Guide: 2- to 5-channel upmix using Dolby Pro Logic II

[bleo]

This procedure is similar in purpose to GUIDE: Stereo to 5-Channel Surround (http://forum.doom9.org/showthread.php?s=&threadid=29277) but uses Dolby Pro Logic II upmixing.

You can upmix any 2-channel audio format as long as you have the appropriate DirectShow decoder filter installed.

Software required:
- InterVideo WinDVD (http://www.intervideo.com/)
- GraphEdit (http://www.doom9.org/Soft21/Audio/graphedit.rar)
- A hex editor e.g. XVI32 (http://www.chmaas.handshake.de/delphi/freeware/xvi32/xvi32.htm)

Procedure:
- In GraphEdit go to File > Render Media File... > Open your 2-channel audio file
- Delete the Default DirectSound Device filter
- Go to Graph > Insert Filters... > DirectShow Filters > InterVideo Audio Processor Fx > Insert Filter
- Also insert the WAV Dest filter and File writer filter with an output filename e.g. output.wav
- Connect the audio decoder to InterVideo Audio Processor Fx, then WAV Dest, then the File writer filter
- Right click the InterVideo Audio Processor Fx filter > Filter Properties... > 6 SPEAKER
- Go to InterVideo Container DMO > InterVideo EXP DMO > Append <---
- Click Play and then immediately click Stop to 'initialise' the filter
- Go back to Filter Properties > InterVideo EXP DMO tab > ProLogic2
- Click Play and wait until the button reactivates
- The output file has an incorrect 2-channel WaveFormatEx header!
- Use a hex editor to change:
- nChannels at 16 from 02 to 06
- nAvgBytesPerSec at 1C from 00 EE 02 to 00 CA 08 if your audio is 48 KHz, or from 10 B1 02 to 30 13 08 if it is 44.1 KHz
- nBlockAlign at 20 from 04 to 0C
- You now have a Dolby Pro Logic II upmixed 6-channel WAV file! (empty .1 channel)
- You can reencode it to AC3, AAC, Ogg Vorbis, RealAudio, DTS, WMA9Pro, et al, multi-channel formats using the appropriate programs

Let me know your feedback/problems! Good luck!

BeSplit:
- The demuxed mono output files also have incorrect WAV headers. Use your handy hex editor to change nBlockAlign at 20 from 06 to 02.
- The channel ordering is FL FR C LFE BL BR

WMA9Pro:
Requires WaveFormatExtensible header for direct loading of 6-channel WAV. Use your hex editor to change:
- riffSIZE at 4, add 0x16 to the DWORD here (note little-endian byte order)
- fmtSIZE at 10 from 12 to 28
- wFormatTag at 14 from 01 00 to FE FF
- cbSize at 24 from 00 to 16
- insert at 26, 10 00 3F 00 00 00 01 00 00 00 00 00 10 00 80 00 00 AA 00 38 9B 71

Notes:
- Feel free to tweak the settings on the InterVideo EXP DMO tab to your liking! In general, use Movie mode for DPL(2) encoded material and Music mode if not.
- InterVideo's filter seems to silence the first second of audio...
- Has anyone got a better solution to the WAV header hacking e.g. an automatic program or a better WAV wrapper?

Edit: argh! :eek: On closer inspection, there appears to be a bug in the CyberLink filter where it only does DPL ONE upmixing i.e., the surround channels are identical. Is there a registry/filter hack to invoke DPL2?

I have changed the guide to use InterVideo's filter

[Sycho]

wouldn't it be better if it was in movie mode
music mode is meant for music, not very presice localization

movie mode is very presice, and the surround left is the information between the left and surround, the right surround in the information between the right and surround

[SallyDog]

@bleo

What are you using to split the 6 chan wav?

I tried besplit and wav2wav6 and got 6 mono files, but there's no volume in them. :confused:

[bleo]

@sycho
From Roger Dressler. Dolby Surround Pro Logic II Decoder Principles of Operation. http://www.dolby.com/tech/l.wh.0007.PLIIops.html

The Movie mode uses sufficient delay in the surround channels to ensure the sounds from the front speakers arrive at least 10 ms before the sounds from the surround speakers. This creates the Haas precedence effect, which helps ensure dialogue and other frontal sounds intended to relate with the on-screen action are actually perceived as originating there.

The goal for music playback is to have the sounds from all the speakers arrive at the listener at the same time, which is known as coincident arrival. This helps prevent any smearing or combing of the sounds as they combine from the various speakers.

There is a mild high-frequency shelf filter provided in the surround channels for the Music mode. It results in a more natural, believable soundfield, since ambient sounds normally have a high-frequency rolloff induced by room reflections and absorption.

Lastly, the autobalance is turned off in Music mode, considering vocalists are sometimes deliberately placed off center in the mix.


So which mode you choose depends on your source and preference!

[bleo]

@SallyDog
...well the truth is I never tried splitting it... but I did try BeSplit just then and got some strange mono WaveFormatEx files that played like chipmunks... :confused: Can anyone else help? Perhaps something along the lines of stripping the header and writing a new mono WaveFormatPCM header?...

[specise_8472]

I found that using Wav2wav6 on the resultent intervideo file splits okay. (just don't use the -44 option as it seems to screw up the file sizes, and surcode complains:))

Remap the resultent files from
FR to C
FL to FL
C to FR
SL to sub (the blank file BTW)
SR to SR
Sub to SL

If these are wrong let me know (I,m working on best guess here)
:p

The files play fine as are.
Also you can use cool edit pro 2 to add a perfct subwoofer channel into the mix.

Take the original stero wave and open it in CE.
Then Edit -> Convert Sample Type -> Mono,50%, 50%, 16bit, 44100.
This creates mono file to work with.
Then Effects -> Filters -> FFT Filter -> Preset to only the subwoofer.
You now have a sub channel - be warned that you may need to de-amplify this file as it can override the sub by being to high in volume.

Hope this helps

[Hello]

Is this really that much better than just takin AC3 audio with 2 channels and enabling Pro Logic or DSP so all 5 speakers create the effect?

[bleo]

@Hello
This is the SAME as what you suggested, BUT allows you to output the 5 channels to a WAV file and process it!

[Hello]

Yeah, but I don't know if it's making true, 5 channels and not some effect that I can enable without this.

[bleo]

@Hello
hmm... I'm not sure I understand what you mean... The procedure will give you the same 5 channels as if you went into WinDVD and played a 2-channel file straight to your 5 speakers using 'Dolby PLII', BUT it writes the audio to a WAV file. I suggest you try WinDVD first and see if you like the output.

The procedure makes 'true' 5 channels, as in, they are all different. BUT, the contents of those 5 channels depends on your source. If it was Dolby Surround 2 encoded, then it will decode to 5 channels as the producer intended. If it wasn't, then DPL2 will attempt to use the phase information present in the audio to create two surround channels. Obviously, the results will vary greatly!

[Sycho]

how exactaly do you change the wav header?

[bleo]

@sycho
Use the hex editor XVI32 to load up your WAV file. Go to the byte addresses (shown in the lower left corner as 'Adr. hex') that I have specified and change the bytes to the correct values. Note, the actual field names, e.g. nChannels, are not seen in the WAV header.

[Sycho]

thanx, i was a little confused at first but now i got it

[Sycho]

what would you change in the headed if it decoded 3/0?

[bleo]

For 3/0:
nChannels = 03
nAvgBytesPerSec (nBlockAlign x samples per second) = 00 65 04 (48 KHz) or 98 09 04 (44.1 KHz)
nBlockAlign (bytes per sample x nChannels) = 06

[Eandtc]

My first post (what's up with the damn 5-day waiting period?)

I'm currently capturing the SW LDs with my Sony digital camcorder (video pass-through), and there's definitely pro-logic steering going on (I have the Logitech Z640 5.1 speaker system)

What I would like to do, however, is capture the LPCM soundtrack directly off the laserdisc and steer the soundtrack into 4 .wav's (L,C,R,S - possibly an LFE as well) so that I can make and remix a DD 5.1 soundtrack (i.e. better surround effects.)

Will Besweet do this? If I want to use Pro-Logic II, is there a program that will do this.

BTW, please don't tell me to run it through my receiver. I want to avoid unnecessary A/D conversions, and my reciever is nowhere next to my computer. Additionally, I don't want to record 2 channels at a time and have to re-sync the files on the computer. I'd like to do all this in the computer as much as possible.

Similarly, I don't want a 2.0 soundtrack, because these usually sound better in Pro-logic mode anyways. I'd rather just have a 5.1 soundtrack, so I don't have to switch modes.

[specise_8472]

You will find a Dolby Pro-Logic VST encoder here

http://multiphonie.free.fr/index_nouveau.htm

Use it in Bidule or other VST 'Enabled' program. EG Nuendo

[Eandtc]

Will this decode as well? I need to split the file, not encode it.

(My firewall at work won't let me see the site - I'll have to wait until I get home.)

[specise_8472]

There is also a decoder as well.

The site is in French. The files you want are towards the bottom of the page.

[Sycho]

can you just link to the plugins please?

[Sycho]

@ Eandtc
why don't you just wait till september for the DVD's?
or captuor the AC3 stream

[Eandtc]

I don't want the SE's.

specise_8472, I've not a clue which file (if any) to download from that site.

If I'm reading it right (I had Babelfish translate it for me), it won't work with Vegas 4.0, at least not completely. Not a huge problem, if I can find something it will work with.

[Umma]

I had a b*tch of a time finding links to the software. Finally, when you get to the page with the software compatability charts, scroll down past the chart until you see a series of links on the left. "OTHERS" will take you to the download page.

It doesn't look like you can link to the page directly.

[specise_8472]

It is fun:p navigating the site.

Also on the front-page is an ENGLISH button. Then most of the site is readable. Not all has been translated, but enouth to know waht each VST does.

But persistance pays off.

Anyway all the ones I've downloaded from the page work perfect in Plogue Bidule.

[Eandtc]

bleo, when I try to append the filter, I get this...

Can NOT append DMO, its mediatype may be in-compatible with up/down stream!!!

What am I doing wrong?

[Eandtc]

Okay, I solved that problem (had to delete the matrix mixer as well, this wasn't in the instructions.)

Now, can someone please tell me how to locate and change these values...
- Use a hex editor to change:
- nChannels at 16 from 02 to 06
- nAvgBytesPerSec at 1C from 00 EE 02 to 00 CA 08 if your audio is 48 KHz, or from 10 B1 02 to 30 13 08 if it is 44.1 KHz
- nBlockAlign at 20 from 04 to 0C
I don't know where these values would be.

[ursamtl]

Originally posted by specise_8472
It is fun:p navigating the site.

Also on the front-page is an ENGLISH button. Then most of the site is readable. Not all has been translated, but enouth to know waht each VST does.

But persistance pays off.

Anyway all the ones I've downloaded from the page work perfect in Plogue Bidule.
Hi species, I was just wondering if you used the 12-channel to 5-channel plugin from the site. I tried it but couldn't get anything to work right with it.

Regards,
Steve.

[Eandtc]

Don't you mean "5 to 12"?

[ursamtl]

Originally posted by Eandtc
Don't you mean "5 to 12"?

Actually, it's the SpatMixer 12 plugin and it's supposed to allow mixing up to 12 inputs to 7 outputs. I was trying to use it for 5.1 outputs but I couldn't get anywhere with it. It's not a question of language since I'm fluent in French. I just couldn't get the darned thing to work! :)

[jda]

Originally posted by Hello
Is this really that much better than just takin AC3 audio with 2 channels and enabling Pro Logic or DSP so all 5 speakers create the effect?

Could you tell me simple how to do this.

I want to enable Prologic from a 2 channel ac3 file.


Thanks

[Sycho]

use besweet to convert the ac3 2.0 to a 2channel wav and then just follow the guide at the begining of this thread.

[Gav]

I don't think the Intervideo Pro Logic II filter in graphedit (using Music Mode) is working as it should. I ripped a single song from a CD and ran in through the graphedit program as described at the beginning of this thread. For the options in InterVideo EXP DMO tab I chose Expansion ON, Algorithm ProLogic2, Disable AutoBalance YES, Surround Chan. Shelf Filter YES, Rs Polarity Inversion YES, Panorama Mode YES, Output Channel Config 3/2, Dimension setting 3, Decode Mode Music Mode, Center Width control setting 7(full phantom - no C output). The wav is created and all the other instructions are followed. The center wav is always very loud compared to the other channels. Also, the left and right surround wav's are identical to each other. I suspect that the Dolby Pro Logic 1 instead of 2 is being used even though 2 is specified which is the same problem that bleo had with the Cyberlink filter.

[Shadowfax3000]

This guide is almost perfect for what I want, except for the fact my WAVE file created from GraphEdit is 3.5GB. Now correct me if I'm wrong, but won't I need at least 3.5GB of virtual memory to open this file in a hex editor? XVI32 only accepts files 2GB or smaller, assuming you have the necessary virtual memory. Since I don't really feel like getting 4GB of RAM just to edit the header of a movie's WAVE file, there has gotta be a way to fix the header without opening the entire file. Anybody have any ideas? I have one, but can't implement it.

1. We know the addresses we want changed, and we know what we want to change them to. Can't there be a program that exists or can be made to edit only the addresses needed?

There's gotta be a way!

[Shadowfax3000]

Alright, figured it out (I mean, Eyes Only did). Don't use XVI32 as your hex editor, which uses virtual memory. Use UltraEdit, which uses your hard drive space. For any movie's 6 channel WAVE this guide creates, you probably won't have enough RAM to edit in XVI32. Use UltraEdit. Simple as that.

[Hironimo]

Hi.

I followed the instructions in the first post of this thead.
Then I wanted to convert the file to ac3 using BeSweet.
This is what I got:

E:\>"j:\Drivers\Applications\Audio\Tools\BeSweetv1.4\BeSweet.exe" -core( -input
"k:\FinalFantasy\test.wav" -output "k:\FinalFantasy\test.ac3" ) -ac3enc( -b 448
)
BeSweet v1.4 by DSPguru.
--------------------------
Using AC3enc.dll v0.2 by Gerard Lantau & Dg (http://ffmpeg.org).

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : k:\FinalFantasy\test.wav
[00:00:00:000] | Output: k:\FinalFantasy\test.ac3
[00:00:00:000] | Floating-Point Process: No
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] +------- AC3ENC ------
[00:00:00:000] | Bitrate method : CBR
[00:00:00:000] | AC3 bitrate : 448
[00:00:00:000] | Channels Mode : 5.1
[00:00:00:000] | Error Protection: Yes
[00:00:00:000] +---------------------
Error 32: No input-data was found (wrong substream?)
Quiting...

What's wrong?

Hironimo

[Hironimo]

Hi.

I found the solution to this.
I used BeSweet 1.4 - in this version it requires 7 channels of input for some reason - or so I read.
Skipping to the latest beta fixed this.

The channels do not seem to be assigned properly, when I play the sound I hear no left channel. How to fix this?

Hironimo

[PDU]

Originally posted by Sycho
can you just link to the plugins please? Managed to find som direct links (finally)

Spatencode.zip (http://multiphonie.free.fr/fichiers/spatencode.zip)
Spatdecode.zip (http://multiphonie.free.fr/fichiers/spatdecode.zip)

Cheers

[joshbm]

I've been using this method for a while and I was wondering, is there a way to process 32 bit float/24 bit WAV files? Because each time I attempt to load them it will freeze up and not allow me to continue.

Regards,
Josh

[ursamtl]

Originally posted by joshbm
I've been using this method for a while and I was wondering, is there a way to process 32 bit float/24 bit WAV files? Because each time I attempt to load them it will freeze up and not allow me to continue.

Regards,
Josh

Does it freeze up when you open the 2-channel audio file after choosing "Render Media File"? If so, there may be a problem with your setup or file because 32-bit files load fine for me this way. The problem I get comes later when I try to connect filters, they can't agree on a connection when I use 32-bit files.

An alternative would be to use the Spatdecode VST in a bidule. Spatdecode appears to produce only Pro Logic I output, but for 2-channel music, all DPLII does is add something called a "Panorama" control, which in reality simply sends some of the front channel to the back to give the illusion of "wrapping the soundfield around the sides of the listener." This also gives the illusion of stereo surrounds but in reality, the actual surround material is mono.

Anyway, just hook Spatdecode up to a 6-channel File Recorded leaving the fourth channel blank (for the empty LFE) and it should work.

Regards,
Steve.

[daphy]

quote:
--------------------------------------------------------------------------------
Originally posted by Sycho
can you just link to the plugins please?
--------------------------------------------------------------------------------

Managed to find som direct links (finally)

Spatencode.zip
Spatdecode.zip

Cheers

is already a part of the VST-Collection on needfulthings

[PDU]

Originally posted by ursamtl
An alternative would be to use the Spatdecode VST in a bidule. Spatdecode appears to produce only Pro Logic I output, but for 2-channel music, all DPLII does is add something called a "Panorama" control, which in reality simply sends some of the front channel to the back to give the illusion of "wrapping the soundfield around the sides of the listener." This also gives the illusion of stereo surrounds but in reality, the actual surround material is mono.

Anyway, just hook Spatdecode up to a 6-channel File Recorded leaving the fourth channel blank (for the empty LFE) and it should work.

Regards,
Steve. Hi Steve,
Since I'm pretty new to bidule, I would appreciate if you could explain how to make the File Recorder start recording when play is activated on a 2 ch File Player. Oh, and another question. What output connectors on the Spatdecode VST produce which surround channel?

Spatdecode out 1 = ?
Spatdecode out 2 = ?
Spatdecode out 3 = ?
Spatdecode out 4 = ?
Spatdecode out 5 = ?

Cheers,
Peter

[ursamtl]

Originally posted by PDU
Hi Steve,
Since I'm pretty new to bidule, I would appreciate if you could explain how to make the File Recorder start recording when play is activated on a 2 ch File Player. Oh, and another question. What output connectors on the Spatdecode VST produce which surround channel?

Spatdecode out 1 = ?
Spatdecode out 2 = ?
Spatdecode out 3 = ?
Spatdecode out 4 = ?
Spatdecode out 5 = ?

Cheers,
Peter Hi Peter,

Here`s the procedure for automating the recording so the starting times are aligned:[list=1] Check the "Toggles signal processing" icon (6th icon from top left corner). If the small square in the upper right corner of the icon is green (for "on"), click on it to toggle the square to red (for "off").
Click on the Tools menu and choose Parameters.
In the Parameters dialog box, expand the Audio file Player in the left Source pane and select Playing.
Expand the Audio file Recorder in the right Target pane and select Recording.
Click the Link button and see that the link is listed in the bottom pane
Click X at top right of screen to close Parameters dialog box.
Providing you're not planning to monitor your playback in bidule, go to the Edit menu and make sure the Offline Processing menu item has a check mark beside it. This uses as much of PC's processing power as possible to complete the conversion as fast as possible. Without this, the file is converted at playback speed; that is, a 5-minute song would take 5 minutes to write the file.
Double-click on the Audio file Recorder object to open its dialog box.
Set your bit depth (16, 24, or 32 bits).
Click the button on the right with three periods. This opens a dialog box for you to name and save your 6-channel file (although the dialog box title is "Select a file"). When you close this, do not click the Start button.
Double-click on the Audio file Player object to open its dialog box.
Click on the button at the lower right corner of the Audio file Player dialog box to Open your source audio file. Navigate to the folder containing your file, select the file, then click OK.
Click the Play button on the Audio file Player. Notice that the Start button in the Audio File Recorder changes to Stop, but the Elapsed time counter remains at 00:00:00.
Click the "Toggles signal processing" icon (6th icon from top left corner) to toggle the small square to green (for "on"). The Audio File Player starts playing the file at the same time as the File Recorder starts writing a file. Notice that in offline mode, the Elapsed time readout does not always increment smoothly. Instead, it jumps rapidly in chunks of several seconds, depending on your PC's power.
[/list=1]As for Spatdecode's outputs, I'm not sure as I'm at work right now. I'll try to check it out and get back to you.

Steve.

[PDU]

Hi Steve,

Thanx a lot. Very detailed instructions. I think I'll be able to make this work. I do however have another question - hope you don't mind. Since my input file is 44.1 KHz but AC-3 files should be in 48KHz I would like to make some kind of samplerate conversion. Can this be done without installing any kind of additional VST plugins? If not, can you recommend a freeware plugin for this task? I guess I could always convert my sourcefile to 48KHz before loading it into bidule but it would be nice to do it "on-the-fly".

Cheers,
Peter

[joshbm]

I don't have Bidule up in running right in front of me, but I believe you are able to set the out file as 48 Khz by going to Preferences.

Regards,
Josh

[PDU]

Yes, I've tried that. Unfortunately it applies the "Mickey Mouse" effect to the resulting output (hope I make myself clear).

[ursamtl]

Probably the best freeware sample rate converter I've seen is Voxengo's R8Brain at http://www.voxengo.com/r8brain/. This is a bit of s slow program, but it produces good results.

By the way, I checked out SpatDecode and the channel layout is as I expected, F, R, C, sL, sR. Thus you could connect it the first three SpatDecode pins to the first three Audio File Recorder pins, leave the 4th Recorder pin unconnected, then connect the SpatDecode pins 4 and 5 to Audio File Recorder pins 5 & 6.

By the way, the surrounds are mono. They're both simply the L-R difference signal with sR feed inverted. This is as per Dolby's specs. By putting the surrounds out of phase, localizing specific sounds is more difficult.

Steve.

[joshbm]

@ursamtl

What about hp_ssrc.exe? I've been using that for my sample rate conversions, does R8Brain yield better results?

PS- The normal specs on an AC3 file is 24-bit, 48KHz... correct? Or do they use 32-bit float? What I am talking about is a normal NTSC DVD Movie, Dolby Digital 5.1 AC3 here.

Thanks!
Josh

[ursamtl]

Originally posted by joshbm
@ursamtl

What about hp_ssrc.exe? I've been using that for my sample rate conversions, does R8Brain yield better results?

PS- The normal specs on an AC3 file is 24-bit, 48KHz... correct? Or do they use 32-bit float? What I am talking about is a normal NTSC DVD Movie, Dolby Digital 5.1 AC3 here.

Thanks!
Josh

Yes, you're right. I forgot about ssrc (I don't know about the "hp_" prefix. I've never heard of it and Google turns up nothing). r8brain is better for those who like a GUI, but SSRC seems fine otherwise. I don't know if anyone has test data or anything otherwise scientific for a comparison of the two. I haven't done much work at all converting to 48kHz because my interest so far has been music. I do have some video footage with soundtracks I eventually plan to upmix to 5.1, so one of these days I'll have to get to the 48kHz stuff.

As per the AC3 bit depth, I'm not really sure. I've seen varying discussions on this but no solid answer. For sure 16-bit files should work on all playback devices but obviously 24- or 32-bit files will give better quality. From what I've read the difference between the latter two is virtually inaudible so if the software you're using only support 24-bit, don't sweat it. As a result, you don't really need dithering when reducing from 32 to 24 bits.

Steve.

[PDU]

Offline samplerate conversion is no problem. Both SSRC.EXE and R8Brain are part of my standard collection:) I was however hoping that it was possible to do the conversion inside bidule. Would make things a lot easier and would not affect the source file(s).

@Steve
Thanks for all your help and the pinout explanation for the Spatdecode VST. Speaking of Spatdecode isn't this really a Dolby Prologic decoder:confused: Exactly what does it do when the input file is not Dolby Surround encoded in the first place but just an ordinary stereofile?

Final question - at least for now:)
I need some guidelines for Soft Encode. I have encoded some files using the V.I. bidule (http://forum.doom9.org/showthread.php?s=&threadid=79862&perpage=20&pagenumber=1). The resulting 6-track wav-file opens up nicely in Soft Encode. Some of the channels have to be rearranged for proper setup but that's not a problem. After encoding to 5.1 AC-3 files i have used Audio DVD Creator to make a DVD-Video disc for playback on my system (can't handle DVD-Audio or AC-3 files burned on an ISO disc like MPEG's). However there's no sound at all, just complete silence. If I play back the AC-3 files in Windows there's no problem. So maybe it's Audio DVD Creator that messes up the files - or I use some wrong settings in Soft Encode. Anyway, a "How to encode 5.1 AC-3 files with Soft Encode" guide would be really helpfull and highly appreciated.

Cheers,
Peter

EDIT:
It seems to be Audio DVD Creator that messes up the AC-3 files when creating it's output. Just extracted the AC-3 stream from the DVD-Video created with Audio DVD Creator and the extracted AC-3 file is only 2.0 - and it's completely silent when played back in Windows. Too bad:sly:

[ursamtl]

Originally posted by PDU
Offline samplerate conversion is no problem. Both SSRC.EXE and R8Brain are part of my standard collection:) I was however hoping that it was possible to do the conversion inside bidule. Would make things a lot easier and would not affect the source file(s).Unfortunately, I know of no way in a bidule to convert the sample rate in real time.@Steve
Thanks for all your help and the pinout explanation for the Spatdecode VST. Speaking of Spatdecode isn't this really a Dolby Prologic decoder:confused: Exactly what does it do when the input file is not Dolby Surround encoded in the first place but just an ordinary stereofile?Glad I could be of help, Peter. Yes, Spatdecode does a kind of basic Pro Logic decoding. I haven't discussed it with the author, but I suspect it doesn't implement all the proprietary Dolby ProLogic processing on the surrounds, such as modified Dolby B noise reduction, a 7kHz low-pass filter, or steering technology, but then again, some of these may be more effective for marketing than for sound reproduction. The basic decoding circuit is there and could easily be duplicated in a bidule group. It consists of L and R channels passed through as is. The center is L+R with attenuation. Dolby specifies -3dB attenuation, although in my tests last evening SpatDecode's center was at -6dB. The surrounds are both L-R but the right surround has its signal inverted to reduce localization, as I explained in my previous message.

What will it do to an ordinary stereo file? Depending on how the source file was mixed, the surrounds will consist of mostly ambience. If the source had instruments, vocals, or any other sounds hard panned to the L or R channel, they will also appear in the rears. So, it will probably be somewhat effective in producing a sense of surround. If you do a search for "Hafler" in this forum, you'll find that I mentioned how I used to achieve this effect with speakers.

You mentioned that you've encoded some files using V.I. <as ursamtl gleefully dons his "shameless self-promotion" hat :devil:> It also takes the ambience information present when one subtracts L-R, but it distributes the ambience throughout the 360° soundfield using a combination of ambisonic and other formulas found around the net. The main difference is that some of the ambience is distributed to the front and some of the direct sound to the rear. The overall effect is the closest I've heard to the 5.1 mixes from DVDs I rent.Final question - at least for now:)
I need some guidelines for Soft Encode. I have encoded some files using the V.I. bidule (http://forum.doom9.org/showthread.php?s=&threadid=79862&perpage=20&pagenumber=1). The resulting 6-track wav-file opens up nicely in Soft Encode. Some of the channels have to be rearranged for proper setup but that's not a problem. After encoding to 5.1 AC-3 files i have used Audio DVD Creator to make a DVD-Video disc for playback on my system (can't handle DVD-Audio or AC-3 files burned on an ISO disc like MPEG's). However there's no sound at all, just complete silence. If I play back the AC-3 files in Windows there's no problem. So maybe it's Audio DVD Creator that messes up the files - or I use some wrong settings in Soft Encode. Anyway, a "How to encode 5.1 AC-3 files with Soft Encode" guide would be really helpfull and highly appreciated.

Cheers,
Peter

EDIT:
It seems to be Audio DVD Creator that messes up the AC-3 files when creating it's output. Just extracted the AC-3 stream from the DVD-Video created with Audio DVD Creator and the extracted AC-3 file is only 2.0 - and it's completely silent when played back in Windows. Too bad:sly: You can usually get great results with the default SoftEncode settings, but if you want to know more, get ready to do a bit of reading. Check out the excellent thread here called GUIDE: How To Properly Encode Dolby Digital Audio (AC3) (http://forum.doom9.org/showthread.php?s=&threadid=56020). It contains a wealth of info on the settings you'll find in SoftEncode. You can also check its help file and information from the Dolby Technical Library (http://www.dolby.com/resources/tech_library/index.cfm). Yeah, reading might not be as fun as actually doing the stuff, but it's been my experience that you can gain a much more thorough appreciation of the whole surround process if you get some idea of what's going on.

Enjoy!