Discussion:
[Sursound] LCR Upmix
Gianni Ricciardi
2004-01-07 21:20:08 UTC
Permalink
Hi, I'm trying to find a simple way to produce a disctrete LCR from a plain
stereo program (it's a cartoon, so music and dialogue are both present).
The target will be Home video on DVD, so I don't want use expensive gear to
do that, but I can't find a simple way to derive the mono content and put it
on the center channel (subtracting it from L and R).
Anyone of you knows if there's a trick to perform that in the digital domain
(possibly without phase distortion)?
I use Nuendo 2.01 Surround Edition.

Thanks a lot!

Gianni

Gianni Ricciardi
Sound design & Music production
[:: www.gianniricciardi.com ::]
Eberhard Sengpiel
2004-01-07 22:39:25 UTC
Permalink
Hi Gianni,

Don't disturb your L/R stereo picture. Therefore add the signal
L + R (that is mono), delay it by 20 milliseconds und put it with
a level of -6 db (to L or R) to the center loudspeaker. Than
there is some signal in the center loudspeaker that does not
narrow the stereo width.
There is an other trick. Do no (!) signal in the center loudspeaker
and everybody is believing that there must be a signal in that speaker.

Cheers

Eberhard Sengpiel
German forum of microphone recordings
and sound studio techniques
http://www.sengpielaudio.com


----- Original Message -----
From: Gianni Ricciardi
To: ***@music.vt.edu
Sent: Wednesday, January 07, 2004 10:20 PM
Subject: [Sursound] LCR Upmix

Hi, I'm trying to find a simple way to produce a disctrete LCR from a
plain stereo program (it's a cartoon, so music and dialogue are both
present).
The target will be Home video on DVD, so I don't want use expensive gear
to do that, but I can't find a simple way to derive the mono content and
put it on the center channel (subtracting it from L and R).
Anyone of you knows if there's a trick to perform that in the digital
domain (possibly without phase distortion)?
I use Nuendo 2.01 Surround Edition.

Thanks a lot!

Gianni

Gianni Ricciardi
Sound design & Music production
[:: www.gianniricciardi.com ::]
Eero Aro
2004-01-08 20:10:39 UTC
Permalink
The AGM ESsEX software is also marketed by Sascom:

http://www.sascom.com/AGM_ESsEX.html

Eero Aro
***@saunalahti.fi
Gianni Ricciardi
2004-01-07 23:42:45 UTC
Permalink
Thank Eberhard for your suggestions.

Yes, after many tries I do, I think I will not "destroy" the original L/R
program.
I wish to add something in the center (not only a phantom center ;o) so I
think I will add L+R as you said.
I've read about the trick to add delay to the C channel in order to preserve
the stereo picture (that elsewhere collapse at the center), but I try with
20 ms and it seems too much for me.. I can perceive it as a true delay
during dialogue.
But if I use smaller values (like 15 or 10) I'm afraid that the AC3 stereo
downmix will have phase problems, doesn't it? The minimum value for center
downmix in the AC3 metadata is -6 (it would be a total of -12db) so... what
do you think about this?

Another question: what do you mean with "put it with a level of -6 db (to L
or R) to the center loudspeaker" ?
You mean that I have to add some L or R in the center? Sorry I don't
understand this step.

Thank you, and sorry for my bad english!

GIANNI

-----Messaggio originale-----
Da: sursound-***@music.vt.edu
[mailto:sursound-***@music.vt.edu]Per conto di Eberhard Sengpiel
Inviato: mercoledì 7 gennaio 2004 23.39
A: Surround Sound discussion group
Oggetto: Re: [Sursound] LCR Upmix


Hi Gianni,

Don't disturb your L/R stereo picture. Therefore add the signal
L + R (that is mono), delay it by 20 milliseconds und put it with
a level of -6 db (to L or R) to the center loudspeaker. Than
there is some signal in the center loudspeaker that does not
narrow the stereo width.
There is an other trick. Do no (!) signal in the center loudspeaker
and everybody is believing that there must be a signal in that speaker.

Cheers

Eberhard Sengpiel
German forum of microphone recordings
and sound studio techniques
http://www.sengpielaudio.com


----- Original Message -----
From: Gianni Ricciardi
To: ***@music.vt.edu
Sent: Wednesday, January 07, 2004 10:20 PM
Subject: [Sursound] LCR Upmix

Hi, I'm trying to find a simple way to produce a disctrete LCR from a
plain stereo program (it's a cartoon, so music and dialogue are both
present).
The target will be Home video on DVD, so I don't want use expensive gear
to do that, but I can't find a simple way to derive the mono content and
put it on the center channel (subtracting it from L and R).
Anyone of you knows if there's a trick to perform that in the digital
domain (possibly without phase distortion)?
I use Nuendo 2.01 Surround Edition.

Thanks a lot!

Gianni

Gianni Ricciardi
Sound design & Music production
[:: www.gianniricciardi.com ::]
Eberhard Sengpiel
2004-01-08 08:22:18 UTC
Permalink
Hi Gianni,

it is really not critcal if you break up the stereo with the 15 to 25 ms
delayed L + R signal and put it into the center. The same is with the
level of this sum signal in relation to the L and R signal. 3 to 6 dB
down
will be sufficient. It will depend on the sort of signal you have.
You may give more level to the center if you take off some treble there.
Play arround and you willl see. My proposal is really no scientifical
solution. It is a practical and cheap one just right for your case.
I see no phase problems there. Play arround.

Cheers

Eberhard Sengpiel
German forum of microphone recordings
and sound studio techniques
http://www.sengpielaudio.com

----- Original Message -----
From: "Gianni Ricciardi" <***@gianniricciardi.com>
To: "Surround Sound discussion group" <***@music.vt.edu>
Sent: Thursday, January 08, 2004 12:42 AM
Subject: R: [Sursound] LCR Upmix


> Thank Eberhard for your suggestions.
>
> Yes, after many tries I do, I think I will not "destroy" the original
L/R
> program.
> I wish to add something in the center (not only a phantom center ;o)
so I
> think I will add L+R as you said.
> I've read about the trick to add delay to the C channel in order to
preserve
> the stereo picture (that elsewhere collapse at the center), but I try
with
> 20 ms and it seems too much for me.. I can perceive it as a true delay
> during dialogue.
> But if I use smaller values (like 15 or 10) I'm afraid that the AC3
stereo
> downmix will have phase problems, doesn't it? The minimum value for
center
> downmix in the AC3 metadata is -6 (it would be a total of -12db) so...
what
> do you think about this?
>
> Another question: what do you mean with "put it with a level of -6 db
(to L
> or R) to the center loudspeaker" ?
> You mean that I have to add some L or R in the center? Sorry I don't
> understand this step.
>
> Thank you, and sorry for my bad english!
>
> GIANNI
Richard G. Elen
2004-01-08 02:17:58 UTC
Permalink
----- Original Message -----
From: Gianni Ricciardi

Hi, I'm trying to find a simple way to produce a disctrete LCR from a plain
stereo program (it's a cartoon, so music and dialogue are both present).
The target will be Home video on DVD, so I don't want use expensive gear to
do that, but I can't find a simple way to derive the mono content and put it
on the center channel (subtracting it from L and R).
Anyone of you knows if there's a trick to perform that in the digital domain
(possibly without phase distortion)?

===

The ideal solution in my view would be to use Geoff Barton's Trifield system
to generate a CF channel that would be full co-ordinated with the LF and RF
to produce true 3-speaker stereo, with a number of advantages over two
speakers and without any drawbacks - except that I don't know what would be
the appropriate software to use (though I can think of two hardware
solutions, the AGM box and a Meridian processor).

Unfortunately these days some people complain about *music* mixes with no CF
content because they think something is broken or that you ought to put
something in there (even if it actually sounds better with a "virtual
centre" - note for example the nasty comments about Mark Linett's in my view
excellent mixes of Pet Sounds on DVD-A) so that is probably not an option.
The other possibility is to put a low-level L+R mix in the CF, maybe delay
it a bit to minimize its impact on the stereo image, but only if you think
people would dislike the effect of no CF information.

--Richard E
espen-b
2004-01-08 09:14:04 UTC
Permalink
>===== Original Message
espen-b
2004-01-08 09:16:50 UTC
Permalink
>----- Original Message -----
>From: Gianni Ricciardi
>
>Hi, I'm trying to find a simple way to produce a disctrete LCR from a plain
>stereo program (it's a cartoon, so music and dialogue are both present).
>The target will be Home video on DVD, so I don't want use expensive gear to
>do that, but I can't find a simple way to derive the mono content and put it
>on the center channel (subtracting it from L and R).

Why not process the stereo signal with Pro Logic II or Lexicon's Logic 7
decoding.


Esp1
espen-b
2004-01-08 09:16:49 UTC
Permalink
>----- Original Message -----
>From: Gianni Ricciardi
>
>Hi, I'm trying to find a simple way to produce a disctrete LCR from a plain
>stereo program (it's a cartoon, so music and dialogue are both present).
>The target will be Home video on DVD, so I don't want use expensive gear to
>do that, but I can't find a simple way to derive the mono content and put it
>on the center channel (subtracting it from L and R).

Why not process the stereo signal with Pro Logic II or Lexicon's Logic 7
decoding.


Esp1
Geoffrey Barton
2004-01-08 09:30:34 UTC
Permalink
<?xml version="1.0" ?><html>
<head>
<title></title>
</head>
<body>
<div align="left"><font face="Comic Sans MS"><span style="font-size:10pt">On 8 Jan 2004 at 2:17, Richard G. Elen wrote:</span></font></div>
<div align="left"><font face="Comic Sans MS" color="#0000ff"><span style="font-size:10pt"><i>&gt; </i></span></font></div>
<div align="left"><font face="Comic Sans MS" color="#0000ff"><span style="font-size:10pt"><i>&gt; The ideal solution in my view would be to use Geoff Barton's Trifield system</i></span></font></div>
<div align="left"><font face="Comic Sans MS" color="#0000ff"><span style="font-size:10pt"><i>&gt; to generate a CF channel that would be full co-ordinated with the LF and RF</i></span></font></div>
<div align="left"><font face="Comic Sans MS" color="#0000ff"><span style="font-size:10pt"><i>&gt; to produce true 3-speaker stereo, with a number of advantages over two</i></span></font></div>
<div align="left"><font face="Comic Sans MS" color="#0000ff"><span style="font-size:10pt"><i>&gt; speakers and without any drawbacks - except that I don't know what would be</i></span></font></div>
<div align="left"><font face="Comic Sans MS" color="#0000ff"><span style="font-size:10pt"><i>&gt; the </i></span></font><font face="Comic Sans MS" color="#ff0000"><span style="font-size:10pt"><i>appropriate
software </i></span></font><font face="Comic Sans MS" color="#0000ff"><span style="font-size:10pt"><i>to use (though I can think of two hardware</i></span></font></div>
<div align="left"><font face="Comic Sans MS" color="#0000ff"><span style="font-size:10pt"><i>&gt; solutions, the AGM box and a Meridian processor).</i></span></font></div>
<div align="left"><font face="Comic Sans MS" color="#0000ff"><span style="font-size:10pt"><i>&gt; </i></span></font></div>
<div align="left"><br/>
</div>
<div align="left"><font face="Comic Sans MS"><span style="font-size:10pt">see www.agmdigital.com for the 'essex' processor; this is a software version of the
Trifield process. Works with broadcast wav files.</span></font></div>
<div align="left"><br/>
</div>
<div align="left"><font face="Comic Sans MS"><span style="font-size:10pt">regds,</span></font></div>
<div align="left"><font face="Comic Sans MS"><span style="font-size:10pt">Geoffrey</span></font></div>
<div align="left"><br/>
</div>
<div align="left"></div>
</body>
</html>
VDOSH
2004-01-08 22:35:53 UTC
Permalink
It's easy and simple to do:
- do an MS decoding of the stereo file, then you'll have 3 channels, left, right and center.

You can do it very easily with Samplitude, send me a mail if you want that I explain the trick to you.

I don't use Nuendo, so I don't know how to do Mid-Side decoding with Nuendo.

cheers,
Viriato de Oliveira
----- Original Message -----
From: Gianni Ricciardi
To: ***@music.vt.edu
Sent: Wednesday, January 07, 2004 10:20 PM
Subject: [Sursound] LCR Upmix


Hi, I'm trying to find a simple way to produce a disctrete LCR from a plain stereo program (it's a cartoon, so music and dialogue are both present).
The target will be Home video on DVD, so I don't want use expensive gear to do that, but I can't find a simple way to derive the mono content and put it on the center channel (subtracting it from L and R).
Anyone of you knows if there's a trick to perform that in the digital domain (possibly without phase distortion)?
I use Nuendo 2.01 Surround Edition.

Thanks a lot!

Gianni

Gianni Ricciardi
Sound design & Music production
[:: www.gianniricciardi.com ::]
Eberhard Sengpiel
2004-01-08 23:41:07 UTC
Permalink
Hallo Viriato de Oliveira,

your proposal will not work, because the M signal is L + R.
That is no separate signal for the center. If you do this, your
stereo image will be very narrow.

Cheers

Eberhard Sengpiel
German forum for microphone recordings
and sound studio techniques
http://www.sengpielaudio.com


It's easy and simple to do:
- do an MS decoding of the stereo file, then you'll have 3 channels,
left, right and center.

You can do it very easily with Samplitude, send me a mail if you want
that I explain the trick to you.

I don't use Nuendo, so I don't know how to do Mid-Side decoding with
Nuendo.

cheers,
Viriato de Oliveira
----- Original Message -----
From: VDOSH
To: Surround Sound discussion group
Sent: Thursday, January 08, 2004 11:35 PM
Subject: Re: [Sursound] LCR Upmix
VDOSH
2004-01-09 04:18:48 UTC
Permalink
Hello Eberhard Sengpiel,

In a stereo listening system, this MS decoding process lead to 3 different
and complementary signals wich can be panned left, center and right, and the
stereo field remains identical to the original; the stereo image can also be
reworked if needed. Simple MS stereo recording leads to 2 signals: one M
panned center, and one S that is splitted in two equals signals panned left
and right, the right being phase inversed for completing the stereo image,
so at the console there is three signals. Here, we can take a stereo file,
decode it on two M and S files, and route and split them in three signals,
L, C, and R for an identical stereo image. I assume that they are a good
base for a calibrated LCR listening system, that I hadn't the chance to
experiment. The process is quite simple.

Regards,
Viriato de Oliveira


----- Original Message -----
From: "Eberhard Sengpiel" <***@t-online.de>
To: "Surround Sound discussion group" <***@music.vt.edu>
Sent: Friday, January 09, 2004 12:41 AM
Subject: Re: [Sursound] LCR Upmix


> Hallo Viriato de Oliveira,
>
> your proposal will not work, because the M signal is L + R.
> That is no separate signal for the center. If you do this, your
> stereo image will be very narrow.
>
> Cheers
>
> Eberhard Sengpiel
> German forum for microphone recordings
> and sound studio techniques
> http://www.sengpielaudio.com
>
>
> It's easy and simple to do:
> - do an MS decoding of the stereo file, then you'll have 3 channels,
> left, right and center.
>
> You can do it very easily with Samplitude, send me a mail if you want
> that I explain the trick to you.
>
> I don't use Nuendo, so I don't know how to do Mid-Side decoding with
> Nuendo.
>
> cheers,
> Viriato de Oliveira
> ----- Original Message -----
> From: VDOSH
> To: Surround Sound discussion group
> Sent: Thursday, January 08, 2004 11:35 PM
> Subject: Re: [Sursound] LCR Upmix
>
>
>
> _______________________________________________
> Sursound mailing list
> ***@music.vt.edu
> http://mail.music.vt.edu/mailman/listinfo/sursound
Eberhard Sengpiel
2004-01-09 07:08:40 UTC
Permalink
Hello Viriato de Oliveira,

what is different of *your* M-signal to a simple MS stereo
matrix, where M = L + R and S = L - R?
This M-signal is the in phase sum of L + R and will
narrow your L/R stereo image if you add it as a third
channel to your stereo. There is easy math that shows it.

Kind regards

Eberhard Sengpiel
German forum for microphone recording
and sound studio techniques
http://www.sengpielaudio.com


----- Original Message -----
From: "VDOSH" <***@noos.fr>
To: "Surround Sound discussion group" <***@music.vt.edu>
Sent: Friday, January 09, 2004 5:18 AM
Subject: Re: [Sursound] LCR Upmix


> Hello Eberhard Sengpiel,
>
> In a stereo listening system, this MS decoding process lead to 3
different
> and complementary signals wich can be panned left, center and right,
and the
> stereo field remains identical to the original; the stereo image can
also be
> reworked if needed. Simple MS stereo recording leads to 2 signals: one
M
> panned center, and one S that is splitted in two equals signals panned
left
> and right, the right being phase inversed for completing the stereo
image,
> so at the console there is three signals. Here, we can take a stereo
file,
> decode it on two M and S files, and route and split them in three
signals,
> L, C, and R for an identical stereo image. I assume that they are a
good
> base for a calibrated LCR listening system, that I hadn't the chance
to
> experiment. The process is quite simple.
>
> Regards,
> Viriato de Oliveira
>
> ----- Original Message -----
> From: "Eberhard Sengpiel" <***@t-online.de>
> To: "Surround Sound discussion group" <***@music.vt.edu>
> Sent: Friday, January 09, 2004 12:41 AM
> Subject: Re: [Sursound] LCR Upmix
>
> > Hallo Viriato de Oliveira,
> >
> > your proposal will not work, because the M signal is L + R.
> > That is no separate signal for the center. If you do this, your
> > stereo image will be very narrow.
> >
> > Cheers
> >
> > Eberhard Sengpiel
> > German forum for microphone recordings
> > and sound studio techniques
> > http://www.sengpielaudio.com
> >
> > It's easy and simple to do:
> > - do an MS decoding of the stereo file, then you'll have 3 channels,
> > left, right and center.
> >
> > You can do it very easily with Samplitude, send me a mail if you
want
> > that I explain the trick to you.
> >
> > I don't use Nuendo, so I don't know how to do Mid-Side decoding with
> > Nuendo.
> >
> > cheers,
> > Viriato de Oliveira
Aldo Bazan
2004-01-09 16:29:19 UTC
Permalink
At 10.34 09/01/2004 -0500, you wrote:


>I would like to make samplers of the the songs I like from my CD
>collection as well as my DTS CDs/DVDs and Dolby Digital DVDs to make
>DVD-R samplers for better party music and enjoyment.

first, dvd specs allow for a 48KHz sample, not the 44.1 of CD, so you will
need to resample them.
second, mixing cd audio and dts cd tracks togheter isn't going to work so
well; some player may find difficult to play a dts track after a audio
track after a dts etc. Issue of that kind has been observed at least with
some Pionieer and Toshiba player in the past.
third, for stereo tracks are you going to "unwrap" them in some way, or
leave it as stereo? this is another issue, since you will have some tracks
that sounds only in the two front speaker, other in all... not pleaseant, imho.

>I will be outputting the final DVD-R to a Meridian 861. Also, my car
>had a DD/DTS decoder and surround sound phantom center but has 4.1 using
>Dynaudio 3 ways all the way around. Hopefully these DVD players will be
>able to read DVD-R?

not all players can do it (the panasonic dvd-audio, for example); what do
you have in the car?

>Also, I should be able to mix and match 44.1 CD with 5.1 DD and DTS
>right? That would be great!!!

there's the serious risk of a total mess... which will output a wonderful
white noise. 8-)


F. Aldo Bazan

e-mail:
***@aug.org
VDOSH
2004-01-11 04:42:39 UTC
Permalink
Hello Eberhard Sengpiel,

I'm afraid I'm not enough involved in maths right now to answer you in this
way, but I took the time to make a little html page to show how can this be
done with the audio editor I use, Samplitude v7. It works, so I just can't
see another way to extract three L,C,R mono signals from a stereo file with
the stereo image remaining unchanged.

The link: http://mapage.noos.fr/fruizelegum/mslcr.htm


Best regards

Viriato de Oliveira


----- Original Message -----
From: "Eberhard Sengpiel" <***@t-online.de>
To: "Surround Sound discussion group" <***@music.vt.edu>
Sent: Friday, January 09, 2004 8:08 AM
Subject: Re: [Sursound] LCR Upmix


> Hello Viriato de Oliveira,
>
> what is different of *your* M-signal to a simple MS stereo
> matrix, where M = L + R and S = L - R?
> This M-signal is the in phase sum of L + R and will
> narrow your L/R stereo image if you add it as a third
> channel to your stereo. There is easy math that shows it.
>
> Kind regards
>
> Eberhard Sengpiel
> German forum for microphone recording
> and sound studio techniques
> http://www.sengpielaudio.com
>
>
> ----- Original Message -----
> From: "VDOSH" <***@noos.fr>
> To: "Surround Sound discussion group" <***@music.vt.edu>
> Sent: Friday, January 09, 2004 5:18 AM
> Subject: Re: [Sursound] LCR Upmix
>
>
> > Hello Eberhard Sengpiel,
> >
> > In a stereo listening system, this MS decoding process lead to 3
> different
> > and complementary signals wich can be panned left, center and right,
> and the
> > stereo field remains identical to the original; the stereo image can
> also be
> > reworked if needed. Simple MS stereo recording leads to 2 signals: one
> M
> > panned center, and one S that is splitted in two equals signals panned
> left
> > and right, the right being phase inversed for completing the stereo
> image,
> > so at the console there is three signals. Here, we can take a stereo
> file,
> > decode it on two M and S files, and route and split them in three
> signals,
> > L, C, and R for an identical stereo image. I assume that they are a
> good
> > base for a calibrated LCR listening system, that I hadn't the chance
> to
> > experiment. The process is quite simple.
> >
> > Regards,
> > Viriato de Oliveira
> >
> > ----- Original Message -----
> > From: "Eberhard Sengpiel" <***@t-online.de>
> > To: "Surround Sound discussion group" <***@music.vt.edu>
> > Sent: Friday, January 09, 2004 12:41 AM
> > Subject: Re: [Sursound] LCR Upmix
> >
> > > Hallo Viriato de Oliveira,
> > >
> > > your proposal will not work, because the M signal is L + R.
> > > That is no separate signal for the center. If you do this, your
> > > stereo image will be very narrow.
> > >
> > > Cheers
> > >
> > > Eberhard Sengpiel
> > > German forum for microphone recordings
> > > and sound studio techniques
> > > http://www.sengpielaudio.com
> > >
> > > It's easy and simple to do:
> > > - do an MS decoding of the stereo file, then you'll have 3 channels,
> > > left, right and center.
> > >
> > > You can do it very easily with Samplitude, send me a mail if you
> want
> > > that I explain the trick to you.
> > >
> > > I don't use Nuendo, so I don't know how to do Mid-Side decoding with
> > > Nuendo.
> > >
> > > cheers,
> > > Viriato de Oliveira
>
>
>
> _______________________________________________
> Sursound mailing list
> ***@music.vt.edu
> http://mail.music.vt.edu/mailman/listinfo/sursound
Christian
2004-01-09 15:34:40 UTC
Permalink
Hi,

I would like to make samplers of the the songs I like from my CD
collection as well as my DTS CDs/DVDs and Dolby Digital DVDs to make
DVD-R samplers for better party music and enjoyment.

I will be outputting the final DVD-R to a Meridian 861. Also, my car
had a DD/DTS decoder and surround sound phantom center but has 4.1 using
Dynaudio 3 ways all the way around. Hopefully these DVD players will be
able to read DVD-R?

Can’t I just use Roxio to make DVD-R samplers instead of CD-R samplers?
Also, Can I now burn my entire CD-R collection to DVD-R??? That would
BE GREAT!!!! With DVD-R I can have so much more play time!!!! How about
it?
Also, I should be able to mix and match 44.1 CD with 5.1 DD and DTS
right? That would be great!!!

Can anyone let me know what is and is not possible?

Thanks!

Chris

-----Original Message-----
From: sursound-***@music.vt.edu
[mailto:sursound-***@music.vt.edu] On Behalf Of Eberhard Sengpiel
Sent: Friday, January 09, 2004 2:09 AM
To: Surround Sound discussion group
Subject: Re: [Sursound] LCR Upmix

Hello Viriato de Oliveira,

what is different of *your* M-signal to a simple MS stereo
matrix, where M = L + R and S = L - R?
This M-signal is the in phase sum of L + R and will
narrow your L/R stereo image if you add it as a third
channel to your stereo. There is easy math that shows it.

Kind regards

Eberhard Sengpiel
German forum for microphone recording
and sound studio techniques
http://www.sengpielaudio.com


----- Original Message -----
From: "VDOSH" <***@noos.fr>
To: "Surround Sound discussion group" <***@music.vt.edu>
Sent: Friday, January 09, 2004 5:18 AM
Subject: Re: [Sursound] LCR Upmix


> Hello Eberhard Sengpiel,
>
> In a stereo listening system, this MS decoding process lead to 3
different
> and complementary signals wich can be panned left, center and right,
and the
> stereo field remains identical to the original; the stereo image can
also be
> reworked if needed. Simple MS stereo recording leads to 2 signals: one
M
> panned center, and one S that is splitted in two equals signals panned
left
> and right, the right being phase inversed for completing the stereo
image,
> so at the console there is three signals. Here, we can take a stereo
file,
> decode it on two M and S files, and route and split them in three
signals,
> L, C, and R for an identical stereo image. I assume that they are a
good
> base for a calibrated LCR listening system, that I hadn't the chance
to
> experiment. The process is quite simple.
>
> Regards,
> Viriato de Oliveira
>
> ----- Original Message -----
> From: "Eberhard Sengpiel" <***@t-online.de>
> To: "Surround Sound discussion group" <***@music.vt.edu>
> Sent: Friday, January 09, 2004 12:41 AM
> Subject: Re: [Sursound] LCR Upmix
>
> > Hallo Viriato de Oliveira,
> >
> > your proposal will not work, because the M signal is L + R.
> > That is no separate signal for the center. If you do this, your
> > stereo image will be very narrow.
> >
> > Cheers
> >
> > Eberhard Sengpiel
> > German forum for microphone recordings
> > and sound studio techniques
> > http://www.sengpielaudio.com
> >
> > It's easy and simple to do:
> > - do an MS decoding of the stereo file, then you'll have 3 channels,
> > left, right and center.
> >
> > You can do it very easily with Samplitude, send me a mail if you
want
> > that I explain the trick to you.
> >
> > I don't use Nuendo, so I don't know how to do Mid-Side decoding with
> > Nuendo.
> >
> > cheers,
> > Viriato de Oliveira



_______________________________________________
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Gianni Ricciardi
2004-01-09 19:14:12 UTC
Permalink
Eberhard Sengpiel : "I see no phase problems there. Play arround."

What you say it's not true..

I've tried routing an L+R to the center channel, with a delay of 17ms while
in the Left and Right channel there's only the stereo original at 0db. The
center channel is at -7db.
What I hear is very little and strange phase effect (a sort of "metal
taste"), but it works fine on 5.1 output because the stereo picture works
well and so the dialogues on the C.

But, as I though, when I try the stereo compatibility with a 6ch->2ch Mixer
(mixing the center channel in the stereo out at -6db with a total
attenuation for the delayed L+R of -13db) I hear the phase effect very loud,
it's like a phaser without modulation, fixed on a single "note", because
both direct and delayed signal are present in both channels.
The frequency of this "note" change as I change the delay time (it becames
lower when I increase the delay time).
The problem is that what I'm producing in this way will be not compatible
with the Dolby Digital Standard because it will not stereo-compatible after
common decoders downmix.
The Dolby Metadata (encoded in the ac3 stream for mixdown) has fixed step of
attenuation for the center channel. The max value is -6db so I can't produce
an AC3 flow that will discard the center channel for the users that doesn't
have five speakers.

Any solution?

Thanx

GIANNI

-----Messaggio originale-----
Da: sursound-***@music.vt.edu
[mailto:sursound-***@music.vt.edu]Per conto di Eberhard Sengpiel
Inviato: giovedì 8 gennaio 2004 9.22
A: Surround Sound discussion group
Oggetto: Re: [Sursound] LCR Upmix


Hi Gianni,

it is really not critcal if you break up the stereo with the 15 to 25 ms
delayed L + R signal and put it into the center. The same is with the
level of this sum signal in relation to the L and R signal. 3 to 6 dB
down
will be sufficient. It will depend on the sort of signal you have.
You may give more level to the center if you take off some treble there.
Play arround and you willl see. My proposal is really no scientifical
solution. It is a practical and cheap one just right for your case.
I see no phase problems there. Play arround.

Cheers

Eberhard Sengpiel
German forum of microphone recordings
and sound studio techniques
http://www.sengpielaudio.com

----- Original Message -----
From: "Gianni Ricciardi" <***@gianniricciardi.com>
To: "Surround Sound discussion group" <***@music.vt.edu>
Sent: Thursday, January 08, 2004 12:42 AM
Subject: R: [Sursound] LCR Upmix


> Thank Eberhard for your suggestions.
>
> Yes, after many tries I do, I think I will not "destroy" the original
L/R
> program.
> I wish to add something in the center (not only a phantom center ;o)
so I
> think I will add L+R as you said.
> I've read about the trick to add delay to the C channel in order to
preserve
> the stereo picture (that elsewhere collapse at the center), but I try
with
> 20 ms and it seems too much for me.. I can perceive it as a true delay
> during dialogue.
> But if I use smaller values (like 15 or 10) I'm afraid that the AC3
stereo
> downmix will have phase problems, doesn't it? The minimum value for
center
> downmix in the AC3 metadata is -6 (it would be a total of -12db) so...
what
> do you think about this?
>
> Another question: what do you mean with "put it with a level of -6 db
(to L
> or R) to the center loudspeaker" ?
> You mean that I have to add some L or R in the center? Sorry I don't
> understand this step.
>
> Thank you, and sorry for my bad english!
>
> GIANNI
Eberhard Sengpiel
2004-01-09 20:24:10 UTC
Permalink
Hallo Gianni,

that sort of metal sounding comb filter effect I never had,
when I add L+R with a delay of more than 15 ms to
the center speaker with the level more than 6 dB
down.
Strange, that I never heard such phase problems.

Cheers

Eberhard Sengpiel
Forum for Microphone recordings
and sound studiotechniques
http://www.sengpielaudio.com


----- Original Message -----
From: "Gianni Ricciardi" <***@gianniricciardi.com>
To: "Surround Sound discussion group" <***@music.vt.edu>
Sent: Friday, January 09, 2004 8:14 PM
Subject: R: [Sursound] LCR Upmix

> Eberhard Sengpiel : "I see no phase problems there. Play arround."
>
> What you say it's not true..
>
> I've tried routing an L+R to the center channel, with a delay of 17ms
while
> in the Left and Right channel there's only the stereo original at 0db.
The
> center channel is at -7db.
> What I hear is very little and strange phase effect (a sort of "metal
> taste"), but it works fine on 5.1 output because the stereo picture
works
> well and so the dialogues on the C.
>
> But, as I though, when I try the stereo compatibility with a 6ch->2ch
Mixer
> (mixing the center channel in the stereo out at -6db with a total
> attenuation for the delayed L+R of -13db) I hear the phase effect very
loud,
> it's like a phaser without modulation, fixed on a single "note",
because
> both direct and delayed signal are present in both channels.
> The frequency of this "note" change as I change the delay time (it
becames
> lower when I increase the delay time).
> The problem is that what I'm producing in this way will be not
compatible
> with the Dolby Digital Standard because it will not stereo-compatible
after
> common decoders downmix.
> The Dolby Metadata (encoded in the ac3 stream for mixdown) has fixed
step of
> attenuation for the center channel. The max value is -6db so I can't
produce
> an AC3 flow that will discard the center channel for the users that
doesn't
> have five speakers.
>
> Any solution?
>
> Thanx
>
> GIANNI
>
> -----Messaggio originale-----
> Da: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu]Per conto di Eberhard Sengpiel
> Inviato: giovedì 8 gennaio 2004 9.22
> A: Surround Sound discussion group
> Oggetto: Re: [Sursound] LCR Upmix
>
>
> Hi Gianni,
>
> it is really not critcal if you break up the stereo with the 15 to 25
ms
> delayed L + R signal and put it into the center. The same is with the
> level of this sum signal in relation to the L and R signal. 3 to 6 dB
> down
> will be sufficient. It will depend on the sort of signal you have.
> You may give more level to the center if you take off some treble
there.
> Play arround and you willl see. My proposal is really no scientifical
> solution. It is a practical and cheap one just right for your case.
> I see no phase problems there. Play arround.
>
> Cheers
>
> Eberhard Sengpiel
> German forum of microphone recordings
> and sound studio techniques
> http://www.sengpielaudio.com
Don Cox
2004-01-09 17:28:49 UTC
Permalink
On 08/01/04, Geoffrey Barton wrote:
> <?xml version="1.0" ?><html> <head>
> <title></title>
> </head>
> <body>
> <div align="left"><font face="Comic Sans MS"><span
> style="font-size:10pt">On 8 Jan 2004 at 2:17, Richard G. Elen

etc

Would you care to resend that in plain text so that I can read it?

Regards
--
Don Cox
***@enterprise.net
Gianni Ricciardi
2004-01-09 20:53:03 UTC
Permalink
Yes Eberhard the problem is only in the STEREO DOWNMIX that all the set top
box DVD players do when no AC3 decoders are connected.
They mixdown the 6 channels into 2 using the information included in the
metadata packed in the AC3 stream.
Mixing the delayed L+R to L and R introduce phase problems.

Best regards

GIANNI RICCIARDI

-----Messaggio originale-----
Da: sursound-***@music.vt.edu
[mailto:sursound-***@music.vt.edu]Per conto di Eberhard Sengpiel
Inviato: venerdì 9 gennaio 2004 21.24
A: Surround Sound discussion group
Oggetto: Re: [Sursound] LCR Upmix


Hallo Gianni,

that sort of metal sounding comb filter effect I never had,
when I add L+R with a delay of more than 15 ms to
the center speaker with the level more than 6 dB
down.
Strange, that I never heard such phase problems.

Cheers

Eberhard Sengpiel
Forum for Microphone recordings
and sound studiotechniques
http://www.sengpielaudio.com


----- Original Message -----
From: "Gianni Ricciardi" <***@gianniricciardi.com>
To: "Surround Sound discussion group" <***@music.vt.edu>
Sent: Friday, January 09, 2004 8:14 PM
Subject: R: [Sursound] LCR Upmix

> Eberhard Sengpiel : "I see no phase problems there. Play arround."
>
> What you say it's not true..
>
> I've tried routing an L+R to the center channel, with a delay of 17ms
while
> in the Left and Right channel there's only the stereo original at 0db.
The
> center channel is at -7db.
> What I hear is very little and strange phase effect (a sort of "metal
> taste"), but it works fine on 5.1 output because the stereo picture
works
> well and so the dialogues on the C.
>
> But, as I though, when I try the stereo compatibility with a 6ch->2ch
Mixer
> (mixing the center channel in the stereo out at -6db with a total
> attenuation for the delayed L+R of -13db) I hear the phase effect very
loud,
> it's like a phaser without modulation, fixed on a single "note",
because
> both direct and delayed signal are present in both channels.
> The frequency of this "note" change as I change the delay time (it
becames
> lower when I increase the delay time).
> The problem is that what I'm producing in this way will be not
compatible
> with the Dolby Digital Standard because it will not stereo-compatible
after
> common decoders downmix.
> The Dolby Metadata (encoded in the ac3 stream for mixdown) has fixed
step of
> attenuation for the center channel. The max value is -6db so I can't
produce
> an AC3 flow that will discard the center channel for the users that
doesn't
> have five speakers.
>
> Any solution?
>
> Thanx
>
> GIANNI
>
> -----Messaggio originale-----
> Da: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu]Per conto di Eberhard Sengpiel
> Inviato: giovedì 8 gennaio 2004 9.22
> A: Surround Sound discussion group
> Oggetto: Re: [Sursound] LCR Upmix
>
>
> Hi Gianni,
>
> it is really not critcal if you break up the stereo with the 15 to 25
ms
> delayed L + R signal and put it into the center. The same is with the
> level of this sum signal in relation to the L and R signal. 3 to 6 dB
> down
> will be sufficient. It will depend on the sort of signal you have.
> You may give more level to the center if you take off some treble
there.
> Play arround and you willl see. My proposal is really no scientifical
> solution. It is a practical and cheap one just right for your case.
> I see no phase problems there. Play arround.
>
> Cheers
>
> Eberhard Sengpiel
> German forum of microphone recordings
> and sound studio techniques
> http://www.sengpielaudio.com
Eberhard Sengpiel
2004-01-09 21:35:49 UTC
Permalink
Thank you Gianni, now I understand.
The problem is the electrical mix back stereo.

Eberhard Sengpiel


----- Original Message -----
From: "Gianni Ricciardi" <***@gianniricciardi.com>
To: "Surround Sound discussion group" <***@music.vt.edu>
Sent: Friday, January 09, 2004 9:53 PM
Subject: R: [Sursound] LCR Upmix


> Yes Eberhard the problem is only in the STEREO DOWNMIX that all the
set top
> box DVD players do when no AC3 decoders are connected.
> They mixdown the 6 channels into 2 using the information included in
the
> metadata packed in the AC3 stream.
> Mixing the delayed L+R to L and R introduce phase problems.
>
> Best regards
>
> GIANNI RICCIARDI
>
> -----Messaggio originale-----
> Da: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu]Per conto di Eberhard Sengpiel
> Inviato: venerdì 9 gennaio 2004 21.24
> A: Surround Sound discussion group
> Oggetto: Re: [Sursound] LCR Upmix
>
>
> Hallo Gianni,
>
> that sort of metal sounding comb filter effect I never had,
> when I add L+R with a delay of more than 15 ms to
> the center speaker with the level more than 6 dB
> down.
> Strange, that I never heard such phase problems.
>
> Cheers
>
> Eberhard Sengpiel
> Forum for Microphone recordings
> and sound studiotechniques
> http://www.sengpielaudio.com
>
>
> ----- Original Message -----
> From: "Gianni Ricciardi" <***@gianniricciardi.com>
> To: "Surround Sound discussion group" <***@music.vt.edu>
> Sent: Friday, January 09, 2004 8:14 PM
> Subject: R: [Sursound] LCR Upmix
>
> > Eberhard Sengpiel : "I see no phase problems there. Play arround."
> >
> > What you say it's not true..
> >
> > I've tried routing an L+R to the center channel, with a delay of
17ms
> while
> > in the Left and Right channel there's only the stereo original at
0db.
> The
> > center channel is at -7db.
> > What I hear is very little and strange phase effect (a sort of
"metal
> > taste"), but it works fine on 5.1 output because the stereo picture
> works
> > well and so the dialogues on the C.
> >
> > But, as I though, when I try the stereo compatibility with a
6ch->2ch
> Mixer
> > (mixing the center channel in the stereo out at -6db with a total
> > attenuation for the delayed L+R of -13db) I hear the phase effect
very
> loud,
> > it's like a phaser without modulation, fixed on a single "note",
> because
> > both direct and delayed signal are present in both channels.
> > The frequency of this "note" change as I change the delay time (it
> becames
> > lower when I increase the delay time).
> > The problem is that what I'm producing in this way will be not
> compatible
> > with the Dolby Digital Standard because it will not
stereo-compatible
> after
> > common decoders downmix.
> > The Dolby Metadata (encoded in the ac3 stream for mixdown) has fixed
> step of
> > attenuation for the center channel. The max value is -6db so I can't
> produce
> > an AC3 flow that will discard the center channel for the users that
> doesn't
> > have five speakers.
> >
> > Any solution?
> >
> > Thanx
> >
> > GIANNI
> >
> > -----Messaggio originale-----
> > Da: sursound-***@music.vt.edu
> > [mailto:sursound-***@music.vt.edu]Per conto di Eberhard Sengpiel
> > Inviato: giovedì 8 gennaio 2004 9.22
> > A: Surround Sound discussion group
> > Oggetto: Re: [Sursound] LCR Upmix
> >
> >
> > Hi Gianni,
> >
> > it is really not critcal if you break up the stereo with the 15 to
25
> ms
> > delayed L + R signal and put it into the center. The same is with
the
> > level of this sum signal in relation to the L and R signal. 3 to 6
dB
> > down
> > will be sufficient. It will depend on the sort of signal you have.
> > You may give more level to the center if you take off some treble
> there.
> > Play arround and you willl see. My proposal is really no
scientifical
> > solution. It is a practical and cheap one just right for your case.
> > I see no phase problems there. Play arround.
> >
> > Cheers
> >
> > Eberhard Sengpiel
> > German forum of microphone recordings
> > and sound studio techniques
> > http://www.sengpielaudio.com
Mike Miles
2004-01-09 23:34:42 UTC
Permalink
> I can't find a simple way to derive the mono content and put
> it on the center channel (subtracting it from L and R).

I think the best you can do is use a simple L+R sum. If it was panned center
at -3 dB in left and right, then your center-sum signal will have the
center-panned components at +3 dB relative to the side components. The level
of course needs to be set correctly. You can improve the overall sound by
reducing the center component in the left and right with inverted
crossmixing. This simple type of matrix will not create artifacts and will
allow precise downmixing (you'll need to set the appropriate center-level
downmix coefficient to correspond with your chosen matrix levels).

I think there are basically two strategies for this:

The Gerzon approach which mathematically-correctly recreates the sound of
the original two-channel mix:

(L, R = inputs, L', C', R' = outputs)

L' = 0.885 * L - 0.115 * R
C' = (L + R) * 0.4511
R' = 0.885 * R - 0.115 * L

This will sound good in the center of the listening area but it removes very
little of the center component from the side signals and therefore does not
solve the problem of off-center listeners hearing center mix components from
the nearer (left or right) loudspeaker (which is a principal reason for
using the center).

Or, you could use the Multisonic(R) approach which is:

L' = L - 0.5 * R
C' = (L + R) * 0.5
R' = R - 0.5 * L

This provides more separation, allowing a very distinct center channel to be
heard while preserving the left-to-right sound stage, and resulting in a
much larger listening area. Preservation of the original two-channel sound
staging is not necessarily "perfect" but it is extremely close. See AES
Preprint 4364.

Mike
___________________
Miles Technology Inc.
Ph 269-683-4400
Fx 269-683-4499
www.milestech.com
Angelo Farina
2004-01-10 08:29:15 UTC
Permalink
Eberhard,
I suppose that the M signal is not simply sent as the center channel,
leaving the L and R equal to the original... New "expanded stereo" versions
of L and R need to be computed from M and S...
My understanding (and what I do when we need to drive a three channel front
from a stereo source, as it is common inside car sound system) is as
follows:
1) Compute the M and S signal as usual (M=L+R, S=L-R).
2) Compute the three new channels from M and S, depending on the value of a
"center dominance" parameter c:
L' = (1-c/4)*M+(1+c/4)*S
C' = c*M
R' = (1-c/4)*M-(1+c/4)*S
3) For any value of c between 0 and 1, if the 3-channels are repacked back
to 2 (L=L'+C'/2, R=R'+C'/2) You get the original L and R signals.
4) The amount of c allowed depends on the source material: for perfectly
coincident recordings, or artifical level-only panned signals, where no time
delay exist between L and R, in thoery you can go up to values of c close to
1. If instead the recording was done with spaced mics (even closely spaced,
such as ORTF or Sphere), the maximum value of c is around 0.5 (which means
that the center channel will be weaker than L' and R' by approximately 6
dB). This is probably the recommended setting for most cases, avoiding that
the center channel becomes too intrusive.
Bye!

Angelo Farina

> -----Original Message-----
> From: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu] On Behalf Of Eberhard Sengpiel
> Sent: 09 January 2004 08:09
> To: Surround Sound discussion group
> Subject: Re: [Sursound] LCR Upmix
>
> Hello Viriato de Oliveira,
>
> what is different of *your* M-signal to a simple MS stereo
> matrix, where M = L + R and S = L - R?
> This M-signal is the in phase sum of L + R and will narrow
> your L/R stereo image if you add it as a third channel to
> your stereo. There is easy math that shows it.
>
> Kind regards
>
> Eberhard Sengpiel
> German forum for microphone recording
> and sound studio techniques
> http://www.sengpielaudio.com
>
>
> ----- Original Message -----
> From: "VDOSH" <***@noos.fr>
> To: "Surround Sound discussion group" <***@music.vt.edu>
> Sent: Friday, January 09, 2004 5:18 AM
> Subject: Re: [Sursound] LCR Upmix
>
>
> > Hello Eberhard Sengpiel,
> >
> > In a stereo listening system, this MS decoding process lead to 3
> different
> > and complementary signals wich can be panned left, center and right,
> and the
> > stereo field remains identical to the original; the stereo image can
> also be
> > reworked if needed. Simple MS stereo recording leads to 2
> signals: one
> M
> > panned center, and one S that is splitted in two equals
> signals panned
> left
> > and right, the right being phase inversed for completing the stereo
> image,
> > so at the console there is three signals. Here, we can take a stereo
> file,
> > decode it on two M and S files, and route and split them in three
> signals,
> > L, C, and R for an identical stereo image. I assume that they are a
> good
> > base for a calibrated LCR listening system, that I hadn't the chance
> to
> > experiment. The process is quite simple.
> >
> > Regards,
> > Viriato de Oliveira
> >
> > ----- Original Message -----
> > From: "Eberhard Sengpiel" <***@t-online.de>
> > To: "Surround Sound discussion group" <***@music.vt.edu>
> > Sent: Friday, January 09, 2004 12:41 AM
> > Subject: Re: [Sursound] LCR Upmix
> >
> > > Hallo Viriato de Oliveira,
> > >
> > > your proposal will not work, because the M signal is L + R.
> > > That is no separate signal for the center. If you do this, your
> > > stereo image will be very narrow.
> > >
> > > Cheers
> > >
> > > Eberhard Sengpiel
> > > German forum for microphone recordings
> > > and sound studio techniques
> > > http://www.sengpielaudio.com
> > >
> > > It's easy and simple to do:
> > > - do an MS decoding of the stereo file, then you'll have
> 3 channels,
> > > left, right and center.
> > >
> > > You can do it very easily with Samplitude, send me a mail if you
> want
> > > that I explain the trick to you.
> > >
> > > I don't use Nuendo, so I don't know how to do Mid-Side
> decoding with
> > > Nuendo.
> > >
> > > cheers,
> > > Viriato de Oliveira
>
>
>
> _______________________________________________
> Sursound mailing list
> ***@music.vt.edu
> http://mail.music.vt.edu/mailman/listinfo/sursound
>
Angelo Farina
2004-01-10 08:47:31 UTC
Permalink
Are You referring to DVD-video discs or DVD-audio discs? In the first case,
any basic DVD authoring tools (even the very cheap one found in the bundle
together with the DVD-burner) is OK for the job. In the second case,
instead, you need a specialized DVD-audio mastering software, such as those
produced by Minnetonka. For surround DVD-audio You also need an MLP encoder,
which is quite expensive.
As Your source material is not hi-resolution (or, better, is not the same
high resolution as a DVD-audio), and considering that you want to play it on
a car, I suppose that a basic DVD-video should be OK.
So You just need to rip the audio track of Your original disks (without
decoding, You will just burn the original AC3 or DTS soundtracks to the new
DVD), and make a compilation of them within the DVD-video authoring
software. If this sofwtare only allows to specify AVI files (not separate
video and audio tracks), You first have to pack Your soundtracks with a very
basic and simple video (a static image, for example), employing a tool such
as NanDub.
The same holds for audio tracks coming from CDs. The DVD-video
specifications allow for pure-PCM stereo soundtracks, so You will be able to
use the ripped waveforms coming from CDs as soundtracks on the DVD-video.
Consider, however, that my experience is just in making DVD-videos starting
from Divx films, and changing their soundtrack (typically from English to
Italian), then making a DVD-video from the repacked AVI... (I do this
because my DVD player does not reproduce directly the Divx or Xvid AVI
files).
However, I managed to do this also employing AC3 soundtracks, and I see no
reasons for which it should not work with DTS soundtracks.
I use a Pioneer DVD-R A05, and these disks work fine on most DVD players.
Bye!

Angelo Farina



> -----Original Message-----
> From: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu] On Behalf Of Christian
> Sent: 09 January 2004 16:35
> To: 'Surround Sound discussion group'
> Subject: [Sursound] Making DVD-R samplers instead of CD-R samplers?
>
> Hi,
>
> I would like to make samplers of the the songs I like from my
> CD collection as well as my DTS CDs/DVDs and Dolby Digital
> DVDs to make DVD-R samplers for better party music and enjoyment.
>
> I will be outputting the final DVD-R to a Meridian 861.
> Also, my car had a DD/DTS decoder and surround sound phantom
> center but has 4.1 using Dynaudio 3 ways all the way around.
> Hopefully these DVD players will be able to read DVD-R?
>
> Can't I just use Roxio to make DVD-R samplers instead of CD-R
> samplers?
> Also, Can I now burn my entire CD-R collection to DVD-R???
> That would BE GREAT!!!! With DVD-R I can have so much more
> play time!!!! How about it?
> Also, I should be able to mix and match 44.1 CD with 5.1 DD
> and DTS right? That would be great!!!
>
> Can anyone let me know what is and is not possible?
>
> Thanks!
>
> Chris
>
> -----Original Message-----
> From: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu] On Behalf Of Eberhard Sengpiel
> Sent: Friday, January 09, 2004 2:09 AM
> To: Surround Sound discussion group
> Subject: Re: [Sursound] LCR Upmix
>
> Hello Viriato de Oliveira,
>
> what is different of *your* M-signal to a simple MS stereo
> matrix, where M = L + R and S = L - R?
> This M-signal is the in phase sum of L + R and will narrow
> your L/R stereo image if you add it as a third channel to
> your stereo. There is easy math that shows it.
>
> Kind regards
>
> Eberhard Sengpiel
> German forum for microphone recording
> and sound studio techniques
> http://www.sengpielaudio.com
>
>
> ----- Original Message -----
> From: "VDOSH" <***@noos.fr>
> To: "Surround Sound discussion group" <***@music.vt.edu>
> Sent: Friday, January 09, 2004 5:18 AM
> Subject: Re: [Sursound] LCR Upmix
>
>
> > Hello Eberhard Sengpiel,
> >
> > In a stereo listening system, this MS decoding process lead to 3
> different
> > and complementary signals wich can be panned left, center and right,
> and the
> > stereo field remains identical to the original; the stereo image can
> also be
> > reworked if needed. Simple MS stereo recording leads to 2
> signals: one
> M
> > panned center, and one S that is splitted in two equals
> signals panned
> left
> > and right, the right being phase inversed for completing the stereo
> image,
> > so at the console there is three signals. Here, we can take a stereo
> file,
> > decode it on two M and S files, and route and split them in three
> signals,
> > L, C, and R for an identical stereo image. I assume that they are a
> good
> > base for a calibrated LCR listening system, that I hadn't the chance
> to
> > experiment. The process is quite simple.
> >
> > Regards,
> > Viriato de Oliveira
> >
> > ----- Original Message -----
> > From: "Eberhard Sengpiel" <***@t-online.de>
> > To: "Surround Sound discussion group" <***@music.vt.edu>
> > Sent: Friday, January 09, 2004 12:41 AM
> > Subject: Re: [Sursound] LCR Upmix
> >
> > > Hallo Viriato de Oliveira,
> > >
> > > your proposal will not work, because the M signal is L + R.
> > > That is no separate signal for the center. If you do this, your
> > > stereo image will be very narrow.
> > >
> > > Cheers
> > >
> > > Eberhard Sengpiel
> > > German forum for microphone recordings
> > > and sound studio techniques
> > > http://www.sengpielaudio.com
> > >
> > > It's easy and simple to do:
> > > - do an MS decoding of the stereo file, then you'll have
> 3 channels,
> > > left, right and center.
> > >
> > > You can do it very easily with Samplitude, send me a mail if you
> want
> > > that I explain the trick to you.
> > >
> > > I don't use Nuendo, so I don't know how to do Mid-Side
> decoding with
> > > Nuendo.
> > >
> > > cheers,
> > > Viriato de Oliveira
>
>
>
> _______________________________________________
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>
Brian C. Peters
2004-01-10 16:34:27 UTC
Permalink
On 1/10/04 3:47 AM, "Angelo Farina" <***@pcfarina.eng.unipr.it> wrote:

Snip

> For surround DVD-audio You also need an MLP encoder,
> which is quite expensive.


Just a quick correction. MLP is an option only if you want 6 channels at
2496 resolution or above. It is not required for 6 channels at either 44.1K
or 48k any bit depth. If you only care about 4 channels you can also encode
4 channels 2496 without MLP. The total disc time gets short, just over 60
minutes, but is usable all the same. As long as you keep the bit rate under
9.6 Mbit/sec you can mix and match all you want. Still saving for the MLP
option though.


Brian C. Peters
Senior Engineer
Dorian Recordings
***@dorian.com
Richard G. Elen
2004-01-11 03:36:28 UTC
Permalink
----- Original Message -----
From: "Brian C. Peters" <***@earthlink.net>

> MLP is an option only if you want 6 channels at
> 2496 resolution or above. It is not required for 6 channels at either
44.1K
> or 48k any bit depth. If you only care about 4 channels you can also
encode
> 4 channels 2496 without MLP. The total disc time gets short, just over 60
> minutes, but is usable all the same. As long as you keep the bit rate
under
> 9.6 Mbit/sec you can mix and match all you want...

I note that Minnetonka have a new version of their DiscWelder software,
"Bronze", coming out at $99 in March.

--Richard Elen
Christian
2004-01-10 17:07:20 UTC
Permalink
Thank-you all.

I use Five Nautilus 802 speakers and a serious custom subwoofer array
which I designed and built along with the help of several engineers in
the industry. I've just decided to build a music server to better enjoy
my music collection. I'm a bit appalled at how many people accept MP3
and other lossy formats.

Quite frankly, I was hoping that with HD-DVD looming in the near future,
we would be replacing DD music videos with MLP encoded music videos.
With the new video compression codec such as VC-9 (WM9) we could have
1080p and DVD-Audio simultaneously. I and many others I know refuse to
buy Dolby Digital and other harsh lossy compression for music and music
videos. I'm not a giant fan of DTS, but at least they allow for less
harsh lossy compression and could be a player in HD-DVD music. However,
that would mean running DTS at near lossless compression, and by then
MLP does a better job, which was why it was chosen in the first place.
I would love to by Pink Floyd Dark Side of the Moon, but I'm not buying
into SACD, nor am I buying the Dolby Digtial DVD-V that is available.
I'm so sick of all the B.S. politics and lossy crap floating around it
makes me nauseated.

In the meantime, I want to enjoy my CD collection by making samplers of
all my music which is quite extensive. Due to the improvement in
computer technology and affordability, I just bought a 120 Gig hard
drive and will begin to make samplers of all my CDs. Let's face it, in
a time when people are working 800-100 hours a week, like me, I cannot
fish through my music collection to find one favorite song on one disc.
I am no longer concerned about QUALITY and NOISE introduced with
Computers because of outboard soundcards and digital outputs. I now see
that I can maintain the integrity 100% of what was encoded on my redbook
audio CD's and simply play them from ny hard drive with no concern of
quality loss. I will pass the digital output to a Meridian 861 in my
living room which requires a 45 foot run of coax digital cable, but this
is not a big deal. Those large optical CD player changers (300-400
disc) always have a HUGE delay when using random, which is a feature in
enjoy, particularly with CD-R samplers of tracks I KNOW I enjoy. There
is no delay when playing from the hard drive. I will not use any
compression, although lossless compression is somewhat appealing.
Although don’t you 'technically' alter/lose something going through the
encoding process? Not a big deal, but technically, lossless compression
doesn’t mean identical to the master, only the ability to make perfect
generational copies once the encoding process is complete? Or am I
wrong? Not a big deal... I'm either using no compression or lossless,
not like the people who use MP3 and other garbage lossy compression on
music! Should be a sin or something if you ask me...

I have to look over the various software on how to best store my DTS
CD/DVD music as well as DD DVD music to my hard drive. Likely I'll have
to dissect them out of the VOB file for DVD stuff and perhaps convert to
44.1, although I don’t understand why that matters 44.1 vs 48 kHz since
the output is going to a Meridian 861, although I'm constantly reading
about re-sampling problems and such... I just don’t get it since what
does the 861 care what it gets on the digital output? It should be able
to switch back and forth accordingly??? I bough a DVD-RW Plextor drive
to make DVD-R stuff for my car which has a DVD player...

On another note, getting slightly off the subject, AOD and Blu-Ray have
identical bandwidth capability and use the same frequency diode/laser.
However, Sony offers more physical storage space at a slightly higher
cost. I have seen a figure mentioned of about 20% retooling cost over
AOD. I feel this extra 20% would be reasonable only if Sony uses this
extra storage space to provide a superior product. However, if Sony
only ends up providing MPEG-2 video compression just like standard DVD,
then we would end up with a more expensive product that would be lower
in quality than AOD (AOD would end up with more relative storage space).
It angers me that Sony would want us to pay more for Blu Ray and
actually give us less than AOD, while at the same time avoiding the
necessity of ONE-FORMAT. Everybody should be seriously angered by Sony
unless Sony provides everything that the AOD group provides and we get
the extra storage space provided by Blu Ray to provide a superior
product, particularly when we only desire ONE-FORMAT, not HD-DVD *and*
Blu-Ray. This still does not address the issue of current
incompatibility between AOD, the now approved HD-DVD format, and
Blu-Ray. How are we going to reach the goal of ONE-FORMAT?

AOD has stated they plan on using H.264/and or VC-9 (WM9) in addition to
MPEG-2. Since H.264 and VC-9 are approximately 3 times as efficient as
MPEG-2 and can provide a superior image than MPEG-2 (particularly VC-9)
then Sony in essence is charging more for an inferior product as well as
squandering precious storage space. Further, it has been suggested by
many people in various forums that VC-9 video compression (WM9) is
significantly superior than H.264. The AOD group has a much more open
attitude and clearly addresses the need for vastly superior audio than
standard DVD, also suggesting that DVD-Audio may/should be part of the
HD-DVD specifications.

There has been talk on the Meridian Forum of Dolby Digital Plus and DVD
2.0 specifications (HD-DVD). One Member of the Meridian Forum says MLP
(DVD-Audio) and AAC techniques would be possible all under the DD Plus
label. Have you heard anything about Dolby Digital Plus?

I'd love to hear your thoughts, clarifications, as well as Sony’s
thought’s on some of these issues.



-----Original Message-----
From: sursound-***@music.vt.edu
[mailto:sursound-***@music.vt.edu] On Behalf Of Brian C. Peters
Sent: Saturday, January 10, 2004 11:34 AM
To: ***@pcfarina.eng.unipr.it, Surround Sound discussion group
Subject: Re: [Sursound] Making DVD-R samplers instead of CD-R samplers?

On 1/10/04 3:47 AM, "Angelo Farina" <***@pcfarina.eng.unipr.it>
wrote:

Snip

> For surround DVD-audio You also need an MLP encoder,
> which is quite expensive.


Just a quick correction. MLP is an option only if you want 6 channels at
2496 resolution or above. It is not required for 6 channels at either
44.1K
or 48k any bit depth. If you only care about 4 channels you can also
encode
4 channels 2496 without MLP. The total disc time gets short, just over
60
minutes, but is usable all the same. As long as you keep the bit rate
under
9.6 Mbit/sec you can mix and match all you want. Still saving for the
MLP
option though.


Brian C. Peters
Senior Engineer
Dorian Recordings
***@dorian.com




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***@music.vt.edu
http://mail.music.vt.edu/mailman/listinfo/sursound

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Aldo Bazan
2004-01-11 18:40:03 UTC
Permalink
>I mostly use lossless WMA encoding, even for storing audio tracks on my
>hand-held USB headphone player (NAPA). They simply sound far better tham MP3
>files...
>Angelo Farina

a 1-min standard redbook cd-audio wav file (which is +-10Mb per minute)
what size become when compressed with WMA lossless?

F. Aldo Bazan

e-mail:
***@aug.org
Christian
2004-01-10 19:04:03 UTC
Permalink
Well,

I have the seagate 120 Gig HD coming, I have the outboard USB 2.0
Plextor DVD-RW coming...

I was thinking of getting the M-Audio outboard USB 2.0 sound card,
M-audio Audiophile USB. http://m-audio.rjmg.com/index.cfm?pid=3372
... Is this good for both a HTPC as well as just for a music server? I
need DTS complince, 24/96, etc... Is it possible to get DVD-Audio passed
through the digital coax output? I don't believe digital coax is limited
in bandwidth simply because DVD-Audio is only 10 Mbit/sec... Any
soundcards that will pass the MLP output because I have a Meridian
861... In other words, it would be great if a sound card simply output
the MLP through the coax digital output. Unless this is an SPDIF format
issue, which is silly because there is a damn coax cable, so it
technically shouldn’t be a bandwidth issue. MLP is encrypted so what's
the big deal... Don’t understand why Meridian needs 3 coax cables for
their smart link.

Either way I would appreciate advice on which high end sound card to
get, or else I'll get the one in the link above I guess. Any problem
running 60 feet of coax from the sound card to the Meridian??

So, that leaves me mostly with picking and chosing through many ripping
software. I'll worry about how to rip DD and DTS stuff later, but for
now, I need help figuring out how to best way to store/play/rip my CD
44.1 music to the hard drive. As someone mentioned, I don’t want to do
this very often, if only once! This is a huge effort.

Since I know I am not interested in any lossy compression, which method
do you recommend: Uncompressed or lossless? Although don’t you
'technically' alter/lose something going through the lossless encoding
process? Not a big deal, but technically, lossless compression doesn’t
mean identical to the master, only the ability to make perfect
generational copies once the encoding process is complete? Or am I
wrong? Not a big deal...

Knowing I desire lossless or uncompressed storage for my PC, what would
you recommend? I have Roxio 6.0 DVD/CD which came with the Plextor
drive. I don’t know what limitations there are, nor do I know what
advantages other software has... I know eventually, very soon I want to
rip DVD to get the music from the VOB files on DVD for both DD and DTS.
I will also likely want to edit the music for better fades. I used to
use Turtle beach software a while back for that... I even had the
original Multisound Card from Turtle beach when it cost $995.00!! It
seems I hit medical school, and then residency, and I have amnesia as
well as don’t know what happened to that wave software, but now it seems
that software for this purpose is abundant and even free... I don’t mind
paying for good software, but if share/freeware is equal or better,
that’s great.

Could someone please summarize what I'll need for my purposes?

1) uncompressed or lossless compression?
2) Sound card?
3) I will use 60 feet of Coax cable from the digital output of the
M-Audio to the Meridian 861 in another room. (Should not be an issue.)
4) Which software to use for said purposes? EAC, FLAC, Windows, Roxio,
aye carumba!!!

I'll get a USB 2.0 drive for backup.

I would really appreciate advice on questions 1, 2, 3, 4

Thanks!

Chris


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Angelo Farina
2004-01-10 20:20:51 UTC
Permalink
> -----Original Message-----
> Don't understand why
> Meridian needs 3 coax cables for their smart link.

Because those are actually three SPDIF interfaces, each one carrying 2
channels. But, due to copy-protection issues, Meridian encripted these SPDIF
interfaces in some way, so that a normal SPDIF receiver cannot get the audio
stream from them...
I am sorry to say that I do not see any way in which the MLP stream can be
digitally transferred to a set of external converters, apart, of course, the
Meridian proprietary solution.
The most clever solution, in my opinion, would be to decrypt the MLP data
directly on the PC, then drive a set of professional external DA converters
(I have the Apogee DAC 16, and they sound really fine - and the new Apogee
Rosetta should be even better...).
But, at the moment, there is no software available for ripping the MLP data
from a DVD-audio...
Bye!

Angelo Farina
Kurt Albershardt
2004-01-10 20:27:32 UTC
Permalink
Why not compress them with FLAC and then play back with an app that can understand it?

Foobar2000, Winamp, etc. all have FLAC plugins now...
Angelo Farina
2004-01-10 20:37:27 UTC
Permalink
> -----Original Message-----
> From: sursound-***@music.vt.edu
> Although don't you 'technically' alter/lose something going
> through the encoding process? Not a big deal, but
> technically, lossless compression doesn't mean identical to
> the master, only the ability to make perfect generational
> copies once the encoding process is complete? Or am I wrong?
Lossless means identical to original, bit by bit. It is just like zipping
(or RARing) computer files. Once unzipped, they ARE the original...
Try what follows. Take a computer file (an exacutable, perhaps), add to it
the leading 44 bytes of a WAV file containing the same number of bytes, and
compress it with WMA encoder. Then open it with Adobe Audition (which
contains the WMA lossless decoder),a nd dsave it back to a new WAV file.
Strip away the leading 44 bytes (Wav header), rename it to the opriginal
extensions (.EXE), and You will be able to run it....
Bye!

Angelo Farina

> There has been talk on the Meridian Forum of Dolby Digital
> Plus and DVD 2.0 specifications (HD-DVD). One Member of the
> Meridian Forum says MLP
> (DVD-Audio) and AAC techniques would be possible all under
> the DD Plus label. Have you heard anything about Dolby
> Digital Plus?

If the DOA supports WM9 for video, why not for audio too? It is more
efficient than MLP, and encoders and decoders are completely free...
I always use WM9 for storing my video and audio recordings, this format is
simply great, and anyone with a PC can play these audio and/or video files.
I mostly use lossless WMA encoding, even for storing audio tracks on my
hand-held USB headphone player (NAPA). They simply sound far better tham MP3
files...
Bye!

Angelo Farina
Christian
2004-01-10 20:37:56 UTC
Permalink
Well, that's why I posted. The last 7.5 years have left me with little
time to keep up to date due to medical school and residency, which I'm
in my last 1.5 years out of a five year residency. I'm afraid that all
I remember is the original Turtle Beach Systems Multisound card that
cost $995.00 and the WAVE/QUAD software :-) Since I have a Meridian
861, I never considered the PC 'audiophile' caliber until recently now
that outboard soundcards are available that strictly pass digital info
and no analog involved... I never got into the MP3 nonsense strictly
because I used a portable DAT to go running with, etc... Now I'm trying
to play catch up and learn what software I can use for ripping my CD
collection to the 120 Gig HD in lossless/uncompressed format... what
else do I have to consider? How do I manage all these songs? I'm so
out of it seems... How to rip, manage, play, organize... Can anyone
suggest a complete software solution?

I already have Roxio 6.0 DVD/CD software... does that have FLAC? I
don’t know what is better about Foobar2000, Winamp, etc... I guess if
FLAC is the best way to losslessly compress audio, I should use it. I
guess there is no difference between the original uncompressed red book
audio and the lossless compressed Flac file?


-----Original Message-----
From: sursound-***@music.vt.edu
[mailto:sursound-***@music.vt.edu] On Behalf Of Kurt Albershardt
Sent: Saturday, January 10, 2004 3:28 PM
To: Surround Sound discussion group
Subject: RE: [Sursound] Finalizing plans for a PC based music player...

Why not compress them with FLAC and then play back with an app that can
understand it?

Foobar2000, Winamp, etc. all have FLAC plugins now...




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http://mail.music.vt.edu/mailman/listinfo/sursound

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Kurt Albershardt
2004-01-10 21:00:38 UTC
Permalink
--On Saturday, January 10, 2004 3:37 PM -0500 Christian <***@comcast.net> wrote:
>
> I already have Roxio 6.0 DVD/CD software... does that have FLAC? I
> don’t know what is better about Foobar2000, Winamp, etc... I guess if
> FLAC is the best way to losslessly compress audio, I should use it. I
> guess there is no difference between the original uncompressed red book
> audio and the lossless compressed Flac file?

Change that to "lossless decompressed Flac file" and you're there.

FLAC is one of several lossless compressors out there, but is the only one that's truly open source and royalty free. http://flac.sourceforge.net/


Foobar2000 is an open source media player <http://www.foobar2000.org/> It has a no-nonsense interface and a lot of power.

Winamp is another popular player, though I find it a little too gadget-like for my tastes.
Ken L. Holder
2004-01-11 01:27:50 UTC
Permalink
At 02:00 PM 1/10/2004, Kurt Albershardt wrote:
>Winamp is another popular player, though I find it a little too gadget-like for my tastes.


I rather like Zinf: http://www.zinf.org/ which is an open source and
cross-platform audio player. No FLAC plug-in apparently.

Ken Holder
Angelo Farina
2004-01-11 17:21:50 UTC
Permalink
Uh, I did not know about FLAC, so I had a look at it. Very nice,
particularly for the fact that it has a CoolEdit (Adobe Audition)
input/output filter which works very smoothly (and is much faster than the
Ogg Vorbis filter).
However, FLAC is just stereo, so it deserves little mention here (most
surround formats require more than 2 channels). All players who support
FLAC are also just stereo...
It seems to me that the Microsoft's WMA9 is still on the edge. In fact, WMA9
allows to compress multichannel streams (up to 8 channels), Windows Media
Player natively supports the multichannel WMA9 files and smoothly plays them
on multichannel sound cards.
Furthermore, going to compression efficiency and compression speed, I have
got the following results (starting from a CD-ripped waveform of 136.6 s at
44100 Hz, 16 bits, stereo):
WAV : save time 5s, file size 24,096,476 bytes
OGG : save time 45s, file size 2,466,793 bytes
MP3 : save time 8s, file size 2,186,344 bytes
WMA9: save time 13s, file size 13,771,215 bytes
FLAC: save time 7s, file size 14,092,210 bytes
This means that FLAC is indeed a very good lossless compressor (I used a
sample with very "dense" music, which is very difficult to compress) - not
as efficient as WMA9-lossless, but significantly faster.... It is really a
pity that it does not support yet multichannel files...
Perhaps we should ask this extensions to the FLAC developers....
As FLAC internally support the Mid-Side format, it should be very efficient
to use it for compressing 4-channels B-format files....
It would be optimal to have FLAC-powered versions of B-player and B-recorder
VST plugins....
Bye!

Angelo Farina

> -----Original Message-----
> From: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu] On Behalf Of Kurt Albershardt
> Sent: 10 January 2004 21:28
> To: Surround Sound discussion group
> Subject: RE: [Sursound] Finalizing plans for a PC based music
> player...
>
> Why not compress them with FLAC and then play back with an
> app that can understand it?
>
> Foobar2000, Winamp, etc. all have FLAC plugins now...
>
>
>
>
> _______________________________________________
> Sursound mailing list
> ***@music.vt.edu
> http://mail.music.vt.edu/mailman/listinfo/sursound
>
Kurt Albershardt
2004-01-11 18:21:28 UTC
Permalink
--On Sunday, January 11, 2004 6:21 PM +0100 Angelo Farina <***@pcfarina.eng.unipr.it> wrote:
>
> Furthermore, going to compression efficiency and compression speed, I have
> got the following results (starting from a CD-ripped waveform of 136.6 s at
> 44100 Hz, 16 bits, stereo):
> WAV : save time 5s, file size 24,096,476 bytes
> OGG : save time 45s, file size 2,466,793 bytes
> MP3 : save time 8s, file size 2,186,344 bytes
> WMA9: save time 13s, file size 13,771,215 bytes
> FLAC: save time 7s, file size 14,092,210 bytes
> This means that FLAC is indeed a very good lossless compressor (I used a
> sample with very "dense" music, which is very difficult to compress) - not
> as efficient as WMA9-lossless, but significantly faster....

I don't know about WMA9, but FLAC is reportedly more efficient than SHN is with 24-bit PCM. SHN is usually a little better on 16-bit.



> It is really a
> pity that it does not support yet multichannel files...
> Perhaps we should ask this extensions to the FLAC developers....

Will you write them, or should several of us perhaps?




Congrats on the Waves deal.
smoerk
2004-01-11 23:08:05 UTC
Permalink
Kurt Albershardt wrote:

> I don't know about WMA9, but FLAC is reportedly more efficient than SHN
> is with 24-bit PCM. SHN is usually a little better on 16-bit.

FLAC is very fast in decoding, which makes it very suitable for
multi-channel usage.

>> It is really a
>> pity that it does not support yet multichannel files...
>> Perhaps we should ask this extensions to the FLAC developers....

afaik it supports up to 8 channels, but there is no option to specify it
further (the file is for 5.1, 5.0, 7.1 or ambisonics, etc).

a solution could be flac codec in ogg or matroska container (i don't
know if this is possible yet or if we have to wait some weeks, months or
years).
James H. Cloos Jr.
2004-01-25 03:25:13 UTC
Permalink
>>>>> "smoerk" == smoerk <***@gmx.de> writes:

smoerk> a solution could be flac codec in ogg or matroska container (i
smoerk> don't know if this is possible yet or if we have to wait some
smoerk> weeks, months or years).

Ogg/flac is a standard option of the main flac tools. And it would
seem extremely were matroska to not support it.

Whether other tools that have plugins et al -- especially the non-
libre ones -- for flac support those options I cannot speculate.

-JimC
Gianni Ricciardi
2004-01-13 20:28:01 UTC
Permalink
Many thanks to everyone the helps me in this hard topic.

Finally I've read all your mails and used a compromise of almost all method,
producing the LCR channels as follows:

C=(L+R)*K1
L'=L-R*K2
R'=R-L*K2

L+R is simply the mono track created mixing togheter L and R
To produce L' and R' I used a "subraction channel": a phase inverted stereo
channel that contains the original stereo with channel swapped in (R) (L).
Then I mixed original stereo and subraction channel to front, and mono
channel to C.
Playing with faders I've adjusted K1 and K2.

Finally I've found the best compromise that doesn't destroy the stereo
picture and has a sufficient dialogue separation on the center.
It's something around -6 db for the subtraction channel (relative to the
original) and -6 for C (In my opinion)

This is like a 0.5 factor of attenuation.

Mixing the subraction at 0db (as sayied by Viriato) would produce a complete
Mid-Side picture that can't be used because we should be sure that LCR
speakers are perfectly positioned in the front of the lister to reconstruct
the original LR.

It's a compromise but it sounds good.

Thanks to All

GIANNI

-----Messaggio originale-----
Da: sursound-***@music.vt.edu
[mailto:sursound-***@music.vt.edu]Per conto di VDOSH
Inviato: domenica 11 gennaio 2004 5.43
A: Surround Sound discussion group
Oggetto: Re: [Sursound] LCR Upmix


Hello Eberhard Sengpiel,

I'm afraid I'm not enough involved in maths right now to answer you in this
way, but I took the time to make a little html page to show how can this be
done with the audio editor I use, Samplitude v7. It works, so I just can't
see another way to extract three L,C,R mono signals from a stereo file with
the stereo image remaining unchanged.

The link: http://mapage.noos.fr/fruizelegum/mslcr.htm


Best regards

Viriato de Oliveira


----- Original Message -----
From: "Eberhard Sengpiel" <***@t-online.de>
To: "Surround Sound discussion group" <***@music.vt.edu>
Sent: Friday, January 09, 2004 8:08 AM
Subject: Re: [Sursound] LCR Upmix


> Hello Viriato de Oliveira,
>
> what is different of *your* M-signal to a simple MS stereo
> matrix, where M = L + R and S = L - R?
> This M-signal is the in phase sum of L + R and will
> narrow your L/R stereo image if you add it as a third
> channel to your stereo. There is easy math that shows it.
>
> Kind regards
>
> Eberhard Sengpiel
> German forum for microphone recording
> and sound studio techniques
> http://www.sengpielaudio.com
>
>
> ----- Original Message -----
> From: "VDOSH" <***@noos.fr>
> To: "Surround Sound discussion group" <***@music.vt.edu>
> Sent: Friday, January 09, 2004 5:18 AM
> Subject: Re: [Sursound] LCR Upmix
>
>
> > Hello Eberhard Sengpiel,
> >
> > In a stereo listening system, this MS decoding process lead to 3
> different
> > and complementary signals wich can be panned left, center and right,
> and the
> > stereo field remains identical to the original; the stereo image can
> also be
> > reworked if needed. Simple MS stereo recording leads to 2 signals: one
> M
> > panned center, and one S that is splitted in two equals signals panned
> left
> > and right, the right being phase inversed for completing the stereo
> image,
> > so at the console there is three signals. Here, we can take a stereo
> file,
> > decode it on two M and S files, and route and split them in three
> signals,
> > L, C, and R for an identical stereo image. I assume that they are a
> good
> > base for a calibrated LCR listening system, that I hadn't the chance
> to
> > experiment. The process is quite simple.
> >
> > Regards,
> > Viriato de Oliveira
> >
> > ----- Original Message -----
> > From: "Eberhard Sengpiel" <***@t-online.de>
> > To: "Surround Sound discussion group" <***@music.vt.edu>
> > Sent: Friday, January 09, 2004 12:41 AM
> > Subject: Re: [Sursound] LCR Upmix
> >
> > > Hallo Viriato de Oliveira,
> > >
> > > your proposal will not work, because the M signal is L + R.
> > > That is no separate signal for the center. If you do this, your
> > > stereo image will be very narrow.
> > >
> > > Cheers
> > >
> > > Eberhard Sengpiel
> > > German forum for microphone recordings
> > > and sound studio techniques
> > > http://www.sengpielaudio.com
> > >
> > > It's easy and simple to do:
> > > - do an MS decoding of the stereo file, then you'll have 3 channels,
> > > left, right and center.
> > >
> > > You can do it very easily with Samplitude, send me a mail if you
> want
> > > that I explain the trick to you.
> > >
> > > I don't use Nuendo, so I don't know how to do Mid-Side decoding with
> > > Nuendo.
> > >
> > > cheers,
> > > Viriato de Oliveira
>
>
>
> _______________________________________________
> Sursound mailing list
> ***@music.vt.edu
> http://mail.music.vt.edu/mailman/listinfo/sursound
Christian
2004-01-21 23:57:08 UTC
Permalink
What do you think of this method to enable one to use the vast storage
capacity of DVD-R instead of CD-R to allow me for longer music samplers
in my car with a DVD player?

http://www.dvdrhelp.com/forum/userguides/193049.php




-----Original Message-----
From: sursound-***@music.vt.edu
[mailto:sursound-***@music.vt.edu] On Behalf Of Aldo Bazan
Sent: Friday, January 09, 2004 11:29 AM
To: Surround Sound discussion group
Subject: Re: [Sursound] Making DVD-R samplers instead of CD-R samplers?

At 10.34 09/01/2004 -0500, you wrote:


>I would like to make samplers of the the songs I like from my CD
>collection as well as my DTS CDs/DVDs and Dolby Digital DVDs to make
>DVD-R samplers for better party music and enjoyment.

first, dvd specs allow for a 48KHz sample, not the 44.1 of CD, so you
will
need to resample them.
second, mixing cd audio and dts cd tracks togheter isn't going to work
so
well; some player may find difficult to play a dts track after a audio
track after a dts etc. Issue of that kind has been observed at least
with
some Pionieer and Toshiba player in the past.
third, for stereo tracks are you going to "unwrap" them in some way, or
leave it as stereo? this is another issue, since you will have some
tracks
that sounds only in the two front speaker, other in all... not
pleaseant, imho.

>I will be outputting the final DVD-R to a Meridian 861. Also, my car
>had a DD/DTS decoder and surround sound phantom center but has 4.1
using
>Dynaudio 3 ways all the way around. Hopefully these DVD players will
be
>able to read DVD-R?

not all players can do it (the panasonic dvd-audio, for example); what
do
you have in the car?

>Also, I should be able to mix and match 44.1 CD with 5.1 DD and DTS
>right? That would be great!!!

there's the serious risk of a total mess... which will output a
wonderful
white noise. 8-)


F. Aldo Bazan

e-mail:
***@aug.org


_______________________________________________
Sursound mailing list
***@music.vt.edu
http://mail.music.vt.edu/mailman/listinfo/sursound

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Christian
2004-01-21 23:58:53 UTC
Permalink
I'm sorry, I forgot to add: does re-sampling from 44.1 to 48 KHz cause
any problems, or does it not sound as good? Or is it simply no big deal
quality wise and should be looked at like upsampling?

-----Original Message-----
From: Christian [mailto:***@comcast.net]
Sent: Wednesday, January 21, 2004 6:57 PM
To: 'Surround Sound discussion group'
Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R samplers?


What do you think of this method to enable one to use the vast storage
capacity of DVD-R instead of CD-R to allow me for longer music samplers
in my car with a DVD player?

http://www.dvdrhelp.com/forum/userguides/193049.php




-----Original Message-----
From: sursound-***@music.vt.edu
[mailto:sursound-***@music.vt.edu] On Behalf Of Aldo Bazan
Sent: Friday, January 09, 2004 11:29 AM
To: Surround Sound discussion group
Subject: Re: [Sursound] Making DVD-R samplers instead of CD-R samplers?

At 10.34 09/01/2004 -0500, you wrote:


>I would like to make samplers of the the songs I like from my CD
>collection as well as my DTS CDs/DVDs and Dolby Digital DVDs to make
>DVD-R samplers for better party music and enjoyment.

first, dvd specs allow for a 48KHz sample, not the 44.1 of CD, so you
will
need to resample them.
second, mixing cd audio and dts cd tracks togheter isn't going to work
so
well; some player may find difficult to play a dts track after a audio
track after a dts etc. Issue of that kind has been observed at least
with
some Pionieer and Toshiba player in the past.
third, for stereo tracks are you going to "unwrap" them in some way, or
leave it as stereo? this is another issue, since you will have some
tracks
that sounds only in the two front speaker, other in all... not
pleaseant, imho.

>I will be outputting the final DVD-R to a Meridian 861. Also, my car
>had a DD/DTS decoder and surround sound phantom center but has 4.1
using
>Dynaudio 3 ways all the way around. Hopefully these DVD players will
be
>able to read DVD-R?

not all players can do it (the panasonic dvd-audio, for example); what
do
you have in the car?

>Also, I should be able to mix and match 44.1 CD with 5.1 DD and DTS
>right? That would be great!!!

there's the serious risk of a total mess... which will output a
wonderful
white noise. 8-)


F. Aldo Bazan

e-mail:
***@aug.org


_______________________________________________
Sursound mailing list
***@music.vt.edu
http://mail.music.vt.edu/mailman/listinfo/sursound

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koen vos
2004-01-22 00:38:30 UTC
Permalink
In principle, such resampling doesn't have to degrade quality.

However, at these lower sampling frequencies, significant quality
differences exist between resampling methods. The problem is that little
public information is available on how to build resamplers of the highest
quality - it's a complex and somewhat unresolved area of research. A
straightforward textbook implementation (let's say, a polyphase
windowed-sinc resampler), may well result in disappointing quality.

All this would make me sceptical of using just anyone's resampler (I have no
experience with the software suggested in your link though). At the same
time, if it's merely for use in a car, you're not likely to hear the
difference, at least while driving ;)

koen.

> -----Original Message-----
> From: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu]On Behalf Of Christian
> Sent: Wednesday, January 21, 2004 3:59 PM
> To: 'Surround Sound discussion group'
> Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R samplers?
>
>
> I'm sorry, I forgot to add: does re-sampling from 44.1 to 48 KHz cause
> any problems, or does it not sound as good? Or is it simply no big deal
> quality wise and should be looked at like upsampling?
>
> -----Original Message-----
> From: Christian [mailto:***@comcast.net]
> Sent: Wednesday, January 21, 2004 6:57 PM
> To: 'Surround Sound discussion group'
> Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R samplers?
>
>
> What do you think of this method to enable one to use the vast storage
> capacity of DVD-R instead of CD-R to allow me for longer music samplers
> in my car with a DVD player?
>
> http://www.dvdrhelp.com/forum/userguides/193049.php
>
>
>
>
> -----Original Message-----
> From: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu] On Behalf Of Aldo Bazan
> Sent: Friday, January 09, 2004 11:29 AM
> To: Surround Sound discussion group
> Subject: Re: [Sursound] Making DVD-R samplers instead of CD-R samplers?
>
> At 10.34 09/01/2004 -0500, you wrote:
>
>
> >I would like to make samplers of the the songs I like from my CD
> >collection as well as my DTS CDs/DVDs and Dolby Digital DVDs to make
> >DVD-R samplers for better party music and enjoyment.
>
> first, dvd specs allow for a 48KHz sample, not the 44.1 of CD, so you
> will
> need to resample them.
> second, mixing cd audio and dts cd tracks togheter isn't going to work
> so
> well; some player may find difficult to play a dts track after a audio
> track after a dts etc. Issue of that kind has been observed at least
> with
> some Pionieer and Toshiba player in the past.
> third, for stereo tracks are you going to "unwrap" them in some way, or
> leave it as stereo? this is another issue, since you will have some
> tracks
> that sounds only in the two front speaker, other in all... not
> pleaseant, imho.
>
> >I will be outputting the final DVD-R to a Meridian 861. Also, my car
> >had a DD/DTS decoder and surround sound phantom center but has 4.1
> using
> >Dynaudio 3 ways all the way around. Hopefully these DVD players will
> be
> >able to read DVD-R?
>
> not all players can do it (the panasonic dvd-audio, for example); what
> do
> you have in the car?
>
> >Also, I should be able to mix and match 44.1 CD with 5.1 DD and DTS
> >right? That would be great!!!
>
> there's the serious risk of a total mess... which will output a
> wonderful
> white noise. 8-)
>
>
> F. Aldo Bazan
>
> e-mail:
> ***@aug.org
>
>
> _______________________________________________
> Sursound mailing list
> ***@music.vt.edu
> http://mail.music.vt.edu/mailman/listinfo/sursound
>
> ---
> Incoming mail is certified Virus Free.
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Christian
2004-01-22 00:48:17 UTC
Permalink
Well, it is hard enough for me to use lossless compression, so the
thought of definite degradation from conversion to 48 kHz from 44.1 kHz
absolutely ruins the appeal of increased storage space. Quantity will
never come before quality. It would be very nice to have one DVD
instead of 5 CD-R's in my car, for a particular genre of music... I was
simply looking to rip some of my favorite tracks from multiple CD's and
make a CD-R sampler instead of a DVD-R sampler for my car, but if
degradation is involved, then that is not an option and I will live with
CD-R...

Chris

-----Original Message-----
From: sursound-***@music.vt.edu
[mailto:sursound-***@music.vt.edu] On Behalf Of koen vos
Sent: Wednesday, January 21, 2004 7:39 PM
To: Surround Sound discussion group
Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R samplers?

In principle, such resampling doesn't have to degrade quality.

However, at these lower sampling frequencies, significant quality
differences exist between resampling methods. The problem is that little
public information is available on how to build resamplers of the
highest
quality - it's a complex and somewhat unresolved area of research. A
straightforward textbook implementation (let's say, a polyphase
windowed-sinc resampler), may well result in disappointing quality.

All this would make me sceptical of using just anyone's resampler (I
have no
experience with the software suggested in your link though). At the same
time, if it's merely for use in a car, you're not likely to hear the
difference, at least while driving ;)

koen.

> -----Original Message-----
> From: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu]On Behalf Of Christian
> Sent: Wednesday, January 21, 2004 3:59 PM
> To: 'Surround Sound discussion group'
> Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R
samplers?
>
>
> I'm sorry, I forgot to add: does re-sampling from 44.1 to 48 KHz cause
> any problems, or does it not sound as good? Or is it simply no big
deal
> quality wise and should be looked at like upsampling?
>
> -----Original Message-----
> From: Christian [mailto:***@comcast.net]
> Sent: Wednesday, January 21, 2004 6:57 PM
> To: 'Surround Sound discussion group'
> Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R
samplers?
>
>
> What do you think of this method to enable one to use the vast storage
> capacity of DVD-R instead of CD-R to allow me for longer music
samplers
> in my car with a DVD player?
>
> http://www.dvdrhelp.com/forum/userguides/193049.php
>
>
>
>
> -----Original Message-----
> From: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu] On Behalf Of Aldo Bazan
> Sent: Friday, January 09, 2004 11:29 AM
> To: Surround Sound discussion group
> Subject: Re: [Sursound] Making DVD-R samplers instead of CD-R
samplers?
>
> At 10.34 09/01/2004 -0500, you wrote:
>
>
> >I would like to make samplers of the the songs I like from my CD
> >collection as well as my DTS CDs/DVDs and Dolby Digital DVDs to make
> >DVD-R samplers for better party music and enjoyment.
>
> first, dvd specs allow for a 48KHz sample, not the 44.1 of CD, so you
> will
> need to resample them.
> second, mixing cd audio and dts cd tracks togheter isn't going to work
> so
> well; some player may find difficult to play a dts track after a audio
> track after a dts etc. Issue of that kind has been observed at least
> with
> some Pionieer and Toshiba player in the past.
> third, for stereo tracks are you going to "unwrap" them in some way,
or
> leave it as stereo? this is another issue, since you will have some
> tracks
> that sounds only in the two front speaker, other in all... not
> pleaseant, imho.
>
> >I will be outputting the final DVD-R to a Meridian 861. Also, my car
> >had a DD/DTS decoder and surround sound phantom center but has 4.1
> using
> >Dynaudio 3 ways all the way around. Hopefully these DVD players will
> be
> >able to read DVD-R?
>
> not all players can do it (the panasonic dvd-audio, for example); what
> do
> you have in the car?
>
> >Also, I should be able to mix and match 44.1 CD with 5.1 DD and DTS
> >right? That would be great!!!
>
> there's the serious risk of a total mess... which will output a
> wonderful
> white noise. 8-)
>
>
> F. Aldo Bazan
>
> e-mail:
> ***@aug.org
>
>
> _______________________________________________
> Sursound mailing list
> ***@music.vt.edu
> http://mail.music.vt.edu/mailman/listinfo/sursound
>
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Angelo Farina
2004-01-22 01:20:14 UTC
Permalink
??? Going from 48 kHz to 44.1 kHz certainly means some degradation, but
going from 44.1 to 48, if done properly, does not causes any degradation...
Many PC soundcards have codecs always working at 48 kHz, and when You play
some 44.1 sources, they upsample on-the-chip...
And Audiophile-grade CD players do the same, but having the codecs running
at 96 kHz, and they claim that this makes the sound more brilliant...
I think it is just necessary to employ proper upsampling algorithms, and no
degradation occurs.
You are placing the content in a more capacitive container (or, better, with
a wider bandwidth), so You do not need to throw away anything of the
original content.
Bye!

Angelo Farina

> -----Original Message-----
> From: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu] On Behalf Of Christian
> Sent: 22 January 2004 01:48
> To: 'Surround Sound discussion group'
> Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R
> samplers?
>
> Well, it is hard enough for me to use lossless compression,
> so the thought of definite degradation from conversion to 48
> kHz from 44.1 kHz absolutely ruins the appeal of increased
> storage space. Quantity will never come before quality. It
> would be very nice to have one DVD instead of 5 CD-R's in my
> car, for a particular genre of music... I was simply looking
> to rip some of my favorite tracks from multiple CD's and make
> a CD-R sampler instead of a DVD-R sampler for my car, but if
> degradation is involved, then that is not an option and I
> will live with CD-R...
>
> Chris
koen vos
2004-01-22 01:35:18 UTC
Permalink
Maybe I sounded a bit too negative then.

Handling music at 44.1 or 48 kHz is a delicate task. That's why there is so
much difference between CD players (besides their analog output stages of
course). Different players have different impulse responses for their
anti-aliasing filters, and some impulse responses sound better than others.
Similar impulse responses are involved in the 44.1 -> 48 kHz conversion. And
just like you probably won't play back CDs on a $39 discman when you desire
the highest quality, you should also be careful in choosing the software
that does the resampling.

Playing 44.1 kHz by first converting to 48 kHz and then feeding it to a
high-end DAC can have a quality very close to playing the 44.1 kHz directly,
as long as you use a high-end sample rate conversion to go to 48 kHz.

If you would be able to upsample from 44.1 kHz to 96 kHz and store that on
DVD, you would be in an even better situation. Then the software resampling
is the only step that is really critical.

koen.

> -----Original Message-----
> From: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu]On Behalf Of Christian
> Sent: Wednesday, January 21, 2004 4:48 PM
> To: 'Surround Sound discussion group'
> Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R samplers?
>
>
> Well, it is hard enough for me to use lossless compression, so the
> thought of definite degradation from conversion to 48 kHz from 44.1 kHz
> absolutely ruins the appeal of increased storage space. Quantity will
> never come before quality. It would be very nice to have one DVD
> instead of 5 CD-R's in my car, for a particular genre of music... I was
> simply looking to rip some of my favorite tracks from multiple CD's and
> make a CD-R sampler instead of a DVD-R sampler for my car, but if
> degradation is involved, then that is not an option and I will live with
> CD-R...
>
> Chris
>
> -----Original Message-----
> From: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu] On Behalf Of koen vos
> Sent: Wednesday, January 21, 2004 7:39 PM
> To: Surround Sound discussion group
> Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R samplers?
>
> In principle, such resampling doesn't have to degrade quality.
>
> However, at these lower sampling frequencies, significant quality
> differences exist between resampling methods. The problem is that little
> public information is available on how to build resamplers of the
> highest
> quality - it's a complex and somewhat unresolved area of research. A
> straightforward textbook implementation (let's say, a polyphase
> windowed-sinc resampler), may well result in disappointing quality.
>
> All this would make me sceptical of using just anyone's resampler (I
> have no
> experience with the software suggested in your link though). At the same
> time, if it's merely for use in a car, you're not likely to hear the
> difference, at least while driving ;)
>
> koen.
>
> > -----Original Message-----
> > From: sursound-***@music.vt.edu
> > [mailto:sursound-***@music.vt.edu]On Behalf Of Christian
> > Sent: Wednesday, January 21, 2004 3:59 PM
> > To: 'Surround Sound discussion group'
> > Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R
> samplers?
> >
> >
> > I'm sorry, I forgot to add: does re-sampling from 44.1 to 48 KHz cause
> > any problems, or does it not sound as good? Or is it simply no big
> deal
> > quality wise and should be looked at like upsampling?
> >
> > -----Original Message-----
> > From: Christian [mailto:***@comcast.net]
> > Sent: Wednesday, January 21, 2004 6:57 PM
> > To: 'Surround Sound discussion group'
> > Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R
> samplers?
> >
> >
> > What do you think of this method to enable one to use the vast storage
> > capacity of DVD-R instead of CD-R to allow me for longer music
> samplers
> > in my car with a DVD player?
> >
> > http://www.dvdrhelp.com/forum/userguides/193049.php
> >
> >
> >
> >
> > -----Original Message-----
> > From: sursound-***@music.vt.edu
> > [mailto:sursound-***@music.vt.edu] On Behalf Of Aldo Bazan
> > Sent: Friday, January 09, 2004 11:29 AM
> > To: Surround Sound discussion group
> > Subject: Re: [Sursound] Making DVD-R samplers instead of CD-R
> samplers?
> >
> > At 10.34 09/01/2004 -0500, you wrote:
> >
> >
> > >I would like to make samplers of the the songs I like from my CD
> > >collection as well as my DTS CDs/DVDs and Dolby Digital DVDs to make
> > >DVD-R samplers for better party music and enjoyment.
> >
> > first, dvd specs allow for a 48KHz sample, not the 44.1 of CD, so you
> > will
> > need to resample them.
> > second, mixing cd audio and dts cd tracks togheter isn't going to work
> > so
> > well; some player may find difficult to play a dts track after a audio
> > track after a dts etc. Issue of that kind has been observed at least
> > with
> > some Pionieer and Toshiba player in the past.
> > third, for stereo tracks are you going to "unwrap" them in some way,
> or
> > leave it as stereo? this is another issue, since you will have some
> > tracks
> > that sounds only in the two front speaker, other in all... not
> > pleaseant, imho.
> >
> > >I will be outputting the final DVD-R to a Meridian 861. Also, my car
> > >had a DD/DTS decoder and surround sound phantom center but has 4.1
> > using
> > >Dynaudio 3 ways all the way around. Hopefully these DVD players will
> > be
> > >able to read DVD-R?
> >
> > not all players can do it (the panasonic dvd-audio, for example); what
> > do
> > you have in the car?
> >
> > >Also, I should be able to mix and match 44.1 CD with 5.1 DD and DTS
> > >right? That would be great!!!
> >
> > there's the serious risk of a total mess... which will output a
> > wonderful
> > white noise. 8-)
> >
> >
> > F. Aldo Bazan
> >
> > e-mail:
> > ***@aug.org
> >
> >
> > _______________________________________________
> > Sursound mailing list
> > ***@music.vt.edu
> > http://mail.music.vt.edu/mailman/listinfo/sursound
> >
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> >
> > _______________________________________________
> > Sursound mailing list
> > ***@music.vt.edu
> > http://mail.music.vt.edu/mailman/listinfo/sursound
> >
>
> _______________________________________________
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> http://mail.music.vt.edu/mailman/listinfo/sursound
>
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>
Christian
2004-01-22 01:45:47 UTC
Permalink
Going to 96 kHz would double the storage space and the source is still
44.1, so at that point, you are starting to take away the appeal of
going to DVD-R.

On another note, Meridian touts their upsampling (861)of CD's from 44.1
to 96 kHz as a big improvement... I never really thought of that as an
improvement, more like unnecessary interpolation... what do you think?

-----Original Message-----
From: sursound-***@music.vt.edu
[mailto:sursound-***@music.vt.edu] On Behalf Of koen vos
Sent: Wednesday, January 21, 2004 8:35 PM
To: Surround Sound discussion group
Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R samplers?

Maybe I sounded a bit too negative then.

Handling music at 44.1 or 48 kHz is a delicate task. That's why there is
so
much difference between CD players (besides their analog output stages
of
course). Different players have different impulse responses for their
anti-aliasing filters, and some impulse responses sound better than
others.
Similar impulse responses are involved in the 44.1 -> 48 kHz conversion.
And
just like you probably won't play back CDs on a $39 discman when you
desire
the highest quality, you should also be careful in choosing the software
that does the resampling.

Playing 44.1 kHz by first converting to 48 kHz and then feeding it to a
high-end DAC can have a quality very close to playing the 44.1 kHz
directly,
as long as you use a high-end sample rate conversion to go to 48 kHz.

If you would be able to upsample from 44.1 kHz to 96 kHz and store that
on
DVD, you would be in an even better situation. Then the software
resampling
is the only step that is really critical.

koen.

> -----Original Message-----
> From: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu]On Behalf Of Christian
> Sent: Wednesday, January 21, 2004 4:48 PM
> To: 'Surround Sound discussion group'
> Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R
samplers?
>
>
> Well, it is hard enough for me to use lossless compression, so the
> thought of definite degradation from conversion to 48 kHz from 44.1
kHz
> absolutely ruins the appeal of increased storage space. Quantity will
> never come before quality. It would be very nice to have one DVD
> instead of 5 CD-R's in my car, for a particular genre of music... I
was
> simply looking to rip some of my favorite tracks from multiple CD's
and
> make a CD-R sampler instead of a DVD-R sampler for my car, but if
> degradation is involved, then that is not an option and I will live
with
> CD-R...
>
> Chris
>
> -----Original Message-----
> From: sursound-***@music.vt.edu
> [mailto:sursound-***@music.vt.edu] On Behalf Of koen vos
> Sent: Wednesday, January 21, 2004 7:39 PM
> To: Surround Sound discussion group
> Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R
samplers?
>
> In principle, such resampling doesn't have to degrade quality.
>
> However, at these lower sampling frequencies, significant quality
> differences exist between resampling methods. The problem is that
little
> public information is available on how to build resamplers of the
> highest
> quality - it's a complex and somewhat unresolved area of research. A
> straightforward textbook implementation (let's say, a polyphase
> windowed-sinc resampler), may well result in disappointing quality.
>
> All this would make me sceptical of using just anyone's resampler (I
> have no
> experience with the software suggested in your link though). At the
same
> time, if it's merely for use in a car, you're not likely to hear the
> difference, at least while driving ;)
>
> koen.
>
> > -----Original Message-----
> > From: sursound-***@music.vt.edu
> > [mailto:sursound-***@music.vt.edu]On Behalf Of Christian
> > Sent: Wednesday, January 21, 2004 3:59 PM
> > To: 'Surround Sound discussion group'
> > Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R
> samplers?
> >
> >
> > I'm sorry, I forgot to add: does re-sampling from 44.1 to 48 KHz
cause
> > any problems, or does it not sound as good? Or is it simply no big
> deal
> > quality wise and should be looked at like upsampling?
> >
> > -----Original Message-----
> > From: Christian [mailto:***@comcast.net]
> > Sent: Wednesday, January 21, 2004 6:57 PM
> > To: 'Surround Sound discussion group'
> > Subject: RE: [Sursound] Making DVD-R samplers instead of CD-R
> samplers?
> >
> >
> > What do you think of this method to enable one to use the vast
storage
> > capacity of DVD-R instead of CD-R to allow me for longer music
> samplers
> > in my car with a DVD player?
> >
> > http://www.dvdrhelp.com/forum/userguides/193049.php
> >
> >
> >
> >
> > -----Original Message-----
> > From: sursound-***@music.vt.edu
> > [mailto:sursound-***@music.vt.edu] On Behalf Of Aldo Bazan
> > Sent: Friday, January 09, 2004 11:29 AM
> > To: Surround Sound discussion group
> > Subject: Re: [Sursound] Making DVD-R samplers instead of CD-R
> samplers?
> >
> > At 10.34 09/01/2004 -0500, you wrote:
> >
> >
> > >I would like to make samplers of the the songs I like from my CD
> > >collection as well as my DTS CDs/DVDs and Dolby Digital DVDs to
make
> > >DVD-R samplers for better party music and enjoyment.
> >
> > first, dvd specs allow for a 48KHz sample, not the 44.1 of CD, so
you
> > will
> > need to resample them.
> > second, mixing cd audio and dts cd tracks togheter isn't going to
work
> > so
> > well; some player may find difficult to play a dts track after a
audio
> > track after a dts etc. Issue of that kind has been observed at least
> > with
> > some Pionieer and Toshiba player in the past.
> > third, for stereo tracks are you going to "unwrap" them in some way,
> or
> > leave it as stereo? this is another issue, since you will have some
> > tracks
> > that sounds only in the two front speaker, other in all... not
> > pleaseant, imho.
> >
> > >I will be outputting the final DVD-R to a Meridian 861. Also, my
car
> > >had a DD/DTS decoder and surround sound phantom center but has 4.1
> > using
> > >Dynaudio 3 ways all the way around. Hopefully these DVD players
will
> > be
> > >able to read DVD-R?
> >
> > not all players can do it (the panasonic dvd-audio, for example);
what
> > do
> > you have in the car?
> >
> > >Also, I should be able to mix and match 44.1 CD with 5.1 DD and DTS
> > >right? That would be great!!!
> >
> > there's the serious risk of a total mess... which will output a
> > wonderful
> > white noise. 8-)
> >
> >
> > F. Aldo Bazan
> >
> > e-mail:
> > ***@aug.org
> >
> >
> > _______________________________________________
> > Sursound mailing list
> > ***@music.vt.edu
> > http://mail.music.vt.edu/mailman/listinfo/sursound
> >
> > ---
> > Incoming mail is certified Virus Free.
> > Checked by AVG anti-virus system (http://www.grisoft.com).
> > Version: 6.0.560 / Virus Database: 352 - Release Date: 1/8/2004
> >
> >
> > ---
> > Outgoing mail is certified Virus Free.
> > Checked by AVG anti-virus system (http://www.grisoft.com).
> > Version: 6.0.560 / Virus Database: 352 - Release Date: 1/8/2004
> >
> >
> > ---
> > Outgoing mail is certified Virus Free.
> > Checked by AVG anti-virus system (http://www.grisoft.com).
> > Version: 6.0.560 / Virus Database: 352 - Release Date: 1/8/2004
> >
> >
> > _______________________________________________
> > Sursound mailing list
> > ***@music.vt.edu
> > http://mail.music.vt.edu/mailman/listinfo/sursound
> >
>
> _______________________________________________
> Sursound mailing list
> ***@music.vt.edu
> http://mail.music.vt.edu/mailman/listinfo/sursound
>
> ---
> Incoming mail is certified Virus Free.
> Checked by AVG anti-virus system (http://www.grisoft.com).
> Version: 6.0.560 / Virus Database: 352 - Release Date: 1/8/2004
>
>
> ---
> Outgoing mail is certified Virus Free.
> Checked by AVG anti-virus system (http://www.grisoft.com).
> Version: 6.0.560 / Virus Database: 352 - Release Date: 1/8/2004
>
>
> _______________________________________________
> Sursound mailing list
> ***@music.vt.edu
> http://mail.music.vt.edu/mailman/listinfo/sursound
>

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koen vos
2004-01-22 02:33:06 UTC
Permalink
> On another note, Meridian touts their upsampling (861)of CD's from 44.1
> to 96 kHz as a big improvement... I never really thought of that as an
> improvement, more like unnecessary interpolation... what do you think?

Upsampling is definetely useful when the subsequent processing is nonlinear
and/or time-variant (such as the THX processing in the 861).
For things like EQ or room compensation, which should be linear
time-invariant, there is no real theoretical improvement, AFAIK.

koen.
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