[Radium]

Hi:

Here are my rules for digital audio:

A. Whether compressed or not, the audio must be monoaural and with a
sample-rate of at least 44.1 khz.

B. The only compression allowed is WMA. No other compression format is
permitted.

C. In its uncompressed form, the audio must have a bit-resolution of at
least 16-bit

D. If compression is used, then the sample-rate of the compressed and
the uncompressed version of the audio must be the same.

E. If compression is used, the only thing that should be decreased is
the bit-resolution. The sample-rate must remain unchanged

Lets say a song that was originally recorded in stereo is given to me.
The song must to be converted to mono* via the following steps:

1. Record audio from CD [or other stereo audio source] into Wavelab,
Adobe Audition [or other audio software] into a file. For simplicity
lets call this file "Track1.wav"

2. Make a copy of Track1.wav and save the copy as "Track1B.wav"

3. Open Track1.wav and reduce the gain of its audio by 77.5%

4. Convert Track1.wav to monoaural audio

5. Save Track.1

6. Open Track1B.wav and reduce its audio gain by 50%

7. Invert the phase of the left channel of Track1B.wav

8. Convert Track1B.wav to mono

9. Save Track1B.wav

10. Create a new stereo wave file whose bit-resolution is 16-bit and
sample rate is 44.1 khz. For simplicity lets call this file
"untitled.wav"

11. Copy and paste the audio of Track1.wav into the left channel of
untitled.wav

12. Copy and paste the audio of Track1B.wave into the right channel of
untitled.wav

13. Convert untitled.wav to mono

14. Save untitled.wav

*Songs that were originally-recorded in stereo need to be converted to
mono via the above 14 steps because different sounds are recorded
differently in the L and R channels. The audio that is originally
panned to the center is significantly louder than the audio whose phase
is different in the
left & right channels. This is why, I reduce the loudness of
non-inverted stereo audio file by 77.5% [before converting it to mono].
In the stereo file whose left channel has its phase inverted, I
decrease the loudness only by 50% and then convert it to mono. Usually
-- the lead vocals, bass, and percussion are recorded identically in
both the left and right channels. The paino, chorus, guitar, and synth
pads are usually recorded differently in the left and right channel.


Regards,

Radium

[Karl Uppiano]

....and this affects me how?

[Don Pearce]

On Sat, 11 Nov 2006 06:00:41 GMT, "Karl Uppiano"
wrote:

...and this affects me how?


Now, now. That was a very good set of rules designed to ensure that
nobody will ever try to steal his music. I applaud him, in fact, for
even being interested in music while stone deaf.

d

--
Pearce Consulting
http://www.pearce.uk.com

[Laurence Payne]

On 10 Nov 2006 21:40:05 -0800, "Radium" wrote:

Hi:

Here are my rules for digital audio:

Fine. Run along and play now.

[Radium]

Soundhaspriority wrote:
Dont forget FLAC -- "Free Lossless Audio Codec."
http://flac.sourceforge.net/

No thanks.

[[email protected]]

Radium wrote:
Soundhaspriority wrote:
Dont forget FLAC -- "Free Lossless Audio Codec."
http://flac.sourceforge.net/

No thanks.

Yeah, that's useful, fact-based information. Radium will
have none of that.

[Radium]

Here is the visual equivalent of the rules of my digital audio:

http://groups.google.com/group/sci.e...59e8739e79f0e3

I know its OT but I posted it anyway.

[Karl Uppiano]

"Radium" wrote in message
ups.com...
Soundhaspriority wrote:
Dont forget FLAC -- "Free Lossless Audio Codec."
http://flac.sourceforge.net/

No thanks.

Oh, so you do respond. I thought your original post might have been drive-by
trolling. So -- what are your rules for, and why should I care?

Do you have a particular goal in mind? For example, someone who is
interested in high quality stereo digital audio would find your rules
completely at odds with their goal.

[Radium]

Karl Uppiano wrote:
Do you have a particular goal in mind?

I am just expressing my thoughts on audio.

For example, someone who is
interested in high quality stereo digital audio would find your rules
completely at odds with their goal.

If I am going to burn something into a CD, the software will
automatically convert it to stereo because CDs requires the audio to be
stereo. So, there you go.

Two additional steps to the 14 steps I listed in the first message of
this thread:

15. Convert untitled.wav back to stereo

16. Burn to CD!

Do you think the stereo-lovers would want this? After all, CDs requires
their audio have two channels, otherwise it is not compatible with the
CD.

[Karl Uppiano]

"Radium" wrote in message
ups.com...
Karl Uppiano wrote:
Do you have a particular goal in mind?

I am just expressing my thoughts on audio.

Ok. But I prefer FLAC. What do you think of that?

For example, someone who is
interested in high quality stereo digital audio would find your rules
completely at odds with their goal.

If I am going to burn something into a CD, the software will
automatically convert it to stereo because CDs requires the audio to be
stereo. So, there you go.

If it is automatic, why do I have to think about it? The software developer
has to think about it. Are your rules for the software developer?

Two additional steps to the 14 steps I listed in the first message of
this thread:

15. Convert untitled.wav back to stereo

16. Burn to CD!

Do you think the stereo-lovers would want this? After all, CDs requires
their audio have two channels, otherwise it is not compatible with the
CD.

Stereo lovers probably want the original stereo, if it is available
anywhere. If they are burning a mono WAV to a CD, then sure, the software
needs to convert it. But it isn't exactly stereo, just dual mono. An equally
valid approach would be to send mono to the left channel and leave the right
channel blank. Or put a completely different program on the right channel.
That would be most efficient, but probably not compatible with stereo
headphones without a switch for left or right mono. I think the CD Audio
format actually supports a monophonic mode, but I do not think it was ever
used. I don't know if a player even has to implement it to be logo-compliant
anymore.

[Eeyore]

Radium wrote:

Hi:

Here are my rules for digital audio:

Radium, you're a lunatic.

No-one gives a damn for your roolz.

Graham

[Eeyore]

Radium wrote:

*Songs that were originally-recorded in stereo need to be converted to
mono via the above 14 steps because different sounds are recorded
differently in the L and R channels. The audio that is originally
panned to the center is significantly louder than the audio whose phase
is different in the
left & right channels. This is why, I reduce the loudness of
non-inverted stereo audio file by 77.5% [before converting it to mono].
In the stereo file whose left channel has its phase inverted, I
decrease the loudness only by 50% and then convert it to mono. Usually
-- the lead vocals, bass, and percussion are recorded identically in
both the left and right channels. The paino, chorus, guitar, and synth
pads are usually recorded differently in the left and right channel.

Funny that most ppl are happy to press the mono button then !

Graham

[Eeyore]

Radium wrote:

Karl Uppiano wrote:
Do you have a particular goal in mind?

I am just expressing my thoughts on audio.

If we shake you can we hear your brain rattle inside your skull ?

Graham

[Kalman Rubinson]

Sigh. I thought I had to ignore Radium only on the neuroscience
newsgroups. He is notorious.

Kal


On Sat, 11 Nov 2006 22:34:22 +0000, Eeyore
wrote:



Radium wrote:

Karl Uppiano wrote:
Do you have a particular goal in mind?

I am just expressing my thoughts on audio.

If we shake you can we hear your brain rattle inside your skull ?

Graham

[George M. Middius]

Poopie is on the rag again.

Radium, you're a lunatic.
No-one gives a damn for your roolz.

Could this attitude possibly be the reason you have your own personal
stalker?





--

Krooscience: The antidote to education, experience, and excellence.

[MiNe 109]

In article ,
Kalman Rubinson wrote:

Sigh. I thought I had to ignore Radium only on the neuroscience
newsgroups. He is notorious.

Thanks for the info.

Stephen

[Geoff]

Radium wrote:
Hi:

Here are my rules for digital audio:

A. Whether compressed or not, the audio must be monoaural and with a
sample-rate of at least 44.1 khz.



We are all very pleased for you.

geoff

[Randy Yates]

"Radium" writes:
[...]
3. Open Track1.wav and reduce the gain of its audio by 77.5%

That will increase the quantization noise by approximately 13 dB.
This is a bad idea.

It is also a bad idea to attempt to "help" the codec compress your
music by this step (which I believe is your goal). A lot of smart
folks have spent a lot of time determining how to best compress two
stereo tracks and I doubt your scheme can do better.
--
% Randy Yates % "Ticket to the moon, flight leaves here today
%% Fuquay-Varina, NC % from Satellite 2"
%%% 919-577-9882 % 'Ticket To The Moon'
%%%% % *Time*, Electric Light Orchestra
http://home.earthlink.net/~yatescr

[Karl Uppiano]

Ok, I'm going to top-post here, because my responses are inline, below. I am
still trying to understand exactly what Radium thinks these rules are
supposed to accomplish.

"Radium" wrote in message
oups.com...
Hi:

Here are my rules for digital audio:

Digital audio for high fidelity? Elevator background music? Studio
mastering? Long distance land line telephone signals? Cellular telephone
data? I-Pod? Archiving to long-term storage? I will submit to you that each
application has remarkably different requirements (and different rules for
meeting those requirements).

A. Whether compressed or not, the audio must be monoaural and with a
sample-rate of at least 44.1 khz.

Why monaural? Suppose I like stereo? A sample rate of 44.1 or higher will
give you 20KHz audio bandwidth. That's nice for hi-fi listening, but may be
more than you need for "You Tube" sound tracks.

B. The only compression allowed is WMA. No other compression format is
permitted.

I like FLAC. What about me? You might not have been aware of this, but
everything is all about me.

C. In its uncompressed form, the audio must have a bit-resolution of at
least 16-bit

Okay. What about dither? Does it need to be dithered? I think it needs to be
dithered at 2/3 LSB (that's my rule).

D. If compression is used, then the sample-rate of the compressed and
the uncompressed version of the audio must be the same.

Even if the compressed and uncompressed versions reside in different zip
codes?

E. If compression is used, the only thing that should be decreased is
the bit-resolution. The sample-rate must remain unchanged

I assume that by this you mean you do not want to reduce the bit rate by
reducing the sample rate, but only by means of bit allocation using a
perceptual coder.

Lets say a song that was originally recorded in stereo is given to me.
The song must to be converted to mono* via the following steps:

1. Record audio from CD [or other stereo audio source] into Wavelab,
Adobe Audition [or other audio software] into a file. For simplicity
lets call this file "Track1.wav"

2. Make a copy of Track1.wav and save the copy as "Track1B.wav"

3. Open Track1.wav and reduce the gain of its audio by 77.5%

4. Convert Track1.wav to monoaural audio

In-phase signals from left and right channels will increase by 6dB when you
sum them. In order to avoid clipping if left and right channels are full
scale, you would need to reduce the level by 50% You said reduce *by* 77.5%,
so I assume you mean drop the level *to* 22.5%. That is too much level
reduction. On the other hand, if you meant drop the level to 77.5% that is
not enough. Besides, the "convert to mono" algorithm might scale the result
for you automatically, in which case any further level correction would be
redundant.

5. Save Track.1

6. Open Track1B.wav and reduce its audio gain by 50%

7. Invert the phase of the left channel of Track1B.wav

8. Convert Track1B.wav to mono

So now you have a sum channel (0.225R + 0.225L) and a difference channel
(0.5R - 0.5L).

9. Save Track1B.wav

10. Create a new stereo wave file whose bit-resolution is 16-bit and
sample rate is 44.1 khz. For simplicity lets call this file
"untitled.wav"

11. Copy and paste the audio of Track1.wav into the left channel of
untitled.wav

12. Copy and paste the audio of Track1B.wave into the right channel of
untitled.wav

Weird, but keep going...

13. Convert untitled.wav to mono

You just removed most of the left channel and flipped its phase (0.5R -
0.5L) + (0.225R + 0.225L) = (0.725R - 0.275L). The right channel is 2.1dB
lower than the original; the left channel is 11dB lower than the original,
and "upside-down".

14. Save untitled.wav

If it was mono you wanted, you had it at step 4. If you really wanted
(0.725R - 0.275L), you could have done that all in four steps: Reduce the
right channel by 72.5%, reduce the left channel by 27.5%, flip the phase on
the left channel, and convert to mono. Try that and see if you don't get the
identical results you got with your 14-step plan.

*Songs that were originally-recorded in stereo need to be converted to
mono via the above 14 steps because different sounds are recorded
differently in the L and R channels. The audio that is originally
panned to the center is significantly louder than the audio whose phase
is different in the
left & right channels. This is why, I reduce the loudness of
non-inverted stereo audio file by 77.5% [before converting it to mono].
In the stereo file whose left channel has its phase inverted, I
decrease the loudness only by 50% and then convert it to mono. Usually
-- the lead vocals, bass, and percussion are recorded identically in
both the left and right channels. The paino, chorus, guitar, and synth
pads are usually recorded differently in the left and right channel.

People have been mixing down to mono from stereo for 50 years or more. You
simply add the left and right channels. Listening in stereo in a room
actually does more or less the same thing too (left and right speakers
working in phase (panned to center) will sum 6dB higher in the room,
depending on the frequency, and where you're standing). Record producers mix
the stereo channels for the proper artistic balance in their professional
opinion. Mixing down to mono should not be a problem.

[Radium]

Randy Yates wrote:
That will increase the quantization noise by approximately 13 dB.

Not as far as I can perceive.

[Radium]

Karl Uppiano wrote:
Digital audio for high fidelity?

Digital audio for any application.

Why monaural?

Because I want both the L and R channels to sound the same.

Suppose I like stereo?

A sample rate of 44.1 or higher will
give you 20KHz audio bandwidth. That's nice for hi-fi listening, but may be
more than you need for "You Tube" sound tracks.

Any digital audio requires 44.1 khz or higher in order to sound
pleasant. Aliasing can be a real earsore.

Okay. What about dither? Does it need to be dithered? I think it needs to be
dithered at 2/3 LSB (that's my rule).

No need for dither.

Even if the compressed and uncompressed versions reside in different zip
codes?

Of course. What do zip codes have to do with this?

I assume that by this you mean you do not want to reduce the bit rate by
reducing the sample rate, but only by means of bit allocation using a
perceptual coder.

Exactly.

In-phase signals from left and right channels will increase by 6dB when you
sum them. In order to avoid clipping if left and right channels are full
scale, you would need to reduce the level by 50% You said reduce *by* 77.5%,
so I assume you mean drop the level *to* 22.5%.

You assume correctly.

If it was mono you wanted, you had it at step 4. If you really wanted
(0.725R - 0.275L), you could have done that all in four steps: Reduce the
right channel by 72.5%, reduce the left channel by 27.5%, flip the phase on
the left channel, and convert to mono. Try that and see if you don't get the
identical results you got with your 14-step plan.

The audio that was in the center channel [lead vocal, bass,
percussions] are too loud while the audio that was in the periphery
[paino, chours, guitar, synth-pads] aren't loud enough.

People have been mixing down to mono from stereo for 50 years or more. You
simply add the left and right channels. Listening in stereo in a room
actually does more or less the same thing too (left and right speakers
working in phase (panned to center) will sum 6dB higher in the room,
depending on the frequency, and where you're standing). Record producers mix
the stereo channels for the proper artistic balance in their professional
opinion. Mixing down to mono should not be a problem.

My technique usually ensures that the sounds that were originally in
the central channel are not significantly louder than the sounds that
were originally in the periphery [and visa versa].

[Karl Uppiano]

"Radium" wrote in message
oups.com...
Karl Uppiano wrote:
Digital audio for high fidelity?

Digital audio for any application.

Bzzt! Nope. I will guarantee that your rules will not make any sense for
digital audio for telephony (for example). Or for my application (high
fidelity). Sorry, I would not do that to my FLAC files. My compressed
formats, where unavoidable, consist of WMA, AAC and MP3. I would not import
my files that way.

Why monaural?

Because I want both the L and R channels to sound the same.

Suppose I like stereo?

A sample rate of 44.1 or higher will
give you 20KHz audio bandwidth. That's nice for hi-fi listening, but may
be
more than you need for "You Tube" sound tracks.

Any digital audio requires 44.1 khz or higher in order to sound
pleasant. Aliasing can be a real earsore.

Done right, you can sample at any frequency without aliasing. The sample
rate only affects the bandwidth you can record. While I can understand
wanting full range audio for listening to music, it would be quite
inappropriate, and a big waste of bandiwdth to use 44.1KHz for telephony
(for example).

Okay. What about dither? Does it need to be dithered? I think it needs to
be
dithered at 2/3 LSB (that's my rule).

No need for dither.

Dither eliminates the distortion due to quantization errors present in any
digital system. I feel that there is a need for dither in high quality
applications.

Even if the compressed and uncompressed versions reside in different zip
codes?

Of course. What do zip codes have to do with this?

I was being facetious. There are compressed and uncompressed versions of all
sorts of things all over the world at many different sample rates. They are
not all going to follow your rules. Perhaps I was taking you too literally.

I assume that by this you mean you do not want to reduce the bit rate by
reducing the sample rate, but only by means of bit allocation using a
perceptual coder.

Exactly.

In-phase signals from left and right channels will increase by 6dB when
you
sum them. In order to avoid clipping if left and right channels are full
scale, you would need to reduce the level by 50% You said reduce *by*
77.5%,
so I assume you mean drop the level *to* 22.5%.

You assume correctly.

If it was mono you wanted, you had it at step 4. If you really wanted
(0.725R - 0.275L), you could have done that all in four steps: Reduce the
right channel by 72.5%, reduce the left channel by 27.5%, flip the phase
on
the left channel, and convert to mono. Try that and see if you don't get
the
identical results you got with your 14-step plan.

The audio that was in the center channel [lead vocal, bass,
percussions] are too loud while the audio that was in the periphery
[paino, chours, guitar, synth-pads] aren't loud enough.

I understand what you are trying to do. My point was that you were taking a
very complicated approach to arrive at what you describe as your end result.
I further said you could get the same result in far fewer steps.

People have been mixing down to mono from stereo for 50 years or more.
You
simply add the left and right channels. Listening in stereo in a room
actually does more or less the same thing too (left and right speakers
working in phase (panned to center) will sum 6dB higher in the room,
depending on the frequency, and where you're standing). Record producers
mix
the stereo channels for the proper artistic balance in their professional
opinion. Mixing down to mono should not be a problem.

My technique usually ensures that the sounds that were originally in
the central channel are not significantly louder than the sounds that
were originally in the periphery [and visa versa].

I am not convinced that your technique accomplishes that goal. I won't deny
that it will change the sound. It might even sound better to you in certain
limited cases.

I will say it a different way: There are millions of hours of AM, FM and TV
broadcasts that simply sum L + R for mono receivers. Are you saying that
everyone has got it wrong for lo these many years?

[Eeyore]

"George M. Middius" wrote:

Poopie is on the rag again.

Radium, you're a lunatic.
No-one gives a damn for your roolz.

Could this attitude possibly be the reason you have your own personal
stalker?

Bertei ?

No, that's something else entirely.

Graham

[Eeyore]

Radium wrote:

Randy Yates wrote:
That will increase the quantization noise by approximately 13 dB.

Not as far as I can perceive.

You can't 'perceive' fact from fiction or even fantasy, so no surprise there.

Graham

[Eeyore]

Radium wrote:

No need for dither.

CRETIN !

[mrlefty]

Who would want lossless compression when you can have lossy "wma"s !?

Have any of Radium's phase cancelled S/N reduced mono monstrosities been
posted anywhere? A listener A/B might be interesting.

mrlefty

wrote in message
ups.com...

Radium wrote:
Soundhaspriority wrote:
Dont forget FLAC -- "Free Lossless Audio Codec."
http://flac.sourceforge.net/

No thanks.

Yeah, that's useful, fact-based information. Radium will
have none of that.

[Radium]

Karl Uppiano wrote:
Are you saying that
everyone has got it wrong for lo these many years?

Not necessarily. However, most stereo-to-mono conversion involve simply
decreasing the amplitude level by 50% and then downmixing to mono. The
problem with this, is that the stuff that was identical in both
channels is much louder than the stuff that was different in each
channel.

[Radium]

mrlefty wrote:
Have any of Radium's phase cancelled S/N reduced mono monstrosities been
posted anywhere? A listener A/B might be interesting.

If you have a valid email address and are interested, I can send you
some songs that have been processed via my "steps"

[Karl Uppiano]

"Radium" wrote in message
oups.com...
Karl Uppiano wrote:
Are you saying that
everyone has got it wrong for lo these many years?

Not necessarily. However, most stereo-to-mono conversion involve simply
decreasing the amplitude level by 50% and then downmixing to mono. The
problem with this, is that the stuff that was identical in both
channels is much louder than the stuff that was different in each
channel.

Ok, well, have fun with your rules. I'm glad you're not running the
engineering standards group at the FCC or something where you could force
everyone to use them. Of course, it wouldn't be anything new, the government
making public policy having the force of law based on junk science.

[Randy Yates]

"Radium" writes:

Randy Yates wrote:
That will increase the quantization noise by approximately 13 dB.

Not as far as I can perceive.

Perception and reality are two different things.
--
% Randy Yates % "How's life on earth?
%% Fuquay-Varina, NC % ... What is it worth?"
%%% 919-577-9882 % 'Mission (A World Record)',
%%%% % *A New World Record*, ELO
http://home.earthlink.net/~yatescr

[Eeyore]

Radium wrote:

Karl Uppiano wrote:
Are you saying that
everyone has got it wrong for lo these many years?

Not necessarily. However, most stereo-to-mono conversion involve simply
decreasing the amplitude level by 50% and then downmixing to mono. The
problem with this, is that the stuff that was identical in both
channels is much louder than the stuff that was different in each
channel.

As was intended.

Graham

[Radium]

Eeyore wrote:
As was intended.

Not if the audio I'm listening to is music

[Walt]

Kalman Rubinson wrote:

Sigh. I thought I had to ignore Radium only on the neuroscience
newsgroups. He is notorious.

Oh. They let their vict^H^H^H^H clients post there too?


//Walt

[Walt]

Eeyore wrote:
Radium wrote:
Karl Uppiano wrote:

Are you saying that
everyone has got it wrong for lo these many years?

Not necessarily. However, most stereo-to-mono conversion involve simply
decreasing the amplitude level by 50% and then downmixing to mono. The
problem with this, is that the stuff that was identical in both
channels is much louder than the stuff that was different in each
channel.

As was intended.

Sorry, but I've got to side with Radium here. The center channel
build-up when taking a mono sum is a real phenomenon, and *not*
desirable or intentional.

It happens because taking a voltage sum of two signals increases the
level by 6db, not 3 db as you might expect. For example, if the
original stereo recording has three singers at equal volume panned hard
left, hard right, and hard center, summing to mono will make the guy in
the center 3 db louder than the other two. It was always thus.

So, what we have with Radium is a guy who likes mono (for whatever
reason - I'm not sure I want to know), but doesn't like how most stereo
programs sum to mono. So far, so good. Unfortunately his technique
doesn't come close to solving this problem - he gets .725(R) - .275(L)
not anything approaching a mono sum. But his problem is an
understandable one. (well, the sum-to-mono center channel buildup
problem at least. I'll refrain from commenting on the others)

What to do? Get used to listening in stereo? Write a signal
processing algorithm to compute a mono sum without the center channel
buildup? (maybe this has already been done?) Perform a mono sum the old
fashioned way by jamming a pencil eraser into one of your ears?

The possibilities are endless.

//Walt

[Randy Yates]

Walt writes:
[...]
For example, if the original stereo recording has three singers at
equal volume panned hard left, hard right, and hard center, summing
to mono will make the guy in the center 3 db louder than the other
two.

Shouldn't he have been 3 dB softer to begin with (in the stereo mix)?
--
% Randy Yates % "The dreamer, the unwoken fool -
%% Fuquay-Varina, NC % in dreams, no pain will kiss the brow..."
%%% 919-577-9882 %
%%%% % 'Eldorado Overture', *Eldorado*, ELO
http://home.earthlink.net/~yatescr

[Walt]

Randy Yates wrote:

Walt writes:

[...]
For example, if the original stereo recording has three singers at
equal volume panned hard left, hard right, and hard center, summing
to mono will make the guy in the center 3 db louder than the other
two.

Shouldn't he have been 3 dB softer to begin with (in the stereo mix)?

Yes and no. In order to sound like they're all three at the same level,
the center guy would be -3db in the left channel and -3db in the right
channel.

Say for the sake of the argument that guys on the outside are recorded
at a signal level of 0 dbu (.775 volts), that would mean the guy in the
center is -3dbu or 0.54837 volts. Do a mono sum and the guys on the
outside are still at .775 volts but the guy in the middle is now at
1.09674 volts, or 3db louder.

This little anomaly comes about because loudness as we perceive it is
proportional to the *square* of the voltage. It's called "center
channel buildup" and has been around for as long as we've been doing stereo.


//Walt

[Randy Yates]

Walt writes:

Randy Yates wrote:

Walt writes:

[...]
For example, if the original stereo recording has three singers at
equal volume panned hard left, hard right, and hard center, summing
to mono will make the guy in the center 3 db louder than the other
two.
Shouldn't he have been 3 dB softer to begin with (in the stereo mix)?

Yes and no. In order to sound like they're all three at the same
level, the center guy would be -3db in the left channel and -3db in
the right channel.

Why is that? In order for a signal s(t) to be perceived at the same
power, it should be split into s(t)/2 for the left and s(t)/2 for
the right. Then at the listening position it combines into

l(t) = s(t)/2 + s(t)/2
= s(t)

Thus the center guy should be 6 dB down (1/2 voltage) to sound the
same at the listening position. No?
--
% 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

[Randy Yates]

Randy Yates writes:

Walt writes:

Randy Yates wrote:

Walt writes:

[...]
For example, if the original stereo recording has three singers at
equal volume panned hard left, hard right, and hard center, summing
to mono will make the guy in the center 3 db louder than the other
two.
Shouldn't he have been 3 dB softer to begin with (in the stereo mix)?

Yes and no. In order to sound like they're all three at the same
level, the center guy would be -3db in the left channel and -3db in
the right channel.

Why is that? In order for a signal s(t) to be perceived at the same
power, it should be split into s(t)/2 for the left and s(t)/2 for
the right. Then at the listening position it combines into

l(t) = s(t)/2 + s(t)/2
= s(t)

Thus the center guy should be 6 dB down (1/2 voltage) to sound the
same at the listening position. No?

I should add that I believe the 3 dB/6 dB issue comes up as follows.

Let the left and right channel signals be denotes l(t) and r(t),
respectively. Also assume that l(t) and r(t) are zero-mean,
stationary signals, E[l(t)] = E[r(t)] = 0. Let them also have identical
power: E[l^2(t)] = E[r^2(t)] = P.

What is the power in their sum? We simply compute it as follows:

Psum = E[(l(t) + r(t))^2]
= E[l^2(t)] + 2*E[l(t)r(t)] + E[r^2(t)]
= 2*P + 2*E[l(t)r(t)].

If the left and right signals are completely uncorrelated, then
E[l(t)r(t)] = 0, and the sum power is 3 dB higher than the individual
channels (2*P).

If the left and right signals are perfectly correlated, then

E[l(t)r(t)] = E[l^2(t)] = E[r^2(t)]
= P

and therefore

Psum = 2*P + 2*P
= 4*P.

In this case the sum power is 6 dB (4*P) higher.
--
% Randy Yates % "She's sweet on Wagner-I think she'd die for Beethoven.
%% Fuquay-Varina, NC % She love the way Puccini lays down a tune, and
%%% 919-577-9882 % Verdi's always creepin' from her room."
%%%% % "Rockaria", *A New World Record*, ELO
http://home.earthlink.net/~yatescr

[Karl Uppiano]

"Walt" wrote in message
...
Eeyore wrote:
Radium wrote:
Karl Uppiano wrote:

Are you saying that
everyone has got it wrong for lo these many years?

Not necessarily. However, most stereo-to-mono conversion involve simply
decreasing the amplitude level by 50% and then downmixing to mono. The
problem with this, is that the stuff that was identical in both
channels is much louder than the stuff that was different in each
channel.

As was intended.

Sorry, but I've got to side with Radium here. The center channel build-up
when taking a mono sum is a real phenomenon, and *not* desirable or
intentional.

It happens because taking a voltage sum of two signals increases the level
by 6db, not 3 db as you might expect. For example, if the original
stereo recording has three singers at equal volume panned hard left, hard
right, and hard center, summing to mono will make the guy in the center 3
db louder than the other two. It was always thus.

So, what we have with Radium is a guy who likes mono (for whatever
reason - I'm not sure I want to know), but doesn't like how most stereo
programs sum to mono. So far, so good. Unfortunately his technique
doesn't come close to solving this problem - he gets .725(R) - .275(L) not
anything approaching a mono sum. But his problem is an understandable
one. (well, the sum-to-mono center channel buildup problem at least.
I'll refrain from commenting on the others)

What to do? Get used to listening in stereo? Write a signal processing
algorithm to compute a mono sum without the center channel buildup? (maybe
this has already been done?) Perform a mono sum the old fashioned way by
jamming a pencil eraser into one of your ears?

The possibilities are endless.

//Walt

I was thinking about this the other day, and it occurred to me that center
channel build-up is likely to be more of a problem with "fake" stereo --
multi solo tracks panned to their apparent position in the mix. A "real"
stereo performance, recorded live, with co-incident microphones probably
would not have this problem, although the performers at the center might be
louder due to their proximity to the microphone. That's probably one of the
reasons orchestras are often arranged in a semicircle.

I cannot think of a simple algebraic means to knock down the center channel,
without causing collateral damage to the un-correlated material in the left
and right channels.

[[email protected]]

Radium's "rules" are sheer made up nonsense. He may do this stuff but
there is no reason anyone else anywhere should follow suit.