with the advent of
TTL and FET analog switches. Both systems also inject a 19 kHz pilot tone at
a 9% modulation level to turn on the "stereo" light, and to give
phase-lock-loop stereo decoders (also sampled systems) something to lock
onto.

"François Yves Le Gal" wrote in message
...
On Mon, 02 Feb 2004 16:52:57 GMT, "Karl Uppiano"

wrote:

I believe you. I realize it isn't considered state of the art anymore,
but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

Sample rate? FM is 100% analog, no smpale rate, but a carrier frequency
modulated in frequency.

Nope, FM stereo is a sampled system. It's analog, but it's definitely
sampled. Left and right sampled alternately at 38 kHz is the same as L+R and
L-R fed into a balanced modulator with a suppressed carrier. The balanced
modulator approach was used in FM stereo generators until the mid 1970's or
so, when the alternately sampled approach became possible with the advent of
TTL and FET analog switches. Both systems also inject a 19 kHz pilot tone at
a 9% modulation level to turn on the "stereo" light, and to give
phase-lock-loop stereo decoders (also sampled systems) something to lock
onto.

"Randy Yates" wrote in message
...
"Karl Uppiano" writes:

"Arny Krueger" wrote in message
...
"Karl Uppiano" wrote in message

"François Yves Le Gal" wrote in message
...
On Sun, 01 Feb 2004 12:53:18 -0800, chung
wrote:

One, DSD/SACD proponents
claim the much wider bandwidth over CD's, and filtering will reduce
significantly that claimed advantage.

Having a gentle low pass filter at 60 or 100 KHz doesn't
significantly reduce SACD's bandwith. You get more than PCM 96 or
192 in both cases!

I would start a 3 dB/octave rolloff at 20kHz or so.

Ironically, most if not all people can't hear the difference a
brickwall
filter at 16 KHz makes, if the filter is well-designed.

Don't believe me?

Listen for yourself at
http://www.pcabx.com/technical/low_pass/index.htm .

I believe you. I realize it isn't considered state of the art anymore,
but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and
a 19 kHz pilot tone added. This, along with the L+R baseband signal, is
then FM modulated onto the carrier.

The balanced modulator is mathematically equivalent to a sampled system in
which the left and right channels are alternately sampled at a 38 kHz rate
and used to directly modulate the carrier. The 19 kHz pilot is derived from
the 38 kHz sample frequency and injected onto the carrier at 9% modulation.
Even without the math, it makes sense if you think about it: Alternately
sampling a mono signal (L = R) will give you no subcarrier (L - R = 0),
which is exactly what happens with AM-DSB-SC. The analysis is a little more
complicated when a stereo signal is encoded, but the results are identical
using either approach.

The balanced modulator approach was used in FM stereo generators until the
mid 1970's or so, when the alternately sampled approach became possible with
the advent of TTL and FET analog switches. The switched stereo generators
require less maintenance. The phase-lock-loop demodulators in most receivers
is the same system in reverse.

Nyquist's sampling theorem applies to FM stereo. Therefore, 19 kHz is the
highest audio frequency theoretically possible, but you need a guard band
for the pilot, so 15 kHz is typical, with some extremely good stereo
generators yielding 16 to 18 kHz.

"Randy Yates" wrote in message
...
"Karl Uppiano" writes:

"Arny Krueger" wrote in message
...
"Karl Uppiano" wrote in message

"François Yves Le Gal" wrote in message
...
On Sun, 01 Feb 2004 12:53:18 -0800, chung
wrote:

One, DSD/SACD proponents
claim the much wider bandwidth over CD's, and filtering will reduce
significantly that claimed advantage.

Having a gentle low pass filter at 60 or 100 KHz doesn't
significantly reduce SACD's bandwith. You get more than PCM 96 or
192 in both cases!

I would start a 3 dB/octave rolloff at 20kHz or so.

Ironically, most if not all people can't hear the difference a
brickwall
filter at 16 KHz makes, if the filter is well-designed.

Don't believe me?

Listen for yourself at
http://www.pcabx.com/technical/low_pass/index.htm .

I believe you. I realize it isn't considered state of the art anymore,
but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and
a 19 kHz pilot tone added. This, along with the L+R baseband signal, is
then FM modulated onto the carrier.

The balanced modulator is mathematically equivalent to a sampled system in
which the left and right channels are alternately sampled at a 38 kHz rate
and used to directly modulate the carrier. The 19 kHz pilot is derived from
the 38 kHz sample frequency and injected onto the carrier at 9% modulation.
Even without the math, it makes sense if you think about it: Alternately
sampling a mono signal (L = R) will give you no subcarrier (L - R = 0),
which is exactly what happens with AM-DSB-SC. The analysis is a little more
complicated when a stereo signal is encoded, but the results are identical
using either approach.

The balanced modulator approach was used in FM stereo generators until the
mid 1970's or so, when the alternately sampled approach became possible with
the advent of TTL and FET analog switches. The switched stereo generators
require less maintenance. The phase-lock-loop demodulators in most receivers
is the same system in reverse.

Nyquist's sampling theorem applies to FM stereo. Therefore, 19 kHz is the
highest audio frequency theoretically possible, but you need a guard band
for the pilot, so 15 kHz is typical, with some extremely good stereo
generators yielding 16 to 18 kHz.

"Randy Yates" wrote in message
...
"Karl Uppiano" writes:

"Arny Krueger" wrote in message
...
"Karl Uppiano" wrote in message

"François Yves Le Gal" wrote in message
...
On Sun, 01 Feb 2004 12:53:18 -0800, chung
wrote:

One, DSD/SACD proponents
claim the much wider bandwidth over CD's, and filtering will reduce
significantly that claimed advantage.

Having a gentle low pass filter at 60 or 100 KHz doesn't
significantly reduce SACD's bandwith. You get more than PCM 96 or
192 in both cases!

I would start a 3 dB/octave rolloff at 20kHz or so.

Ironically, most if not all people can't hear the difference a
brickwall
filter at 16 KHz makes, if the filter is well-designed.

Don't believe me?

Listen for yourself at
http://www.pcabx.com/technical/low_pass/index.htm .

I believe you. I realize it isn't considered state of the art anymore,
but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and
a 19 kHz pilot tone added. This, along with the L+R baseband signal, is
then FM modulated onto the carrier.

The balanced modulator is mathematically equivalent to a sampled system in
which the left and right channels are alternately sampled at a 38 kHz rate
and used to directly modulate the carrier. The 19 kHz pilot is derived from
the 38 kHz sample frequency and injected onto the carrier at 9% modulation.
Even without the math, it makes sense if you think about it: Alternately
sampling a mono signal (L = R) will give you no subcarrier (L - R = 0),
which is exactly what happens with AM-DSB-SC. The analysis is a little more
complicated when a stereo signal is encoded, but the results are identical
using either approach.

The balanced modulator approach was used in FM stereo generators until the
mid 1970's or so, when the alternately sampled approach became possible with
the advent of TTL and FET analog switches. The switched stereo generators
require less maintenance. The phase-lock-loop demodulators in most receivers
is the same system in reverse.

Nyquist's sampling theorem applies to FM stereo. Therefore, 19 kHz is the
highest audio frequency theoretically possible, but you need a guard band
for the pilot, so 15 kHz is typical, with some extremely good stereo
generators yielding 16 to 18 kHz.

"Randy Yates" wrote in message
...
"Karl Uppiano" writes:

"Arny Krueger" wrote in message
...
"Karl Uppiano" wrote in message

"François Yves Le Gal" wrote in message
...
On Sun, 01 Feb 2004 12:53:18 -0800, chung
wrote:

One, DSD/SACD proponents
claim the much wider bandwidth over CD's, and filtering will reduce
significantly that claimed advantage.

Having a gentle low pass filter at 60 or 100 KHz doesn't
significantly reduce SACD's bandwith. You get more than PCM 96 or
192 in both cases!

I would start a 3 dB/octave rolloff at 20kHz or so.

Ironically, most if not all people can't hear the difference a
brickwall
filter at 16 KHz makes, if the filter is well-designed.

Don't believe me?

Listen for yourself at
http://www.pcabx.com/technical/low_pass/index.htm .

I believe you. I realize it isn't considered state of the art anymore,
but I
wonder how many people realize that FM stereo uses a 38 kHz sample rate?

Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and
a 19 kHz pilot tone added. This, along with the L+R baseband signal, is
then FM modulated onto the carrier.

The balanced modulator is mathematically equivalent to a sampled system in
which the left and right channels are alternately sampled at a 38 kHz rate
and used to directly modulate the carrier. The 19 kHz pilot is derived from
the 38 kHz sample frequency and injected onto the carrier at 9% modulation.
Even without the math, it makes sense if you think about it: Alternately
sampling a mono signal (L = R) will give you no subcarrier (L - R = 0),
which is exactly what happens with AM-DSB-SC. The analysis is a little more
complicated when a stereo signal is encoded, but the results are identical
using either approach.

The balanced modulator approach was used in FM stereo generators until the
mid 1970's or so, when the alternately sampled approach became possible with
the advent of TTL and FET analog switches. The switched stereo generators
require less maintenance. The phase-lock-loop demodulators in most receivers
is the same system in reverse.

Nyquist's sampling theorem applies to FM stereo. Therefore, 19 kHz is the
highest audio frequency theoretically possible, but you need a guard band
for the pilot, so 15 kHz is typical, with some extremely good stereo
generators yielding 16 to 18 kHz.

"Karl Uppiano" wrote in message
...

"Randy Yates" wrote in message
...
"Karl Uppiano" writes:

"Arny Krueger" wrote in message
...
"Karl Uppiano" wrote in message

"François Yves Le Gal" wrote in message
...
On Sun, 01 Feb 2004 12:53:18 -0800, chung
wrote:

One, DSD/SACD proponents
claim the much wider bandwidth over CD's, and filtering will
reduce
significantly that claimed advantage.

Having a gentle low pass filter at 60 or 100 KHz doesn't
significantly reduce SACD's bandwith. You get more than PCM 96 or
192 in both cases!

I would start a 3 dB/octave rolloff at 20kHz or so.

Ironically, most if not all people can't hear the difference a
brickwall
filter at 16 KHz makes, if the filter is well-designed.

Don't believe me?

Listen for yourself at
http://www.pcabx.com/technical/low_pass/index.htm .

I believe you. I realize it isn't considered state of the art anymore,
but I
wonder how many people realize that FM stereo uses a 38 kHz sample
rate?

Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier
and
a 19 kHz pilot tone added. This, along with the L+R baseband signal, is
then FM modulated onto the carrier.

The balanced modulator is mathematically equivalent to a sampled system in
which the left and right channels are alternately sampled at a 38 kHz rate
and used to directly modulate the carrier. The 19 kHz pilot is derived
from
the 38 kHz sample frequency and injected onto the carrier at 9%
modulation.
Even without the math, it makes sense if you think about it: Alternately
sampling a mono signal (L = R) will give you no subcarrier (L - R = 0),
which is exactly what happens with AM-DSB-SC. The analysis is a little
more
complicated when a stereo signal is encoded, but the results are identical
using either approach.

The balanced modulator approach was used in FM stereo generators until the
mid 1970's or so, when the alternately sampled approach became possible
with
the advent of TTL and FET analog switches. The switched stereo generators
require less maintenance. The phase-lock-loop demodulators in most
receivers
is the same system in reverse.

Nyquist's sampling theorem applies to FM stereo. Therefore, 19 kHz is the
highest audio frequency theoretically possible, but you need a guard band
for the pilot, so 15 kHz is typical, with some extremely good stereo
generators yielding 16 to 18 kHz.

By the way, the transition band for the analog anti-aliasing filters
(required for any FM stereo generator) need to go from 0 dB at 15 kHz to -60
dB at 19 kHz. Stereo decoders in most receivers usually don't bother to
control the high frequency energy very well, which is why many cassette
decks with Dolby noise reduction have "MPX filters" to prevent this energy
from messing up the Dolby encoding.

"Karl Uppiano" wrote in message
...

"Randy Yates" wrote in message
...
"Karl Uppiano" writes:

"Arny Krueger" wrote in message
...
"Karl Uppiano" wrote in message

"François Yves Le Gal" wrote in message
...
On Sun, 01 Feb 2004 12:53:18 -0800, chung
wrote:

One, DSD/SACD proponents
claim the much wider bandwidth over CD's, and filtering will
reduce
significantly that claimed advantage.

Having a gentle low pass filter at 60 or 100 KHz doesn't
significantly reduce SACD's bandwith. You get more than PCM 96 or
192 in both cases!

I would start a 3 dB/octave rolloff at 20kHz or so.

Ironically, most if not all people can't hear the difference a
brickwall
filter at 16 KHz makes, if the filter is well-designed.

Don't believe me?

Listen for yourself at
http://www.pcabx.com/technical/low_pass/index.htm .

I believe you. I realize it isn't considered state of the art anymore,
but I
wonder how many people realize that FM stereo uses a 38 kHz sample
rate?

Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier
and
a 19 kHz pilot tone added. This, along with the L+R baseband signal, is
then FM modulated onto the carrier.

The balanced modulator is mathematically equivalent to a sampled system in
which the left and right channels are alternately sampled at a 38 kHz rate
and used to directly modulate the carrier. The 19 kHz pilot is derived
from
the 38 kHz sample frequency and injected onto the carrier at 9%
modulation.
Even without the math, it makes sense if you think about it: Alternately
sampling a mono signal (L = R) will give you no subcarrier (L - R = 0),
which is exactly what happens with AM-DSB-SC. The analysis is a little
more
complicated when a stereo signal is encoded, but the results are identical
using either approach.

The balanced modulator approach was used in FM stereo generators until the
mid 1970's or so, when the alternately sampled approach became possible
with
the advent of TTL and FET analog switches. The switched stereo generators
require less maintenance. The phase-lock-loop demodulators in most
receivers
is the same system in reverse.

Nyquist's sampling theorem applies to FM stereo. Therefore, 19 kHz is the
highest audio frequency theoretically possible, but you need a guard band
for the pilot, so 15 kHz is typical, with some extremely good stereo
generators yielding 16 to 18 kHz.

By the way, the transition band for the analog anti-aliasing filters
(required for any FM stereo generator) need to go from 0 dB at 15 kHz to -60
dB at 19 kHz. Stereo decoders in most receivers usually don't bother to
control the high frequency energy very well, which is why many cassette
decks with Dolby noise reduction have "MPX filters" to prevent this energy
from messing up the Dolby encoding.

"Karl Uppiano" wrote in message
...

"Randy Yates" wrote in message
...
"Karl Uppiano" writes:

"Arny Krueger" wrote in message
...
"Karl Uppiano" wrote in message

"François Yves Le Gal" wrote in message
...
On Sun, 01 Feb 2004 12:53:18 -0800, chung
wrote:

One, DSD/SACD proponents
claim the much wider bandwidth over CD's, and filtering will
reduce
significantly that claimed advantage.

Having a gentle low pass filter at 60 or 100 KHz doesn't
significantly reduce SACD's bandwith. You get more than PCM 96 or
192 in both cases!

I would start a 3 dB/octave rolloff at 20kHz or so.

Ironically, most if not all people can't hear the difference a
brickwall
filter at 16 KHz makes, if the filter is well-designed.

Don't believe me?

Listen for yourself at
http://www.pcabx.com/technical/low_pass/index.htm .

I believe you. I realize it isn't considered state of the art anymore,
but I
wonder how many people realize that FM stereo uses a 38 kHz sample
rate?

Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier
and
a 19 kHz pilot tone added. This, along with the L+R baseband signal, is
then FM modulated onto the carrier.

The balanced modulator is mathematically equivalent to a sampled system in
which the left and right channels are alternately sampled at a 38 kHz rate
and used to directly modulate the carrier. The 19 kHz pilot is derived
from
the 38 kHz sample frequency and injected onto the carrier at 9%
modulation.
Even without the math, it makes sense if you think about it: Alternately
sampling a mono signal (L = R) will give you no subcarrier (L - R = 0),
which is exactly what happens with AM-DSB-SC. The analysis is a little
more
complicated when a stereo signal is encoded, but the results are identical
using either approach.

The balanced modulator approach was used in FM stereo generators until the
mid 1970's or so, when the alternately sampled approach became possible
with
the advent of TTL and FET analog switches. The switched stereo generators
require less maintenance. The phase-lock-loop demodulators in most
receivers
is the same system in reverse.

Nyquist's sampling theorem applies to FM stereo. Therefore, 19 kHz is the
highest audio frequency theoretically possible, but you need a guard band
for the pilot, so 15 kHz is typical, with some extremely good stereo
generators yielding 16 to 18 kHz.

By the way, the transition band for the analog anti-aliasing filters
(required for any FM stereo generator) need to go from 0 dB at 15 kHz to -60
dB at 19 kHz. Stereo decoders in most receivers usually don't bother to
control the high frequency energy very well, which is why many cassette
decks with Dolby noise reduction have "MPX filters" to prevent this energy
from messing up the Dolby encoding.