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Dolby ProLogic 2 (DS2)
user
post May 18 2002, 20:29
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Please excuse me, if we go slightly off topic with DS2, but it belongs a little bit to multichannel.



Hi DSPGuru,

Yea, of course Besweet will use latest azid.

So I suggested some time ago at doom9, that Midas tries to implement DS2 into azid.
That would be very useful. DA told me, that he has told Midas the way it works.

Perhaps you could encourage Midas to do it, please.

Of course DS2 is not defined publicly by a company biggrin.gif , but DA (mixing DS2 stereo waves) and me (listening tests regarding channel separation and compatibility to normal DS) have found a way how it works.
Meanwhile several people have confirmed my findings, there is channel separation between rear L and R (if decoded by Logic7 or DPL2, Circle surround), and these music/soundtracks are very compatible, as well as they would have been coded as normal Dolby ProLogic, if decoded by normal Dolby Surround decoders.


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DSPguru
post May 18 2002, 20:37
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QUOTE
Originally posted by user
Please excuse me, if we go slightly off topic with DS2, but it belongs a little bit to multichannel.
we really shouldn't discuss this here..

QUOTE
Perhaps you could encourage Midas to do it, please.
yea, i have some influence on Midas. but i'm afraid he doesn't want to be too active in our scene.
but you know what ? i'll give it a try & mail him.

QUOTE
Of course DS2 is not defined publicly by a company biggrin.gif..
please pay attention to my above statement.
DS2 defines encoding process. it's a strict defenition. AAC doesn't.
DS2 doesn't define the decoding process. AAC does (strict).
that's a big difference.

i don't have a ds2 decoder at home, so can't test, but if you'll tell me what are the downmix coefficients, i can write a plugin that implements it.
(BeSweet supports external plugins, remember wink.gif ?)


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user
post May 19 2002, 09:16
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hmm, DA has not told me how it works exactly, he sent me sound for testing.
He should have sent the knowhow to Midas.
Perhaps you can ask Midas to send it to you ?

Since some weeks (1 or 2) DA has not appeared at any known group in internet, like doom9 or German www.dvdboard.de.
And his mailbox is full, so I cannot send him mail. Perhaps DA is on holiday.


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DSPguru
post Jun 8 2002, 18:37
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i've released a plugin for BeSweet that should do something like DPL2 downmix. you're invited to download it from my webpage and test it.
the sources are included, so you can modify whatever you like.


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user
post Jun 8 2002, 19:36
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I will see, I will try with latest Besweet&GUi.
DA has told me, that all information would be included in the thread at doom9:
Additionally I have found in old mails, that we had figured out 3db as difference between left and right rear.


DA:
Hey Midas,

it would be great if you could add DS for DPL2 support in azid ("Surround2"?). Unfortunately you didn't reply to my mail, so I am trying it here:

According to my experiments (with help of "user/Anwender") you need to do following to get seperation for the rears with DPL2/CircleSurround/Logic7 amp:

The rears must be mixed down with a difference of 3db ie.

instead of mixing SR into Lt and Rt with -3db, I guess it needs to be done like -1.5db into Rt and -4.5db into Lt
same applies to SL: -1.5db into Lt and -4.5db into Rt.

Of course I am not sure about the absolute values (-4.5db and -1.5db), but I know that the difference should be at least 2db, best 3db. This downmix is still compatible with DPL, at least the speparation on the front axis seems to be good enough.

As I said I am not sure about the factors. Maye -1.5db and -4.5db is correct, maybe 0db and -3db. I have to test further.


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DSPguru
post Jun 8 2002, 19:49
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here's the matrix i used :
CODE
buffer[j]   =buffer[i++]* 0.1441;    // fl. -16db -> l.  

buffer[j]  +=buffer[i]  * 0.6436;    // c.   -3db -> l.

buffer[j+1] =buffer[i++]* 0.6436;    // c.                 -3db -> r.

buffer[j+1]+=buffer[i++]* 0.1441;    // fr.         -16db -> r.

buffer[j]  +=buffer[i]  * 0.1284;  // sl  -17db -> l.

buffer[j+1]+=buffer[i++]*-0.0909;  // sl      -20db -> r.

buffer[j]  +=buffer[i]  * 0.0909;  // sr  -20db -> l.

buffer[j+1]+=buffer[i++]*-0.1284;  // sr      -17db -> r.


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user
post Jun 8 2002, 21:18
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From Downmix 0.1 I have just to unzip BS_Downmix.dll (and ? BS_Downmix.c ) to BS's directory ?


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DSPguru
post Jun 8 2002, 21:25
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extract the dll, put it in the BeSweet v1.4b1 directory, and append the following string to your commandline :
CODE
-plugin( -name BS_Downmix.dll -func NoLFE -6ch )


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user
post Jun 8 2002, 22:37
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"d:Eigene DateienFilmeDVD Video Audio ProgrammeBeSweet ac3 to mp2BeSweet.exe" -core( -input "d:t AC3 T01 2_0ch 192Kbps DELAY 0ms.wav" -output "d:Eigene DateienFilmewav soundstemp.mp2" ) -plugin( -name BS_Downmix.dll -func NoLFE -6ch -ssrc( --rate 44100 --twopass ) -2lame( -m s -p 1 -b 224 ) -profile( ~~~~~ Default Profile ~~~~~ )

hmm, unfortunately the first trial was not really successful, but I have to check against output of DS2/Headac3he.
The rears had the same output, like old DPL mode.


To explain meaning of commmandline for interested people in generally, but You will know it better:

"d:Eigene DateienFilmeDVD Video Audio ProgrammeBeSweet ac3 to mp2BeSweet.exe"
= location of Besweet.exe

-ssrc( --rate 44100 --twopass )
= 48 kHz to 44,1 kHz wav, for burning CDDA on CD-RW for test purposes, CDDA digitally into HK AVR 5000 with 3 tested modes: 1. pure stereo, 2. "old" DPL, 3. Logic7, same as DPL2, for decoding DS2.

-2lame( -m s -p 1 -b 224 ) :
My standard toolame command line is not used, as I specify in batch mode of BS, that only ac3 to wav 44,1 is applied.

As test 5.1 ac3 I am using 1 minute with spoken content: left, center,right, right surround, left surround, LFE.


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DSPguru
post Jun 8 2002, 22:42
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post your logfile. it looks like you forgot to close the parenthesis of the -plugin( ) section.


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user
post Jun 8 2002, 22:53
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Ah, you think the missing ")" in my above posted command line will be the reason.
I am performing next trial...


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user
post Jun 8 2002, 23:39
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"d:Eigene DateienFilmeDVD Video Audio ProgrammeBeSweet ac3 to mp2BeSweet.exe" -core( -input "d:t AC3 T01 2_0ch 192Kbps DELAY 0ms.wav" -output "d:Eigene DateienFilmewav soundstemp.mp2" ) -plugin( -name BS_Downmix.dll -func NoLFE -6ch) -ssrc( --rate 44100 --twopass ) -2lame( -m s -p 1 -b 224 )


hm, this was not succesful, too.
I have heard no separation between rears.

Perhaps you can try out some Downmix-parameter-matrix-settings with your DPL amp, too.
As I have tried, you can hear the separation of rear left and right, if you switch the mode of your amp to pure stereo.
Then I have heard the rear channels left and right out of front left and right.

And mixing to the rears you can test with DPL setting. Of course you will only be able hear, that it comes out of rears, not if there is really separation.

This you can try out with samples DPl2 at ff123#s page:
http://www.ff123.net/samples/dpl2/

http://www.ff123.net/samples/dpl2/DownMix.ape
http://www.ff123.net/samples/dpl2/Roll.ape
http://www.ff123.net/samples/dpl2/rear-both-left-right.ape

These 3 waves have all more or less separation between left and right rears.


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DSPguru
post Jun 8 2002, 23:58
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as said before.. I NEED THAT LOGFILE !

copy-paste-run the following commandline :
QUOTE
"d:Eigene DateienFilmeDVD Video Audio ProgrammeBeSweet ac3 to mp2BeSweet.exe" -core( -input "d:t AC3 T01 2_0ch 192Kbps DELAY 0ms.wav" -output "d:Eigene DateienFilmewav soundstemp.wav" -2ch -logfile "d:Eigene DateienFilmeBeSweet.txt" ) -plugin( -name BS_Downmix.dll -func NoLFE -6ch ) 
this will create a 2ch-wav file you can test.

and make sure that your input is an AC3 file.


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Frank Klemm
post Jun 9 2002, 00:23
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QUOTE
Originally posted by DSPguru
here's the matrix i used :
CODE
buffer[j]   =buffer[i++]* 0.1441;    // fl. -16db -> l.  

buffer[j]  +=buffer[i]  * 0.6436;    // c.   -3db -> l.

buffer[j+1] =buffer[i++]* 0.6436;    // c.                 -3db -> r.

buffer[j+1]+=buffer[i++]* 0.1441;    // fr.         -16db -> r.

buffer[j]  +=buffer[i]  * 0.1284;  // sl  -17db -> l.

buffer[j+1]+=buffer[i++]*-0.0909;  // sl      -20db -> r.

buffer[j]  +=buffer[i]  * 0.0909;  // sr  -20db -> l.

buffer[j+1]+=buffer[i++]*-0.1284;  // sr      -17db -> r.


Where can I get the proposed download matrices for
3.0/4.0/4.0 quadrophonic/5.0/5.1/7.0/7.1 channel audio
to 2.0 stereo?

--
Frank Klemm


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DSPguru
post Jun 9 2002, 00:29
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sorry Frank, i don't have official info. it's only a trial-and-error process sad.gif.


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user
post Jun 9 2002, 00:34
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BeSweet v1.4b1 by DSPguru.
--------------------------
Using azid.dll v1.8 (b825) by Midas (midas@egon.gyaloglo.hu).
Using BS_Downmix.dll v0.1 by DSPguru (http://DSPguru.doom9.org).

Logging start : 06/09/02 , 01:27:14.

d:Eigene DateienFilmeDVD Video Audio ProgrammeBeSweet ac3 to mp2BeSweet.exe -core( -input d:Eigene DateienMy MusicAC3tests 5.151 AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3 -output d:Eigene DateienMy MusicAC3tests 5.1ds2.wav -2ch -logfile D:Eigene DateienFilmeDVD Video Audio ProgrammeBeSweet ac3 to mp2BeSweet.log ) -plugin( -name BS_Downmix.dll -func NoLFE -6 ) -azid( -s surround -L 0db -S 0db --maximize ) -profile( ~~~~~ Default Profile ~~~~~ )

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : d:Eigene DateienMy MusicAC3tests 5.151 AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3
[00:00:00:000] | Output: d:Eigene DateienMy MusicAC3tests 5.1ds2.wav
[00:00:00:000] | Floating-Point Process: No
[00:00:00:000] +-------- AZID -------
[00:00:00:000] | Output Stereo mode: Dolby surround compatible
[00:00:00:000] | Total Gain: 14.9dB, Compression: None
[00:00:00:000] | LFE levels: To LR 0.0dB, To LFE 0.0dB
[00:00:00:000] | Center mix level: BSI
[00:00:00:000] | Surround mix level: 0.0db
[00:00:00:000] | Dialog normalization: No
[00:00:00:000] | Rear channels filtering: No
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] +---------------------
[00:01:00:000] Conversion Completed !
[00:01:00:000] Actual Avg. Bitrate : 512kbps
[00:00:13:000] <-- Transcoding Duration
Logging ends : 06/09/02 , 01:27:27.

Here it is.

I have seen that my first trials had one big misconception:
I used batch encodng with .exe proggies.
Of course no use of new DPL2-dll !

Then I used Besweet mode, first with ssrc.dll for 48 to 44,1, then without downsampling to create a 48 kHz wave.
Because the 44,1 wave was too short, only 3,8 MB instead of 10 MB ( 1minute) !
But now the 48 kHz wave is still too short, it has resultedt to only 3,8 MB, too..

I need some sleep now, Goodnight...
Tomorrow I have to do a lot work, I will see, when I have time to deal with it again.

My email:


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DSPguru
post Jun 9 2002, 00:36
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wrong commandline sad.gif...
you wrote "-6" instead of "-6ch".
btw, downsampling will just waste your time, and nonrelevant for this test wink.gif.


@frank
http://dspguru.notrace.dk/effects.html


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user
post Jun 9 2002, 00:38
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hmm, should I delete just (-6ch) ?

-plugin( -name BS_Downmix.dll -func NoLFE ) is this correct ?


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user
post Jun 9 2002, 00:42
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BeSweet v1.4b1 by DSPguru.
--------------------------
Using azid.dll v1.8 (b825) by Midas (midas@egon.gyaloglo.hu).
Using BS_Downmix.dll v0.1 by DSPguru (http://DSPguru.doom9.org).

Logging start : 06/09/02 , 01:40:14.

d:Eigene DateienFilmeDVD Video Audio ProgrammeBeSweet ac3 to mp2BeSweet.exe -core( -input d:Eigene DateienMy MusicAC3tests 5.151 AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3 -output d:Eigene DateienMy MusicAC3tests 5.1ds2.wav -2ch -logfile D:Eigene DateienFilmeDVD Video Audio ProgrammeBeSweet ac3 to mp2BeSweet.log ) -plugin( -name BS_Downmix.dll -func NoLFE ) -azid( -s surround -L 0db -S 0db --maximize ) -profile( ~~~~~ Default Profile ~~~~~ )

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : d:Eigene DateienMy MusicAC3tests 5.151 AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3
[00:00:00:000] | Output: d:Eigene DateienMy MusicAC3tests 5.1ds2.wav
[00:00:00:000] | Floating-Point Process: No
[00:00:00:000] +-------- AZID -------
[00:00:00:000] | Output Stereo mode: Dolby surround compatible
[00:00:00:000] | Total Gain: 14.9dB, Compression: None
[00:00:00:000] | LFE levels: To LR 0.0dB, To LFE 0.0dB
[00:00:00:000] | Center mix level: BSI
[00:00:00:000] | Surround mix level: 0.0db
[00:00:00:000] | Dialog normalization: No
[00:00:00:000] | Rear channels filtering: No
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] +---------------------
[00:01:00:000] Conversion Completed !
[00:01:00:000] Actual Avg. Bitrate : 512kbps
[00:00:13:000] <-- Transcoding Duration
Logging ends : 06/09/02 , 01:40:27.

still too short, 3,8 MB , it sounds as it is played much too fast.


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DSPguru
post Jun 9 2002, 00:46
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QUOTE
Originally posted by user
hmm, should I delete just (-6ch) ?

-plugin( -name BS_Downmix.dll -func NoLFE )  is this correct ?
looks like you really need some sleep !
the commandline should include :
CODE
-plugin( -name BS_Downmix.dll -func NoLFE -6ch )  

that's why you got :
QUOTE
[00:01:00:000] Actual Avg. Bitrate : 512kbps
instead of 1.5mbps..

why can't you just copy and paste smile.gif ?


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user
post Jun 9 2002, 00:55
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BeSweet v1.4b1 by DSPguru.
--------------------------
Using azid.dll v1.8 (b825) by Midas (midas@egon.gyaloglo.hu).
Using ssrc.dll v1.28 by Naoki Shibata & Dg (shibatch@geocities.co.jp).
Using BS_Downmix.dll v0.1 by DSPguru (http://DSPguru.doom9.org).

Logging start : 06/09/02 , 01:47:36.

d:Eigene DateienFilmeDVD Video Audio ProgrammeBeSweet ac3 to mp2BeSweet.exe -core( -input d:Eigene DateienMy MusicAC3tests 5.151 AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3 -output d:Eigene DateienMy MusicAC3tests 5.1ds2.wav -2ch -logfile D:Eigene DateienFilmeDVD Video Audio ProgrammeBeSweet ac3 to mp2BeSweet.log ) -plugin( -name BS_Downmix.dll -func NoLFE -6ch ) -azid( -s surround -L 0db -S 0db --maximize ) -ssrc( --rate 44100 --twopass )

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : d:Eigene DateienMy MusicAC3tests 5.151 AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3
[00:00:00:000] | Output: d:Eigene DateienMy MusicAC3tests 5.1ds2.wav
[00:00:00:000] | Floating-Point Process: No
[00:00:00:000] +-------- AZID -------
[00:00:00:000] | Output Stereo mode: Dolby surround compatible
[00:00:00:000] | Total Gain: 7.5dB, Compression: None
[00:00:00:000] | LFE levels: To LR 0.0dB, To LFE 0.0dB
[00:00:00:000] | Center mix level: BSI
[00:00:00:000] | Surround mix level: 0.0db
[00:00:00:000] | Dialog normalization: No
[00:00:00:000] | Rear channels filtering: No
[00:00:00:000] +-------- SSRC -------
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] | Dest. Sample-Rate: 44.1KHz
[00:00:00:000] | Attenuation : 0.0db
[00:00:00:000] +---------------------
[00:00:59:968] Conversion Completed !
[00:00:59:968] Actual Avg. Bitrate : 1411kbps
[00:00:47:000] <-- Transcoding Duration
Logging ends : 06/09/02 , 01:48:23.

In a minute I am listening to it, but then I finish for today.


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user
post Jun 9 2002, 01:07
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I think, there is a little bit separation between left and right surround.
But not much. Perhaps you can play with some parameters ?


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DSPguru
post Jun 9 2002, 01:12
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QUOTE
Originally posted by user
I think, there is a little bit separation between left and right surround.
But not much. Perhaps you can play with some parameters ?
if you have a compiler, you can change the params by yourself, and keep testing...


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user
post Jun 9 2002, 01:18
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I don't have a compiler...
"user" in this case...
Some time ago i compiled some fortran77 programs, very funny.

BTW, from June22 to 29th I will be at Lyngby, Koebenhavn, Denmark. Occasionally near by you ?


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DSPguru
post Jun 9 2002, 19:12
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fortran smile.gif ? hehe, cool.
i'm mostly using matlab.. biggrin.gif.

ps,
i never meet with i-net friends. sorry.. ohmy.gif


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RSS Lo-Fi Version Time is now: 27th November 2014 - 12:19