New Guide - Stereo To 5-Channel Surround - 2004-05-01 PDF

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Eye of Horus
Senior Member
Registered: Jun 2002
Location: The Netherlands
Posts: 255
Stereo to 5 channel Surround with Ambisonics : NEW METHOD !
New method to make surround from stereo with Ambisonics
An Eye of Horus guide, rev. 1.01 from 08-31-2003
01. INTRODUCTION
Almost 2 years ago, Dr. Freudiana send me a copy of Alan Parsons' CD "Stereotomy" converted to DTS format with Ambisonics.
This sounded so excellent, that I wanted to try it myself. With the help of Dr. Freudiana I made my first conversion. (Jeff
Wayne's War of the World). Again an excellent result and I was sold and decided to publish the method so everyone could give
it a try. This was more than one year ago and that guide is still online here. The last year however, it seems like Ambisonics got
into a new spotlight. A lot of new software and methods are developed.
Time to take advantage of these new methods and write a complete new guide !
02. THANKS
- dr. Freudiana
- Kempfand , who did the most of the hard work and coding and who was the one to put the ideas into something working ! We
had a fun time trying out each others ideas and listening to the results !
- Nicky Lois , for going where no one entered before, exploring new horizons and thus being a source of inspiration
- DSPGuru , for making all these excellent tools and for supplying the space for this guide !
- The makers of all these wonderful tools, like Dave Malham and his team. For more information :
http://www.york.ac.uk/depts/music/dgm.htm
and
http://www.dmalham.freeserve.co.uk/vst_ambisonics.html
- And all the other makers of tools used in this guide
03. GOAL OF THE GUIDE
To make 5 mono Wav files from a stereo source with Ambisonics, that can be feed into a surround encoder like Surcode CD or
Soft Encode.
Before you start however with the conversion-process you must decide your goal.
http://forum.doom9.org/showthread.php?s=8a0d2886427e26c4a3aa22881ff9ec9c&threadid=60137 (1 von 55)01.05.2004 01:01:52
1. stereo to DTS CD with use of Ambisonics (reguires 44.1 Khz stereo waves)
2. stereo to AC3 or DTS stream to use on DVD with Ambisonics (requires 48 Khz stereo waves)
Information about Ambisonics can be found here :
http://members.tripod.com/martin_le...t.html#SECTION4 or on other places .
My native language is not English. The advantage is that my English can be easy understood by all other non English speaking
people, but can sometimes lead to confusion for the native English speaking part of the world :-) That's why I include pictures
for every step that's not quite clear and strongly advice every reader to ask if something is not clear. Sometimes a simple
rephrase can do wonders !
In this guide the standard settings are for 44.1 Khz . If a 48 Khz setting differs, it will be mentioned. If nothing is mentioned,
the routines for both samplerates are the same.
04. DOWNLOAD NEEDED SOFTWARE
The following software is needed.....
1. Plogue Bidule (current version v0.600): PlogueBiduleInstaller.exe
Source: http://www.plogue.com/bidule/
Link: http://www.plogue.com/download_form...leInstaller.exe
Although the software is still freeware and in beta, the makers want you to register by email before you can download !
Just fill in your name and emailaddress and you will receive an email with a link to the downloadpage.
2. B-pan Ambisonic Encoder
Link : http://www.dmalham.freeserve.co.uk/pc_vst_ambisonic.zip
3. B-Proc Ambisonic B-Format Processor
Link: http://www.york.ac.uk/inst/mustech/...vst/yorkvst.zip
4. Emigrator
Link : http://www.gerzonic.net/zips/Emigrator1.0rc_win32.zip
5. Besweet latest stable version (1.4)
Link : http://dspguru.doom9.net
6. R8Brain 1.6 This is additional software for up- and downsampling
Link : http://www.voxengo.com/downloads/#r8brain
05. INSTALL AND SETUP SOFTWARE
The setup of the software is the most difficult part of this whole guide. So please pay some extra attention to this. You only
need to do it once.
Pre-setup : your stereo files need to be in the samplerate for the goal you have in mind.
If you want to make AC3 streams or DTS streams for DVD, they need a conversion to 48 Khz.
We included a freeware tool "R8brain" that does a nice job and has a built-in batch routine.
Of course you can use every soundeditor you like, i.e. Besweet, Sound Forge, Cool Edit, Soundwave. They are all capable of
converting 44.1 Khz to 48 Khz.
1. install Plogue Bidule in the recommended directory
2. in this directory you'll find a directory called "VSTPlugins" .
Unzip all other software except Besweet/split/sure to this directory
3. Unzip and/or install Besweet/besure/besplit
4. the setup :
Run Bidule, it will do a search for installed VST plugins and you will end up in the start screen.

However, if you have another product installed with a directory called VSTPlugins, it will take the plugins from that directory
instead of the right ones. Steinberg software is an example of this kind. So we must be sure we have the right plugins.
- Click "Close" on the 'Tip Of The Day' box
- From the "Edit"-menu chose "Preferences"
- If "VST Plugins Path" doesn't point to the VSTPlugins directory of Bidule, double click on the setting and you can change it to
point to the right directory.
- In this same screen you can also chose if you want to work with 44.1 Khz or 48 khz !
- Click "OK" , the program starts scanning again for VST plugins and we can continue the setup.
- Remove everything from the main screen, by clicking on an item and pressing the "delete" button on your keyboard. The
square in the right hand bottom corner cannot be
deleted :-)
- When all is deleted, you will see that the square has become black.
- If you changed the bitrate setting : Close and restart the program !
Now we need to place the tools on screen that we need.
- Goto the small window left from the main screen , called "Palette"
- chose "audio file" , "player", "2 channels" . select it and drag and drop it in the main screen.....
- now chose "VST", "Swiss Center" , "BPan_e_gui". Select it and drag and drop to the main screen.
- Do the same with "VST", "York Music dept.", "bfprocedit"
- And "VST", "gerzonic", "emigrator"
- Go on with "Audio File", "Recorder" , "6 channels"
Now we must make connections between everything on screen. This is a bit difficult to put in writing (a pictures says more than
a thousand words) but I give it a try. Move the boxes on you r main screen a bit , to make room
.
The little squares on top of the boxes are inputs, the ones on the bottom are outputs.
- step 1 : connect the two outputs of "Audio File Player" with the inputs of "BPan_e_gui"
(click on the left square and drag it to the left square of Bpan_e_gui, then do the same with the right one)
- step 2 : connect all 4 outputs of "BPan_e_gui" with the inputs of "bfprocedit"
- step 3 : connect the outputs of "bfprocedit" with the four most left side inputs of "emigrator"
- step 4 : connect the 6 most left outputs of "bfprocedit" to the 6 inputs of "Audio File Recorder" as follows:
Output Pin Emigrator > Input Pin Audio File Recorder
1 > 3 = C
2 > 1 = FL
3 > 5 = SL
4 > 6 = SR
5 > 2 = FR
6 > 4 = LFE (an empty wav)
- step 5 : check all your connection again. Your Screen should look like this :
Now it's time for some fine-tuning of the modules....
- step 1 : double-click on module "Bpan_e_gui". In the middle of the upcoming screen , you see the two panners. Turn the left
panner to the left (90 degrees counter-clock-wise), and turn the right panner to the right (90 degress clock-wise). Leave all
other settings as they are. . You can close the "Bpan_e_gui" window.
- step 2 : Double-click on the "bfprocedit" module. Leave the default setting, but turn the left-most wheel a bit to the left (by 36
degrees counter-clock-wise) . Close the window.
- step 3 : double-click on "emigrator". In the upper part of the module is a selection box. Standard it says "stereo". Change this
to "Pentagon". .
At the bottom you see a slide bar. Move the button completely to the right.
. You can close the window now.
- step 4 : Now we'll link the playing in "Audio File Player" with the recording in "Audio File Recorder" . Goto the "Tools" menu
and chose "Parameters". . In the "source" window, select "Audio File Player",
"playing" and then in the target windows select "Audio File Recorder", "recording". Now click on "Link".
. You can close the window.
- step 5 : In the "file" menu chose "Save as" and save it with the name "stereo2surround.bidule".
Setup is finished !
06. CONVERSION TO AMBISONIC
- step 1 : Double-click on "Audio File Player". In the opened player, click on the most right square at the bottom (blue down
pointing arrow) and select a stereo wav .
Close the Player.
- step 2 : Double-click on "Audio File Recorder". In the recorder window you can chose the name of your output file.
. I gave it the name "result.wav"
Close the Recorder.
- step 3 : Press the Signal Processing Button (6th icon from the left in the toolbar)
The small red square on top of the icon , turns into a green one... .
- step 4 : open the "Audio File Player" and click on "Play". .
The process starts........
When the process is finished, you'll end up with "result.wav" which is a Wav file containing 6 channels.
Time for a DSPGuru tool :-)
Goto the directory where you have those tools and run Besweet with this line when you want to have 6 mono wavs in 44.1 Khz
(for DTS CD): (of course you need to replace my L:\stereo2surround directory and my result.wav and my temp- prefix, with
your own choices !)
BeSweet.exe -core( -input L:\stereo2surround\result.wav -output L:\stereo2surround\temp- -type wav -6ch ) -ota( -G max )
.
Now go to your directory and........... there they are..... ! : 6 complete mono files !
. Ready to encode to DTS CD or when you
used 48 Khz one : AC3 or DTS stream .
FINISHED !!!!!
07. PECULIARITIES
- When you use Bidule for a second time, you will see it doesn't come back with your last settings. You have to reload these. In
"05. Install and Setup " we saved our settings under the name "stereo2surround.bidule". Goto the "file" menu, chose "open"
and load in "stereo2surround.bidule".
- You can avoid this by going to the "edit" menu and chose "Set current patch as default".
Every next time you start Bidule, it will start with "our" screen.
- Some individual modules on your screen, didn't save their settings too :-( . So you have to redo them.
>> bfprocedit : you have to place the wheels again in their desired positions. (see above)
>> emigrator : chose Pentagon again and move slider to the right (see above)
>> link and samplerate : check both to be certain they are set as intended
>> Audio File Player and Audio File Recorder : Of course you need to give new names for your original and end result wave's.
(see above)
08. TIPS AND TRUCS
- Don't like the sound of the end result : Experiment
Every module has a lot of settings. The settings in this guide are the ones we prefer. You can change them and always can go
back to the defaults of this guide. Or add other modules ! Just experiment !!
- You can listen to the results of every small change in the settings in real-time ! Add the "Microsoft Sound Mapper" from the
"Palette" window, "Audio Devices", "Output", "MME", "Microsoft Sound mapper"
to the main screen and
connect it to 4 outputs of "emigrator". From the output of "emigrator" you now have 9 connections. 5 to the "Audio File
Recorder" and 4 to the "Microsoft Sound mapper". Start these 4 connections at position 2. Pos. 1 stays empty. Pos. 2 and 3
connect to Pos. 1 of the "Microsoft Sound mapper" and Pos. 4 and 5 to Pos. 2 of the "Microsoft Sound mapper". A picture makes
it a lot more clear :-)
09. END
If you still have questions or remarks..... Just post them here.
Have a fun time !!! We did !!!
__________________
"If it ain't Dutch, It ain't much :-)"
Last edited by Eye of Horus on 6th September 2003 at 22:59
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16th July 2002 22:40

sycho
rebmeM
Registered: Mar 2003
Location: Ontario Canada
Posts: 108
can this new method be used with the equations from this page for better results on Dolby Surround material http://forum.
doom9.org/showthread.php?threadid=60331
__________________
"If you can't find em, Grind em!"
30th August 2003 04:04

bitsnbytes
Member
Registered: Aug 2003
Location:
Posts: 56
Curiosty bit me so I downloaded Besplit.
Using commandline
BeSplit -core( -input Gene.wav -prefix C:\channel -type wav -demux )
Rendered me 6 wav files named channels 1-6
but they were 44,100 so i resampled in cooledit to 48000.
44,100 would not play on my system
I have to say this was a guess but I went with this:
channel 1 = Front Left
channel 2 = Center
Channel 3 = Front Right
Channel 4 = Surround Left
Channel 5 = Surround Right
Channel 6 = LFE -empty
Its late at night so I wasnt able to fully test
or the wife would be pissed heh
but all speakers seemed to render a audible level
equal to each other.
further test tomarrow.
Oh yeah if my channel mapping is wrong please post
Thanks
30th August 2003 09:15

Eye of Horus
Senior Member
Posts: 255
just some more tips
A search on Google for "vst plugin freeware" gives you a lot of plugins that can be added.
I even saw one (equalizer) that can simply put the same effects on your files as the SAD5.1 (KpeX) method.
You want more reverb : experiment with the settings in Bpan or download a reverb plugin.
Possibilities are endless
__________________
30th August 2003 11:09

kempfand
Senior Member
Registered: Oct 2001
Location: Switzerland
Posts: 194
@ bitsnbytes:
If you use BeSplit, be aware that the audio is not normalized, so you have to turn up the volume. But in principle, it should not
matter
Using BeSplit is fine (do it also sometimes), but the correct channel ordering for this method (i.e. the Pentagon-rig with
Emigrator) is as follows:
channel 1 = Center
Channel 3 = Surround Left
Channel 4 = Surround Right
Channel 5 = Front Right
Channel 6 = LFE -empty
Andreas
30th August 2003 11:41

downloada
Member
Registered: Sep 2002
Location: Germany
Posts: 59
hi,
i found a way to get an lfe-channel out of the 5 channels generated by emigrator. while searching for freeware vst-plugins, i
found a filter called "HNM_filter". it is a low-, band and high-pass-filter.
download it here.
i integrated it into the processing chain as shown on this picture. as it is only a 1-channel-filter, you need 5 instances in our
case.
i saved the settings to a file which you can get here. if it doesn't work for you, here are the values you need to fill in, from top
to bottom:
filter: 0.000
cutoff: 0.023 (kind of a weird value, but the statusbar tells me that it means 81Hz)
resonance:0.500 (the standard value)
slope: 0.000
inamp: 0.500
outamp: 0.500
but you can just load the file i give you ("load preset" in the options of the filter), but you have to load it for EVERY instance of
the filter. of course you can change the cutoff-frequency and experiment with the other values (but i don't know what they
exactly do, the readme is kind of "short"). also i don't know what the second output-pin does, i have to test that.
another suggestion: couldn't we just rearrange the links between the output of emigrator and the input of the recorder so that
besweet demuxes them with the correct names?
cu
30th August 2003 16:48

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by downloada
hi,
i found a way to get an lfe-channel out of the 5 channels generated by emigrator. while searching for
freeware vst-plugins, i found a filter called "HNM_filter". it is a low-, band and high-pass-filter.
download it here.
i integrated it into the processing chain as shown on this picture. as it is only a 1-channel-filter, you
need 5 instances in our case.
i saved the settings to a file which you can get here. if it doesn't work for you, here are the values you
need to fill in, from top to bottom:
filter: 0.000
cutoff: 0.023 (kind of a weird value, but the statusbar tells me that it means 81Hz)
resonance:0.500 (the standard value)
slope: 0.000
inamp: 0.500
outamp: 0.500
but you can just load the file i give you ("load preset" in the options of the filter), but you have to load
it for EVERY instance of the filter. of course you can change the cutoff-frequency and experiment with
the other values (but i don't know what they exactly do, the readme is kind of "short"). also i don't
know what the second output-pin does, i have to test that.
another suggestion: couldn't we just rearrange the links between the output of emigrator and the
input of the recorder so that besweet demuxes them with the correct names?
cu
Excellent catch on the LFE !!
a. 81 Hz, shouldn't that be higher (Between 120 and 200 Hz ?)
b. the reason why we didn't include LFE in the guide : a lot of people don't have a subwoofer or don't have a dedicated
powered one (One with it's own output from the receiver). In all these cases a separate LFE channel is overkill. My experience
is that most modern receivers even give a better sound output on the LFE from 5.0 with their own routines, than software
methods to make a dedicated one.
If your's however gives better results : just use it :-) !!
The suggstion about the rearrange is so simple, cannot believe we didn't think of it :-) BUT....... there is probably one situation
where we want the output as it is now. We will keep you posted on that one... probably tonight or tomorrow......
Anyway thanks for your input....it's highly appreciated !
EoH
__________________
30th August 2003 17:39

downloada
Member
Location: Germany
Posts: 59
hi,
i couldn't do a real comparison between the result with and without lfe, but the short test i did sounded promising, so i thought
i'd share my discovery with others so they can test for themselves.
about the lfe-frequency: AFAIR 120hz is the THX-guideline but a 120hz-lfe has got too much middletone-frequencies in it. and
who cares for thx anyways? it's after all a specification for movie-playback, that's why thx-speakers aren't the best at playing
music. if i had bigger speakers, i would set the cutoff even lower, like 40hz or 30hz so that only the real deep bass is being
played on the sub. but i think that's a matter of taste. and my receiver doesn't create an lfe on multichannel-input (only if i let
it downmix it to stereo and then use pl2, but that would be non-sense), so this is the only way i can get bass on my system.
edit: i almost forgot: thanks for the great guide, it makes the whole thing so much easier and gives great results!
cu
Last edited by downloada on 30th August 2003 at 18:47
30th August 2003 18:31

kempfand
Senior Member
Posts: 194
downloada: Thanks also from my side for the LEF-option. Although I have no LFE, we thought about adding it to the method.
However, there are a few reports already on the topic which advise not to use LFE on decoded b-files (= 'Ambisonic' files).
I'm not saying that results will be bad, and I'm keen to learn about your judgement after you compared the 'without vs with',
I'm just saying 'careful'.
Also, none of the First and Second Order Ambisonic Decoding Equations suggest a LFE.
As said: We are very keen to learn how it compares.
Kind regards,
Andreas
30th August 2003 20:22

KpeX
P.O.W.T.R.C.
Location: Great Lakes, USA
Posts: 964
@Andreas and EoH
Thanks for your work.
1. Worked fine for me, I used the channel mapping as suggested in the first post of this thread by EoH, is that correct?
2. I might have missed this somewhere, but at what bit depth is the editing done?
3. Interesting fill effect. First impression is that it gives a very even, consistent channel mix across all channels. To me the
major drawback is renaming those channels . I will definitely have to add an LFE channel for my musical taste; haven't
experimented with that yet.
Cheers,
__________________
KpeX
Audio FAQs: General | BeSweet | SVCD/MP2 | MP3 | Vorbis | AC3 | DTS | AAC
P4 1.99 Ghz 768 MB DDR | Nvidia GeForce 4 MX 420 | Philips Sonic Edge 5.1 (CM8738 based) | Windows XP / Fedora Core 2 Dev. (2.6.6-RC3)
30th August 2003 21:19

DSPguru
BeSweet Author
Location: Every transfinite
consistent multiplicity
Posts: 3446
BeSweet's naming convention is :
Channel 1->"FL"
Channel 2->"FR"
Channel 3->"C"
Channel 4->"LFE"
Channel 5->"SL"
Channel 6->"SR"
A. i can add another switch with different namings.
in the meantime, you can remap the channels in the graph, or simply make a nice batchfile that does the renamings.
B. what bug did you encouter with BeSweet v1.5 ?
v1.4 is good and stable, but i still want to know about v1.5 bugs.
C. i believe the guide should include info about using BeSweet for encoding that wave directly to dts. (kempfand ?)
D. how about publishing your "stereo2surround.bidule" ? wouldn't it save some precious time to other users ?
__________________
FAQs : BeSweet, Audio, AC3
Homepage : DSPguru's Webpage
Guides : Multilingual Guides of my tools
Last edited by DSPguru on 30th August 2003 at 22:53
30th August 2003 21:48

bitsnbytes
Member
Location:
Posts: 56
Not Bad at all.
Just listned to jimmy wale out along the watchtower, with the addition of the 6th channel LFE set @ 60hz as my sub is set at
that crossover. I made one small adjustment to the
http://home.wanadoo.nl/appyhappy/18.jpg
i moved the centre slider up about a 1/4 inch, and slid vol up to top. I used besweet to demux and upsample to 48000. and
surcode to dts at 24/48. jimmy seemed to be singing from the center
as guitars seemed to wale out the lefts n rights.
The other method I use doesnt generate as much from the center.
questions:I thought one of the goals was to work in 32bit mode.. I take it besweet and besplit only handle 16 bit?
Is there a alternative?
The real bugger to watch is the bprocedit it seems to keep my fist dial setting but the third dial is always default to the left a
bit.
volume is down from original is that the normalizing? would like to remove it.
OK lets Rock!
Last edited by bitsnbytes on 30th August 2003 at 23:24
30th August 2003 23:10

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by bitsnbytes
Not Bad at all.
Just listned to jimmy wale out along the watchtower, with the addition of the 6th channel LFE set @
60hz as my sub is set at that crossover. I made one small adjustment to the
http://home.wanadoo.nl/appyhappy/18.jpg
i moved the centre slider up about a 1/4 inch, and slid vol up to top. I used besweet to demux and
upsample to 48000. and surcode to dts at 24/48. jimmy seemed to be singing from the center
as guitars seemed to wale out the lefts n rights.
The other method I use doesnt generate as much from the center.
questions:I thought one of the goals was to work in 32bit mode.. I take it besweet and besplit only
handle 16 bit?
Is there a alternative?
We have been busy with it, as B-pan for example can handle even 64 bits !! But somehow along the track it just worked nice
with 16 bits and there were some troubles finding a good 32 bits solution.
Nice to see you experiment with the different modules. The standard setting is some kind of "average for all" setting. It would
be nice if we could start some database with the best settings for every CD....
quote:
The real bugger to watch is the bprocedit it seems to keep my fist dial setting but the third dial is
always default to the left a bit.
Had some contact with the programmer of Bidule : the settings-fault will be corrected in the next version !
quote:
volume is down from original is that the normalizing? would like to remove it.
OK lets Rock!

EoH
__________________
Last edited by Eye of Horus on 30th August 2003 at 23:43
30th August 2003 23:40

kempfand
Senior Member
Posts: 194
@ DSPGuru:
A.
quote:
you can remap the channels in the graph
That is indeed the most easy & elegant solution Why didn't we think at that before Anyhow: Big thanks
B. BeSweet -core( -input In-6ch.wav -output Out_ -type wav -6ch ) -ota( -G max ) -ssrc( --rate 48000 )
For BeSweet v1.5b20, this produces garbled (clicking) mono-wav's on 2 XP machines I tested. Same EoH & bitsnbytes. It
works for v1.4.
C. Will do, as soon as we understand better why above (B) command-line works only for BeSeet v1.4 for some of us.
D. I believe in learning by doing, but point taken
@ bitsnbytes & KpeX:
Doing the B-Proc/P-Pan/Emigrator steps @ 32 bits gives slightly better results in our ears, with an emphasis on 'slightly' (i.e.
one has to really carefully listen, and know the piece well).
The Ambisonic-VST even work at 64 bit, and the author suggested to use the highest possible bit-resolution.
In order to keep the method 'workable', we thought the current 16-bit is a good start and avoids 1st going into an application
or use tools such as SOX. I personally do it @ 32 bit.
quote:
The real bugger to watch is the bprocedit it seems to keep my fist dial setting but the third dial is always default to
the left a bit.
--> Watch out for upcoming versions of Bidule, which hopefully will fix that (i.e. settings are kept).
Kind regards,
Andreas
Last edited by kempfand on 31st August 2003 at 15:24
30th August 2003 23:42

downloada
Member
Location: Germany
Posts: 59
hi,
i just got up, didn't do my listening comparison yet.
but i looked at the emigrator settings a little bit and i saw that there's a preset called "surround" which i think should be ideal
for most users because it has the same speaker-arrangement as most home theaters have today.
i will definitely include that in my comparison but the problem is that i don't exactly know the channel-mapping of that preset.
do the numbers near the speakers in emigrator represent the output-pins from left to right? then it would be easy.
cu
31st August 2003 07:47

Eye of Horus
Senior Member
Posts: 255
quote:
hi,
i just got up, didn't do my listening comparison yet.
but i looked at the emigrator settings a little bit and i saw that there's a preset called "surround"
which i think should be ideal for most users because it has the same speaker-arrangement as most
home theaters have today.
i will definitely include that in my comparison but the problem is that i don't exactly know the channel-
mapping of that preset. do the numbers near the speakers in emigrator represent the output-pins from
left to right? then it would be easy.
cu
I can immediately help you out of the dream : it's not ideal for most users. If it was we would have used it. The method is
optimized to the Pentagon, because according to one of the inventors of Ambisonics, this is the ideal setting for a 5.0 or 5.1
system.
This discussion was in the former thread on the old method also a "hot" item. Use the pentagon. It gives the best result. We
didn't "oversee" the surround rig !!
regards,
EoH
__________________
31st August 2003 09:40

downloada
Member
Location: Germany
Posts: 59
hi,
ok, just compared the different settings. my test track was "trinity dream" from the matrix reloaded score mainly because it's
rather short (1:56) but not too short and because i just had it ripped on my hdd
on my setup, the pentagon with lfe sounded best to my ears. it has a wide sound stage and the bass is as good as i thought.
although i could get my receiver to create an lfe out of the 5.0-signal, it wasn't as good as the dedicated lfe.
you're right about the surround-rig, but it's always better to hear it with your own ears. it sounded sort of compressed and too
small to me (i can't really express it in english as it isn't my native language either).
so, in the future i will use the guide with my lfe-addition and be happy with the results
cu
31st August 2003 09:56

Eye of Horus
Senior Member
Posts: 255
quote:
hi,
ok, just compared the different settings. my test track was "trinity dream" from the matrix reloaded
score mainly because it's rather short (1:56) but not too short and because i just had it ripped on my
hdd
on my setup, the pentagon with lfe sounded best to my ears. it has a wide sound stage and the bass is
as good as i thought. although i could get my receiver to create an lfe out of the 5.0-signal, it wasn't as
good as the dedicated lfe.
you're right about the surround-rig, but it's always better to hear it with your own ears. it sounded
sort of compressed and too small to me (i can't really express it in english as it isn't my native
language either).
so, in the future i will use the guide with my lfe-addition and be happy with the results
cu
Excellent ! I wasn't even so far I could test the dedicated LFE. Busy, busy, busy..... :-( But I sure will do that, as I have one
and want to hear the difference myself too. But I guess a lot has to do with the receiver, as mine gives excellent results on 5.0
inputs. Yamaha RV 496 : not exactly an expensive one, but probably for the money the best you could get.....
Ok, on with the rewriting..... this afternoon we're publishing revision 1.01 :-)
kind regards,
EoH
__________________
31st August 2003 10:12

daphy
Miles Freak
Registered: Apr 2003
Location: Germany
Posts: 175
Hi,
Ive done so much testings and comparisons between the SAD5.1, Kpex and this method (->EOH whats the new name for this
) and for me ambisonic system wins the match
The addon of the LFE is definitivly a must have because its really the bass what misses.
quote:
For those who want to save their time Ive uploaded the complete VSTPlugins (the same as in the guide + lfe
patch) in one file.
Start archiv and extract the files in the right folder (maybe you have to change the location from C:\Programme
\... in C:\Program Files\... etc.)
The only thing left to do is to scan the VSTPlugins with Plogue Bidule and load the settings!!!
@ EOH
really fine work and even for newbies an essential recomondation
One question - is it possible to make Plogue Bidule faster? Now it works only in real time -> the CPU usages is less then ~12%
much more posibilities with those machines nowadays
CYA Daphy
Last edited by daphy on 17th March 2004 at 18:00
31st August 2003 14:10

kempfand
Senior Member
Posts: 194
quote:
whats the new name for this [method]
- Visual Ambisonics
- Hitchhiker's Guide to Ambisonics
Andreas
31st August 2003 15:12

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by kempfand
- Visual Ambisonics
- Hitchhiker's Guide to Ambisonics
Andreas
Hey you, I thought we should give it a Country & Western title ??
LOL !!!
EoH
__________________
31st August 2003 17:02

Mug Funky
now with Lossy memory!
Location: flavour country
Posts: 611
holy crap! eye of horus, you've given me something to do in idle moments... i love this stuff!
right now i'm listening to Portishead's "Humming" off 180g vinyl, routed through bidule and simultaneously outputting to a 5ch
wav.
this is the awesomest thing since i got a 16 bit sound card in 1993!
thanks everyone who contributed... if i find some sweet settings, i'll let y'all know.
__________________
Everybody knows the war is over. Everybody knows the good guys lost.
31st August 2003 18:39

daphy
Miles Freak
Location: Germany
Posts: 175
Beware!!!!!!!!!!!!!!!!!!!!!!!!
Hi folks,
found a bug in this routine, dont know exactly what program produces it:
after the besweet part you have single 6 wavs (I use the DTS transcoding into 44.1 khz). And sometimes, I dont now why I
got a clipping noise in each channel. First time I explored this problem to late while sitting in my favorit chair in front of my
stereo - whooom
This noise only takes 1~2 seconds and then the normal sound starts - but it could be enough for your speakers - mine
survived this attack (gladly)
Bug search:
I tried the new Besweet 1.5 B20 and the old 1.4 -> Im shure this cant be the reason , also Surcode isnt guilty because the
6 mono files already have this damage
last but not least it must be Bidule itself! Ive got this error two times EOH shurely not, so maybe it depends on the lfe
transcoding
Ill do some more testing for finding the bug!
One thing I noticed last time I got this error the transcoding with Bidule hang at the beginning -> about one second after
this it works normal...
As advise:
check at least one of the mono files with a WAV-editor like cool edit. If there is an error in this file at the beginning transcode it
again.
CYA Daphy
Last edited by daphy on 16th January 2004 at 10:45
31st August 2003 18:52

kempfand
Senior Member
Posts: 194
quote:
Beware!!!!!!!!!!!!!!!!!!!!!!!!
I experienced this once, an it was at the very beginning (timeline) of the soundfile. Looks like your's also at the beginning of
the track.
If it's in the beginning, a quick fix is to add the 'Sound Mapper' to the Bidule layout (see section 8: tipps & tricks), so you
will instantly hear it during the 'transcode'.
My impression was that it had sth to do with the overall 'interplay' of the sound-drivers, sound-SW etc on the system, but this
was just a guess.
Regards,
Andreas
31st August 2003 19:35

daphy
Miles Freak
Location: Germany
Posts: 175
thx for the suggestion
I added this profile to the layouts
(see the connections)
If this is the only problem - nevermind
Fine toy!
Im wondering to make a DVD instead of many CDs.
What format (WAV in 44.1 or 48 khz) are needed to make a Audio DVD on DTS?
CYA Daphy
31st August 2003 20:06

SallyDog
Who Knows (who cares?)
Registered: Dec 2002
Location: Florida USA
Posts: 50
Re: Beware!!!!!!!!!!!!!!!!!!!!!!!!
quote:
Originally posted by daphy
Hi folks,
found a bug in this routine, dont know exactly what program produces it:
after the besweet part you have single 6 wavs (I use the DTS transcoding into 44.1 khz). And
sometimes, I dont now why I got a clipping noise in each channel. First time I explored this problem
to late while sitting in my favorit chair in front of my stereo - whooom
This noise only takes 1~2 seconds and then the normal sound starts - but it could be enough for your
speakers - mine survived this attack (gladly)
Error.gif
Bug search:
I tried the new Besweet 1.5 B20 and the old 1.4 -> Im shure this cant be the reason , also Surcode
isnt guilty because the 6 mono files already have this damage
last but not least it must be Bidule itself! Ive got this error two times EOH shurely not, so maybe it
depends on the lfe transcoding
Ill do some more testing for finding the bug!
One thing I noticed last time I got this error the transcoding with Bidule hang at the beginning ->
about one second after this it works normal...
As advise:
check at least one of the mono files with a WAV-editor like cool edit. If there is an error in this file at
the beginning transcode it again.
CYA Daphy
I had the same problem, was driving me nuts. Then I realized that Bidule was not maintaining the "stereo2surround" default
setting. I solved the problem by making sure to load the "S2S" setting at the beginning of each session. Only takes an extra 2
seconds. Maybe this is causing your problem too.
SallyDog
31st August 2003 20:56

Eye of Horus
Senior Member
Posts: 255
version 1.01 now online
The latest version is now online !
Suggestions from previous messages are all in the guide now.
I got some private emails from people asking for the best ways to convert to AC3 and DTS. The guide was too long already to
include that. I pointed them to all the websites already online with explanations and of course Doom9 , but some questions
were more specific about Ambisonics and encoding to AC3.
We thought about putting that in the guide , so you got it all from input to output, but it would be a lot of repeating . There are
however a few important things. I will write another chapter That should be in the guide, but lack of space "forces" me to put
it here...... (that is : next message from me :-))
EoH
__________________
31st August 2003 21:19

erymax
Junior Member
Registered: Jan 2003
Location: UK
Posts: 2
Can you post a link for the latest version ?
thanks
erymax
31st August 2003 22:04

daphy
Miles Freak
Location: Germany
Posts: 175
@ erymax
read the posting right from the start

CYA Daphy
31st August 2003 22:19

erymax
Junior Member
Location: UK
Posts: 2
@daphy
thanks, I've got it now, sorry for my mistake.
erymax
31st August 2003 22:39

Eye of Horus
Senior Member
Posts: 255
Re: i would suggest the following :
quote:
Originally posted by DSPguru
1.i never saw a comparison graph of R8Brain's quality compared to ssrc. one thing for sure - ssrc is
GOOD, and i believe it can be fed with 32bit waves.
One of the progammers of Bidule said that all things you process can better be done on 48 khz directly than on 44.1 Khz and
upsample them afterwards to 48 Khz. That could even effect the Ambisonics effect. Who am I to judge ? :-) Anyway I thought
that doing the resam,pling at the start would simplify the proces a lot ! Now we talk about upsampling, not downsampling. I
just wonder, also because my Japanese is not what it used to be :-), when I load a 16 bit 44.100 Hz stereo file into Sound
Forge, I can save it immediately as 16 bit 48 Khz. Every other processing or program accepts it as 48 Khz. So why the
upsample and not just "save as" ??? If I remember correctly that isn't on the Japanese site too !
But main reason was the opinion of the Bidule programmer !
quote:
2. suggestion for graph update : place six "audio file recorder"s, and create directly those 6 mono
waves.
Thought about that. But I guess you have to read my next message to understand why we didn't include it.....
quote:
3. for DTS authors, i would suggest to :
3.1 omit the resampling at start
3.2 use BeSweet with ".mux" input. BeSweet will normalize all channels and automaticly resample if
needed (depending on the source sample-rate and the target bitrate).
the 2gb barrier is not a problem when using ".mux" since now you have a (virtual) limit of 2gb for
each channel !
Enough to think about :-)
Thanks for the input......
EoH
__________________
31st August 2003 22:54

Eye of Horus
Senior Member
Posts: 255
Missing Chapter ?
The "missing" chapter
Best ways to encode output from Ambisonics method
At the start of the whole proces, you already decided what you would do with your output by chosing 44.1 Khz or 48 Khz.
Now lets get a bit into detail. I won't explain or repeat every method here. There are plenty guides on the subject,
I will try to give some tips for this specific method (Ambisonics) and one you will not find anywhere else yet,
but I am sure it will find it's way to many methods. I hope DSPGuru can build it into his methods too :-)
01. 44.1 Khz from stereo CD to make a DTS
Use EAC (Exact Audio Copy) to rip as one big WAV with CUE-sheet . Transcode with the guide to 6 mono waves. Load these
into Surcode CD Pro 1.09 , encode them to DTS and use the CUE sheet to burn to CD.
02. 48 Khz to make a DTS track for DVD
Transcode with guide. Load the 6 mono waves into Surcode DVD pro 1.2x and encode to a DTS track
These were the already known steps.
03. 44.1 Khz from stereo CD to make a Dolby Digital 5.1 CD
Use EAC (Exact Audio Copy) to rip as one big WAV with CUE-sheet . Transcode with the guide to one big (very big!) 6 in 1
wave.
Load this wave into Softencode and.... you can encode it to Dolby Digital 5.1 CD ! There is no need to make 6 mono waves
first !
Use the CUE-sheet again to burn the CD with tracks.
See below for the best settings......
04. 48 Khz stereo CD in separate tracks to make AC3's for DVD
Use EAC (Exact Audio Copy) to rip the tracks. Keep their names simple like track01.wav, track02.wav etc.
At the end of the proces you can rename the final AC3.
Transcode all to 6 mono wavs for each song. This will take a lot of space and I am sure someone can come up later with a
better
way to handle this :-) But for now do it this way. So your directory will look like track01FL.wav, track01FR.wav, track01C.wav
etc.etc.
When you're always using the same temporary names, you can use the batch-routine in Softencode and save it. Everytime you
start Softencode, the previous used batch-routine is still there !
Only a one time setup, use the batch several times and advantage : you can do a lot of processing while you're sleeping or
working or whatever :-)
See below for the best settings......
05. 48 Khz stereo CD in separate tracks to make AC3's for DVD
Again ? Yes, but this time with a new idea. This does NOT improve on the previous method, but it is a new one with an already
old program and
I am glad my friend Nicky Lois found out about it. I am sure DSPGuru can include this in his routines just like he did with
Surcode !
Ok.... let's start.
- Remove Softencode from your PC
- Re-install it, but install it in c:\Sonic (no longer in c:\program files\Sonic Foundry Soft Encode\). No other name, just like
that !
- From now on you can use the batchroutine of Softencode !!!
- It has some limitations : all files must be in the root directory of your drive > in this case all temporary waves must be in c:\ !
- It is very sensitive about the naming of the files. It won't process track01-FL.wav, but will track01_FL.wav.
- The ideal settings, which are below, can be put in an .INI file. Everytime you start SE with a commandline, it will search for
the. INI file.
- This .INI file contains only the "not default" settings. Place the .INI file in the root too.
- A commandline will look like this :
C:\Sonic\Sftencdd.exe -A -P c:\encode.ini -o c:\test.ac3 -L c:\track01FL.wav -R c:\track01FR.wav -C c:\track01C.wav -l c:
\track01sl.wav -r c:\track01sr.wav -e c:\track01LFE.wav You input the 6 mono waves of track01 and end up with test.ac3 !
- This is the i.e. encode.INI, you have to make (copy and paste into notepad and save as i.e. encode.INI) :
[parameters]
BSMode=1
SurMix=2
Cut3dB=On
CompressionType=3
This .INI has the same ideal settings as the screens below !
Settings for SE for Ambisonic encoded waves
Make sure all screens look like these. It's too much to tell every setting, just look at the screens.
screen 1 :
screen 2 :
Make sure you save the settings !
happy encoding !!
EoH
__________________
31st August 2003 22:57

Eye of Horus
Senior Member
Posts: 255
quote:
Hi,
@ EOH
really fine work and even for newbies an essential recomondation
One question - is it possible to make Plogue Bidule faster? Now it works only in real time -> the CPU
usages is less then ~12%
much more posibilities with those machines nowadays
CYA Daphy
Don't know ! I didn't write it :-)
OTOH..... you cannot listen realtime when the processing isn't realtime :-)
Perhaps the programmer reacts to this when he sees it.....
kind regards,
EoH
__________________
31st August 2003 23:20

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by Mug Funky
holy crap! eye of horus, you've given me something to do in idle moments... i love this stuff!
right now i'm listening to Portishead's "Humming" off 180g vinyl, routed through bidule and
simultaneously outputting to a 5ch wav.
this is the awesomest thing since i got a 16 bit sound card in 1993!
thanks everyone who contributed... if i find some sweet settings, i'll let y'all know.
Thanks for the compliments !!!
kind regards,
EoH
__________________
31st August 2003 23:22

sycho
rebmeM
Location: Ontario Canada
Posts: 108
is there any way at all of using these equations for converting stereo to b-format:
Dolby MP:
W' = 0.36(S - jD(a /180))
X' = 0.54(S + jD(a /180))
Y' = 1.00D(a /180)
where:
S = Left + Right
D = Left - Right
a = Width control, varying between 0 and 180
j = 90 phase shift
taken from:
http://www.geocities.com/ambinutter..._equations.html
if usde it sould provied better results on dolby surround
__________________
"If you can't find em, Grind em!"
Last edited by sycho on 31st August 2003 at 23:55
31st August 2003 23:26

kempfand
Senior Member
Posts: 194
@ sycho: Can't answer your Q directly, but here are 2 suggestions:
1) Have a look at the non-GUI-version of B-Pan-e, and especially at the related documentation . There are much more settings
there to play around with.
2) I'm sure the author (Dave Malham) would be able to answer. You might send him an email.
Keep us posted on your findings ...
Regards,
Andreas
31st August 2003 23:57

Eye of Horus
Senior Member
Posts: 255
DVD usage
My good old friend Nicky Lois wrote me a nice email from which I will translate a few things here :
DVD with only a LPCM 48 Khz soundstream ?
Make it surround with Visual Ambisonics !
Rip the soundstream , convert to AC3 or .DTS and remux the sound and video together . Works excellent !
And thanks for the compliments Nicky ! Don't forget your own part in it :-)
EoH
__________________
1st September 2003 00:04

bitsnbytes
Member
Location:
Posts: 56
High End Test
I spent A good portion of my late night making files from cd and mp3. I finished late morning
with A good mix of songs all encoded the same.
centers up a 1/4 LFE at 500's @60hz using the 16bit method. Besweet to demux and resample @ 48000. DTS encoded to
24/48000.
Now I traveled 30 mile to my trusted highend stereo shop where the gave me the room with about 60-$80,000 system the
pricer one wouldnt play my files heh.
surfing the disk on there $5000 dvd audio player was troublesome only rendering track1,2,3 etc no folders...{they didnt have
the model I bought on floor} but anyway played 5 or 6 diff tracks.
The sounds drew A bit of a crowd as salesmen brought customers in and volume was at A loud setting they started asking
who, what is this?
All were thoughly impressed and enjoyed children of the sun huge. I know this guy runs comparison test and room
environment sweet spot test... so I kept asking for his proffesional opinion I think I got it he, just kept saying it sounded sweet
and asking how I made them and would I do his favorite CD he uses to setup sytems... with the music he likes best
So now im sure this is the way to do surround sound.
Good job guys and as they say the proof is in the pudding... now about doing this 24 bit
PS he loved the bass coming from the 15in subwoofer and asked if I turned it up... um er i used .500 i said heh LFE baby yeah
__________________
[System] AVP-Rotel RSP-1066; DVD Audio/Video-Rotel RDV-1060; AMP-Rotel RMB-1075; Speakers JBL, Rock Solid And Polk
Audio

bitsnbytes
Member
Location:
Posts: 56
ok lets see if I can get on to better.
so lets say i want to work at 32 bits and end up with 24 bit... also @48000
so i tried making 6 recorders @24 bit but rendered some of the channels invalid or with un recoverable errors.
can this be done in the bidule?
If so an example please :-)
__________________
[System] AVP-Rotel RSP-1066; DVD Audio/Video-Rotel RDV-1060; AMP-Rotel RMB-1075; Speakers JBL, Rock Solid And Polk
Audio

kempfand
Senior Member
Posts: 194
@ bitsnbytes: Thanks for the nice feedback.
The Ambisonic VST's (B-Pan, B-Proc, Emigrator) work at 32 bit (even 64). For quick tests, I suggest you use CoolEdit (or your
preferred app) to bring things up to 32 bit precision and 48 kHz before you feed into the Bidule scheme.
To my knowledge, the encoders (ac3 or dts) will request 16 bit input, so just use the 'Audio File Recorder' in Bidule with the 16-
bit depth.
Regarding the LFE: I have not tested this yet, but this might only work in 16 bit (and therefore the errors you got). So you'd
have to use a more complex set-up scheme to do things in one run.
Regards,
Andreas

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Eye of Horus
Senior Member
Posts: 255
quote:
thx for the suggestion
I added this profile to the layouts
(use save as.. option).
(see the connections)
If this is the only problem -
nevermind
Fine toy!
Im wondering to make a DVD
instead of many CDs.
What format (WAV in 44.1 or 48 khz)
are needed to make a Audio DVD on
DTS?
CYA Daphy
Thanks for the module you made !
Nice work
EoH
__________________
http://forum.doom9.org/showthread.php?s=35a738356...8949990dc5&threadid=60137&perpage=40&pagenumber=2 (1 von 32)01.05.2004 01:03:54

Eye of Horus
Senior Member
Posts: 255
small test..... not scientific
quote:
@ bitsnbytes: Thanks for the nice
feedback.
The Ambisonic VST's (B-Pan, B-Proc,
Emigrator) work at 32 bit (even 64).
For quick tests, I suggest you use
CoolEdit (or your preferred app) to
bring things up to 32 bit precision
and 48 kHz before you feed into the
Bidule scheme.
To my knowledge, the encoders (ac3
or dts) will request 16 bit input, so
just use the 'Audio File Recorder' in
Bidule with the 16-bit depth.
Regarding the LFE: I have not tested
this yet, but this might only work in
16 bit (and therefore the errors you
got). So you'd have to use a more
complex set-up scheme to do things
in one run.
Regards,
Andreas
Did some comparisons this morning between several
upsamplers...
Do not use a "save as..." function. !
Best results :
- R8brain with a setting medium - very high
- Sound Forge 6.0 with a setting between 2 - 4
@kempfand
I still have some problems when I input a 32 bits file
and output (in the file recorder) as 16 bits..... It won't
make a file....
Outputiing in 32 bits gave no problems......
I will try again this evening....
@kempfand
Do you prefere "normalise" on all files (with a VST
plugin or a preprocessor like R8brain) before the
Ambisonics conversion or afterwards on all 6 mono
wavs ?
@bitsnbytes.....
Nice work !!!! LOL !!!
EoH
__________________

specise_8472
Newbie
Location: New Zealand
Posts: 101
EOH
For those that own the Nuendo Dolby DIgital Encoder.
How about just plugging in this baby and then just
straight encode into AC3 from Bidule.
Also has Decode for AC3 in package - decode DD AC3
2.0 from some dvds and then viola, Ambisonic AC3
Am waiting with baited breath for the Nuendo DTS
encoder to appear - Also when I can afford it
Hopefully it also contains a Decoder for DTS.
__________________
Don't blame me! I only work here.

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by specise_8472
EOH
For those that own the Nuendo Dolby
DIgital Encoder.
How about just plugging in this baby
and then just straight encode into
AC3 from Bidule.
Also has Decode for AC3 in package -
decode DD AC3 2.0 from some dvds
and then viola, Ambisonic AC3
Am waiting with baited breath for the
Nuendo DTS encoder to appear - Also
when I can afford it Hopefully it
also contains a Decoder for DTS.
I know, we thought about it, but .....it's nit freeware,
huh ?? LOL !!
AC3 2.0 to Ambi AC3 5.1 : See former message about
LPCM :-)
I think you can forget about the decoder :-(
I like the possibility (if it is possible !) to have a DTS
encoder on the Bidule screen ! OTOH DTS has not
much options, so for quality I doubt anything will
change.......
BTW good to see you here too
EoH
__________________

kempfand
Senior Member
Posts: 194
@EoH:
quote:
Do you prefere "normalise" on all files
(with a VST plugin or a preprocessor like
R8brain) before the Ambisonics
conversion or afterwards on all 6 mono
wavs ?
Normalise (with BeSweet -ota( -G max ) ) after b-file
decoding. After all, this is what the 'old' Command-
Line Ambisonic Decoder does (where the -g switch is
disabled by default).
quote:
I still have some problems when I input a
32 bits file and output (in the file
recorder) as 16 bits.....
Strange ... always works here...
Andreas

Eye of Horus
Senior Member
Posts: 255
experiment !!!!
Today I did so many experiments...... I lost count
That's the nice thing with this system. There are
hundreds of freeware VST plugins and they are so
easy to use in Bidule and on "our" Ambi project.
You can immediately hear the effect.....
Here are a few sites with loads of free plugins.....
http://usitweb.shef.ac.uk/~mup01jrm/vst/vstlinks.htm
http://acidfreak.future.easyspace.com/
http://www.vstcentral.com/results.a...
&Submit2=Submit
For those who are real ambitious : make your own
VST !
http://www.steinberg.net/en/ps/supp...nload/index.
php
EoH
__________________

bitsnbytes
Member
Location:
Posts: 56
I converted the .wav to 32bit 48000 then loaded it in
the player and set recorder to 16bit.
Result was a 16/44 and sound was muffled, female
voice was mannish.
Perhaps the setup doesnt handle 48000?
I played the original file sounds fine.
__________________
[System] AVP-Rotel RSP-1066; DVD Audio/Video-
Rotel RDV-1060; AMP-Rotel RMB-1075; Speakers JBL,
Rock Solid And Polk Audio

kempfand
Senior Member
Posts: 194
@ bitsnbytes: I might have given a wrong suggestion
earlier. In summary, and more concrete:
What I tested and what works:
- Increase precision to 32-bit before Biduel scheme
- Record to 16 bit multichannel wav
- Use BeSweet to upsample to 48 kHz after the
Bidule scheme, using:
BeSweet -core( -input In-6ch.wav -output Out_ -type
wav -6ch ) -ota( -G max ) -ssrc( --rate 48000 )
What I didn't test, is the 48 kHz upsampling before
running the Bidule scheme. I just assumed it works...
Maybe that's not the case, which would explain the
muffled sound you mention.
@ EoH: Hope that doesn't mean Guide v1.02
LOL
Andreas

Eye of Horus
Senior Member
Posts: 255
quote:
I converted the .wav to 32bit 48000
then loaded it in the player and set
recorder to 16bit.
Result was a 16/44 and sound was
muffled, female voice was mannish.
Perhaps the setup doesnt handle
48000?
I played the original file sounds fine.
Did you change the 44100 to 48000 in the settings
screen ???
EoH
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
@ bitsnbytes: I might have given a
wrong suggestion earlier. In
summary, and more concrete:
What I tested and what works:
- Increase precision to 32-bit before
Biduel scheme
- Record to 16 bit multichannel wav
- Use BeSweet to upsample to 48 kHz
after the Bidule scheme, using:
BeSweet -core( -input In-6ch.wav -
output Out_ -type wav -6ch ) -ota( -G
max ) -ssrc( --rate 48000 )
What I didn't test, is the 48 kHz
upsampling before running the Bidule
scheme. I just assumed it works...
Maybe that's not the case, which
would explain the muffled sound you
mention.
@ EoH: Hope that doesn't mean Guide
v1.02 LOL
Andreas
Nope
Author of Bidule said : 48 Khz before using it ! But.....
don't forget you also have to change in the settings-
menu !!
EoH
__________________

bitsnbytes
Member
Location:
Posts: 56
ok got this sucker workin now
cool edit I resample to 32/48
set recorders to 24bit
Load 6 Recorders only link 1 at a time
6th recorder is LFE/Optional
record one channel at a time slow but I think results
are worth looking at,the lack of "normalizing" is a
different sound all together.
Ok Emigrator Mapping for those crazy enough to
follow heh
from left to right
channel l = Center
channel 3 = Rear Left
channel 4 = Right Rear
channel 5 = Front Right
channel 6 = LFE /optional
Really hope they can fix this so we can output to 6
recorders at a time
MY HNM Low Pass Filter settings for Subwoofer
Crossover @ 60hz
Filter: .060
CutOff: .017
Resonance: .500
Slope: .000
InAmp: .500
OutAmp: .333 im lookin at increasing this
slightly... .500 was very intense I had to reduce SW
to -5db .333 is nice but I want A inbetween ;-)
__________________
Last edited by bitsnbytes on 2nd September 2003 at 05:41

kempfand
Senior Member
Posts: 194
Try using 'Audio FIle Recorder' (6 channels) for the 5
pins from Emigrator (as outlined in the guide), and
use the proven DSPGuru tools (BeSweet ... -6ch -
OTA). For the LFE, try using one 1-channel 'Audio File
Recorder'.
Good luck,
Andreas
2nd September 2003 00:51

kempfand
Senior Member
Posts: 194
@ EoH / all:
quote:
Today I did so many experiments...... I
lost count
More food:
B-Plane Mirror
B-Mic
Bpan3 - Third Order Ambisonic Encoder
Let's test
Andreas

TerraForce1
Member
Location:
Posts: 82
Food is nice
Hi,
First of all I would like to thank everybody pointing
out on this.
At first glance I wanted to test if this method worked
out for me. I have the following problem and it keeps
doing it over and over.
The pitch of the created 6 channel wav is higher
compared to the original stereo WAV file, so I end up
with a 14 seconds less than the original. The songs
ends like it has to end but the pitch is higher
compared to its original. The created multichannel
wav is like 2:45 as for the original is 2:59. I dont
know if anybody has experienced this problem.
I know I just could alter the pitch, but I find it better
that the program works flawless.
Could somebody point me out what I am doing wrong
or is it bidule itself?

Eye of Horus
Senior Member
Posts: 255
Re: Food is nice
quote:
Originally posted by TerraForce1
Hi,
First of all I would like to thank
everybody pointing out on this.
At first glance I wanted to test if this
method worked out for me. I have
the following problem and it keeps
doing it over and over.
The pitch of the created 6 channel
wav is higher compared to the
original stereo WAV file, so I end up
with a 14 seconds less than the
original. The songs ends like it has to
end but the pitch is higher compared
to its original. The created
multichannel wav is like 2:45 as for
the original is 2:59. I dont know if
anybody has experienced this
problem.
I know I just could alter the pitch,
but I find it better that the program
works flawless.
Could somebody point me out what I
am doing wrong or is it bidule itself?
You're not feeding it with 48 Khz, while the settings in
the preferences are 44.1 ?
(First thing to come in mind and else : I don't
know.....)
__________________

TerraForce1
Member
Location:
Posts: 82
Well, I love the way your mind works
Hi, well as the post subjects says it all:
I love the way your mind works. Meaning you are
right.
Thx.

daphy
Miles Freak
Location: Germany
Posts: 175
@ all
If you have a new profil that works fine for you please
contact me via PM. Than send me your settings (***.
bidule -file) via email, so I can give others the
opportunity to download it and make further
development possible
CYA Daphy

TerraForce1
Member
Location:
Posts: 82
Awesome results
Hi,
I want to report about the awesome results I got. I
own an Eagles DVD. You can choose between DTS or
PCM. I have expirmented on the song California Hotel
(Acoustic version). Because I own the DVD, I
downloaded a mp3 of this song and tried the guide.
Unbelievable, you can hear the same effects as they
come on the original DTS track as with the new
encoded DTS sound. I don't know how, but it looks
like the same. But however there is a but, and that is.
In the original DVD I hear the BassTablets from Front
Left, as with the encoding they come from Rear Right.
As from the original DVD I get the hihats from Rear
Right as from the Encoded they come from Front Left.
Somehow it looks like the sound is swapped between
the Speakers. The center and right front speaker also
look to be just the way around. I had not much time
for more analyzing. But the sound of the original DVD
gives with the original speaker setting a better
harmonic between the sounds.
My first thought is the setting of the Emigrator where
you can set the conrolled opposites. But I may be
wrong.
For your information.
TerraForce1
Last edited by TerraForce1 on 2nd September 2003 at 23:37

Eye of Horus
Senior Member
Posts: 255
Re: Awesome results
quote:
Hi,
I want to report about the awesome
results I got. I own an Eagles DVD.
You can choose between DTS or PCM.
I have expirmented on the song
California Hotel (Acoustic version).
Because I own the DVD, I
downloaded a mp3 of this song and
tried the guide. Unbelievable, you can
hear the same effects as they come
on the original DTS track as with the
new encoded DTS sound. I don't
know how, but it looks like the same.
But however there is a but, and that
is. In the original DVD I hear the
BassTablets from Front Left, as with
the encoding they come from Rear
Right. As from the original DVD I get
the hihats from Rear Right as from
the Encoded they come from Front
Left. Somehow it looks like the sound
is swapped between the Speakers.
The center and right front speaker
also look to be just the way around. I
had not much time for more
analyzing. But the sound of the
original DVD gives with the original
speaker setting a better harmonic
between the sounds.
My first thought is the setting of the
Emigrator where you can set the
conrolled opposites. But I may be
wrong.
For your information.
TerraForce1
I like these experiments, but...... with a but......
I don't think you should compare these two.
It's a bit what we want to avoid, because Ambisonics
is a complete different system. The DTS or AC3 are
only used here as containers for the format. Just like
putting an old Quad album on DTS, doesn't make it
DTS, it's still Quad !
Another thing is...... (IMHO !!) never use mp3's !!
An MP3 is a compressed format. You miss information
when you put it back to WAV. Perhaps you don't hear
it, but it can be essential information for the whole
process.
About the difference from where you hear the
sounds : It has , I think, nothing to do with emigrator.
(That is : You did make the latest correction ??? Guide
version 1.01 !) .
Mixing in stereo is complete different from mixing in
DTS.
Ask Alan Parsons !!
kind regards,
EoH
__________________
3rd September 2003 00:34

kempfand
Senior Member
Posts: 194
@ TerraForce1: Thanks for the nice feedback. Three
comments:
1) The channel-mapping with the method in the guide
is correct (we checked this again and again before
publishing). Despite the 'good' results with this
method, it is very likely that results are different from
those optained with special 5.1-mix (btw: I know
California Hotel from the DVD very well too ).
2) The Ambisonic method is more about creating a
'sound-image', not about isolating effects &
instruments to discrete channels. Given this, having
effects coming out of the same speaker as with the
DVD-5.1-mix (with emphasis on mix) should not be
expected.
Another way to put this: B-Pan creates an Ambisonic
1st-order B-file (=WXYZ). The fact that we decided
for the guide to outline decoding to 5.0 is just
because most of us have a 5.1-system. However,
decoding could be easily made for 4.0, 8.0, 12.0, etc.
etc. speaker rigs
3) You can experiment with the `controlled
opposite' (vs. 'strict soundfiled'). Richard Furse finds
that the `controlled opposite' equations produce a
larger listening area at the expense of some
directional information (Ambisonic Decoding
Equations . Keep us posted on your preference.
Regards,
Andreas

Eye of Horus
Senior Member
Posts: 255
quote:
EOH
For those that own the Nuendo Dolby
DIgital Encoder.
How about just plugging in this baby
and then just straight encode into
AC3 from Bidule.
Also has Decode for AC3 in package -
decode DD AC3 2.0 from some dvds
and then viola, Ambisonic AC3
Am waiting with baited breath for the
Nuendo DTS encoder to appear - Also
when I can afford it Hopefully it
also contains a Decoder for DTS.
BTW I just did some tests with the Nuendo, but I
don't get it to output anything..... Do you have an
idea ?
kind regards,
EoH
__________________

specise_8472
Newbie
Posts: 101
quote:
Originally posted by Eye of Horus
BTW I just did some tests with the
Nuendo, but I don't get it to output
anything..... Do you have an idea ?
kind regards,
EoH
It was just an idea - but am working on it at the
moment
Also, is there any good reason why in Bidule you
cannot just define 6 single channel outputs and output
each channel as a single wave file. Then at the end
you have your 6 waves all ready to go
__________________

bitsnbytes
Member
Location:
Posts: 56
quote:
It was just an idea - but am working
on it at the moment
Also, is there any good reason why in
Bidule you cannot just define 6 single
channel outputs and output each
channel as a single wave file. Then at
the end you have your 6 waves all
ready to go
You can but it is time consuming.
I tried 6 recorders but some generated invalid files.
So I set the six recorders unlinkig all but the first...
then unlink the first and link the second...etc
then in my encoder it warned me that there were files
of different data lenghths... but song encoded
correctly and sounded right.
ps I only did it this way to go from 32bit/48000
to 24 bit/48000 directly
with this manner you dont get normalizing
so a 5 min song gonna take you 30 min.....
results are not worth the extra time I dont think....
but maybe later when they fix it ;-)
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
It was just an idea - but am working
on it at the moment
Also, is there any good reason why in
Bidule you cannot just define 6 single
channel outputs and output each
channel as a single wave file. Then at
the end you have your 6 waves all
ready to go
But..... I want to make AC3's from the 6-in-1 wav.....
I can load this file directly into Soft Encode !!
&@Kempfand
These are the pros and cons of hitchhiking :-)
EoH
__________________

bitsnbytes
Member
Location:
Posts: 56
New Test Subject
I got The Edgar Winter Group - "They Only Come Out
AT Night" CD from guy at work... I have to say zzz
Frakenstein is A masterful piece and made sweet love
to my ears.... originally encoded for quadraphonics...
I now see why
__________________
Last edited by bitsnbytes on 3rd September 2003 at 23:53

specise_8472
Newbie
Posts: 101
Interesting find.
Load the resultant 6 channel wave into Adobe
Audition and it automatically splits the file into 6
waves. Then add them to multitrack view.
The only problem then is that the only multichannel
output formats supported at the moment are Windows
Media 9 Multichannel. Will also export 6 wavs and
multichannel wave.
Am still playing around with various options and
programs.
BTW have you tried the TAPEIT vst plugin (get from
vst central) as the recorder for individual wavs.
__________________
4th September 2003 00:01

kempfand
Senior Member
Posts: 194
TapeIt
Last edited by kempfand on 4th September 2003 at 00:17

bitsnbytes
Member
Location:
Posts: 56
tapeit worked worse than the on board recorder
rendering 1kb files when configured 6 times
__________________

bitsnbytes
Member
Location:
Posts: 56
I pulled the .wav out of a divx to convert but cant
seem to get it done.
its microsft acm waveform mpeg layer 3?
cant get it into bidule
__________________

daphy
Miles Freak
Location: Germany
Posts: 175
quote:
its microsoft acm waveform mpeg layer 3?
I always thought bidule works only with non
compressed files?
Use for example cooledits save as option to save into
a non compressed WAV (PCM 16bit 48/44.1khz)
CYA Daphy

TerraForce1
Member
Location:
Posts: 82
a long the 24 bit 32 bit story
Hi,
Along the 24 bit 32 bit story I have the problem that
the recorded amplitude with Bidule is very low. I don't
know if it is possible to get the recorded audio at a
higher amplitude. I know that Besweet is capable of
doing this (btw I don't know the acceptable settings
guess -13 db will do), but is there somehow a way in
Bidule to get it correct from the beginning ? As for my
second question, while maintaining a higher amplitude
of the surround speakers I want my LFE output to be
of a lower amplitude than the current settings. It's
really too loud. I have to lower the input output ratio
on my subwoofer otherwhise I get a distorted effect.
It's like going to a houseparty with gabbers or to put
it in other words real hardcore bassdrums. Well and
that ain't what you are expecting when you listen to
slow rock.
Watching the waveforms with Cooledit gives a clear
image that the amplitude of the LFE is really much
bigger than the other waveforms. So for the moment
I have to manually cut 6 db every time.
As my conclusion is: from how I want to have it, the
amplitude of the LFE had to be the amplitude of my
surround speakers. And the amplitude of my surround
speakers had to be the amplitude of my subwoofer.
Is there somehow an amplitude plugin for Bidule. If
yes how do we integrate this in the current model.
@ daphy
Also compressed wave file are accepted. I saved my
wave file in an ADPCM format and it played the file.
Mp3's however dont.
TerraForce1
Last edited by TerraForce1 on 4th September 2003 at 08:31

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
Hello
1) There is allready a Gain module in Bidule,
in the toolbar (or right mouse button menu) under
mixing/gain...
its a simple 2 in 2 out gain module
2) Sadly while our multichannel wav files are following
the wav standard to the letter, since that format is
uncommon, not many apps read them properly (ex
MS Media player). I knew about Cool Edit and
Audacity, but i didnt know about Adobe Audition,
which is good to hear.
3) Bidule is a realtime application, and while we are
very happy to learn its being used in this fashion (im
waiting to set up my own 5.1+ setup to try this very
soon, when my home theatre is done), It wasnt
made specifically for this task.
That said, this thread has waken up the need for a
Offline (render) mode which would allow to process
much faster than real time
(say, if in realtime bidule reports 10% cpu use, the
offline mode would able - in theory process that 10
times faster) But you wouldnt be able to monitor it at
the same time.
4)I have to look at why the settings are not saved
properly, it seems its a bug that seem to affect soem
VSTs and not others.
Cheers

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
Sorry other things just crossed my mind
5)we dont support mp3 natively, but a vst plugin does
that: mp3play
(the site however down to protest against software
patents) which we also are against.
6) on Bit depth. Bidule uses 32bit IEEE floating points
data internally
and thats the signal that's passed though the graph.
So even if a plugin uses 64 bits internally, it wouldnt
make much of a difference outside of it. Secondly,
wav files stored at 24 bits (linear) contain as much bit
headroom as a Float 32 bit IEEE wav file.
(techy talk
A 32 bit Float used for audio uses the same precision
as a 24 (linear) integer, since the remaining 8 bits are
used for exponent.

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
7) for a list of natively supported audio files in bidule,
have a look at this list:
http://www.zipworld.com.au/~erikd/libsndfile/
#Features
(bidule uses this library in order to read/write audio
files)

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by davidv@plogue
Hello
1) There is allready a Gain module in
Bidule,
in the toolbar (or right mouse button
menu) under mixing/gain...
its a simple 2 in 2 out gain module
Hi David,
But.... how much gain do you have to give ?
Is there a way like OTA's -g max ??
quote:
2) Sadly while our multichannel wav
files are following the wav standard
to the letter, since that format is
uncommon, not many apps read them
properly (ex MS Media player). I
knew about Cool Edit and Audacity,
but i didnt know about Adobe
Audition, which is good to hear.
Adobe Audition = Coold Edit Pro 2.1 (They bought the
program.)
quote:
3) Bidule is a realtime application,
and while we are very happy to learn
its being used in this fashion (im
waiting to set up my own 5.1+ setup
to try this very soon, when my home
theatre is done), It wasnt
made specifically for this task.
But it does a good job !!!!
quote:
That said, this thread has waken up
the need for a Offline (render) mode
which would allow to process much
faster than real time
(say, if in realtime bidule reports
10% cpu use, the offline mode would
able - in theory process that 10 times
faster) But you wouldnt be able to
monitor it at the same time.
The value on the bottom of the screen gives
sometimes strange results.... i don't know what really
happens then, or if it's a bug somewhere in the
"memory and cpu management".
load a module and use it...... cpu : 2.6 %
reload the module, do some modifications (other file
names) : cpu 80 % and once even 105.67 % > hard
reset to be able to work again....
quote:
4)I have to look at why the settings
are not saved properly, it seems its a
bug that seem to affect soem VSTs
and not others.
Cheers
And why it changes one setting without that setting
being touched !
kind regards,
EoH
__________________

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
>But.... how much gain do you have to give ?
you can do -oo to +10 dB (nominal - unchanged -
gain by default)
>Is there a way like OTA's -g max ??
You mean normalize?
Im afraid thats impossible to normalise a signal in
a live (streaming situation) unless someone invents
crystal-ball VST,
as an audio signal normalizing algorithm must
1)first scan the audio signal entirely
2)remember the highest (peak) value it encountered
3)find a gain value x so that high val = max
4)apply gain x to whole file
none of which can be done in a live streaming app,
like this.
(but are easily done in offline sound editors)
Engineers use Dynamics Compressors in realtime
(with which they can use to prevent sudden
unexpected peaks),
but these, well change the source dynamics.. so you
might not want that here.
The best way would be with trial and error, run a
couple of source
files and find a gain setting that doesnt clip the signal.
(this plugin might help: http://www.pspaudioware.
com/plugins/vmeter.html)
Our audio file player extracts the file at nominal
volume, and
audio files have a max "normalized" value they can
support
(well the 32 bit float can go beyond that and
represent signal
out of the normalised [-1;1] range, but i dont want to
get too techy again), So im sure there is a propper
gain setting, but the overall gain setting might be
influenced by the settings of any of the other plugins
in the chain.
>Adobe Audition = Coold Edit Pro 2.1 (They bought
the program.)
Yeah, youre right, my plogue collegue just told me.
>But it does a good job !!!!
Hey thanx!
>The value on the bottom of the screen gives
sometimes strange results....
>i don't know what really happens then, or if it's a
bug somewhere in the "memory and cpu
management".
>load a module and use it...... cpu : 2.6 %
>reload the module, do some modifications (other file
names)
>: cpu 80 % and once even 105.67 % > hard reset to
be able to work again....
it can be many things, pentium 4 denormalisation bug
in (any) VST,
I/O disk thread code having problems (bidule bug).
Or something else weve never encountered before.
If you can find specific steps to reproduce these
errors, we would glady
try to fix them.
Last edited by davidv@plogue on 4th September 2003 at 22:29

joshbm
Member
Location:
Posts: 67
First off, I would like to say EoH you are awesome! I
have been working around with KpeX's SAD5.1 and it
is very nice it does a good job, yet it was lacking
something. Your new Ambisonics method works great!
Hats off to you :-)!!!
If you guys are wondering how to get 6 mono wavs,
well it is quite simple to do using CDP Multichannel
Toolkit.
You can download it from there website here:
http://www.cs.bath.ac.uk/~rwd/mctools.html
or a direct link to the zip file here:
http://www.cs.bath.ac.uk/~rwd/mctools.zip
Now once you download it, you will be working with
the command-line tool called "channelx". Since it is a
command-line tool you will need to create a new
batch file with the following in it:
@ECHO OFF
channelx -och.wav YOUR_6CH_WAV.wav 1 2 3 4 5 6
This will separate all the channels within a 6 channel
wav file. You can also specify which channels to take
out with this tool. This is definately a handy set of
tools.
Last edited by joshbm on 4th September 2003 at 22:34

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by joshbm
First off, I would like to say EoH you
are awesome! I have been working
around with KpeX's SAD5.1 and it is
very nice it does a good job, yet it
was lacking something. Your new
Ambisonics method works great! Hats
off to you :-)!!!
If you guys are wondering how to get
6 mono wavs, well it is quite simple
to do using CDP Multichannel Toolkit.
You can download it from there
website here:
http://www.cs.bath.ac.uk/~rwd/
mctools.html
or a direct link to the zip file here:
http://www.cs.bath.ac.uk/~rwd/
mctools.zip
Now once you download it, you will
be working with the command-line
tool called "channelx". Since it is a
command-line tool you will need to
create a new batch file with the
following in it:
@ECHO OFF
channelx -och.wav YOUR_6CH_WAV.
wav 1 2 3 4 5 6
This will separate all the channels
within a 6 channel wav file. You can
also specify which channels to take
out with this tool. This is definately a
handy set of tools.
Thanks for the kind words......
I had these tools already since more than a year, but
now they're a bit outdated.....
You know, there's also a tool called wav2to6.exe.....
does exactly the same :-)
And the Besweet tools, Soft Encode also....there are
more...
As long as there's no alternative (except doing it in a
soundeditor) to the OTA -g max command..... I'm
more than happy with Besweet !!!
But it's always good to look at other tools too !
So....thanks for the links !
kind regards,
EoH
__________________

bitsnbytes
Member
Location:
Posts: 56
Speaker Arrangement
Now looking at the pentagon layout in emigrator
speakers should proly be adjusted to match... Has
anyone tried this different speaker layout?
dunno if the wife would let me turn the room into a
pentagon hahahahahah
@plogue david
Is the 6 seperate recorders a bug or a processor /
resources problem?
__________________

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
You create 6 separate mono recorders and sync their
start parameters toghether?
sadly this trick is not very sample accurate.
the only trick i know is to press rec on the first one
from series of linked recs and then start the audio
processing (circle icon on top of the app) then do the
inverse after.
Even then all files might not have the same end point,
idd need to check.
What we tought about doing was a single module that
saved multiple mono files OR a multichannel one, so
all mono files would be synched.
We might add this in a future version.
Cheers

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kempfand
Senior Member
Posts: 194
@ davidv; Good to see you here. Bidule (despite beta) is indeed a great tool, and I would like to say
'Big Thanks' for it. If you / your team can implemnet an offline (render) mode, that would really rock :-)
@ EoH:
quote:
I'm more than happy with Besweet !!!
The 'old' method with Ambidec (which does normalisation by default) and BeSplit achieves exactly the
same. In other words: After B-Pan & B-Proc, the B-file could be saved with B-Rec, with subsequent
Ambidec & BeSplit, and this would achieve the same as Emigrator & BeSweet -ota( -G max ).
@ bitsnbytes: Good catch. Using only B-Pan & Emigrator (w/o B-Proc), you would have to re-arrange
the speaker set-up, as there would be a SC (surround center, in your back so-to-speak).
In order to avoid this, we used a 'trick': Rotating the sound-field by -36 degrees counter-clock-wise with
B-Proc. Result is that you don't need a SC (surround center) speaker.
I know it might sound a bit confusing. It might help if you just draw the speaker-setup from Emigrator
on a piece of paper. Channel-1 is slightly turned counter-clock-wise (by -37 degrees). Channel-3 is the
surround-center (at the end of the minux-X axis).
Then, by literally turning the sound-field by -37 degress (counter-clock-wise), channel-1 becomes C,
channel-2 becomes FL, cannel-3 becomes SL, etc.
http://forum.doom9.org/showthread.php?s=&threadid=60137&perpage=40&pagenumber=3 (1 von 31)01.05.2004 01:04:31
Regards,
Andreas
Last edited by kempfand on 4th September 2003 at 23:56

bitsnbytes
Member
Location:
Posts: 56
Re: a long the 24 bit 32 bit story
quote:
Hi,
As for my second question, while maintaining a higher amplitude of the surround
speakers I want my LFE output to be of a lower amplitude than the current
settings. It's really too loud. I have to lower the input output ratio on my
subwoofer otherwhise I get a distorted effect. It's like going to a houseparty with
gabbers or to put it in other words real hardcore bassdrums. Well and that ain't
what you are expecting when you listen to slow rock.
TerraForce1
IF you use the HNM filter for the LFE there is a AmpOut slider Try .333 seems pretty mellow setting for
the sub you can also set the sub to cut off at say 60hz so its not generating subwoofer at higher range
you can see my HNM filter specs on another previous post
__________________
[System] AVP-Rotel RSP-1066; DVD Audio/Video-Rotel RDV-1060; AMP-Rotel RMB-1075; Speakers JBL,

DSPguru
BeSweet Author
Posts: 3446
Hello David, Welcome,
quote:
That said, this thread has waken up the need for a Offline (render) mode which
would allow to process much faster than real time
(say, if in realtime bidule reports 10% cpu use, the offline mode would able - in
theory process that 10 times faster) But you wouldnt be able to monitor it at the
same time.
that would be nice indeed
quote:
A 32 bit Float used for audio uses the same precision as a 24 (linear) integer,
since the remaining 8 bits are used for exponent.
that's true, but the extra 8bit used for exponent representation expands your dynamic range.
in fact, judging by the waves created in this particular process, it seems to save something like 2 extra
bits out of the linear 24bits/16bits, due to gain loss of something like 13db.
quote:
you can do -oo to +10 dB (nominal - unchanged - gain by default)
i unerstand that all modules are supposed to be floating-point-based processors, hence, unsensitive to
pregain of 10db, but i doubt that's true.
applying 10db amplification before the audio file recorder seems like a reasonable act.
quote:
You mean normalize?
Im afraid thats impossible to normalise a signal in a live (streaming situation)
i agree.
anyway, i've noticed that the product wave includes a "PEAK" chunk, so that's good enough for me
Cheers,
Dg.
__________________
Last edited by DSPguru on 5th September 2003 at 10:31

bitsnbytes
Member
Location:
Posts: 56
Thanks For The Gain!
just what I was missing... I added 3 gains
utilizing 5 slots ... NOT using the 6th one for LFE as the HNM filter adds OutAmp ...Gain
I set it to +10db much better
__________________

specise_8472
Newbie
Posts: 101
Playing around with Emigrator.
Used rig layout Oct2 with speaker mappings
1 - C
2 - FL
3 - FL
4 - LS
5 - LS and RS
6 - RS
7 - FR
8 - FR
Just connect them to the right channels on the Recorder Out.
Very good results as more info per channel. In DTS sounds great. Also I am fortunate enouth to have
an amp that allows ES on any 5.1 sound, so when in 6.1 mode it works great as the 5 speaker would be
the Center surround posistion. Now only if we could get a 6.1 encoding program - either AC3 or DTS
Am working on the 3D rigs at the moment.
__________________

DSPguru
BeSweet Author
Posts: 3446
quote:
might I have a copy of ur commandline im using surcode dvd... and do you use the
gain or just go into besweet? I never get the spike only once on first few tries.
you'll find my commandline on Q32 at the BeSweet FAQ.
since i now work in floating-point, i don't set a pregain value in the Bidule graph, i just add "-ota( -g
max )" to the commandline from the BeSweet FAQ.
note that a spike will prevent BeSweet from normalizing, that's why you might wanna add a "-split( -
start 1 )" suffix.
__________________

kempfand
Senior Member
Posts: 194
@ DSPguru: Can't wait to get home in 10 hrs to use the new new BeSweet v1.5b21. Great job
Regarding the 1-2 sec spike (which someone else too described earlier): Try starting the play for 2-3
secs, stop it, move the slider back to the start, and play again. This has avoided the spike for me when
I had it.
Andreas

DSPguru
BeSweet Author
Posts: 3446
the spike at the beginning cannot be avoided, since it's actually data - not samples, and in fact, this is
useful data : it's the peak value of all channels.
quote:
BeSweet.exe -core( -input L:\s2s\result.wav -output L:\s2s\temp- -6ch ) -ota( -g max )
if you use v1.5b21, this would be a more advised commandline (faster/better quality):
quote:
BeSweet.exe -core( -input L:\s2s\result.wav -output L:\s2s\temp- -6chfloat ) -ota( -g
peak )
but if you don't feel like sitting infront of the computer until the BeSweet process ends, use the
commandline from the BeSweet FAQ to transcode directly to DTS. (end-to-end floating-point process)
note : don't forget to create logfiles - for future discussions!
__________________

DSPguru
BeSweet Author
Posts: 3446
/me wonder, perhaps there's a vst plugin for shell execute ?
the idea is to write the BeSweet commandline in this kind of plugin, and link its trigger with the end of
the recorder.
this way, we'll have a fully automated process.
(stereo to multichannelwave,multichannelwave to dts)
David ?
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
the idea is to write the BeSweet commandline in this kind of plugin, and link its
trigger with the end of the recorder.
David ?
Please don't forget about us 48 Khz AC3 fans !!!!
How about the Besweet line for encoding with the commandline of Softencode ????????????
EoH
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DSPguru
BeSweet Author
Posts: 3446
don't expect it in the near future..
anyway, how about updating your guide to use v1.5b21 with -ota( -g peak ) instead of v1.4 with -ota( -
g max ) ?
__________________

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
quote:
the idea is to write the BeSweet commandline in this kind of plugin, and link its
trigger with the end of the recorder.
David ?
While that sounds like a nice VST idea, then the command line would have
to be fixed (custom), as the thought of someone entering
"deltree c:\winnt" in a empty text field,
then sharing this bidule on the net, scares me a bit
Hehe

DSPguru
BeSweet Author
Posts: 3446
you're right. no problem, a fixed one !!
all you need to configure in this plugin :
1. BeSweet.exe Path
2. Surcode.exe Path
3. Input FileName (the same one you write with the recorder)
4. Temp Directory for mono wave files
5. Output filename
and a "start" button, to be linked with the "stop" button of the recorder.
it will generate the commandline by itself.
if you could attach here a zip file with a sketch source-code, i'll do the rest..
forgive me, but i don't have the time to read the VST specs..
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
the spike at the beginning cannot be avoided, since it's actually data - not
samples, and in fact, this is useful data : it's the peak value of all channels.
I had a spike today of 6 seconds... !!! ???
EoH
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
don't expect it in the near future..
anyway, how about updating your guide to use v1.5b21 with -ota( -g peak )
instead of v1.4 with -ota( -g max ) ?
I'm sorry to hear that ! I think it would be a useful addition and I'm certain a lot of us want that
option......
Back to good old batchprocessing, huh ?
I will adjust to 1.5b21 after confirmation this is the stable one now ??
BTW...... why change that OTA option ?
Or is it an addional option ?
EoH
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bitsnbytes
Member
Location:
Posts: 56

__________________
Last edited by bitsnbytes on 6th September 2003 at 01:17

specise_8472
Newbie
Posts: 101
Surcode DTS 1.021
If any programmers are interested, I have pulled apart the Registry settings and the .SSF files. If
interested I can give the info if wanted. May be easier to change the registry settings then call surcode.
As do not have to load in .SSF file. But surcode keeps track of the last used .SSF file in the registry
also. Does it load it back in on open?.(Better check this one out). I also found that if you change the .
SSF file you have to restart Surcode, as it goes through the motions of loading again, but ignores the
changes - unless restarted-reloaded.
I was going to program something but have got the Delphi version of the VST templates and am
investigating some interesting options. So sidetracked.
Have made great strides in a DTS decompiler, but nothing to give yet. Actually I don't know if I will
release it as it uses the routines and ideas from the Patent and ETSI tech specs. So copyright
issues???? (I only attack this project when in the mood as it does not seem to be as easy as first
thought )
__________________

DSPguru
BeSweet Author
Posts: 3446
what's going on with your brackets ?
reminder : every section should start with an openning bracket and end with a closing bracket.
it better look like this :
BeSweet.exe -core( -input track01.wav -output "C:\BeSweetv1.5b20\mono\temp-" -logfile c:\dts.txt ) -
ota( -g peak ) -surdvd( -b 1536 -output "C:\DTS Rips\Besweet\track01.dts" -path "C:\Program Files
\Minnetonka Audio Software\SurCode DVD DTS" ) -split( -start 1 )
@EoH
assuming you created a 32bit float wave with Bidule, -ota( -g peak ) will normalize that wave just like -
ota( -g max ), but will do it in a single pass instead of two-pass (=faster).
for confirmation, i guess we'll have to wait to kempfand's logfiles.
btw, multichannel, 32bit floating-point waves bigger than 2gb are supported as well, so i don't see any
reason why not to create 32bit waves. (unless you're short in H.D. space)
quote:
I had a spike today of 6 seconds... !!! ???
interesting.
can you attach the first 10kbytes in here ?
@specise_8472
BeSweet doesn't use a .ssf file, it sets the registry keys and call surcode.
__________________

Eye of Horus
Senior Member
Posts: 255
Ambisonics .....
I doubted a long time if I should post this, but I do.....
Kempfand and I developed an other method to make surround sound from stereo material with the use
of Ambisonics. We use several tools to get to that with the help of the Plogue Bidule package. We wrote
the guide from 2 channels to 5 or 6. I made a small (very limited) additional chapter about ways to
convert these 5 or 6 channels to AC3 or DTS. But that's only there for completion !
What we see happen now, is that this is becoming the xxth thread about commandlines in Besweet.
That's not where we wrote this guide for and not what we would like to see in this thread. IMHO it's off
topic , because there are already several threads about that topic. So why start another one here ?
(gee, feel like a "mod" now )
It's not my intention to sound pedant or to insult anyone, because I admire the works of DSPGuru a lot
and I am a regular user of his tools. But this is IMHO not the right thread.
This current thread is for discussing the pros / cons of the method itself, and how to use & apply the
tools (B-Proc, P-Pan, etc).
I hope to bring some structuring into this, helping the readers to keep the overview and get quick
answers
For BeSweet usage, there are the DSPGuru / Doom9 guides in addition to plenty of threads.
And I think that for BeSeet development, a new one should be opened with a meaningful header.
I know there are a lot connections between all kind of soundprocessing-methods and one or other
"Besweet"-family member, but I hope we can bring this back a bit to the start..... :
How to get the ultimate method to convert stereo to surround with software !?
with kind regards,
Eye of Horus
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
interesting.
can you attach the first 10kbytes in here ?
I can't..... I don't keep the failures :-) I wish I had, but it was between 5 and 6 seconds one large block
that was not equal on all channels. 4 were the same, the other two were a fraction shorter.
The file I was converting was a stereo WAV with a length of 7 minutes.
In future I will keep this kind of faults in a separate folder....
Anyway I discovered more strange things today.
I converted a complete ripped stereo album to a 6CH file.
Shoot....... forgot to reset to 44.1 Khz in Bidule.
I changed the setting to 44.1 Khz, ended and restarted the program as suggested. Looked again at the
preferences : indeed 44100 and started the 50+ minutes file for the second time. And....... ended again
with a huge 6CH file in........48 Khz !!
I completely reset the machine (OFF/ON !), started Bidule again, checked the preferences and did the
file again.
This time it was fine. (150 minutes !) Something's not quite right :-)
About the real-time versus speed : wouldn't it be simple to implement : no output module loaded
(soundcard, midi, ms-mapper) then : SPEED. Output module loaded : real time. (I'm not a
programmer, but this came into my mind.....).
grtz,
EoH
__________________

DSPguru
BeSweet Author
Posts: 3446
Re: Ambisonics .....
quote:
I made a small (very limited) additional chapter about ways to convert these 5 or
6 channels to AC3 or DTS. But that's only there for completion !
yes, you are right.
the main issue to deal with is the 2ch to 5ch/6ch process.
i myself don't enjoy those commandline-discussions threads, mainly, Because it's either documented or
supported by DD's GUI.
note that my only motivation was to enhance my tool to complete your needs, as far as i understand
them. (+have the time+have the interest).
anyway, thread splitted.
Cheers ,
Dg.
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bitsnbytes
Member
Location:
Posts: 56
HNM Filter
The readme was short n sweet so I dug up A page on this kind of filtering... I believe the best choice for
music is the lowpass filtering
which would set it @ 80hz or lower some may like to test 100 and 125 but I think would be too bassy. I
set mine to 60hz.125 would proly be suitable for explosive movie.
so in HNM filter first slider "filter" set to desired hz range so 100 hz would let 100 or less
pass.
the "cutoff filter" I believe sets too small of A slope. set to "0".
"resonance filter" Another steep cutoff filter
again I recommend against it. set to "0"
"slope" I have no idea but I presume its A no. set to "0"
"InAmp" I presume A pre Gain I leave at .500
"OutAmp" I presume a post Gain I set it to .380
I suggest some more research on these options
here's one http://home.cdsnet.net/~roberth/bandpass.htm
For those on A MAc I found some free VST :-(
http://www.arboretum.com/download/d...d_freeware.html
A filter info page from above site:
http://www.arboretum.com/support/ma...oc_filters.html
__________________

ilmanu
Junior Member
Registered: May 2003
Location:
Posts: 14
when i try to convert wav file in a 6 mono wav besweet crash....on win2k....
why?
help me please
ps: i use
C:\audio\beswe\BeSweet.exe -core( -input f:\6ch.wav -output f:\wav\final- -type wav -6ch ) -ota( -G
max )
Last edited by ilmanu on 11th September 2003 at 21:38

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by ilmanu
when i try to convert wav file in a 6 mono wav besweet crash....on win2k....
why?
help me please
ps: i use
C:\audio\beswe\BeSweet.exe -core( -input f:\6ch.wav -output f:\wav\final- -
type wav -6ch ) -ota( -G max )
I have no idea, but perhaps DSPGuru does.
In the meanwhile, you can try this program to get the 6 wav's.
Only you don't have the 'gain' option.... but that can be done with other programs too.
http://akson.sgh.waw.pl/~wj23277/wav2wav6.zip
good luck,
EoH
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ilmanu
Junior Member
Location:
Posts: 14
well, now there is a new problem plogue generate a corrupt file, the original is 45:34:700 but the new
6wav file is 74:56:650 why is too long then original?

bitsnbytes
Member
Location:
Posts: 56
A little note when playing music back.
Ambisonics is not limited to creating a reverberant rear sound field, and requires a different
arrangement of speakers. Also, with Ambisonics all speakers cooperate to localise sounds, so the front-
rear time delay is unnecessary (and would be detrimental).
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
well, now there is a new problem plogue generate a corrupt file, the original is
45:34:700 but the new 6wav file is 74:56:650 why is too long then original?
Is it +74 minutes after you made the 6 mono files ?
EoH
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ilmanu
Junior Member
Location:
Posts: 14
yes after....but i have found the problem.....i made a 16bit wav 2ch and try to convert in 5.1 24bit...

Eye of Horus
Senior Member
Posts: 255
quote:
yes after....but i have found the problem.....i made a 16bit wav 2ch and try to
convert in 5.1 24bit...
OKAY
EoH
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ilmanu
Junior Member
Location:
Posts: 14

well, my english is good like your italian

Umma
Member
Location:
Posts: 57
Eye of Horus:
Might it be possible to have your original thread on stereo-to-dts conversion archived somethere on this
site? I find myself referring back to it for the absolutely fascinating information that you all put in there
in the course of that year that it was posted.
All the posts here have been extremely informative and I've been pleased with my own ambisonics
results. I'm still playing with the plogue bidule...I guess I'm just used to the 'old' way...I'll prolly feel
better when I get the batch mode to work in Besweet. Drivin' me nuts right now.
Thanks everyone for your contributions! I've been lurking here for quite some time, now, and have
learned quite a bit (though I'm no pro...)!
Last edited by Umma on 15th September 2003 at 05:51

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by Umma
Eye of Horus:
Might it be possible to have your original thread on stereo-to-dts conversion
archived somethere on this site? I find myself referring back to it for the
absolutely fascinating information that you all put in there in the course of that
year that it was posted.
All the posts here have been extremely informative and I've been pleased with my
own ambisonics results. I'm still playing with the plogue bidule...I guess I'm just
used to the 'old' way...I'll prolly feel better when I get the batch mode to work in
Besweet. Drivin' me nuts right now.
Thanks everyone for your contributions! I've been lurking here for quite some
time, now, and have learned quite a bit (though I'm no pro...)!

Thanks for the kind words !
That playing in Plogue Bidule is something I still do everyday. It's so nice to hear every change you
make in real time ! I only wish some of these VST's were better documented !
Here is the link to the old one :
http://forum.doom9.org/showthread.php?s=&threadid=29277
(BTW a simple search on "ambisonics" would have brought you there too )
kind regards,
EoH
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TRILIGHT
Moderator
Location:
Posts: 1217
Thanks. I finally got it running with this command line...
code:
BeSweet.exe -core( -input E:\TEST.wav -output E:\temp- -6chfloat ) -ota( -g peak )
I encoded using SoftEncode but I'm not overly impressed with what I'm hearing. Sure, it's clean and
comes out all of the speakers (I was expecting something a bit more intelligent but vocals are in the rear
also), but it is not as clear as the original. I can compare the two and the extracted WAV obviously has
more clarity than what I'm hearing in the AC3 that was created. Any idea what could be causing this loss
of clarity?
Just to clarify some info, what I did to run my particular test was to extract Track 01 on my Norah Jones
CD. I resampled the WAV in SoundForge to 48kHz (if I didn't do this, my final result was pitch shifted
up). After, I processed the WAV in Bidule as per the guide, then I split with the above BeSweet
commmand line, and then encoded in SoftEncode. Any advice? Thanks!!
__________________
Das UberGuide lives! Check it out! --> HTTP:/ / DVDGUI DES.TRI LI GHT.NET
16th September 2003
20:49

kempfand
Senior Member
Posts: 194
@ TRILIGHT. Glad the technical side now worked out.
To the method: Depending on what you expect from Ambisonics, you will be delighted or disappointed.
This method does not attempt to seperate isolated instruments (or voice) to discrete channels. It's
more about producing a 'sound-image' in a defined space (with dimensions, walls, reflections etc) and
then 'recording' that in what is called an Ambisonic B-file (1st order in this case).
I suggest you take some various samples (only 1 treack each) and see what it does for you. For my
personal taste, I had some excellent results with Kate Bush, Pink Floyd, Massiv Attack, Hooverphonic,
just to name a few). I had bad results on recordings that already were heavily 'manipulated' during a/o
after recording (echo etc).
Regarding the method, some additional reading can be found at:
Richard Elen's page
Martin Leese's FAQ page
Ambiophonics
Technical Papers
Kind regards,
Andreas

bitsnbytes
Member
Location:
Posts: 56
Bit of An odd question.
I have A Mono .wav file... what do you suggest I do to get it mixed in surround?
Ok got it. Made a copy of the mono wav selected 2 1 channel players linked playing/playing.
first 1 also linked to recorder playing/recording.
so my subject :Though it was recorded on two-track tape, only the mono master of 'Love Me Do' was
kept, so the song has never been available in true stereo.
as I just listened to the finished piece not bad not bad, one thing I've noticed Ambisonic seems a little
bassier .. like I make A 2ch +LFE from the same wav.. not so much bass hmmm
maybe LFE should not run through ambisonic
but straight to the recorder what u think?
Hah beat you. Yes Straight to the recorder For LFE
Otherwise over bassy response is made.
IE That soft rock gonna sound like a Mad Chatterhouse or whatever the bloke said that cracked me up

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Eye of Horus
Senior Member
Posts: 255
bidules
@bitsnbytes
Any new bidules ???
please post them......
TIA.....
EoH
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bitsnbytes
Member
Location:
Posts: 56
I have A website I havnt updated in sometime
Its A gaming website for quake3 Arena. My online buddy makes quake3 trick movies and I demuxed his
avi and converted it to 5.1 And muxed it back w/ fairly nice results and file size was much smaller than
his .mp3 track . So I'm gonna make A page this weekend for gamers to use these fine tools as well
as post my bidules w/ links to helpful info and tools.
I havnt set my comp up to work with my website as we recently moved and I bought some stereo
eqiupment
Then I can just post links to my bidules here and small sample files for the Guru.
Question have you tried the BPan_m_gui if so
your thoughts? I'm using it to go to stereo havnt tried it to 5.1... How's testing going on the 3rd order
tools?
And A help page http://www.dmalham.freeserve.co.uk/bpan_help.html
Thanks Again for this wonderful guide
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
I have A website I havnt updated in sometime
Its A gaming website for quake3 Arena. My online buddy makes quake3 trick
movies and I demuxed his avi and converted it to 5.1 And muxed it back w/ fairly
nice results and file size was much smaller than his .mp3 track . So I'm gonna
make A page this weekend for gamers to use these fine tools as well as post my
bidules w/ links to helpful info and tools.
I havnt set my comp up to work with my website as we recently moved and I
bought some stereo eqiupment
Then I can just post links to my bidules here and small sample files for the Guru.
Question have you tried the BPan_m_gui if so
your thoughts? I'm using it to go to stereo havnt tried it to 5.1... How's testing
going on the 3rd order tools?
And A help page http://www.dmalham.freeserve.co.uk/bpan_help.html
Thanks Again for this wonderful guide
THNX !
I think we need a webpage where we can post our bidules. (Daphy are you still with us ??)
And I think every bidule should have a screen picture with it !
In case there are some special settings : add them preferred with a screen dump too.......
So... for every bidule some sort of mini guide :-)
I can supply the webspace if necessary......
About BPan_M_gui and the 3th order tools.....
I didn't experiment with them yet. I think you "should" know by now, that Kempfand is the expert of
the format and I think he can answer that immediately; I am only the writer of the guide to (try to)
make it understandable for everyone
I do experiment a lot with all the tools however. And I like it a lot.... This Bidule system is so easy !!
And there are so many nice free plugins available !
I have been experimenting a lot with stereo spreaders just to see what kind of effect they have on the
end 5.1......
And I combined the KpeX (SAD5.1) method with the Ambisonics method to see if that was an
improvement.... It wasn't ! On their own they both do fine, but it's no use to combine them. Actually
the theories of Kempfand were confirmed....
So, Daphy, are you still there.... ? Can we still "dump" everything to you ? Or shall I open my own
space for bidules, screendumps and their explanations ?
kind regards,
EoH
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TRILIGHT
Moderator
Location:
Posts: 1217
I just wanted to thank you as well, Eye of Horus. If it were not for your guide, I would not have even
considered still playing with the whole 2ch to 5ch thing. I've not had a chance to put any real time into
testing but I know it's out there now.
__________________

kempfand
Senior Member
Posts: 194
quote:
Question have you tried the BPan_m_gui if so your thoughts?
If have, but not in a systematic way, and therefore I will not comment here. Something for the
upcoming winter-months
For sure, the 'Mirror type' with B_Pan_m is sth to experiment with, as is B-Plane Mirror , as well as
'improvement' of Quad-recordings.
quote:
Kempfand is the expert of the format
It was clearly a joint effort (an excellent example of team-work so-to-speak), and after all, my interest
in Ambisonics got triggered by EoH's 'old' guide and what it delivers in terms of sound.
Andreas

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Author Thread
joshbm
Member
Location:
Posts: 67
Hello all,
I have created a group for Bidule (instead of a project file). Download these files and put them in the Bidule "Group"
folder.
5.1 Surround
6 Channel Gain
The first one is the 5.1 surround. After you put it in your folder called "Group". You can access it from your "palette"
and under "Groups" called "5.1 Surround". Instead of a mess like this, you get a nice group like this with exactly 2
inputs and 6 outputs.
I also included something called 6ch Gain which allows you to control the gain of 6 channels at once instead of 2.
Problems:
Bidule currently does move the attributes around as you saw in the tutorial and have to manually set it to
"Pentagon" under Emmigrator and move the right most notch on B-Pan to 0 degrees. To do this you can simply
right click on the group (not in the palette) and hit "Grouping" and "Expand" then change your controls and then to
get back go to "Grouping" again and then "Parent Group".
For me it helps because it makes it less confusing.
- joshbm
Last edited by joshbm on 19th September 2003 at 04:51

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
Nice groups!
Just a quick note, the next release of bidule saves the emigrator
settings properly and has a brand new "OFFLINE" mode in order to
render files as fast as your machine can process.
Thanx for your interrest in Bidule!
And watch out for the next version around november...
Cheers

TRILIGHT
Moderator
Location:
Posts: 1217
quote:
Just a quick note, the next release of bidule saves the emigrator
settings properly and has a brand new "OFFLINE" mode in order to
render files as fast as your machine can process.
AWESOME!!!! Now I can really start playing! Thanks for the update, David!
__________________

bitsnbytes
Member
Location:
Posts: 56
I think Besweet GUI changes the layout A bit
instead of
1 > 3 = C
2 > 1 = FL
3 > 5 = SL
4 > 6 = SR
5 > 2 = FR
as per your guide... BesweetGUI uses
1 > 2 = C
2 > 1 = FL
3 > 4 = SL
4 > 5 = SR
5 > 3 = FR
6 > 6 = LFE
I say this for my page, as I'm sure most will opt for ac3enc.dll and besweet GUI is the quickest way.
Please correct me if i'm wrong.
PS cant wait for new bidule! Thanks
__________________
[System] AVP-Rotel RSP-1066; DVD Audio/Video-Rotel RDV-1060; AMP-Rotel RMB-1075; Speakers JBL, Rock Solid
And Polk Audio

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by TRILIGHT
Thanks. I finally got it running with this command line...
code:
BeSweet.exe -core( -input E:\TEST.wav -output E:\temp- -6chfloat ) -ota( -g peak )
I encoded using SoftEncode but I'm not overly impressed with what I'm hearing. Sure, it's
clean and comes out all of the speakers (I was expecting something a bit more intelligent but
vocals are in the rear also), but it is not as clear as the original. I can compare the two and the
extracted WAV obviously has more clarity than what I'm hearing in the AC3 that was created.
Any idea what could be causing this loss of clarity?
Sorry for the late reaction, must have overseen this....
If I understood it correct, you are comparing an original WAV with an AC3. Every WAV will sound better than any AC3
made of that WAV !
It's the compression !!! Just like a WAV always will sound better than a MP3. Of course a lot of us won't hear the
difference between a WAV and a MP3 in 320 kbs, but...... make a WAV again from that mp3 and compare with the
original WAV.....
Try encoding it in 44.1 Khz to DTS CD and compare that with the original WAV. Still compression but a lot less than
in AC3's.
Anyway : More clarity can be reached
a. by converting the WAV to 32 bits before the processing
b. when you upsample in SF, do you really upsample or use the "save as..." function ? When you upsample, try
adding a clarify filter.
"process" menu , "smooth/enhance" .
It helped me......
kind regards,
EoH
__________________
20th September 2003
14:00

Eye of Horus
Senior Member
Posts: 255
quote:
I think Besweet GUI changes the layout A bit
instead of
1 > 3 = C
2 > 1 = FL
3 > 5 = SL
4 > 6 = SR
5 > 2 = FR
as per your guide... BesweetGUI uses
1 > 2 = C
2 > 1 = FL
3 > 4 = SL
4 > 5 = SR
5 > 3 = FR
6 > 6 = LFE
I say this for my page, as I'm sure most will opt for ac3enc.dll and besweet GUI is the
quickest way.
Please correct me if i'm wrong.
PS cant wait for new bidule! Thanks
I won't correct you :-)It's already confusing enough :-))
I think this question is for the Besweet thread :-)
Anyway, what you quoted from the guide is the way the channels are recorded by the File-recorder from 6 inputs to
one WAV containing these 6 channels. Right ?
And these settings are correct.
So any way you use a piece of software to extract the 6 separate mono's from that one wav, gives you back the 6
original wavs...
And they should be the same !
I used the last STABLE Besweet version (1.4 !).
If anything changed, please ask again in a Besweet Thread..... That's DSPGuru's department :-) :-)
kind regards,
EoH
__________________

TRILIGHT
Moderator
Location:
Posts: 1217
Hey EoH! Thanks for the advice. I know what I'm hearing is not compression artifacts. I'm not using any DRC either
so that can't be to blame either. I'm encoding at 384kbps which I thought was high enough. I've never been able to
tell much of a difference between 384 and 448 before.
I guess I'm comparing it to the MP3 encoding I do using LAME. I challenge anyone to hear the difference in the
MP3's I encode and the original. If the AC3 encoding is really to blame, is Dolby really that screwed up?? I can't
imagine they would come up with algorithms worse than that found in the free LAME encoding engine but maybe
they did?
Thanks for the tips on processing, EoH! I will try converting to 32bit first. I do upsample correctly but I will try
adding a filter to it to see if it will help in the final output.
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
Hello all,
I have created a group for Bidule (instead of a project file). Download these files and put
them in the Bidule "Group" folder.
5.1 Surround
6 Channel Gain
The first one is the 5.1 surround. After you put it in your folder called "Group". You can
access it from your "palette" and under "Groups" called "5.1 Surround". Instead of a mess
like this, you get a nice group like this with exactly 2 inputs and 6 outputs.
I also included something called 6ch Gain which allows you to control the gain of 6 channels
at once instead of 2.
Problems:
Bidule currently does move the attributes around as you saw in the tutorial and have to
manually set it to "Pentagon" under Emmigrator and move the right most notch on B-Pan to
0 degrees. To do this you can simply right click on the group (not in the palette) and hit
"Grouping" and "Expand" then change your controls and then to get back go to "Grouping"
again and then "Parent Group".
For me it helps because it makes it less confusing.
- joshbm
Hi Joshbm,
I was experimenting this evening and used your 5.1 surround group.
I made sure settings were okay, only I didn't look at the HNM filters, assuming they would be correct.
Anyway, to make a short story long.... I loaded the 6 files into CE (I refuse to use the name Audition ) and the
LFE was completely clipped..... really terrible !
Just FYI....
Any idea ?
EoH
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
Hey EoH! Thanks for the advice. I know what I'm hearing is not compression artifacts.
You said it sounded dull.... that's what happens with processing in 16 bits AND compression.
quote:
I'm not using any DRC either so that can't be to blame either. I'm encoding at 384kbps which
I thought was high enough. I've never been able to tell much of a difference between 384
and 448 before.
I guess I'm comparing it to the MP3 encoding I do using LAME. I challenge anyone to hear
the difference in the MP3's I encode and the original. If the AC3 encoding is really to blame,
is Dolby really that screwed up?? I can't imagine they would come up with algorithms worse
than that found in the free LAME encoding engine but maybe they did?
No they didn't ! but ...it's the combination of bitrate, bitdepth and processing..... Every step on the process should
be done in the highest possible quality to get an excellent result.
BTW... compare an original WAV with a WAV extracted from AC3. Not by your ears but bit-by-bit.....
What's out by compression, will not get in again !!
And (in my experience) with MP3 you're only safe with bitrates higher than 160.........
Lower onces will give a hearable difference in the comparison test, higher ones only when comparing bit-by-bit......
quote:
Thanks for the tips on processing, EoH! I will try converting to 32bit first. I do upsample
correctly but I will try adding a filter to it to see if it will help in the final output.
The 32 bits is the most important ! (Just like Kempfand said : Steinberg is even suggesting 64 bits now !)
Only use the filter when it still sounds dull..... but it shouldn't !
Hope it's a bit clear.... after a whole day trying and experimenting with different settings, my head is like a strainer
kind regards,
EoH
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
Hi Joshbm,
I was experimenting this evening and used your 5.1 surround group.
I made sure settings were okay, only I didn't look at the HNM filters, assuming they would be
correct.
Anyway, to make a short story long.... I loaded the 6 files into CE (I refuse to use the name
Audition ) and the LFE was completely clipped..... really terrible !
Just FYI....
Any idea ?
EoH
A bit more up-to-date information. I used wav2wav6.exe to make the 6 WAV's. But..... this program has (again)
another order. So I used the good old Besweet again and.... now the channels were in the right order and... the
clipping is in the SR channel (instead of the LFE). Of course I will have a look at the settings in Bidule, but I thought
you would like to know too .
grtz,
EoH
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
A bit more up-to-date information. I used wav2wav6.exe to make the 6 WAV's. But..... this
program has (again) another order. So I used the good old Besweet again and.... now the
channels were in the right order and... the clipping is in the SR channel (instead of the LFE).
Of course I will have a look at the settings in Bidule, but I thought you would like to know
too .
grtz,
EoH
Pppppffffffff........
I finally found out what's wrong.....
When you expand the group, you will see that the connections are not what they should be !
"The first one is the 5.1 surround. After you put it in your folder called "Group". You can access it from your
"palette" and under "Groups" called "5.1 Surround". Instead of a mess like this, you get a nice group like this with
exactly 2 inputs and 6 outputs." . this is from your first announcement. When you check the expanded group
against the "mess like this" picture, you will see the difference..... Only 5 channels instead of 6.
My fault was that I remapped the channels from your output again, which was already taken care of in the group.....
Anyway.... can you correct it ? TIA !!!
kind regards,
EoH
__________________

joshbm
Member
Location:
Posts: 67
Yeah, I was fooling around with some of the settings, :-\. I just tested it again and it sounds terrible lol. I will
upload my other group that isn't clipping.
Regards,
- joshbm

Umma
Member
Location:
Posts: 57
I'm hoping I missed something in this voluminous thread...but is there a batch method that we can batch files prior
to Plogue Bidule?

joshbm
Member
Location:
Posts: 67
The updated version can be redownloaded from the first link I gave you:
5.1 Surround
You can also make your own group simply by enhancing mine or you could select all of your work area (except of
course the player and recorders) and right click (once it's all highlighted) on an object that is highlighted and goto
"Grouping" and "Group Selected Objects".
Than once your done you can go to "Grouping" again and "Save Selected Group".
Regards,
- joshbm

Eye of Horus
Senior Member
Posts: 255
quote:
I'm hoping I missed something in this voluminous thread...but is there a batch method that
we can batch files prior to Plogue Bidule?
Where would you use a batch for ?
EoH
__________________

Umma
Member
Location:
Posts: 57
"Where would you use a batch for ?"
Well, say if I got 10 files I want to load into Pogue Bidule. Do I just do them one at a time or can I line them up
some kind of way?

Eye of Horus
Senior Member
Posts: 255
quote:
"Where would you use a batch for ?"
Well, say if I got 10 files I want to load into Pogue Bidule. Do I just do them one at a time or
can I line them up some kind of way?
I'm sorry. I guess I didn't see the reason , but was rather shortsighted.... :-)
Indeed, there is no batch for 10 different files.
I guess it can be done somehow....
- make one big file from the 10 (WAV/CUE ?)
- process that big one
- make 10 files from that big one with the CUE
It is a solution and I guess it can be made into something automatic....
Anyone else with ideas ???????????
grtz,
EoH
__________________

Umma
Member
Location:
Posts: 57
quote:
make 10 files from that big one with the CUE
I like the sound of that...okay.
To keep this thread on task, can you point me to someplace that explains how to make a cue file? I know what they
are, I think KPex or Andreas posted one in a thread somewhere on Doom9, but to make them is beyond me.
Thanks for the kind replies, I'm deleting, defragging, and getting ready to do a another CD.
D

daphy
Miles Freak
Location: Germany
Posts: 175
CUE instead of batch

@Umma
thats easy:
1. use Exact Audio Copy V0.95 prebeta 3 (works fine) to rip as one WAV + CUE
2. work on the WAV (ambisonic -> DTSWAV/AC3WAV)
3. burn the CUE as CDIMAGE with NERO/CDRWIN...
alternativ
1. use Cooledit open your single WAVs append to one big WAV
2. work on the WAV (ambisonic -> DTSWAV/AC3WAV)
3. use CUESHEET Editor or a simple Texteditor to make your own CUE in the EAC stil - you will find out, its not that
difficult
4. burn the CUE as CDIMAGE with NERO/CDRWIN... (howto and further informations)
-> this method is the only method I know to make f.e. a live concert on DTS/AC3 not to stop (for a very little, but
noticeable moment) between the tracks
further questions? send PM
CYA Daphy
PS: you can also use the CUE-file with more the one WAV and you can also create a CD-TEXT CD
Last edited by daphy on 23rd September 2003 at 12:39

Calabuig
Junior Member
Location:
Posts: 22
Sorry if it's off-topic...
I want to make a big applause for EoH and friends' work...
I'm new to this method and I'd like to try it.
But I have a problem in the Bidule installation: when it starts a message says that the program have found 46
groups and then it crashes. A new message says "access violation" and then the programs closes.
This is the last part of the bidule's log:
Now enumerating Slave Devices...
-------------------------------------------------------------------------------
SysTimer created
Starting ReWire!
Enumeration ReWire Devices:
-------------------------------------------------------------------------------
OnInit Opening Main Window
OnInit BiduleMainWindow in
OnInit maximize
OnInit SetTopWindow
OnInit buildGUI
BiduleMainWindow::buildGUI() in
BiduleMainWindow::buildGUI() new BiduleDocument in
BiduleDocument ctor in
BiduleDocument looking for default.bidule
opening default
--Loading C:\Programmi\Plogue\Bidule\layouts\default.bidule
XMLSerializer::load Building DOC in
XMLSerializer::build in
building children:0
building children:1
building children:2
building children:3
XMLSerializer::buildBidule of type : AudioOutput_0 with ID : UID_1026570523
AudioDevice: AudioOutput_0 id:3 master:1 ins:0 outs:2 allocated!
AudioDevice ctor out
AudioDevice::init in
AudioDevice::init out
AudioDevice: AudioOutput_0, is now acquired
XMLSerializer::buildBidule of type : MidiInput_0 with ID : UID_1026570524
MidiDevice: MidiInput_0 id:5 master:0 ins:1 outs:0 allocated!
MidiDevice: MidiInput_0, is now acquired
XMLSerializer::buildBidule of type : com.plogue.Comment with ID : UID_1026758624
XMLSerializer::buildBidule of type : com.plogue.Mixer with ID : UID_1044475375
building children:4
building children:5
building children:6
XMLSerializer::load Building DOC out
--Loading C:\Programmi\Plogue\Bidule\layouts\default.bidule done!
BiduleDocument ctor out
BiduleMainWindow::buildGUI() new BiduleDocument out
BiduleMainWindow::buildGUI() _biduleDoc->getGroupRoot()->getGeometry() in
BiduleMainWindow::buildGUI() _biduleDoc->getGroupRoot()->getGeometry() out
BiduleMainWindow::buildGUI() new Patchbay in
Severe Bidule Crash! docOpen:1 intransv:0 pbDirty:1
StackTraceExceptionFilter in
Unhandled Exception
Code: c0000005
Description: EXCEPTION_ACCESS_VIOLATION
Call stack:
------------------------------------------------------------------
------------------------------------------------------------------
StackTraceExceptionFilter out
Is there anyone who can help me?
Thanks in advance!

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
quote:
EXCEPTION_ACCESS_VIOLATION
Could you send the full Bidule.log to support@plogue.com?
Cheers

Calabuig
Junior Member
Location:
Posts: 22
Ok, I can use Bidule changing the screen colors depth to 256!
It's Matrox G400TV's fault!
(Anywhere, if is there anyone who can help me...!)
Another question: I own a home theatre system and I'd like to make a 5.1 CD (with LFE). Could you suggest me
"good" values for LFE's HNM_filter?
Thanks in advance!
@daphy
The link to your 2.1to5 layout doesn't work (I used "save as..." option). Could you fix it? Thanks
Last edited by Calabuig on 23rd September 2003 at 19:37

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by Calabuig
Ok, I can use Bidule changing the screen colors depth to 256!
It's Matrox G400TV's fault!
(Anywhere, if is there anyone who can help me...!)
Another question: I own a home theatre system and I'd like to make a 5.1 CD (with LFE).
Could you suggest me "good" values for LFE's HNM_filter?
Thanks in advance!
@daphy
The link to your 2.1to5 layout doesn't work (I used "save as..." option). Could you fix it?
Thanks
I would almost say : RTFT !!!!
(Read The F.....g Thread !)
But I won't.........
Joshbm posted a 5.1 group !
kind regards,
EoH
__________________

bitsnbytes
Member
Location:
Posts: 56
Managed to start the page but have not expanded the page so as not to confuse the gamers.
Yes this page is for gamers so its based on ac3enc.dll.
I will be expanding and dressing it up as real life allows I will Be adding A Bidule page ... Groups etc.
Again thanks
http://www.planetquake.com/filterfactory/3daudio.html
__________________
And Polk Audio

vtwin0001
CEO and owner of VTCorp.
Registered: Jul 2003
Location: Somewhere in Mars
Posts: 67
Hello, I just tried the method given, but I am getting strange results.
Its a VHS to DVD cnversion, so I took the .mpa file and converted it to 48Khz WAV (its already 2ch) with Headac3he.
I sucesfully converted the Stereo WAV file to 6 ch WAV; the resulting file was called result.wav and its size is 3.71
Gb
When I tried to run BeSweet on command line, I've got this:
D:\AC3 Audio from Stereo 2 5.1\BeSweetv1.4>BeSweet.exe -core( -input c:\result.wav -output c:\tempor\temp- -
type wav -6ch ) -ota( -G max )
BeSweet v1.4 by DSPguru.
--------------------------
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : c:\result.wav
[00:00:00:000] | Output: FL, FR, SL, SR, C, LFE
[00:00:00:000] | Floating-Point Process: No
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] +---------------------
[00:00:00:000] Conversion Completed !
[00:00:00:000] <-- Transcoding Duration
Visit DSPguru's Homepage at :
http://DSPguru.doom9.net/
D:\AC3 Audio from Stereo 2 5.1\BeSweetv1.4>
as you can see the time it took to convert it was zero, and the resulting files are 44kb each, therefore, I think there
was no conversion at all
What am I doing wrong with Besweet?
Please help
Thanks.

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by vtwin0001
Hello, I just tried the method given, but I am getting strange results.
Its a VHS to DVD cnversion, so I took the .mpa file and converted it to 48Khz WAV (its
already 2ch) with Headac3he.
I sucesfully converted the Stereo WAV file to 6 ch WAV; the resulting file was called result.
wav and its size is 3.71 Gb
When I tried to run BeSweet on command line, I've got this:
D:\AC3 Audio from Stereo 2 5.1\BeSweetv1.4>BeSweet.exe -core( -input c:\result.wav -
output c:\tempor\temp- -type wav -6ch ) -ota( -G max )
--------------------------
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : c:\result.wav
[00:00:00:000] +---------------------
Visit DSPguru's Homepage at :
http://DSPguru.doom9.net/
D:\AC3 Audio from Stereo 2 5.1\BeSweetv1.4>
as you can see the time it took to convert it was zero, and the resulting files are 44kb each,
therefore, I think there was no conversion at all
What am I doing wrong with Besweet?
Please help
Thanks.
I've had the same problems lately with files larger than 2 GB.
I used wav2wav6.exe instead to get the 6 mono files.
Only one thing : the channel naming is different.
The resulting files must be renamed like this :
fl > fl
fr > fr
c > c
sl > lfe
sr > sl
lfe > sr
kind regards,
EoH
__________________

daphy
Miles Freak
Location: Germany
Posts: 175
@ EOH
I did much encoding last week -> it now works perfect for me without LFE (youre right, no need of this hn-filter)

I also managed to integrate the results in DTS to VOBs - this wasnt a nice job, cause programs like IFOEDIT doesn
t do the expected actions (wasting sooo much time on it) so I had to switch to a commercial program (maybe I
will write a guide for this ).
some little hints
-> DTS-WAV CDs(PCM) work perfect on 32/44.1 no need to transform back to 16/44.1
-> 32 bit encoding needs to be checked every single channel for the click at the beginning
-> you can also mux DTS-WAV PCM (16/48) to DVD - but you will have only trouble with that (synchronize movie
etc.)
-> beware of a too big file (6 channel WAV) because Besweet wont accept them. Until now I dont know the
exactly size when you have to split your source file in to two pieces ~ around one hour at 32/48 I would
recommend this means less than ~5GB
CYA Daphy
Last edited by daphy on 30th September 2003 at 10:56

daphy
Miles Freak
Location: Germany
Posts: 175
quote:
I used WAV2WAV6.EXE instead to get the 6 mono files.
fl > fl
fr > fr
c > c
sl > lfe
sr > sl
lfe > sr
kind regards,
EoH
quote:
like:
c:\wav2wav6.exe 6wav.wav
I guess somethings missing
I havent tried this so far, but I remember this as part of the former ambi-guide
CYA Daphy

SallyDog
Posts: 50
Besweet should be able to handle files larger than 2g using 32 bit floating, although I have yet to succeed with this
method.
@EoH - thanks for the wav2wav6 mapping info. I've been trying to figure that out for the last week.
SallyDog

Eye of Horus
Senior Member
Posts: 255
commands wav2wav6
quote:
CYA Daphy
This is the commandline :
(you can put it in any directory you like, so here c:\test)
c:\test\wav2wav6.exe input.wav -44
This wil give 6 mono wav, with names frontleft,frontright,center,surroundleft,surroundright,subwoofer.
in 16 bit 44.1 Khz in the same directory.
( -48 is the default input )
I don't know if it works with 32 bits though ......
kind regards,
EoH
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by SallyDog
Besweet should be able to handle files larger than 2g using 32 bit floating, although I have
yet to succeed with this method.
@EoH - thanks for the wav2wav6 mapping info. I've been trying to figure that out for the last
week.
SallyDog
Hi Sallydog,
I don't know what's wrong, but every version of Besweet I tried (from 1.3 to 1.5b21) crashes on files larger than 2
GB (16 bits !) on Window 2000 pro or on XP pro. (Both with the latest SP's).
It crashes with differences though :
1. it starts and goes through the whole proces. At the end : crash
2. it starts and is ready within 2 seconds.
I tried several things like auto channel, 6 channels... makes no difference.
On my PC the versions worked without problem on smaller files......
grtz,
EoH
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
@ EOH
I did much encoding last week -> it now works perfect for me without LFE (youre right, no
need of this hn-filter)
I also managed to integrate the results in DTS to VOBs - this wasnt a nice job, cause
programs like IFOEDIT doesnt do the expected actions (wasting sooo much time on it)
so I had to switch to a commercial program (maybe I will write a guide for this ).
@Daphy
I also did a few albums and..... here it is the other way round :-)
I made one with a LFE that sounded better than without..... !!
I guess it just has too much to do with the "playing" equipment.
DTS to VOBs > Do you mean something similar like DVDArchitect does with AC3 ? Making music albums ?
Which program did you use ? I would love to see a guide !!
And now we're busy..... hey, I had some other questions for you, but you didn't respond. Perhaps overseen ???
Question was : do you still provide the webspace to post the Bidules and screenshots ? If not...please let me know,
because I have the intention to built one myself, but don't want to invent the wheel twice :-)
OTOH it won't be before the new Dibule version is finished !
kind regards............
EoH
__________________

SallyDog
Posts: 50
@EoH
See the BeSweet FAQ thread in this forum, question #30.
Regards,
SallyDog
Last edited by SallyDog on 30th September 2003 at 14:35

rasta21
german.doom9
Registered: Nov 2001
Location:
Posts: 39
quote:
I used wav2wav6.exe instead to get the 6 mono files.
fl > fl
fr > fr
c > c
sl > lfe
sr > sl
lfe > sr
kind regards,
EoH
...i think you have to change some front to center, too.
fr > c
c > fr
...or fl...if you prefer this way
__________________
CoVeR YoUr AsS
rasta21 @ doom9.de board
1st October 2003 05:20

TerraForce1
Member
Location:
Posts: 82
2nd pass
Well hi,
my question, as stupid as it may sound, is the following. Can the output (let's say a 32 bit recording)of Bidule be
used as an input again. I really like the ambisonic sound image that is created by Bidule. I don't know if it is
possible to feed this sound image again trough bidule to achieve a better sound than the first. Or will it stay the
same. (I look at it like a 2nd pass for Bidule)
I have done no testing, because I am now converting a movie soundtrack, but suddenly it came up in my mind.
1st October 2003 21:28

daphy
Miles Freak
Location: Germany
Posts: 175
quote:
quote:
CYA Daphy
This is the commandline :
(you can put it in any directory you like, so here c:\test)
c:\test\wav2wav6.exe input.wav -44
This wil give 6 mono wav, with names frontleft,frontright,center,surroundleft,surroundright,subwoofer.
in 16 bit 44.1 Khz in the same directory.
( -48 is the default input )
I don't know if it works with 32 bits though ......
kind regards,
EoH
Hi folks,
I did some testing with wav2wav6.exe -> result 32bit/48KHz splitting ended always in 6 empty wavs so we cant
use this with greater file size, what means we stick at besweet and the 2GB barrier.
6th October 2003 10:28

daphy
Miles Freak
Location: Germany
Posts: 175
Promised Guide
DVD sound track guide:
If you want to do some tracks for DVD this means much work in splitting and mergin files.
A. Sound
1. I would recommend to split the 32/48 source WAV at about ~40-45 min or less
2. then run the ambisonic part on each file
3. splitt the 6wav-file (result from bidule) with besweet (the file would be around 2GB),
4. lower (NOT CUTTING because of the synchronization) the volumen if necesary at the beginning of
EACH of the channel files because of the noisy scratch produced mostly in 32bit(?),
5. merge all channel files with f.e. 'open apend' (monoFL1 + monoFL2 + monoFLxx -> monoFL ... )
6. then transform via encoder to DTS/AC3.
B. Movie
1a. Use f.e. Smartripper to demux the movie track from DVD -> result a single file (MV2) - (you can also
use ChaptorXtraktor to make a chapterfile in DVD Maestro styl)
1b. Use a MPEG Encoder like TMPEG Plus / MainConcept / ProEncoder... to transform your AVI-file into a
MV2/MPV/MPG ... file (basic MPEG Stream -> no sound input selected, this makes the rest little easier).
If necessary you can merge more than one AVI before doing this with VirtualDub <open> AVI1 <open
apending segment> AVI2 ... then saving using source file modus.
C. Merging (not authoring in Hollywood style)
Because of being so much impressed of the DTS vs. AC3 comparison I do always work with DTS.
The only program I found which is EASY to use for our needs is DVD Maestro (2.9 not supported
anymore, maybe someone would find a as easy method with a other program)
-> Im also tired of messing around and wasting time with Scenarist, THIS IS DEFINITIVLY NO EASY TO
USE PROGRAM, IF YOU WANT TO SUGGEST THIS MAKE YOUR OWN GUIDE AND POST EM BEFORE
SUGGESTING IT
1. open the app and dragndrop the MV2 and the DTS, AC3 (beware of naked PCM-Tracks because of
the sampling rate) to the box down right.
Now the program prozesses each single file, depending of the file size you can go in the kitchen and
make another cup of cafe
2. Now dragndrop each file to the top (right), position zero, that means beware of not exact positions
because of the synchronization
(3. Chaptors: use the chaptorfile from ChaptorXtraktor - but you have to check them, found a bug in
DVD Maestro)
(4. Make your Hollywoodstyl Menue, else)
5. delete the menues in DVD Maestro recommended)
6 Complile to VOB/IFO/BUB
D. Make it fit to DVD
I noticed with DTS Tracks a lag of space on DVD since I have no DVD-RAM writer there only two
oportunities left:
a. Shrink
Use DVD Shrink/Nero Recode or what ever on the files to make it fit!
I strongly recommend to use the reauthor combined with the deeper scan option - do not shrink lower
than 70% in my opinon the quiality the is still acceptable.
b. Split
Use DVDFab to split the files, dont forget to use a nice splitting graphic
Thats it
For the first take calculate a weekend on a two hour film
Any suggestions - or ideas to explain it easier?
CYA Daphy
__________________
CYA Daphy
________________________________________
The New Ambisonic Server (needfulthings) can be found here!,
further info thread; upload folder ; (sometimes down please send PM in this case)
Last edited by daphy on 12th February 2004 at 08:56

DSPguru
BeSweet Author
Posts: 3446
quote:
I did some testing with wav2wav6.exe -> result 32bit/48KHz splitting ended always in 6
empty wavs so we cant use this with greater file size, what means we stick at besweet
and the 2GB barrier.
your attention to q30..
@EoH
any chance that you update your guide according to this ?
__________________
Last edited by DSPguru on 6th October 2003 at 13:15

Eye of Horus
Senior Member
Posts: 255
quote:
your attention to q30..
@EoH
any chance that you update your guide according to this ?
I still need to update the guide, but on this point :
I am a bit confused. Besweet (all versions I have) crashed when I tried to extract 6 mono wavs from a bigger than 2
GB file. But Wav2wav6 did the job !
@Daphy : you make a strange conclusion : it doesn't work on 32/48, so we must find another solution for files
bigger than 2 GB. But these both have nothing to do with each other ! It doesn't work on 32/48, but it does work on
files bigger than 2 GB ! (with the right OS and filesystem, of course !) I had no problem at all on Windows 2000 with
a 3.6 GB 16/48 file on NTFS.
So am I right in this conclusion :
File in 16 or 32 bits and smaller than 2 GB > Use Besweet
File in 16 bits and smaller or bigger tahn 2 GB > use wav2wav6
File in 32 bits and larger than 2 GB > use Cooledit ??
kind regards,
EoH
__________________
Last edited by Eye of Horus on 6th October 2003 at 13:42

daphy
Miles Freak
Location: Germany
Posts: 175
quote:
@Daphy : you make a strange conclusion : it doesn't work on 32/48, so we must find another solution
for files bigger than 2 GB. But these both have nothing to do with each other ! It doesn't work on
32/48, but it does work on files bigger than 2 GB ! (with the right OS and filesystem, of course !) I had
no problem at all on Windows 2000 with a 3.6 GB 16/48 file on NTFS.
Hi EoH
shurely strange but it works fine for me.
Still the main problem is if you transcode the whole 2 hour 32/48 WAV in one go, youll get a about 5.8 GB file
(6wav-file ambisonic).
And now suddendly something very surprising happend: you start besweet to split and it works but
finished after (it differs everytime) a splitting time from f.e. 25min -> you get 6 six files with 25min -
definitivly no crash at all -> cause with the GUI commandline besweet tries to start the surcode app -
this wouldnt happen if besweet crashes, am I right?).
I tried to find out where the mistake is, ok I do know the 2GB barrier, but those 6 single WAVs are alright but only
unfinished.
I use cooledit on the 6wavfile (ambisonic):
the scanner stops after 25min (again!!!!) and shows me a 25min 32/48 mono file
quote:
So am I right in this conclusion :So am I right in this conclusion :
File in 16 or 32 bits and smaller than 2 GB > Use Besweet
File in 16 bits and smaller or bigger tahn 2 GB > use wav2wav6
File in 32 bits and larger than 2 GB > use Cooledit ??
sorry EoH
-> for me it seams that cooledit wouldnt solve the big filesize problem
why 25min or 3:37min... (take1/take2...)
I start believing that this is not meanly a besweet/cooledit problem maybe more a bidule problem.
Ok this is why I start splitting the source file (stereo file) up to 4 times - transcode the pieces to 6wavs (ambisonic),
split into 6 single mono wavs and then again merge them to 6 single 2h 32/48 mono wavs (as mentioned in the
guide) (FL/FR/C/SL/SR)
Maybe the guy from plodge can help us to find out if there is a second size barriere caused by bidule
CYA Daphy
PS: hope someone would understand what I mean

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-- Audio encoding
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Channel Surround
Author Thread
Calabuig
Junior Member
Location:
Posts: 22
Hi, I've a problem with Besweet when I try to create
single 6 mono wav files. It produces 6 files but when I
open one of them, the player says that his lenght is
6:12:00 hours (the original and the bidule result file is
1:59:55 and larger than 2Gb).
I try wav2wav6 but the resulted files' names are
wrong.
Eye of Horus suggest to rename them like this:
fl > fl
fr > fr
c > c
sl > lfe
sr > sl
lfe > sr
but Rasta21 says we must also rename:
fr > c
c > fr
"...or fl...if you prefer this way"
Can you help me to make it right?
Thanks in advance!

daphy
Miles Freak
Location: Germany
Posts: 175
quote:
Hi, I've a problem with Besweet when I
try to create single 6 mono wav files. It
produces 6 files but when I open one of
them, the player says that his lenght is
6:12:00 hours (the original and the bidule
result file is 1:59:55 and larger than 2Gb).
Question
what source file format do you use?
16bit/32bit - 44.1/48KHz
CYA Daphy

Calabuig
Junior Member
Location:
Posts: 22
16bit - 48Khz
Wav2wav6 is ok, but I'm in a mess with the resulted
files' names...

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by Calabuig
16bit - 48Khz
Wav2wav6 is ok, but I'm in a mess
with the resulted files' names...
I will do some tests and give you the definitive answer
ASAP....
kind regards,
EoH
__________________

Eye of Horus
Senior Member
Posts: 255
layout with wav2wav6
This is how the layout should be :
fl > fl
fr > c
c > fr
sl > lfe
sr > sl
lfe > sr
(Thanks Rasta21 !)
kind regards,
EoH
__________________

Boulder
Pig on the wing
Location: Jyvaskyla, Finland
Posts: 525
Thanks EoH, it's a very fine guide indeed I'm
converting some older VHS tapes to DVD and I'm
using the guide to make a 5.1 AC3 track - the tapes
have a surprisingly good stereo sound despite the fact
that the material was shot on a handheld camera long
time ago.
EDIT: Thanks for the Soft Encode command line tip, I
couldn't get the program working in the GUI mode, it
just crashed after a short while. The command line
works perfect!
Also a note to davidv@plogue: I can reproduce one
situation where the CPU usage hits over 100%.
1) Start Bidule
2) Load a layout
3) Open audio file player and enter a file
4) Open audio file recorder and enter the destination
file
5) Activate signal processing
Bang, 100% it is
If you close the audio file player before going to the
audio file recorder and close the recorder before
activating signal processing, the CPU usage will
remain normal.
Hope this helps.
__________________
And if the band you're in starts playing different tunes
I'll see you on the dark side of the Moon...

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by Boulder
Thanks EoH, it's a very fine guide
indeed I'm converting some older
VHS tapes to DVD and I'm using the
guide to make a 5.1 AC3 track - the
tapes have a surprisingly good stereo
sound despite the fact that the
material was shot on a handheld
camera long time ago.
Thanks !!
Love your signature !!!!
Please take a look at my homepage :-)
EoH
__________________
10th October 2003 21:00

24hourloop
Senior Member
Location: Texas
Posts: 185
How do you burn?
Maybe I am asking a question with an obvious answer:
a.) One method suggested using EAX, pump the thing
through the conversion and then burn with the same
CUE file. Unfortunately the WAV is now 1.6GB and I
could locate no tool that would load the CUE. What
should I use?
Using Softencode I would wind up with an AC3. How
would I burn the AC3? For a regular DVD I need video
or none of the tools I know will create a structure, not
even IFOEDIT.
What am I missing?
13th October 2003 20:25

Eye of Horus
Senior Member
Posts: 255
Re: How do you burn?
quote:
Originally posted by 24hourloop
Maybe I am asking a question with an
obvious answer:
a.) One method suggested using EAX,
pump the thing through the
conversion and then burn with the
same CUE file. Unfortunately the WAV
is now 1.6GB and I could locate no
tool that would load the CUE. What
should I use?
First : so this won't fit onto a cd :-) download Daemon
tools and see if you can load the CUE into that ....
quote:
Using Softencode I would wind up
with an AC3. How would I burn the
AC3? For a regular DVD I need video
or none of the tools I know will
create a structure, not even IFOEDIT.
What am I missing?
You can burn AC3's with Sonic AC3 DVD Burner (max
of 99 tracks) or use DVD Architect to make a music
DVD. You can burn as much as you want (well : 99 x
99 songs maximum !). I love this program !
Here is a link to a guide I wrote :
Make a music DVD with DVD Architect
http://www.app.demon.nl/guides.htm
Good luck !
__________________
Last edited by Eye of Horus on 1st March 2004 at 12:10
13th October 2003 22:35

24hourloop
Senior Member
Location: Texas
Posts: 185
Great, thanx.
The DVD Architect method looks promising ;-)
14th October 2003 02:30

Umma
Member
Location:
Posts: 57
Question: Couldn't we use dts with Maestro or
Scenarist in the same way you suggest composing an
AC3 DVD with Architect? I would love to try...but
learning those progs is a b*tch...especially Scenarist.
14th October 2003 03:02

Eye of Horus
Senior Member
Posts: 255
quote:
Question: Couldn't we use dts with
Maestro or Scenarist in the same way
you suggest composing an AC3 DVD
with Architect? I would love to try...
but learning those progs is a b*tch...
especially Scenarist.
I never tried it, but I think it should....
As video you use a static picture that doesn't change
throughout the song. And I agree about the learning
curve.....
Perhaps the newer programs can handle DTS too ?
DVD Lab, TMPG Author ?
It's a pity DVD Architect became part of a package....
but I still hope they will bring out an update for DTS
and where you can use AC3 and DTS, without gaps !
EoH
__________________
14th October 2003 15:17

vtwin0001
Posts: 67
Can somebody please point me out where can I find
Wav2wav6.exe ???
I couldn't find it here nor in google.
Thanks.
14th October 2003 23:15

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
New Bidule
Hi there.
A new beta of bidule is available on plogue.com
We fixed some issues with disc streaming and
parameter saving.
This doesnt include however, the Offline mode yet.
its available here:
http://plogue.com/download_form.php...er6500test1.
exe
Please try it out if you experienced problems.
Davidv
14th October 2003 23:31

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by vtwin0001
Can somebody please point me out
where can I find Wav2wav6.exe ???
I couldn't find it here nor in google.
Thanks.
It IS in this thread :-)
Well, anyway here's the link again :
EoH
__________________
14th October 2003 23:45

vtwin0001
Posts: 67
Thank you, I guess I didnt searched that well...
14th October 2003 23:56

Boulder
Pig on the wing
Posts: 525
Re: New Bidule
quote:
Hi there.
A new beta of bidule is available on
plogue.com
We fixed some issues with disc
streaming and parameter saving.
This doesnt include however, the
Offline mode yet.
its available here:
http://plogue.com/download_form.
php...er6500test1.exe
Please try it out if you experienced
problems.
Davidv
Thanks, David.
Would you happen to have a changelog somewhere?
I'd like to see what's changed before I upgrade
Did you notice my bug report a bit earlier in this
thread?
@EoH: Thanks, I'll definitely have to check your
website out Ain't it funny how all the PF and Alan
Parsons fans seem to fiddle with this surround stuff?
__________________
Last edited by Boulder on 15th October 2003 at 11:19
15th October 2003 09:36

Eye of Horus
Senior Member
Posts: 255
Re: Re: New Bidule
quote:
@EoH: Thanks, I'll definitely have to
check our website out Ain't it funny
how all the PF and Alan Parsons fans
seem to fiddle with this surround
stuff?
LOL !!
I think that's because WE heard the Quad version of
DSOTM and the DTS of On Air !!
grtz,
EoH
__________________
15th October 2003 11:16

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
Yes i saw the bug report. we are investigating.
Allthough im unable to reproduce it.
We've changed some file handling code in the latest
version,
mind to have a go?
revision history:
http://plogue.com/bidule/experimental.html
Cheers
15th October 2003 18:23

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
speaking of DSOTM in quad.
Anyone has the SQ quad vinyl somewhere?
I have the schematics for a .bidule file that decodes
CBS SQ.
But since im not yet 'setup" to do quad/5.1, i never
tested it using
my copy of walter carlos's "switched on back" SQ
Quad.
I guess if i had an example of say the intro to money
in SQ,
idd speed up my setup
Cheers
15th October 2003 18:27

Boulder
Pig on the wing
Posts: 525
Thanks, upgrading it is then I have some stereo-
>5.1 work to do this week so it'll be a nice test too.
The bug could well be system dependent, those are
hard to trace. We'll see if I can reproduce it with the
latest version.
__________________
15th October 2003 18:30

Vidi_It
Junior Member
Registered: Feb 2003
Location: Denmark
Posts: 1
quote:
speaking of DSOTM in quad.
Anyone has the SQ quad vinyl
somewhere?
I have the schematics for a .bidule
file that decodes CBS SQ.
But since im not yet 'setup" to do
quad/5.1, i never tested it using
my copy of walter carlos's "switched
on back" SQ Quad.
I guess if i had an example of say the
intro to money in SQ,
idd speed up my setup
Cheers
I have a copy of DSOTM in sq. Tell me about the
Bidule schematic as I would also like to convert it to
5.1.
16th October 2003 17:34

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
Hi
Its basically to create a bidule group that has 2 ins
and 4 outs
and does the maths that are on this page:
http://www.mtsu.edu/~dsmitche/rim45...ad_Formats.
html
the only "obscure" part is doing a 90degree phase
shift,
But just use the BiduleFIR, and do a "hilbert
transform"
Ill try to to the bgrp for you if i get the chance...
17th October 2003 17:36

daphy
Miles Freak
Location: Germany
Posts: 175
sounds interesting
hope you find the necessary time
17th October 2003 20:40

Tantulus
Junior Member
Location: Massachusetts
Posts: 11
Re: layout with wav2wav6
quote:
This is how the layout should be :
fl > fl
fr > c
c > fr
sl > lfe
sr > sl
lfe > sr
(Thanks Rasta21 !)
kind regards,
EoH
I'm confused with the Wav2Wav6 layout. First of all
my files are named Front_Left.wav etc. instead of fl
etc. Do these files need to be renamed? Second, is it
simply a matter of just inserting them into surcode be
remapping them?
23rd October 2003 03:00

Eye of Horus
Senior Member
Posts: 255
Re: Re: layout with wav2wav6
quote:
Originally posted by Tantulus
I'm confused with the Wav2Wav6
layout. First of all my files are named
Front_Left.wav etc. instead of fl etc.
Do these files need to be renamed?
Second, is it simply a matter of just
inserting them into surcode be
remapping them?
Front_Left =FL etc.
You can remap them by putting them in the right
place in Surcode.
But you can bet you will forget at some point and put
them in without the right mapping
EoH
__________________
23rd October 2003 10:07

echooff
old and clueless
Location: The Two Rivers
Posts: 549
Has anyone run DSOTM thru bidule. Can anyone
reccomend other albums that compare with sterotomy
after bidule.
__________________
Live like there is no tomorrow!
Plan for your 100th birthday.
23rd October 2003 15:14

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by echooff
Has anyone run DSOTM thru bidule.
Can anyone reccomend other albums
that compare with sterotomy after
bidule.
Stereotomy was (AFAIK) the only rock album
recorded with the special Ambisonic microphones.......
EoH
__________________
23rd October 2003 15:49

Tantulus
Junior Member
Posts: 11
Re: Re: Re: layout with wav2wav6
quote:
Front_Left =FL etc.
You can remap them by putting them
in the right place in Surcode.
But you can bet you will forget at
some point and put them in without
the right mapping
EoH
You're absolutely right. I get confused when I try to
put the files in the right place in Surcode. Thanks for
your sage advice.
24th October 2003 18:32

TerraForce1
Member
Location:
Posts: 82
MPA vst-plugin
Hi, is there a freeware .mpa vst-plugin avalaible. I
would like to run some mp2 audio files trough bidule.
26th October 2003 03:32

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
Re: MPA vst-plugin
quote:
Hi, is there a freeware .mpa vst-
plugin avalaible. I would like to run
some mp2 audio files trough bidule.
Hum there is a decoder yes
"Read MPEG 1&2 (layer 1,2,3) files (i.e. .mp1, .mp2, .
mp3). "
(so rename your mpa to whatever the format, prolly
mp2)
Get it here
http://askywhale.free.fr/vst/
Warning, this plugin seems to load the decompressed
file in ram!!
Now since Bidule only process uncompressed audio
streams (and will always be that way), you would
have to reencode the output to mpa again after the
processing.
So imho you should maybe uncompress the mpa ->
wav then use the Bidule
file player and recorder.
26th October 2003 15:24

echooff
old and clueless
Posts: 549
@Eye of Horus
I guess what I was meaning to ask was a little input
on what cd's people have tried and good results. It's
my understanding from reading the thread that some
produce poor results. I have recently upgraded my
stereo system and I want to quickly out put a few that
I can enjoy while I tinker with settings and cds
__________________
26th October 2003 15:59

Umma
Member
Location:
Posts: 57
Pink Floyd gives good results. I've gotten good results
on everything except U2. The Unforgettable Fire is
heavy in the reverb, anyway, and I have a hard time
keeping it from sounding muddy. The Beatles don't
seem to sound right for me, either. I'm working on
the White Album.
Peter, Paul and Mary, Crosby, Stills, and Nash came
out great...
26th October 2003 17:15

echooff
old and clueless
Posts: 549
Thanx, I'm going to play with my new toys tonite!
Anybody try tubular bells or any van halen?
__________________
Last edited by echooff on 26th October 2003 at 17:40
26th October 2003 17:34

kempfand
Senior Member
Posts: 194
Updated / New Ambisonics VST-plugins
Gerzonic has published 1 update (Emigrator) and 2
new toys:
Panorama
quote:
Panorama is a 2nd Order Ambisonic
Panner with a very simple user interface
in the tradition of bPan.
InterGate8 Demo
quote:
The demo of InterGate8 is online! A
virtual 100% ambisonic mixing desk.
Limitations of the Demo: only the first
two channels can be used. The Demo is
first order only. Full versions shall be
available in a few days.
Emigrator 1.0 (latest built)
quote:
A new build of Emigrator is available. It
fixes one bug: The UI was not updated
according to the internal state when
loading a project in some hosts. This has
been fixed for OSX and win32. Maybe this
release will work in Nuendo 2.0, who
knows..
Kind regards,
Andreas
26th October 2003 18:40

ChristianHJW
Matroska Team
Location: Bavaria - Germany
Posts: 1274
Please explain a bit more in detail ? What exactly can i
do with those plugins ? Could it help us using Vorbis
surround some day ?
__________________
Support the future of video and audio encoding :
matroska as container, USF as subtitles standard
and CoreAPI as codec interface API in future If you
are a developer and plan to contribute to any of the
projects, pls. visit us on irc.corecodec.com ,
#matroska , as a user pls. turn to the Doom9
Guides or to the Matroska Support Pages to get
more information on how to use matroska and USF
for your encodings
26th October 2003 19:58

echooff
old and clueless
Posts: 549
Do a search on ambisonics in the forum. Several
extensive threads on the subject.
__________________
26th October 2003 20:14

kempfand
Senior Member
Posts: 194
Btw: I meant to add this to the sticky-post /
Ambisonic guide, so I made a mistake here
No big deal, but if one of the Moderators can move /
end-add it there, the better.
Regards,
Andreas
26th October 2003 23:37

daphy
Miles Freak
Location: Germany
Posts: 175
Hi Andreas,
have you made some testings with the new
emigrator?
quote:
A new build of Emigrator is available. It
fixes one bug: The UI was not updated
according to the internal state when
loading a project in some hosts. This has
been fixed for OSX and win32.
for me this bug didnt apear again since Bidule
0.6500test1
CYA Daphy
Last edited by daphy on 27th October 2003 at 15:52
27th October 2003 15:49

kempfand
Senior Member
Posts: 194
No tests yet from my side on the latest built of
emigrator
Too much at professional work-place currently
ongoing for me
I'll wait with further tests until the final Bidule 6.5 is
out, which is rumored to offer a warp-speed offline-
mode
Andreas
27th October 2003 16:21

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Tantulus
Junior Member
Posts: 11
quote:
Originally posted by echooff
@Eye of Horus
I guess what I was meaning to ask was a little input on what cd's people have tried and good
results. It's my understanding from reading the thread that some produce poor results. I have
recently upgraded my stereo system and I want to quickly out put a few that I can enjoy while
I tinker with settings and cds
If you like Orchestral music I strongly suggest The Beethoven Nine Symphonies conducted by H. von Karajan. Feels
like your in a concert hall.
28th October 2003 18:11

kempfand
Senior Member
Posts: 194
You might want to try your preference from the following list. They have been reported in the quadraphonicquad forum
section to give impressive surround effects when played back through various high end decoders. I'm sure this is a good
starting-point for the Ambisonic method(s)
Albums:
code:
Aphrodite's Child -666 - 1969 Vertigo LP - 838 430-2
Beasty Boys-Hello Nasty QS Decoder
Beatles - Revolver (especially "Tomorrow Never Knows") QS Decoder
Beatles - Sgt Pepper QS Decoder
Beatles - Magical Mystery Tour
(except Walrus) QS Decoder
Beatles - White Album QS Decoder
Beatles - Yellow Submarine Original version- QS Decoder
Big Brother And The Holding Co - Cheap Thrills QSD-1
Blue Cheer - Outside Inside QSD-1
Captain Beefheart-Trout Mask Replica SQ Decoder
Paula Cole - This Fire SQ Logic
John Cooper Clarke - Best Of SQ Logic
Julian Cope-Rite 1, Rite 2, Rite Now Sansui Variomatrix-Surround setting
Julian Cope - St. Julian QSD-1
Dead Can Dance-Into The Labyrinth Sansui Variomatrix-Surround Setting
Thomas Dolby- She Blinded Me With Science QSD-1
Bob Dylan - Highway 61 Revisited QS decoder
Bob Dylan - Blonde On Blonde QS decoder
Fleetwood Mac - Then Play On QSD-1
Fortran 5 - Bad Head Park Tate II, surround mode
John Foxx- Metamatic Virgin CDV2146 QS decoder-Surround Setting
Peter Gabriel - Secret World Live SQ Logic
Rory Gallagher - Wheels within Wheels (2nd half of album) SQ Logic
Rory Gallagher - Let's Go To Work (Box Set) SQ Logic
Godspeed You Black Emperor-F#A# Sansui Variomatrix-Surround setting
Grateful Dead - Anthem Of The Sun QSD-1
Joji Hirota -The Gate SQ Logic
Iron Maiden - Dance of Death Sansui Variomatrix-Surround setting
Jefferson Airplane - After Bathing At Baxter's QSD-1
Jimi Hendrix-Are You Experienced? Sansui Variomatrix-Surround setting
Jimi Hendrix-Electric Ladyland Sansui Variomatrix-Surround setting
Jimi Hendrix Experience - Axis Bold As Love QSD-1
Jethro Tull-Benefit QSD-1
King Crimson-In The Wake Of Poseiden Sansui Variomatrix-Surround setting
Paul McCartney-Ram QS Decoder
Monster Magnet - Powertrip QSD-1
Moody Blues - In Search Of The Lost Chord QSD-1
The Mothers of Invention -Uncle Meat QS Decoder 1968 Bizarre/Warner Reprise LP - 2MS 2024
Nektar - Journey To The Center Of The Eye QSD-1
John Parish and Polly Jean Harvey-Dance Hall at Louse Point Sansui Variomatrix-Surround setting
Pink Floyd-Piper At The Gates of Dawn Sansui Variomatrix-Surround setting
Pink Floyd-Pompeii Sessions Sansui Variomatrix-Surround setting
Pink Floyd-Saucerful of Secrets Sansui Variomatrix-Surround setting
Pink Floyd-Ummagumma (studio sides) Sansui Variomatrix-Surround setting
Robert Plant & the Strange Sensation - Dreamland
SQ Logic
Procol Harum - A Salty Dog QSD-1
Psychedelic Furs - Book Of Days QSD-1
Radiohead - Hail To The Thief Tate II, surround mode
Rolling Stones - Their Satanic Majesties Royal Request QS Decoder
The Sensational Alex Harvey Band - Rock Drill SQ Logic
Simon And Garfunkel - Bookends QS Decoder
Soundgarden - A-Sides QSD-1, surround mode
Spirit - The Twelve Dreams of Dr. Sardonicus QSD-1
Tangerine Dream-Alpha Centuri Sansui Variomatrix-Surround setting
Tangerine Dream-Zeit Sansui Variomatrix-Surround setting
United States Of America - United States Of America QS Decoder - 1967 Columbia LP CS9614
Andreas Vollenweider - Book of Roses SQ Logic
Andreas Vollenweider - White Winds SQ Logic
Andreas Vollenweider - Dancing With the Lion SQ Logic
Rick Wakeman & The English Rock Ensemble-Out of the Blue Sansui Variomatrix-Surround setting
Yes-Tales From Topographic Oceans Sansui Variomatrix-Surround setting
Frank Zappa-Lumpy Gravy Tate-SQ Mode
Regards,
Andreas
29th October 2003 18:02

echooff
old and clueless
Posts: 549
Thank you for the input. Gonna have fun tonite.
__________________
29th October 2003 18:36

Gradius
Junior Member
Location:
Posts: 13
I'm trying to do a 5.1, but I just get an empty wave on LFE channel.
Check my config here.
Please, someone post a clean LFE guide.
Gradius
30th October 2003 06:42

PeterC
Junior Member
Location:
Posts: 3
As a newbie to this technique I'd like to clarify one thing ?
Is it important to convert...16bit 44.1khz material
that I rip from audio cd's to 32 bit before playing
it through Bidule ?
I did mix a few Queen albums
( Queen and Queen II ) with ambisonics as a test and created
a DTS-cd combining the two using this guide.
Most of the mixes hardly gained anything from it with the exception
of a few songs (shame really), mind me
I have the oldest remasters of the set that are
said to be poor compared to the current 24bit remasters.
(I actually prefer my vinyl versions to the cd's I have..they are that..cold sounding)
But as a plus side. Live recordings sounds really
nice usually in ambisonics...or whatever the proper term for this would be. Ambi-dts ? I tested on several stereo
sourced fm-radio broadcast recordings and other quality live material with stereo separation (fm/soundboard sourced
bootlegs mainly) and was pleased with what I heard.
Anyone done any experimenting with their BPAN etc settings for live recordings etc ? I'm curious to what has given
good results other than
the default settings displayed on this guide.
Thanks.
And a big thank you to 'Eye of Horus' for this guide.
12th November 2003 00:34

iSpy
Junior Member
Location:
Posts: 4
where do i find the program wav2wav6.exe
I was just wondering where I could find the program wav2wav6.exe that is spoken of in this thread. Thanks, iSpy

jorel
Senior Member
Location: Brasil - MG
Posts: 657
link from Maxiuca posted here:
http://forum.doom9.org/showthread.p...xe&pagenumber=2
maybe is working:
http://tommers.w.interia.pl/wav2wav6.zip
Last edited by jorel on 13th November 2003 at 02:17

Umma
Member
Location:
Posts: 57
Gradius...did you ever solve your problem?
Peter C...
I'm experimenting right now with some live stuff, but I haven't really manipulated any of the ambisonic settings. I had
to run the wav through cool edit to try and clear up the muddle of the mid-range, first, but I'm having trouble cleaning
up. I don't experiment with the ambisonic settings because in order to hear the results I have to burn it, and I went
through too many CDs, too fast.
What live stuff are you working with? ROIOs?
Eye of Horus...if you get the chance to read this...do you think you can convince Eric to do another APP album in
ambisonics? Better yet, bring him here and ask him to post.

Eye of Horus
Senior Member
Posts: 255
quote:
Gradius...did you ever solve your problem?
Peter C...
I'm experimenting right now with some live stuff, but I haven't really manipulated any of the
ambisonic settings. I had to run the wav through cool edit to try and clear up the muddle of the
mid-range, first, but I'm having trouble cleaning up. I don't experiment with the ambisonic
settings because in order to hear the results I have to burn it, and I went through too many
CDs, too fast.
What live stuff are you working with? ROIOs?
Eye of Horus...if you get the chance to read this...do you think you can convince Eric to do
another APP album in ambisonics? Better yet, bring him here and ask him to post.
S..t !!!
Just came back from a radio recording in the studios in Hilversum - the Netherlands. Had an excellent time with Eric !
But..... did not ask this :-(
Well..... pictures will be on the website within a few hours !
grtz,
EoH
__________________

TerraForce1
Member
Location:
Posts: 82
Problems with windows 98 se
Hi,
I am experiencing some problems with windows 98 se and bidule. The program gets stuck at the opening screen.
I tried some GL test , they worked allright. Does somebody has a clue?

Umma
Member
Location:
Posts: 57
Bidule's home page says that it is mostly for Win2K though some people have gotten it to work with Win98.

Umma
Member
Location:
Posts: 57
So, uh...when is the new Bidule coming out?
23rd November 2003 04:31

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
quote:
So, uh...when is the new Bidule coming out?
For bidule-specific questions, you might have better chances on posting on our own phorums here http://plogue.com/
phorum/list.php?f=1
However for questions about the usage on bidule in the context
of a 5.1 mix, here is fine.
To answer the question though we will release 0.6500 final
before the first half of december. I might include the
"Offline rendering" mode.

Eye of Horus
Senior Member
Posts: 255
quote:
For bidule-specific questions, you might have better chances on posting on our own phorums
here http://plogue.com/phorum/list.php?f=1
However for questions about the usage on bidule in the context
of a 5.1 mix, here is fine.
To answer the question though we will release 0.6500 final
before the first half of december. I might include the
"Offline rendering" mode.
Might ???? You'd better !!!!!
Eye of Horus
__________________

Umma
Member
Location:
Posts: 57
Thanks for the reply, David. I can't WAIT for that offline rendering mode!
Yup. EoH already got the guide written up...
You know, guys, I've been using Virtual Microphone almost exclusively after processing through Bidule. The way I've
been doing it is to just record the 4 channels that come out of bfprocedit as a wxyz file, then run it through VVM with
the master gain at 3.1, using the 5.1 option (and the link boxes checked). VVM is FAST! Spits out 6 mono files in no
time at all. Gotta shut it down after about 4 big conversions and start it right back up, though. I guess it's not
releasing the memory, but the speed of the program MORE than cancels this out as an annoyance. Starts up fast and
processes fast.
I'm not saying it's better than Besweet. You can customize your output a lot more intricately with Besweet, if you need
to (and I have/do, with the DVD wavs).
Sometimes I find the final dts wav sounds better if I leave the subwoofer out (as in the original method, especially for
live Floyd, anyway), but I assign the wavs as
FL = 1
FR = 3
C = 2
SUB = 4 (if I use it at all)
SL = 5
SR = 6
I'd be interested to hear what others think of this program.
Last edited by Umma on 25th November 2003 at 07:18

Eye of Horus
Senior Member
Posts: 255
quote:
Thanks for the reply, David. I can't WAIT for that offline rendering mode!
Yup. EoH already got the guide written up...
You know, guys, I've been using Virtual Microphone almost exclusively after processing
through Bidule. The way I've been doing it is to just record the 4 channels that come out of
bfprocedit as a wxyz file, then run it through VVM with the master gain at 3.1, using the 5.1
option (and the link boxes checked). VVM is FAST! Spits out 6 mono files in no time at all. Gotta
shut it down after about 4 big conversions and start it right back up, though. I guess it's not
releasing the memory, but the speed of the program MORE than cancels this out as an
annoyance. Starts up fast and processes fast.
I'm not saying it's better than Besweet. You can customize your output a lot more intricately
with Besweet, if you need to (and I have/do, with the DVD wavs).
Sometimes I find the final dts wav sounds better if I leave the subwoofer out (as in the original
method, especially for live Floyd, anyway), but I assign the wavs as
FL = 1
FR = 3
C = 2
SUB = 4 (if I use it at all)
SL = 5
SR = 6
I'd be interested to hear what others think of this program.
Tried it a long time ago...... Don't know if that has changed, but the disadvantage was that it only worked with 16 bit
files and thus you could hear the results were less bright than the original sound files........ And in this program it
certainly was a hearable difference !
EoH
__________________

Shayne
Senior Member
Location: northern canada
Posts: 193
Hi thanks this is great i need my cd's back past Mp3
i new i should keep soft encode in the tickle truck when they trashed it.
I am trying to make 32 bit 48 hz ac3 files and the following is my procedure:
1) Rip cd 44.1
2) convert to 32 n 48 in ce pro
3) pin thru Your great program with 32 out
4) split with the command
D:\Video\AAC_Encode\BeSweet.exe -core( -input "D:\Surround\Temp\Temp6Ch.wav" -output "D:\Surround\Temp
\temp_" -type wav -6chfloat ) -ota( -g peak )
5) load into soft enode and all appear to play fine.
I am getting a real loud aaaaaaaaaaaa at the beginning of the recording when spdif out
It appears that some of (not all)the split five wave files have a clipped click almost like a record click at the first .2
seconds and if this is edited out all works fine. But this is a lot of work.
What do you think
Thank you very much for the guide and have pretty well followed it to the letter. One thing that did work for me was to
add gain filters to pin 3 and 4 out and reduce the gain of the backs by a couple of db for my setup

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by Shayne
Hi thanks this is great i need my cd's back past Mp3
i new i should keep soft encode in the tickle truck when they trashed it.
I am trying to make 32 bit 48 hz ac3 files and the following is my procedure:
1) Rip cd 44.1
2) convert to 32 n 48 in ce pro
3) pin thru Your great program with 32 out
4) split with the command
D:\Video\AAC_Encode\BeSweet.exe -core( -input "D:\Surround\Temp\Temp6Ch.wav" -output
"D:\Surround\Temp\temp_" -type wav -6chfloat ) -ota( -g peak )
5) load into soft enode and all appear to play fine.
I am getting a real loud aaaaaaaaaaaa at the beginning of the recording when spdif out
It appears that some of (not all)the split five wave files have a clipped click almost like a record
click at the first .2 seconds and if this is edited out all works fine. But this is a lot of work.
What do you think
Thank you very much for the guide and have pretty well followed it to the letter. One thing that
did work for me was to add gain filters to pin 3 and 4 out and reduce the gain of the backs by a
couple of db for my setup
Do a search again here on Plogue and spike. It seems that some have this problem and some don't. I have it only the
first time I convert. So I only convert a few seconds and then start again and the spike is gone. Others have no one at
all...........
Those things are so nice to do experiments with !
grtz,
EoH
__________________

Shayne
Senior Member
Posts: 193
I think we have read all pages here but thanks for the insight what we are looking for is a cure / workaround

Eye of Horus
Senior Member
Posts: 255
quote:
I think we have read all pages here but thanks for the insight what we are looking for is a
cure / workaround
If there was a cure, I would give it. I gave the workaround in my last message ! (at least the one working for me !)
EoH
__________________

Shayne
Senior Member
Posts: 193
Thank you for your quick response
your previous post is not a work around here but maybe i am doing something wrong?
I have try recording one and then the next, the next also has this click.
I have tried recording a few seconds of one and then record from start, no good
I have tried recording a few seconds and then rerecording from then on
I have dragged it past start next and all combinations of the above and still a click after besweet demux.
It appears to be besweet and i know you split this thread but it is definitely a bug with your guide that uses besweet. If
i bring the 6ch into soft encoder all is well and i have added gains to all channels and the results are good.
Does anyone know an alternate method of normalizing? Virtual Mic?
I was wondering if someone can reproduce this with 32 bit 48000 Hz files it appears to me that the statement some
have this click and some don't is hard to believe since it is a software bug and not a hardware bug i am using win xp
and besweet stable and/or latest. I can send the wave form which starts at the being and lasts approx 0.002 seconds.
Thanks for your help and this is truly a great step in audio here.
Peace

Eye of Horus
Senior Member
Posts: 255
quote:
<snip>
I was wondering if someone can reproduce this with 32 bit 48000 Hz files it appears to me that
the statement some have this click and some don't is hard to believe since it is a software bug
and not a hardware bug i am using win xp and besweet stable and/or latest. I can send the
wave form which starts at the being and lasts approx 0.002 seconds.
Thanks for your help and this is truly a great step in audio here.
Peace
Hard to believe, but true
My "workaround" never fails here. Win 2000 ...
Others have no spike at all..... (Is in this or the "other part" of the thread !)
The spike has nothing to do with Besweet, but with Plogue. (I guess you need to do some re-reading !! )
And there (In Plogue) is where I start a few seconds, stop and restart.....
EoH
__________________

Shayne
Senior Member
Posts: 193
quote:
The spike has nothing to do with Besweet, but with Plogue. (I guess you need to do some re-
reading !! )
EoH
If this is so how come when i open a 6 ch wav (one file) produced in Plogue into soft encoder this spike is not present?
However the spike does exist when i demux with besweet.
-start(..) = click moves to x seconds
most likely something i am doing that is not right but i really can not find it, where should we re-read?
Thanks for the help
Peace
Ps edit typo

Eye of Horus
Senior Member
Posts: 255
quote:
If this is so how come when i open a 6 ch wav (one file) produced in Plogue into soft encoder
this spike is not present? However the spike does exist when i demux with besweet.
-start(..) = click moves to x seconds
most likely something i am doing that is not right but i really can not find it, where should we
re-read?
Thanks for the help
Peace
Ps edit typo
With 20 messages a page :
page 2 (first message by Daphy about the problem....) and page 5 with some comments by Kempfand.
grtz,
EoH
__________________
1st December 2003 01:44

Shayne
Senior Member
Posts: 193
@ Daphy
Dont thing so try open 6ch in soft encoder as is .... no problem
It happens at the split stage ---- Besweet i think? Redue still click but i will retry more than 2x's
@ kempfand
This method doesn't seem to work here.
To solve this problem i think the program doing it needs to be determine and a fix/alt determined. I have to say its not
Plogue if i open 1 file 6ch.wav there is no click but it occurs at the besweet stage. (32 bit 48 hz)
Edit:
did most my tests off Jethro Tull but full album eagles coming up.
For sure work around.
add 3 gains to front(2) , rear (2) and center adjust gains to 7 , 4 , 8 respec or whatever suits u. set to - to determine
ur pin outs.
open 6ch.wav in soft ender (by pass besweet)
encode
RnR
Peace
Last edited by Shayne on 1st December 2003 at 06:18

Umma
Member
Location:
Posts: 57
There is a thread at the Bidule site that mentions the click at the beginning, but I'm not sure it applies here.
Might...
Might not...
4th December 2003 00:46

joshbm
Member
Location:
Posts: 67
I have a problem:
I have gotten Bidule to work quite nicely in the past. And just when I need it, I can't get the thing to work!
It gives me blank audio. I record it and it is blank... all 6 channels :-\.
Could someone help me out here?
Specs:
- PIII 600Mhz - Shouldn't be the prob, b/c I have gotten it to work in the past
- Bidule 0.6500test
- Using my group I created (following tutorial to a tee)
- Linked Audio Player (Playing) and Audio Recorder (Recording).
Thanks!
- joshbm
Last edited by joshbm on 9th December 2003 at 21:28

joshbm
Member
Location:
Posts: 67
Re: Re: MPA vst-plugin
quote:
Hum there is a decoder yes
"Read MPEG 1&2 (layer 1,2,3) files (i.e. .mp1, .mp2, .mp3). "
(so rename your mpa to whatever the format, prolly mp2)
Get it here
http://askywhale.free.fr/vst/
Warning, this plugin seems to load the decompressed file in ram!!
Now since Bidule only process uncompressed audio streams (and will always be that way), you
would have to reencode the output to mpa again after the processing.
So imho you should maybe uncompress the mpa -> wav then use the Bidule
file player and recorder.
I tested this before... the only problem is linking the Recorder with this MP3 Plugin (Playing). It will not work. Is there
a work around?
Thanks!
- joshbm

Shayne
Senior Member
Posts: 193
Thanks for the link Umma but this toggle is not the guilt here i don't think.
I should clarify click its a wave form clipping that clips out for approximately 0.00025 seconds and then dies like a
decaying sine wave.
On the side
This is a good look into the future. What we need is all this put together in a MP3 player that joins all these filters and
outputs a play list to a temp.dts file in small chucks and then sends them out spdif. The 3 gain filters can be user
adjusted and voil a player that would definitely out play anything we have now. I am sure its thought of and in the
works.
Peace
Last edited by Shayne on 4th April 2004 at 05:30

joshbm
Member
Location:
Posts: 67
Hello everyone,
I posted before my Bidule Group called "5.1 Surround". I now have a newer version (lol, not sure if anyone really
cares):
Download here
What is new:
-> Removed the HNM lowpass plugin
-> Added another great VST alternative called Bass Manager
-> Saves a lot of CPU Usage (60% on my PIII 600Mhz lol).
-> Added volume parameters for each channel.
This Bass Manager is great and is actually meant for 5.1 surround input. Goto Kelly Industries and download Bass
Manager. Also another VST plugin is the LCRS Matrix Encoder-- have yet to try.
VST PLUGINS:
1. B-pan Ambisonic Encoder
Link : http://www.dmalham.freeserve.co.uk/pc_vst_ambisonic.zip
2. B-Proc Ambisonic B-Format Processor
Link: http://www.york.ac.uk/inst/mustech/...vst/yorkvst.zip
3. Emigrator
Link : http://www.gerzonic.net/zips/Emigrator_1.0_win32.zip
4. Bass Manager
Link: http://www.kellyindustries.com/down...manager_vst.zip
ADD:
You can add the "5.1 Surround" by downloading it and saving it to the "Groups" folder. Then from there open Bidule
and you can drag "5.1 Surround" from your Palette under "Groups".
BUGS:
-> Settings to keep for plugins: bfproedit and Bass Manager. In order to fix these settings you must (at the moment,
may be fixed later) right click on your "5.1 Surround" Group >> Grouping >> Expand. Then double click on "bfproedit"
and the right most dial turn so it is vertical strait up and down like | .
Then from there double click the "Bass Manager" and drag the bar at the bottom to the very right (or critique your
settings the way you want them) and click "100" under "X-Over Frequency" then back to "120". From there you can
get out by right clicking on the background and Grouping >> Parent Group.
From there connect up everything as normal.
I hope these bugs can be fixed in the future, but right now it looks like we have to live with them.
Thanks,
- joshbm

Eye of Horus
Senior Member
Posts: 255
wrong thread ??
Nice post Josh, but don't you think it is more usable in the other thread ?
The thread split up and this part is only about the usage of Besweet.
The other one has all kind of Plogue related subjects in it....
kind regards,
EoH
__________________

Umma
Member
Location:
Posts: 57
The latest Bidule is out!

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
quote:
The latest Bidule is out!
And has an Offline mode so you can render your 5.1 files
faster than realtime.
Watch out for "Edit/Offline Processing"
int the new version available on plogue.com/bidule

Eye of Horus
Senior Member
Posts: 255
quote:
Excellent David !
Can't wait to give it a try......
Thanks !
kind regards,
EoH
__________________

Umma
Member
Location:
Posts: 57
Much faster...MUCH faster!
I had stopped doing the conversions because of the time involved, but now...well, it begins anew!
Thank you very much, David! To you AND your team...

joshbm
Member
Location:
Posts: 67
quote:
david... I downloaded the new Bidule... looks great, but I have had the most problems with it. For instance when I
have a connection and try to remove the connection by selecting the connection and hitting the [DEL] key, it freezes
and I have to do a cold restart by holding the power button of my computer. This has happened consecutively for
about 15 times now. Is anyone else experiencing this?
Thanks,
-joshbm

specise_8472
Newbie
Posts: 101
quote:
david... I downloaded the new Bidule... looks great, but I have had the most problems with it.
For instance when I have a connection and try to remove the connection by selecting the
connection and hitting the [DEL] key, it freezes and I have to do a cold restart by holding the
power button of my computer. This has happened consecutively for about 15 times now. Is
anyone else experiencing this?
Thanks,
-joshbm
I have the same problem, but after testing I find that it only seems to happen after processing a file. Not before.
A workaround that seems to work is to make sure that processing mode is off. (red). Works for me.
A very annoying bug, as I am testing out a new plugin I wrote for MatrixMixing to use in a new method of encoding
5.1. I thought that I had a broken plug
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
I have the same problem, but after testing I find that it only seems to happen after processing
a file. Not before.
A workaround that seems to work is to make sure that processing mode is off. (red). Works for
me.
A very annoying bug, as I am testing out a new plugin I wrote for MatrixMixing to use in a new
method of encoding 5.1. I thought that I had a broken plug
Sounds exciting !
Please keep us posted !
kind regards,
EoH
__________________

joshbm
Member
Location:
Posts: 67
quote:
I have the same problem, but after testing I find that it only seems to happen after processing
a file. Not before.
A workaround that seems to work is to make sure that processing mode is off. (red). Works for
me.
A very annoying bug, as I am testing out a new plugin I wrote for MatrixMixing to use in a new
method of encoding 5.1. I thought that I had a broken plug
Yes, keep us updated on it! Can't wait to see it!
You're right about being out of processing mode-- it does that *less* frequently when I am out of it. Yet it still drives
me crazy, my bidule file I am working with. I am out of processing mode and this will be about the 23rd I think time I
have to do a cold shutdown (holding the button on my comp). I hope it can be fixed in later versions :-).
edit: Make that 25th :-P
Thanks!
- joshbm

specise_8472
Newbie
Posts: 101
quote:
Yes, keep us updated on it! Can't wait to see it!
You're right about being out of processing mode-- it does that *less* frequently when I am out
of it. Yet it still drives me crazy, my bidule file I am working with. I am out of processing mode
and this will be about the 23rd I think time I have to do a cold shutdown (holding the button on
my comp). I hope it can be fixed in later versions :-).
Thanks!
- joshbm
I am using win2000pro, all I have to do is wait for the program not responding box or Three Finger Salute (CTRL ALT
DEL) and manually close it. Never have had to reboot.
__________________

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Author Thread
joshbm
Member
Location:
Posts: 67
quote:
I am using win2000pro, all I have to do is wait for the program
not responding box or Three Finger Salute (CTRL ALT DEL) and
manually close it. Never have had to reboot.
Yeah, it's too bad my WinXP totally freezes up :-\.

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
Hum
Im trying to reproduce this without success.
Ca you guys send me your .bidule file and step by step
instruction on how to freeze it?
Thanx
davidv@plogue.com

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
joshbm, precise,
Could you try this version (just an updated exe)
This changes some things with disk thread priority,
that may improve usage.
http://plogue.com/bidule/6501test.zip

joshbm
Member
Location:
Posts: 67
Okay...
Just look at delthis.jpg in the zip file provided. And open "New.bidule" and select
that and hit [DEL].
It freezes my WinXP up to the point of cold restart.
PS- I think for the most part to you can delete any connection when processing
mode is on and it will freeze. The one above I tried deleting it both in and out of
Processing Mode. The crazy thing is, it worked in 6500test, but doesn't work in this
new version.
(PMed you the link to download it)
Thanks for looking into it,
- joshbm

joshbm
Member
Location:
Posts: 67
Just tried it... No go :-\. Still had to do a cold restart.
Regards,
-joshbm

Eye of Horus
Senior Member
Posts: 255
first experience........
Not that good.....
A lot of crashes (freezes) on Win XP.....
Offline or Online makes no difference here.
When you "load" a bidule, it freezes.
When you clean the space and "import" it does not freeze.
This with the standard 2 to 5.1 setting.
Speed was as expected : 5-6 times normal speed > Excellent !
More experiments soon........
grtz,
EoH
__________________

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
ouch
You were right there is a evil bug in 6501
We've just fixed it, i hope it solves your issues.
You can either decide to redownload the full thing or just the exe
which is here http://plogue.com/bidule/65011.zip
(the new version should read 0.6501.1)
Please inform us of more problems directly on our phorums
as i might not be able to monitor all forums for a while.
Cheers

joshbm
Member
Location:
Posts: 67
quote:
ouch
You were right there is a evil bug in 6501
We've just fixed it, i hope it solves your issues.
You can either decide to redownload the full thing or just the
exe
which is here http://plogue.com/bidule/65011.zip
(the new version should read 0.6501.1)
Please inform us of more problems directly on our phorums
as i might not be able to monitor all forums for a while.
Cheers
Thank you so much! I have been troubled with this (I bet my PC is relieved :-D)...
No more cold shutdowns!
Works great!
Thanks again,
-joshbm

kempfand
Senior Member
Posts: 194
@david: Big Thank You for the new version and making the offline processing
available. This baby rocks !!!
@All: I had some crashes too (but not to the point of cold-reboot). I then started
with a "blank" template and did the set-up again (i.e. placed the Bidules, and re-
did the wireing). So far all works well (touch wood).
Kind regards,
Andreas

puzio
Junior Member
Location:
Posts: 20
Sorry but with the new version the offline process is slower than online.
Last edited by puzio on 15th December 2003 at 00:20

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
quote:
Originally posted by puzio
Sorry but with the new version the offline process is slower
than online.
Hum sorry about that. I was experimenting on ways to make the
app/pc responsive (too a certain extent) even though it was rendering.
Perhaps in a future version ill add a virtualdub-like drop down list where you can
specify the priority of the rendering.
For now you could improve the speed in two ways.
1)set your bidule buffer size to 512 (larger chunks will be streamed at once, so less
graph overhead and less inter-block "idling")
2)If this is still too slow you could - on top of that - use this temporary build:
http://plogue.com/bidule/65011offline.zip
(it has the same fixes the one i did earlier, but it uses the original 6501 offline
speed - allthough i still think the app is not responsive enough, but if you dont
mind that, yes it renders pretty fast
Sorry about all the probs. Offline processing is a really new inside
bidule and hasent been tested much by our usual testing group, which are mostly
realtime audio/musicians
Thans again for the amount of feedback you provide us with.
And please tell me how it goes

joshbm
Member
Location:
Posts: 67
Thanks for the update david!
Bidule has become an asset in terms of 2.0 to 5.1 upmixing and a different
assortment of audio effects and is a great tool! All that is needed to make Plogue
near-perfect is an execution addition.
I think it would be a cool feature if file execution was added. For instance the
dialogue in the Audio Recorder would have an added option that says:
Post-Execute: [ ] (Browse...)
This you can browse for a file or enter a command line string that it will execute
(thinking in terms of .bat, .exe, .vbs, etc). That way after our 6 channel wavs were
completed it could automatically either encode them directly to Dolby Digital 5.1 or
DTS (or handed off to any other program that we might need to process the
material). It would even be cool to have:
Pre-Execute: [ ] (Browse...)
under Audio Player so that it could easily be done to prepare an audio stream
before the actual Bidule operation starts (ie I like to change my file from 16-bit
44100 to a 32-bit float, 48000 before I put it in Bidule). After it was done executing
the .bat file it would somehow set that 32-bit, 48000 file somehow as the input file.
Also maybe a switch like for command-line parameters could be read for instance
in the File... under Audio Recorder, the filename chosen could also be used in the
command line Post-Execute as %bidule_file01% (path and name of wav for
Channel 1 out of 6, of course Create Files for Each New Record would have to be
checked to use this) or %bidule_file% (path and name of wav for all 6 channels).
Just my thoughts...
PS- Thanks again for Bidule 0.6501.1 and the offline fix!
Kindest Regards,
- joshbm

puzio
Junior Member
Location:
Posts: 20
Thank You David.
It works now. But in offline process when signal processing is on ( Audio Player
runs or stoped) I can't run another progs (I mean I have to wait long time). So to
run another progs I have always to turn signal processing off.
THX

Eye of Horus
Senior Member
Posts: 255
??
quote:
Thank You David.
It works now. But in offline process when signal processing is
on ( Audio Player runs or stoped) I can't run another progs (I
mean I have to wait long time). So to run another progs I have
always to turn signal processing off.
THX
I don't quite understand this......
In offline mode the whole process is 5-6 times as fast as online on my PC.
(Pentium 4 2400 Hz). So a whole CD with 72 minutes is done in 12-14 minutes.
Why the heck do you need to start another program then ? For 12-14 minutes ?
Go drink some coffee !
(no pun intended !)
EoH
__________________

puzio
Junior Member
Location:
Posts: 20
Thank You EoH.
I mean when I turn on "offline processing" and "signal process", and it is not
important I run the convertion or not , Win XP is blocked.
THX

davidv@plogue
Junior Member
Location: Montreal
Posts: 20
quote:
Thank You EoH.
I mean when I turn on "offline processing" and "signal
process", and it is not important I run the convertion or not ,
Win XP is blocked.
THX
Thats what i meant by "trying to make it responsive"
I also noted that close to 100% of the whole PC is used when offline+proc are both
on, and while it renders fast, its a bit hard
and slow to respond even to basic stop commands.
On the suggestions on making it easier to render files, and starting batch
programs. Youll have to understand that while we do think its a good idea, this
offline rendering mode in bidule is roughly 5% of our market and we have lots of
other features to support.
Bidule is a realtime experimental music creation studio primarily
We'll think of an easy way to do such a thing that woulndt take too much time to
implement. Suggestions are welcome.

Umma
Member
Location:
Posts: 57
David,
I'm not saying that I'm one of them, but don't be surprised if you get volunteers
from this site to help you out with the batching suggestions, etc.
BTW, I have had no trouble at all on my Win2K box. Bidule hauls *ss!

joshbm
Member
Location:
Posts: 67
quote:
On the suggestions on making it easier to render files, and
starting batch programs. Youll have to understand that while
we do think its a good idea, this offline rendering mode in
bidule is roughly 5% of our market and we have lots of other
features to support.
Bidule is a realtime experimental music creation studio
primarily
We'll think of an easy way to do such a thing that woulndt take
too much time to implement. Suggestions are welcome.
I understand.
Thanks for listening to my suggestion,
- joshbm

daphy
Miles Freak
Location: Germany
Posts: 175
Hi folks,
back from a hard (and I hope last business trip this year) I noticed much happened
on my favourite board!
Thx to David for the new bidule! I am looking forward to do some testings this
week.
joshbm wish list
for me one feature on this list is most attracktive:
optional output with 6 single monofiles.
To integrate a AC3/DTS encoder would be a nice thing in the future, but for now
this output option would save much time!
@joshbm
did you some tests with the new bidule version and your 5.1 solution?
is some rewiring necessary?
something I remarked:
5.0 with DTS on all tested Amp (mostly Yamaha) sounds fine - even the bass. I
also find out that most of the DVDs in DTS are 5.0.
5.0 with AC3 suffers under less bass, so for this case a 5.1 transcoding methode
would bring some exciting improvements in bass.
Thats why I also did some further experiments with 5.1.
CYA Daphy

Shayne
Senior Member
Posts: 193
Shoot what a speed increase. Been playing for a couple of days now flawless and
fast!
Test are greatly increased and still looking at tweaks and my new amp to arrive till
batch mode kicks in.
Thanks again great work here

joshbm
Member
Location:
Posts: 67
@daphy:
Yeah, some rewiring was necessary. I have a really excellent 5.1 solution as well. I
would release it, but there are 2 or 3 more plugins required besides the ones
already listed (mostly for the bass and to make the sound more clear).
If you would like me to post it I can. I added the following:
- Group version does not work for me because it takes up too much CPU usage (on
my comp), so it is a .bidule file.
- Reverb option
- Baxxpander (cool wave expander-- making your sounds more crisp including the
bass)
- X-cita, which defines the treble sound.
If I could make a 32-bit float ac3 (with ac3enc, which I don't know how to do) I
would post a sample.
Regards,
- joshbm

puzio
Junior Member
Location:
Posts: 20
Hello Joshbm.
Can You post the new way to do stereo2dd5.1? How do You get 6channel
32bits wave file?
As EoH says don't use never free ac3enc because it is now very bad encoder.
Forget it.
THX

daphy
Miles Freak
Location: Germany
Posts: 175
@ joshbm
Ive sent you a PM
CYA Daphy

kempfand
Senior Member
Posts: 194
@ joshbm:
I'd apprciate if you share your *.bidule(s).
Kind regards,
Andreas

PeterC
Junior Member
Location:
Posts: 3
I have tried the new Bidule and all the exe updates
posted here, but I have this one rather
annoying problem.
I get 4xnormal speed with my 1,7ghz (running Xp Pro) when I set the
buffer size to 512k, at 256k it is as slow as ever.
But, with the speed increase, the program skips
the first 4 or so seconds (I'm trying to do
a 32bit float, 48khz file of 300mb) of the wav file giving
a result which is missing a few seconds...and this
is a major problem when the audio is
destined to go to DVD ;-)
I could not get past this
by any other way than slowing the thing down to
the good..erhm..bad old real time processing, where
the file is processed to the full length and nothing
gets skipped.

Umma
Member
Location:
Posts: 57
Just an FYI on something I just did...
Took a 32-bit 73 minute - 1.5GB wav and Biduled it with the normal 5.1 setup,
offline processing, into a 16-bit multichannel wav.
Resulting file was 2.2GB.
Out of curiosity, I opened the file in Cool Edit 2.0...took a few minutes, but six
wavs opened, and in single-wav view, each wav was a mono 379MB file. The
naming scheme used by cool edit was AID1 (original file name), then AID1(2), AID1
(3)...and so on to the number 6. The number 4 wav was empty, but each other
wav was mono and just a little different-looking than the last. Used the SAVE AS
option to save the first one as FL, the second as FR, the third as C, the 5th and 6th
as SL and SR, respectively. Bully on the blank one, I leave the LFE empty in
Surcode, anyway.
(I quit using Besweet-to-dts automation for now...Surcode has been crashing when
called by Besweet lately. Don't know why.)
Surcode rejected the FL file, the first file saved (and without a number) telling me
it needed a mono file, so when I looked at it in Explorer it showed up as a 2.2 GB
file whereas the rest were 379 MB files (like they were supposed to be).
That FL-file still showed up as a 379 MB mono file in cool edit when I opened it...
and it split into five other mono wavs just as the original one did before I saved as
speaker assignments. Weird how this file is still considered to be a multi-channel
file, but yet also as a mono, as well.
Don't roll your eyes at me, Scully...might just have to do with the different header-
naming scheme that DSPGuru spoke of.
I ran that "mono" FL file through cool edits "resample to mono" option, saved it
again, and it finally showed up in Explorer as a 379 MB mono file, was accepted by
Surcode, encoded with the others into a dts file, and the heavens screamed with
joy.
But I'm out of blank CD-Rs, can't try it out as a dts CD on the DVD player yet.
Anyway, I like to read every bit of information on everyone's experiences, so I just
thought I'd share.

joshbm
Member
Location:
Posts: 67
@daphy, kempfand, puzio:
I will e-mail or PM you guys tomorrow... I will try to post my bidule file on my
server, but right now I can't (extremely busy over the holidays). I will try to make
time to post it tomorrow (Dec 22).
Thanks guys for your interest!
Regards,
- joshbm
22nd December 2003 03:57

daphy
Miles Freak
Location: Germany
Posts: 175
THX

Eye of Horus
Senior Member
Posts: 255
quote:
@daphy, kempfand, puzio:
I will e-mail or PM you guys tomorrow... I will try to post my
bidule file on my server, but right now I can't (extremely busy
over the holidays). I will try to make time to post it tomorrow
(Dec 22).
Thanks guys for your interest!
Regards,
- joshbm
HEY,
Don't forget me !!
Happy Christmas everyone !!
EoH
__________________
23rd December 2003 17:54

daphy
Miles Freak
Location: Germany
Posts: 175
Ambisonic ftp server installed
Hi folks,
me and @ndy have installed a special ftp server (with upload option) to give
everyone of you the posibility to spend his bidules/groups/infos/own guides ... (no
illegal stuff of course, please!)
have a nice time with it
Merry X-Mas
Ambisonic Server(new link)
CYA Daphy & @ndy
PS: if the server is down, please send me PM

kempfand
Senior Member
Posts: 194
@daphy: Thanks for the initiative.
As a 1st contribution, I'm adding a newly designed bidule which uses the
Panorama-VST. It's currently in the "/Ambisonic/upload"-directory, but you might
want to sort it differently.
Some explanations:
code:
I think it basically achieves the same as
the 'old' one. Maybe it is slightly better, as it
processes internally as 2nd-order-B-format (within
Panorama), whereas B-Pan "only" does
1st-order-B-format processing. I don't yet have
enough samples to really call this a "firm" statement,
so it's just a working assumption currently.
Some notes to the settings (see 2 attached JPG's for
illustration):
(1) Elevation: I put that from 0 to 0.55 (L & R), to
tilt the soundfield slightly upwards by 33 degrees.
This is because my speakers are all very close to the
floor.
(2) Distance: I out that from 3.70 to 5.05 (L & R),
which slighy enlarges the size of the sound-sphere. If
enlarged too much, the VST calculates too much
reverberation for my taste.
(3) Azimuth: For "L", set to +2.20, for "R" to -0.94.
This only applies to when decoded to Pentagon.
For Square-decoding, it would be +1.57 for "L", and
-1.57 for "R".
Why the change (Square --> Pentagon) ?
- "L": +1.57 --> +2.20
- "R": -1.57 --> -0.94
Remember that we turned the soundfield
counter-clock-wise by 36 degrees, inorder to have it
correctly aliged to the 5.0 (or 5.1) speakers in people's
living-rooms.
We have to do the same here, i.e. rotate
counter-clockwise by 36 degrees, which corresponds to
an an increase of +0.63. Meaning:
- "L": +1.57 + 0.63 = +2.20
- "R": -1.57 + 0.63 = -0.94
Comments & suggestions welcome here in the forum.
Happy Xmas to all
Andreas
P.S. joshbm: don't forget us ...

Umma
Member
Location:
Posts: 57
quote:
Some notes to the settings (see 2 attached JPG's for
Um...what two attached jpgs?

kempfand
Senior Member
Posts: 194
The jpg's are within the _Panorama_-_Pentagon_.RAR I uploaded.
Andreas

Shayne
Senior Member
Posts: 193
Just a follow up since i rambled on about this problem for so long in this thread.
It appears that the clicking problem has been fixed in Besweet by DSPguru in the
latest b24. Clipping at the first .002 seconds on 32 bit float waves is gone.
Since there is two versions of the latest beta you need the latest.
http://forum.doom9.org/showthread.p...7052#post417052
or home
Edit:
With regard to the process and the guide. I find the LFE input nice on dts and
therefore have been using it. I remember the warnings about using it in the
beginning on this thread.
What i have found is you need to watch to make sure that the lfe channel is not the
controlling wave in the normalization process. This is the case with the graph
suggested. This is dangerous and really does not normalizes. What i have done is
add gains to the lfe channel between HNM and the recorder (3 would be required).
I have found the default -3.33 db is a good setting but my testing is young (ed : -5
in most conditions?). This ensure that the normalization is carried out on C, F, or
rears and not "pump up the volume".
Peace
Last edited by Shayne on 4th April 2004 at 04:41

kempfand
Senior Member
Posts: 194
Conversion between UHJ and B-format using SIR (Super Impulse reverb)
From Shayne Dec 11:
quote:
What we need is all this put together in a MP3 player that joins all these
filters and outputs a play list to a temp.ac3 file in small chucks and then
sends them out spdif. The 3 gain filters can be user adjusted and voil a
player that would definitely out play anything we have now. I am sure
its thought of and in the works.
I think we are basically there. At least for "wav/mp2-in" --> "6-channel-out", it
works beautifully here with Bidule and SIR (Super Impulse reverb).
A few (selected) details on the SIR-VST:
code:
- This is a reverb device that works with "impulse-responses".
Impulse-responses are generated by recording short, broadband
signals and their corresponding room-reflections, and by cutting
out the initial signal. What is left contains the information
needed to reproduce the same room with other signals.
- SIR is a stereo plugin. To take full advantage of
stereo impulses for mono signals, you must first convert
the track to stereo.
- Features:
- performance optimised
- wave-file import 24-32 bit
Angelo Farino already published an outline Conversion between UHJ and B-format
using SIR for AudioMulch, which is pretty self-explanatory, so I wont' dulpicate here.
I've written a small bidule which does the same as outlined Audiomulch contraption
(I can upload it to Daphy's ftp if there is interest).
In summary, this achieves the same as the initial Ambisonic Guide by EoH ,
but with the new tools in real-time (or "hyper-speed" with the offline-mode).
A few notes:
- make sure you use the "right" impulses with your source (i.e. 44.1kHz 32-bit
pulses on source=stereo_wav with same properties)
- "M2Buss" in AudioMulch translates into "AudioMatrix" for Bidule
- I personally prefer Bidule, but this is my personal taste (think it's more intuitive to
use, more straight-forward)
- Filtering-impulses are also available from A. Farina's Site
- You might want to play with the settings for 'dry' and 'wet'. As a starter, I
recommend 'dry'= 0 dB, and 'wet0= -6 dB.
Cheers,
Andreas

puzio
Junior Member
Location:
Posts: 20
Hello Kempfand.
Thank You for your work with Panorama Vst plugin. Can You upload Bidule with SIR
VST Plugin?
I think "initial Ambisonic Guide by EoH" gives the best DD5.1 output,
How do You think?
THX

kempfand
Senior Member
Posts: 194
Ratings:
- I like Panorama slightly better than B-Pan & B-Proc
- Aurora gave excellent results, but I only used it on 3 CD-conversions, which for
me is not enough to make a statement "it's better" than B-Pan.
- However: Given the speed-advantage of the Bidule-setup, I'd clearly give that
more 'ranking-weight' than the "maybe slightly better" of Aurora.
- SIR (Super Impulse Reverb) -VST: I've just started to play with that and need to
do more tests before I can make a statement.
Andreas

puzio
Junior Member
Location:
Posts: 20
Hello Kempfand.
Thank You for your work.
Do You see that in Farina's guide with Audiomulch Y-signal is connected with 4-
input of B_Decoder , NOT 3-input of B_Decoder.
So We have W,X,Z(silence),Y !!!!! in input of B_Decoder, NOT W,X,Y,Z.
In Bidule as You wrote I did convertion with "Bette Midler - Wind Beneath My
Wings" I get "Front Left" loud, but "Front Right" is very low. With "Panorama" way
they are similar.
?????
THX

kempfand
Senior Member
Posts: 194
Hi puzio,
quote:
Do You see that in Farina's guide with Audiomulch Y-signal is
connected with 4-input of B_Decoder , NOT 3-input of B_Decoder.
So We have W,X,Z(silence),Y !!!!! in input of B_Decoder, NOT W,X,Y,Z.
Good catch. I personally believe it is a mistake on A. Farina's page.
As you correctly note, the reverb is done to create W,X,Y. No need for Z (as there
is no information on Z in the starting stereo-wav).
On the B_decoder, the pin-sequence is W, X, Y, Z (as you correctly note).
According to my understanding, reverbed W should connect to W-pin of B-decoder,
X to X, and Y to Y. This is also how EoH's original guide works.
quote:
I get "Front Left" loud, but "Front Right" is very low. With "Panorama"
way they are similar.
I get perfect reults here (both with SIR & Panorama). I believe that this problem
(and similar ones, such as 'wrongly' rotated sound-fields) are related to how the
channel-mapping & demuxing is done.
Some suggestions:
a) Make sure B-Proc is configured as in the guide
b) Use BeSweet to demux the 6-ch-wav into the mono-wav's. If you use wav2wav6.
exe, you need to either re-name the channels, or use a different channel-mapping.
Good luck,
Andreas

Umma
Member
Location:
Posts: 57
Has anyone every contacted Angelo Farina about us using his guidelines over here?

Thread
Forum Jump:

-- Audio encoding
conversion.

Channel Surround
Author Thread
kempfand
Senior Member
Posts: 194
I'm not sure if I read your question ('his' guidelines ?)
correctly (bear in mind that my native language is not
English).
- If you refer the puzio's question about the wireing of
the reverbed W, X & Y, then the answer is no (it's on
my todo list).
- If you refer about potential 'ownership' of guidelines,
then let me state that Farina's page was just recently
updated, and that the bidule I outline above does
exactly what what already described here in July 2002
( A Guide from Stereo to 5 channel Surround ! .
As I said: My native language is not English, so I
might mis-read.
Andreas

puzio
Junior Member
Location:
Posts: 20
.

Umma
Member
Location:
Posts: 57
quote:
I'm not sure if I read your question
('his' guidelines ?) correctly (bear in
mind that my native language is not
English).
- If you refer the puzio's question
about the wireing of the reverbed W,
X & Y, then the answer is no (it's on
my todo list).
Sorry about the ambiguity. I should have been clearer
as the wiring would have been a topic to touch with
him, that and...
quote:
- If you refer about potential
'ownership' of guidelines, then let me
state that Farina's page was just
recently updated, and that the bidule
I outline above does exactly what
what already described here in July
2002 ( A Guide from Stereo to 5
channel Surround ! .
I know it was with a couple of others (and you, too,
right?) that EoH put together the first stereo to 5.1
guide, but he also referenced Farina's site in the use
of Aurora to create the b-wave. I'm not saying Farina
'owns' the method of conversion, but by providing us
with directions in using Farina's Aurora program (and
with a big nod to Farina's own guide to using Cool Edit
and Aurora), EoH indicated that Farina was a big
inspiration at least in the beginnins of the conversion
process. Of course it is a dynamic thing and it will
constantly be changing as newer stuff comes out, but
I guess my question should have been: Has Farina
ever seen how his suggestions and use of his Aurora
software have inspired people? Or something like
that...
I didn't mean any offense, Andreas. I was basically
wondering if anyone had contacted him about
anything over the last year or so.
Pax,
D

kempfand
Senior Member
Posts: 194
quote:
Has Farina ever seen how his suggestions
and use of his Aurora software have
inspired people?
My guess is no. I'll write him and give the
background. I've been a bit sloppy here and should
have done it earlier. I agree that he should be given
the right credit.
// off-topic on //
Reason I didn't do it yet: I've sent a really nice & kind
email to Dave Malham, saying "thank you" for the
nice tools he and his team provide for free (B-Pan, B-
Proc, etc). Never heard back a single word ...
Needless to say that I was very disappointed (fully
understanding that he might have personal reasons
which make him focus on other things). Similar
experience (on different topics) with Mark Taylor on
tooLame (exe/dll).
// off-topic off //
Peace and kind regards,
Andreas
Edit/Add: Also wrote an email to Christian Knufinke,
the author of the SIR-VST, saying "thank you".
Without the effort from people such as him, all this
wouldn't be possible.
Last edited by kempfand on 28th December 2003 at 23:48

Eye of Horus
Senior Member
Posts: 255
quote:
My guess is no. I'll write him and give
the background. I've been a bit
sloppy here and should have done it
earlier. I agree that he should be
given the right credit.
// off-topic on //
Reason I didn't do it yet: I've sent a
really nice & kind email to Dave
Malham, saying "thank you" for the
nice tools he and his team provide for
free (B-Pan, B-Proc, etc). Never
heard back a single word ... Needless
to say that I was very disappointed
(fully understanding that he might
have personal reasons which make
him focus on other things). Similar
experience (on different topics) with
Mark Taylor on tooLame (exe/dll).
// off-topic off //
Peace and kind regards,
Andreas
Edit/Add: Also wrote an email to
Christian Knufinke, the author of the
SIR-VST, saying "thank you". Without
the effort from people such as him,
all this wouldn't be possible.
I mailed Farina for at least 3 times and never got a
response back !
grtz,
EoH
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
I didn't mean any offense, Andreas. I
was basically wondering if anyone
had contacted him about anything
over the last year or so.
Pax,
D
I try to contact him long before I wrote my FIRST
guide with the Aurora plugins. Because he didn't
answer any of my emails, I decided to go on without
him and luckily I found someone who helped me in
the process.
The second guide was written without any of his
software.
This year I emailed him for the third time and again
got no response.
I gave up :-)
To me it's not quite clear where the problem is......
Guidelines like Farina's are on so many sides, that it's
still the question if these were his' in the first place ! I
mean.... he didn't invent Ambisonics ! And everyone
can write guidelines about how to use tools. (even I !!)
Don't get me wrong : the guy is a genius, but I don't
think it's necessary to "ask him permission" to write
guidelines !
I would have loved however if any of these guys came
on this board and supplied us with more tips and help.
But..... after emailing (Andreas did some emailing
too !) and no replies, we gave up hope...
What rests is that we have to find it all out by
ourselves :-))
CanI add a nasty one ? If not allowed the moderator
can remove it : I am not surprised that when people
are selling stuff, but don't reply to emails, the
potential buyers will look for illegal versions. If I ask if
a routine can be explained, before buying and I get no
answer, I won't buy it ! But I will try to find it out by
myself with a not so legal version ! Seeing the
enormous response my 2 guides have on this board,
the makers of these software should think again.
That's why I will give 10M kudos to David from
Plogue ! When he start selling, I want to be one of the
first to buy his program !! And...... don't forget it
works two ways. We act as his beta-testers :-)
my 2 eurocents !
EoH
__________________

kempfand
Senior Member
Posts: 194
Comments from Angelo Farina
Great news: I've just got mail from Angelo Farina ,
which I share with his permission.
quote:
Subject: Re: UHJ t B : VST-method
using SIR & Bidule
Hi,
I have seen Your postings, and it was
evident that in my original AudioMulch
patch there were severe errors (patching
errors and improper gain settings
in SIR). So I corrected everything (I
hope)...
I also tested the system with Bidule, and
added a corresponding image
on my web page:
Also usage of Audiomulch/Bidule and SIR
(Super Impulse reverb) are explained
I also saved the Bidule patch on my Public
server area: Link
Please, check if everything now is OK....
Regarding the concern about the usage of
UHJ-to-Surround conversion for
not-UHJ stereo recordings, I can explain
what follows:
1) Most coincident or near-coincident (i.e.
ORTF) microphonic techniques
still work reasonably well in a UHJ-
decoder, as "artificial" mixing (as
most pop/junk music is) designed for
Dolby Surround (as almost anything
found on VHS tapes or stereo DVD-
video). Spaced or binaural recordings,
instead, are not optimally treated by UHJ
decoding networks.
2) The decoding impulse responses DO
NOT CONTAIN any room information: the
surround channels are decoded from the
original recording only, and contain
information which appear "out of phase"
on the Left-Right channels. This is
very similar to the "Dolby Surround"
approach, but here the result is stereo
instead of mono, and is full-band instead
of band-limited. This outperforms
original Dolby Surround, Dolby ProLogic
and similar 2-to-5 channels decoders
(Circle Surround, SRS, etc.). The only
serious competitor which can rival or
even sound better than UHJ decoding is
actually Dolby Pro Logic II
(available only in very recent surround
processors).
3) For very dry original recordings there
is no "surround" to be extracted
at all, and none of the above-mentioned
methods (nor UHJ) can work. The
better solution, in these cases, is to
employ convolution with room reverb.
I suggest, in these cases, to employ the
Ambiophonics decoding method
(www.ambiophonics.org). Similar results,
but less mathematically rigorous,
can be obtained with some Lexicon
hardware tools (such as the 960L).
You can, of course, post this info also on
the doom9 forum....
Bye!
Angelo Farina
The mentioned Bidule is correct. Based on my and
EoH's listening tests, which compared different
decoding-layouts (Square, ITU 5.1 loudspeaker,
Pentagon) and different VST-decoders (B-Dec,
Emigrator, VVMic), we prefer the Pentagon-layout, but
feel free to make your own tests.
Kind regards,
Andreas

Umma
Member
Location:
Posts: 57
Wow. What a response. At least we know he read
these posts. Strange timing on my part to ask about
contacting him...
I did some comparisons with 5.1 and Pentagon, and
Pentagon wins out, too. Not just with me, but to
about three others who listened to some Simon and
Garfunkel that I did. Pentagon won.
I really don't know what is meant by "dry"
recordings, though.
And it is still not clear why the ambiophonics method
should be used. I read through some of the stuff on
the web site, but then I think "ambiophonics," I think
about what Jonas was saying about using a room
impulse created in the listening environment...I don't
remember seeing anything about that at the
ambiophonics web site.

specise_8472
Newbie
Posts: 101
Have just uploaded to the FTP site a VST plugin to do
ambiophonics.
At the moment just does the six mentioned encodings.
On a lot of recordings there is a big difference from
ambiosonics decoding. One I tried it out on was The
Music Of The X_Files. Track 6 contains chiming bells.
The reverb through the rears was spectacular.
Sounded good in Emigrator too, but was not quite as
pronounced, and dropped off sooner.
Ambiophonics was produced as a way of producing
true 5.1 from the information in an WXYZ file. This file
contains the inforamtion needed for decoding
X = Front back
y = Left right
z = Up down (when included)
w = pressure signal (basicaly the input sound at -3db.
Any suggestions improvements etc. Let me know.
On another note, I have just got Roomverb M2 from
www.spinaudio.com
Putting it between the input signal and B-Proc
produces very good results. This is a very good reverb
engine that as heaps of presets and full control over
every setting. Some recordings are really brought to
life. There is a demo avaliable.
__________________

kempfand
Senior Member
Posts: 194
specise_8472: Thanks for sharing this. This looks
really
It's funny: Yesterday, I was just reading the
corresponding paper and thought this would be great
to try. I don't habe the necessary coding experience
though, to put this into dll.
I will try it tomorrow :-)
Thanks again and kind regards,
Andreas

kempfand
Senior Member
Posts: 194
Update: Specise's ambiophonics-VST ROCKS ! I
just did a testing series with one song (Armik: Rubia;
guitar music).
Created 9 tracks as follows:
code:
1) B-Pan, B-Proc, Emigrator (Pentagon)
2) Panorama, Emigrator (Pentagon)
3) SIR, Emigrator (Pentagon)
4) Ambio I
5) Ambio I~
6) Ambio J
7) Ambio J~
8) Ambio K
9) Ambio K~
All were batch-BeSweet'ed to dts with normalization
(@DSPGuru: BeSweet working in batch-mode like a
charm).
Based on this one-song-only-test, B-Pan is very good,
but for my ears, "Ambio J" wins (more discrete
impression, nicer & broader sound-image). In general,
I preferred the I , J, K -type over the "~"-type, but this
is my personal impression and based on 1 initial song
only.
In summary @ All: Highly recommend you give it a
try.
If you want more technical background, read the
corresponding papers from Robin Miller , especially
- Transforming Ambiophonic + Ambisonic 3D Surround
Sound to & from ITU 5.1/6.1
- Scalable Tri-play Recording for Stereo, ITU 5.1/6.1
2D, and Periphonic 3D (with Height) Compatible
Surround Sound Reproduction
@specise_8472:
quote:
Any suggestions improvements etc. Let me
know.
I'm just thinking on howto achieve a workable &
affordable solution to include the SC (surround Center)
speaker. My amp (Yamaha 630) has a speaker-
connection for SC.
If I understand this correctly, SC with dts is matrixed
into the SL&SR. But this is currently probably a 'dead-
end' as I'd guess that no affordable encoders are
around.
Any chance or idea on howto 'achieve' this packaging
of SC (into SL&SR) already within the VST ? Just
thinking lound ...
Kind regards,
Andreas

specise_8472
Newbie
Posts: 101
quote:
Update: Specise's ambiophonics-VST
ROCKS ! I just did a testing series
with one song (Armik: Rubia; guitar
music).
I'm just thinking on howto achieve a
workable & affordable solution to
include the SC (surround Center)
speaker. My amp (Yamaha 630) has a
speaker-connection for SC.
If I understand this correctly, SC with
dts is matrixed into the SL&SR. But
this is currently probably a 'dead-
end' as I'd guess that no affordable
encoders are around.
Any chance or idea on howto
'achieve' this packaging of SC (into
SL&SR) already within the VST ? Just
thinking lound ...
Kind regards,
Andreas
As is mentioned in the papers, 6.1 AC3 is matrixed
center surround, and so does not work properly. At
the moment I have done as suggested and mixed the
SC channel into the LS RS at -3db. The only way to do
a proper SC is to use DTS Discrete 6.1 which there is
currenly no software for that I know of. Nuendo and
Surcode only support 5.1.
I can output the SC if wanted, if you have a reason
for it?
I am looking at writing a PLI/II encoder in the future
to use. Also I am looking now into a filewriter that
thakes the output from the Steinberg VST AC3
encoder and puts it to disk as a valid AC3 stream. Just
think from Source to AC3 in one hit.
Also in the papers are mentioned that you can recover
the original WXYZ file from the 5.1 track. Would there
be a use for this facility to turn original AC3 tracks
(and DTS) into WXYZ files for processing through
emigrator to more or less speakers? Just need to
reverse the plugin and change the co-efficients.
About the difference between ijk ijk~, the ~ are
basically the same but with the microphone tilted -
30degrees.
Just thinking out loud
__________________
Last edited by specise_8472 on 31st December 2003 at 05:49

Ace_V
Junior Member
Posts: 5
Hi,
I think I've set up every thing correctly but my front
and rear channels sound very similar. There shouldn't
be as much vocals in the rear channels as in the front,
right?
Can anyone help me? I must have done something
wrong.
Also just tried it with a mono file which I would have
expected to only come through the center channel or
at the most the front channels, but it has pretty much
the same sound in all channels except the LFE.
thanks
Last edited by Ace_V on 1st January 2004 at 07:25

kempfand
Senior Member
Posts: 194
quote:
The only way to do a proper SC is to use
DTS Discrete 6.1 which there is currenly
no software for that I know of.
I had to do some reading an am still getting 'up to
speed' on the topic. Regarding dts ES, I understand
there are 2 flavours: DTS-ES Discrete , for which I
also understand there is currently no encoder, and
DTS ES matrix. See e.g. technical_literature.
For the "DTS ES matrix", I understand all that is
needed is:
(a) 45 degree phase shift applied to LS and RS signals
(b) CS at -3db
See above source, as well as dts-es.pdf
Update / Edit added below
Now:
quote:
As is mentioned in the papers, 6.1 AC3 is
matrixed center surround, and so does
not work properly
I didn't find that concrete recommendation (can
you provide a link, maybe I was just having bricks in
front of my eyes). If it that's what is stated, we
should test. I'm stating this in analogy to the 5.0-
decoding rigs (Pentagon vs. ITU 5.1 layout), where
'recommendations go for 5.1, and personal tests often
go for Pentagon.
---
quote:
Would there be a use for this facility to
turn original AC3 tracks (and DTS) into
WXYZ files for processing through
emigrator to more or less speakers?
Still thinking about this. I could see a potential
application for 'down-mixing' into 2.0. Would be keen
to see how this compares to other methods (e.g.
azid).
Don't want to 'make you work', but I'd be very keen
and happ to test, if it's done without too much effort
(realising here that nothing is 'easy').
---
Regarding PanAmbio 4.1: Robin Miller mentions this
as
quote:
For a single listener, PanAmbio is superior
to 5.1 in accurate 360 localization,
spatial impression, and envelopment - a
benchmark of excellence
Any idea on howto achieve this witha bidule ?
---
All this is also thinking out loud
Kind regards,
Andreas
P.S. Have done more tests with your Ambi-rc1.dll. It
still 'rocks' and wins !
Edit: I'm assuming that "DTS ES matrix" can be
encoded with current SurCode version, if conditions
(a) & (b) above are fullfilled.
I did a test as follows; BeSweet/azidts-demux a "DTS
ES matrix" encoded audio (had to use "-ota( -fs
44100"), and SurCode-re-encoded. FOrcing "DTS ES
Matrix" on the amp resuited in the same audible
siganl in each speaker (also SC)
Last edited by kempfand on 2nd January 2004 at 03:06
2nd January 2004 02:36

specise_8472
Newbie
Posts: 101
Okay, one at a time
quote:
For the "DTS ES matrix", I understand all
that is needed is:
(a) 45 degree phase shift applied to LS
and RS signals
(b) CS at -3db
True, but Surcode does not allow a true
implementation of this, as in it will encode okay, but
will not set the flags needed in the frame headers to
indicate that the front or/and back left-right pair/s are
matrixed encoded with the appropriate center signal.
Writing a program to go through and do this after
could be done. But I have found that the published
specs and the actual specs are just different enouth to
cause trouble on this point. There are unknown flags
that are being used
quote:
For a single listener, PanAmbio is superior
to 5.1 in accurate 360 localization,
spatial impression, and envelopment - a
benchmark of excellence
I have not found anything majorly useful at the
moment to look a adding this feature (yet ). I am
still testing/playing around. A lot of this centers
around crosstalk cancellation, which I am
investigating.
quote:
P.S. Have done more tests with your
Ambi-rc1.dll. It still 'rocks' and wins !
As Robin Miller points out, Ambisonics suffers from
lack of spaciousness. But a hybrid system of the two
prtoduces the best results. Basically what I am trying
to acheive. Per-Ambio from Ambisonic WXYZ. Also
Ambiophonics really only caters for the single listener
sitting in the "sweet" spot. Whereas Ambisonics has a
large "sweet" spot.
__________________
2nd January 2004 03:59

specise_8472
Newbie
Posts: 101
Have just uploaded Version 1 of ambiophonics
decoder.
Have added 6 new presets, and deleated the 3 that
did not work good.
The new presets are based upon proper Ambisonic
decode equations, the two versions of each are s =
spherical decode and p = controled opposites.
Attention Daphy - just noticed that my old DLL was
gone from the site. Was this you, or are you going to
have the age old problem of people deleating stuff for
the hell of it
__________________
3rd January 2004 02:22

@ndy
Junior Member
Location: Germany
Posts: 12
@specise_8472
quote:
Attention Daphy - just noticed that my old
DLL was gone from the site. Was this you,
or are you going to have the age old
problem of people deleating stuff for the
hell of it
That was Daphy and me
So what I can say is that we have to make some rules
for the FTP- Sever.
If your file was named "Ambi-rc1.dll" you`ll find it in
"\Ambisonic\VST-Plugin\plugins-031231.rar".
The best way to get along is, that all of you upload
the files with your nicknames. Instaead of zipping the
files there should be a categorized folder in which you
can find the uploads as well as an zipped file with all
available files in there.
I don`t want you to make double postings, so the
threads "stereo to 5 channel surround" and "dedicated
ambisonic server online" should be separated.
Feel free to post your uploads of bidules, dlls and
somewhat else in the ambisonic post.
All technical problems should be posted in the server
posting.
The structure of the FTP- Server is quite easy:
you`ll find all your uploads as soon as possible in the
folders named to the subject. I agree that the uploads
should not be zipped to find them a little easier (Note:
I unzipped the file plugins-031231.rar)
If you have other suggestions of organizing the FTP-
Server please let me know
BTW: except of daphy and me has no one the
(official) right to delete anything
Last note for this posting:
please upload your files with your nickname in the file
file like Ambi-rc1_specise_8472.dll".
CU
Andy
Last edited by @ndy on 3rd January 2004 at 03:07

specise_8472
Newbie
Posts: 101
thanks to @ndy I have now uploaded version 1.0 of
the Ambiophonic DLL to the server. In a directory
named (strangely enouth) ambiophonics.
Take note that the server will be down from 5 - 9 Jan.
@kempfand
hope to upload tomorrow (time permitting, but gotta
do things with the partner once in a while ) the
reverse process I was talking about. The easiest way
to use it will be to use Hypercubes HYPERCUBE
decoder, can decode AC3 and DTS directly to 6
channel wave
I will be having the inputs to match its output.
__________________

kempfand
Senior Member
Posts: 194
@ specise_8472 :
Thanks for the Ambi 1.0.dll . I will be happy to test
the reverse process. Don't kill yourself (know that
partner & friends need time too ). A few questions/
comments:
(1) With reference to First and Second Order
Ambisonic Decoding Equations I understand what
'Controlled Opposite" and "Spherical Harmonics"
means. I'm a bit unclear however as to what you
mean with the "O", "P", and "S" -options. Would
appreciate if you clarify a bit.
(2)
quote:
The easiest way to use it will be to use
Hypercubes HYPERCUBE decoder, can
decode AC3 and DTS directly to 6 channel
wave
Preferring BeSweet here, as it well supported and
documented
(3) Regarding DTS ES Matrix and
quote:
Surcode does not allow a true
implementation of this
Agree. That's also why I had to force 44.1kHz in order
to BeSweet-demux a
'44.1_kHz_DTS_ES_Matrix' (using "-ota( -fs 44100" ).
Also saw your previous threat on the topic. Regarding
a proggie to set the correct flags, there was a
previous threat with regard to AC3: Downmix 6.1 EX
to normal 5.1, where member 'S_O' talked about a
small proggie 'ac3ex'. Might be useful here.
Kind regards,
Andreas

specise_8472
Newbie
Posts: 101
O P S are just Octagon Centered, Pantagon and
Surround decode matrix.
The only difference to Emigrator is that Pentagon is
different, in that there is no C channel as such. I re-
computed for center and also the other channels so as
not to need to do a 30 degree rotate to bring
speakers in line.
I noticed that the LR channels in these decodings are
far to loud and overpower the surround/c channels.
Have fixed this and am implimenting a decent
crosstalk cancellation on the LR pair.(hopfully). The
IJK are unchanged.
About Surcode. As I said then, DIE ANOTHER DAY
uses Matrixed SC channel.(Must do as it does not use
the extended channel specs - but all other 6.1DTS I
have do). But the flags that are in the specs are
obviously not the ones used. Have not got around to
investigating this properly yet. At the moment when I
get time, I will check out the Sum/Difference flags to
see if and how they work.
After reading up on DTS ES again, combined with all
the reading I have been doing recently on Dolby
surround, I have found a new path to follow regarding
this. Thanks for pointing me back down this path.
Will follow up next week.
__________________
4th January 2004 01:38

Umma
Member
Location:
Posts: 57
quote:
I noticed that the LR channels in
these decodings are far to loud and
overpower the surround/c channels.
Have fixed this and am implimenting
a decent crosstalk cancellation on the
LR pair.(hopfully). The IJK are
unchanged.
I did a "K" encode of live concert and what bothered
me was the rear surrounds seemed too loud. I split
the Bidule wav with Besweet into 6 mono files and
opened up the files in CEP. The SL wav was HUGE,
going to the top and bottom of the scale, and the SR
wav was much narrower. I wondered about this, but
didn't want to say anything (I am not proficient in
audio geekspeak ). I thought maybe there was a
reason for it (though I couldn't see it).
Kempfan is right: It rocks, dude. Thank you very
much! I like the K encode. We really didn't hear a
difference between the others you added.
And I really appreciate it, man!

specise_8472
Newbie
Posts: 101
You are right. I introduced a big bug when adding the
Ambisonics decodes into the mix. For now please keep
using the old RC1, which are correct for what you
want - IJK. I will add the SC channel into the
Ambisonic decodes as well.
As you may have read,I have found how to do DTS6.1
and will be adding it into the mix next release.
Actually, looking at my old sourcecode to RC1 I have
actually done this mix to create 6.1.
It should work. If not, try putting the streams directly
into surcode without any post-processing with
besweet. Just split them out unchanged. I have a
feeling that things like Besweets Normalization etc
may change the signal enouth to confuse the decoder?
As I don't use Besweet, I would like someone to check
this out. I have succeded in just resampling from 44.1
to 48.0 to make DVD spec audio okay. But the huge
mind-blowing surprise was finding that you can get
6.1 on 44.1 cd audio
(Now I suppose everyone who wants will have to redo
all their hard work - assuming anyone wants 6.1
audio cd's
Btw I am using a Marantz 7300ose amp for testing.
Works perfectly everytime.
__________________

@ndy
Junior Member
Location: Germany
Posts: 12
quote:
[B]You are right. I introduced a big bug
when adding the Ambisonics decodes into
the mix. For now please keep using the
old RC1, which are correct for what you
want - IJK. I will add the SC channel into
the Ambisonic decodes as well.
I searched for the old RC1.
It is now available again in theses folders on our FTP-
Server : \ambiophonics and \incomming
\ambiophonics.
Have fun
@ndy
__________________
If you`re not a part of the solution, you`re a part of
the problem.

kempfand
Senior Member
Posts: 194
quote:
But the huge mind-blowing surprise was
finding that you can get 6.1 on 44.1 cd
audio
Same experience & surprise here (also played with
"007: Die Another Day").
There also is a 6.1 dtswav at SR 5.1 MultiKanal which
can be used for testing. On that page, scroll down to
"Saga". It says "Coded in DTS-ES, 6.1 Discrete. (All in
Swedish)", but my feeling is that it is 'matrixed SC
(based on demux and SurCode-reencode).
quote:
I have a feeling that things like Besweets
Normalization etc may change the signal
enouth to confuse the decoder?
I always use BeSweet's PreGain Normalization
to 100% ( "-ota( -g max )" ), and for me, it
worked very good with I, J, & K (RC-1 dll-
versions). I realize that DSPGuru recommends using
the exact "PreGain Extract Peak chunk" option ( "-ota
( -g peak )" ), but I had some problems not doing it
"correctly" (which I attributed to the work-in-progress
on how Bidule's filewriter actually writes the data).
Regards,
Andreas
Last edited by kempfand on 4th January 2004 at 15:36

Umma
Member
Location:
Posts: 57
quote:
(Now I suppose everyone who wants
will have to redo all their hard work -
assuming anyone wants 6.1 audio
cd's)
SOME of us po' peoples are still stuck in the dark
ages and are financially limited to their once-
expensive Yamaha 5240 that has ONLY 5.1...
I'm at work now, and will be for a while, but I have a
question. If I add a gain control to the RC1 bidule on
the last two going into the sound recorder...the two
rear surrounds (so I can lower the rear gains -3dB or
so)...AND use Besweet's -ota(-g max), will the
resulting mono SR and SL keep their lower gain after
splitting the multichannel wav? Or should I stick with -
g peak? I only want to do it for some of the live
concerts, depending on the clarity of the recording,
and this new bidule really adds some nice, startling
dimensions to the sound.
Thanks,
D

specise_8472
Newbie
Posts: 101
Have posted V1.5 and bidule to match.
Major change is removed Ambisonic decodes as feel
that emigartor does a good job. It is possible for the
author of emigratr to add the SC channel into rigs
also.
Have added the z input to the mix as I am now
testing the ability to add SC to existing 5.1 original
mixes by using the reverse plugin (which computes a
z channel)and either redoing whole file this way or
just using the reversed wxyz to only compute the SC
channel and adding it back to the original file.
I have a lot of testing to do yet on different
methods
On adding gains to outputs - play - and let your ears
be the judge. What it sounds like to you is based upon
your system (speakers, amp, dvd/cd, cable etc etc
etc). Play until you get what sounds good to you.
Attn moderators - is it possible to thread split this
disscussion on ambiophonics and my plugins out as I
feel it is a different way as opposed to EOH method?
__________________

Umma
Member
Location:
Posts: 57
quote:
On adding gains to outputs - play - and
let your ears be the judge. What it sounds
like to you is based upon your system
(speakers, amp, dvd/cd, cable etc etc
etc). Play until you get what sounds good
to you.
That wasn't what I was asking, for that is what I do.
My question was more about Besweet after Bidule,
and whether or not Besweet will neutralize any gain
changes I make in Bidule. As far as I understand it,
both -g max and -g peak both normalize the wavs
some kind of way...right? I couldn't find the answer
using the search.
Did you fix the bug in the new release you posted this
morning, or should I still use the RC1 release? I ought
to be home in about three more hours...
Pax,
D

kempfand
Senior Member
Posts: 194
@ All: If you want to get the latest Swiss Center /
York Ambisonic VST's, go here: Please note - this is a
temporary download page . Please reas the info,
especially the "You must inform users of its source
and acknowledge its use" (which is only fair, as these
people really deserve according credit).
@ specise_8472: I like the logo . On a more
serious basis, does v1.5 decode as 'controlled
opposites' (which is what I preper) ? If not, could you
add an option ? Not trying to 'ask' here, just want to
help to improve, test etc.
Kind regards,
Andreas

specise_8472
Newbie
Posts: 101
quote:
On a more serious basis, does v1.5
decode as 'controlled opposites' (which is
what I preper) ? If not, could you add an
option ? Not trying to 'ask' here, just
want to help to improve, test etc.
Controlled opposites only works on Ambisonic
decodes. Ambiophonics uses a different method. I
removed the ambisonic part out of 1.5. I feel that
emigrator does a good job. I might consider doing a
different plugin based on Ambisonic alone, with the
DTS mixing added in. The SC channel can be added
into the set Rigs easy enouth. I am always looking for
ways to improve on my plugin. And I removed the
ambisonics also so as to concentrate on other things I
want in it.
__________________

Eye of Horus
Senior Member
Posts: 255
And another one.........
Hi all,
I am pretty much behind on the latest developments,
due to my illness.
But I have a lot of contact with Kempfand and
yesterday we did some tests to compare the Specise
Ambiophonics with a new one we found online.
These tests are posted on alt.binaries.sounds.dts.
Download hot.wav from me and the repost hcDTS.
wav from Kempfand and compare. Let's see if you all
come to the same conclusion as I did....
Now you want to know about this new method, huh ?
It's a VST plugin you can use in Plogue.
The name is CS II VST and you van download a 7
days trial at http://www.srslabs.com.
Don't look at the price :-) Quality comes with a high
pricetag !!
Here are the details to make a bidule (the FTP site is
in Egypt now..... 0.0 K/s connection :-))
1. Install the plugin
2. Use the decoder
3. Use a stereo file player and a 6 channel recorder
4. The settings of the plugin :
- circle surround : on
- rear center : off (we keep it 5.1 because of the 6
channel recorder !)
- center : on
- post/process : ON (if you want to use True Bass and
Dialog clarity)
- True bass LFE : on (5.1) with a setting of 5.0 (this
has nothing to do with number of the channels !)
- True bass front : off (other way round for 5.0 !)
- Dialog clarity : on setting : 5.0
5. The pin connections from CS to the recorder :
(first=CS second=recorder)
1-1
2-2
3-3
4 not used
5-5
6-6
7-4
6. Demux with Besweet
To me this sounds as one of the best stereo to
surround conversions I ever heard !!
I'm anxious to hear your opinions.........
kind regards,
EoH
__________________
Last edited by Eye of Horus on 6th January 2004 at 13:13

specise_8472
Newbie
Posts: 101
I had a look at this a short time ago. As you say good.
I use the Windows Media player version all the time.
Does a good job.
If you want, I can make a vst module to mix the
outcoming signal into a valid DTS neo6 signal?
I already basically have this as I wrote it to test out
my DTS theory.
SC in and mixed with LS RS for output.
Just tell me if you want a particular order of input to
output. I can make a simple 3in 2 out if you want.
BTW nice to see you back on deck EOH.
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
I had a look at this a short time ago.
As you say good.
I use the Windows Media player
version all the time. Does a good job.
If you want, I can make a vst module
to mix the outcoming signal into a
valid DTS neo6 signal?
I already basically have this as I
wrote it to test out my DTS theory.
SC in and mixed with LS RS for
output.
Just tell me if you want a particular
order of input to output. I can make a
simple 3in 2 out if you want.
BTW nice to see you back on deck
EOH.
It will take at least another 5.5 months to fully
recover, but at least I can concentrate now for more
than 30 minutes :-)
There is a huge difference between the WMP or
Winamp plug, compared to the VST plug !
BTW I only have and use 5.1, but be my guest and
come up with some more good work
grtz,
EoH
__________________

DSP8000
Member
Location:
Posts: 46
.
.
Last edited by DSP8000 on 6th January 2004 at 15:06

Eye of Horus
Senior Member
Posts: 255
layout adjusted in the original CS VST message !
Kempfand showed me the channel mapping can be
done without my extra step.
This reminds me again that I am not completely
healed yet :-)
I edited the message !
EoH
__________________

bitsnbytes
Member
Location:
Posts: 56
Been awhile but hey lifemoves you in different
directions heh.
well I just tried the circle surround thing... I'm not
saying its bad. but my observation finds that it
deminishes the fullness of a song... but yes it does
direct the vocals more to the center channel... my
test subject was limp biscuit behind blue eyes, so
nothing outstanding "musicaly".
I wanna call it a softer take on surround sound.
I did play the reg ambisonic one right after the circle
surround... I think that they proly have there best and
worst uses each... not considering that everyone has
different taste.
I guess it would be the soundstage is larger in reg.... I
would not recommend circle surround to any heavy
music but for softer music proly a nicer touch. and if
you're not doing music with vocals theres no sense in
it.
Glad to see you guys are still hammering away at
this... my 3 version dvd turned out sweet.. as i did 3
versions of a couple albumns 1 was 2 channel DTS
and 1 was 3ch DTS and of course 6 ch dts... took
some time not having offline mode but now I see I
can get it done quikly ;-)
and almost time to upgrade my rotel
__________________

bobn4burton
Junior Member
Location:
Posts: 10
Hi, I've been following this thread as best as I can...
although I will admit that I get a little lost with the
more technical stuff. I have followed the original guide
in the first post and gotten good results.
It seems that there have been a few updates in the
last few pages of this thread. Not understanding all of
them very well, I'm just wondering if I should still be
following the guide as shown in the first post, or are
there updates that I should be considering that yield
"better" results??

kempfand
Senior Member
Posts: 194
If you want safe & quick results, follow the guide on
the 1st page of this threat.
The recent pages are on experimental "work-in-
progress", which you might wanna try if you really are
into it, but it needs a pretty good understanding of
things to allow you understand "what's happening
where".
I'm sure there will be new guides sooner or later,
which describe howto do things. EoH already provided
a quick one on the CS II VST the page before.
For the AmbioPhonic VSTs, the posted files contain a
bidule to get you started. I've gotten some superb
results with this one.
Andreas

specise_8472
Newbie
Posts: 101
I have just uploaded a Bidule to do 6.1 output from
CSII to 6.1 DTS compatable.
I.E. SC mixed into LS and RS.
ftp://daphy.mine.nu/Ambiophonics/circle2ne06.rar
__________________
11th January 2004 04:39

specise_8472
Newbie
Posts: 101
I have just uploaded a bidule to do reverse
engineering of 5.1 sounds.
Read the Post 5.1 to WXYZ and Back to get a better
idea what this is about.
I posted this here so as subscribed users will get
notified
__________________
12th January 2004 11:15

daphy
Miles Freak
Location: Germany
Posts: 175
direct links
quote:
I have just uploaded a bidule to do
reverse engineering of 5.1 sounds.
Read the Post 5.1 to WXYZ and Back to
get a better idea what this is about.
I think with this link its easier to find
@ all
please do not link directly to files on the FTP-Server,
because if we change anything like moving files etc.
this link could be dead!
@specise_8472 & kempfand
this weekend I will tryout all the bidules you and
kempfand have uploaded!!! (promised)
CYA Daphy
Last edited by daphy on 18th January 2004 at 18:24
16th January 2004 12:10

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dannyv
450 movies and counting
Location: dvdland
Posts: 65
where to get this VTS
quote:
Have just uploaded to the FTP site a VST plugin to do ambiophonics.
At the moment just does the six mentioned encodings.
@Specise_8472
I can't seem to get on te FTP site to get this new ambisonic vts plug in. Can you tell me how I can get this vts
I would realy like to try it out.
Thanks,
Dan
__________________
AMD XP 2700+ 333fsb on a giga-byte 7vaxp/512 pc2700 ram 360 gb of hard drive space ati radion 9000 pro/
pioneer dvr-106
Last edited by dannyv on 16th January 2004 at 23:33
16th January 2004 23:31

@ndy
Junior Member
Location: Germany
Posts: 12
Re: where to get this VTS
quote:
Originally posted by dannyv
@Specise_8472
I can't seem to get on te FTP site to get this new ambisonic vts plug in. Can you tell me
how I can get this vts I would realy like to try it out.
Thanks,
Dan
You`ll probably find it here
@ndy
__________________
If you`re not a part of the solution, you`re a part of the problem.
17th January 2004 13:14

kempfand
Senior Member
Posts: 194
It's always good to have the latest updates
- Bidule v0.6502 : Bidule Site
- Swiss & York Ambisonic VST's January 13th, 2004: this is a temporary download page
Cheers,
Andreas
22nd January 2004 22:57

daphy
Miles Freak
Location: Germany
Posts: 175
quote:
- Swiss & York Ambisonic VST's January 13th, 2004
will be uploaded to the ftp ASAP
Here the direct download link for plogue bidule 0.6502 (I see no use in uploading this to the ftp, because
plogue has a much better i-net connection as we have!)
CYA Daphy
Edit:
done
__________________
CYA Daphy
________________________________________
Last edited by daphy on 23rd January 2004 at 14:52
23rd January 2004 09:29

desertrat
Junior Member
Location: Germany
Posts: 8
Hi,
I am new to this forum... been mostly reading since Novemeber.
I've done about 10 - 15 ambisonic mixes since then, about half of them in the DTS CD format and the other
half in AC3 to make a larger compilation on DVD (haven't found a good easy to use DVD authoring program
that will support DTS and is easy for making music compilations - if anyone has any recommendations from
what they've tried... it would be appreciated). Most of these ambisonic mixes that I have done so far are on
instrumental music. (Works exceptionally well with Ambient music... Brian Eno, JM Jarre, etc).
I have also tried the I, J, & K and CS II VST. The CS II VTS mix I can't say I was very impressed with the one
track (bjork - joga) that I did. The Ambisonic mix actually sounded better to my ears. I am assuming that the
CS II mix would need to be encoded in AC3 and not DTS (which is what I just tried) and will be trying that
next to see if the results aren't better.
I was also wondering if anyone knows where there is a explanation of the differences between the I, J & K
options. Having an understanding of what they do might make it easier to choose which option is the right one
for a certain type of music.
Lastly I would like to thank everyone (especially Eye of Horus for his guide) who has been involved in this
discussion for the knowledge that I have acquired here
Time to get back to the CS II VST since I've only got seven days with it
Cheers,
desertrat
30th January 2004 10:08

specise_8472
Newbie
Posts: 101
quote:
Originally posted by desertrat
Hi,
I was also wondering if anyone knows where there is a explanation of the differences
between the I, J & K options. Having an understanding of what they do might make it
easier to choose which option is the right one for a certain type of music.
Cheers,
desertrat
Welcome to the fray
I suppose I had better elaborate on this:
I = The microphone array is placed at source level (L,R), below acoustic shell reflections (C), EG an outdoor
amphitheater event, with audience low and behind (SL,SR) and raked upward (SC)
J = The array is more closely placed before a small ensemble at source level for direct sound and early floor
and sidewall reflections (L,R), higher direct solo and ceiling reflections (C), and hall ambience from back-up
(SL,SR) and back down (SC)
K = The microphone array is in an arena with sports play-action or musical instruments at microphone level (L,
R), and with good high front (C) and back (SC) crowd sounds or ceiling ambience
Obviously you will not get these results perfect, but gives you a good indication of the difference between
them.
The overall feedback that comes through is that J is the best. But let your ears be the judge of that.
__________________
30th January 2004 10:54

desertrat
Junior Member
Location: Germany
Posts: 8
Thanks for the info specise_8472
Also now that I've done the CS II VTS mix with AC3 instead of DTS it sounds much better to my ears. Will do
a bit more with the VST before I make any judgement.
Cheers,
desertrat
31st January 2004 07:41

trooper11
Member
Location:
Posts: 42
Im new to the forum, bu tive bene reading article here for a few months. I have been experimenting with this
guide to get tracks for dd 5.1 files. what i want to do is rip tracks forma cd, convert each track to dd 5.1 and
play those on my pc, since i have surround speakers. I got the output of the 6 wav files, but which program is
best to convert that to a dd 5.1 type of file and i wasnt sure if i could play that in any player, or if only some
players will do that. Thanks.
31st January 2004 18:20

Shayne
Senior Member
Posts: 193
@Trooper11
Quote
"03. GOAL OF THE GUIDE
To make 5 mono Wav files from a stereo source with Ambisonics, that can be feed into a surround encoder
like Surcode CD or Soft Encode."
I think you need to read and search a bit hard. Type sonic or surcode in google. Try them and let us know.
Peace
5th February 2004 03:18

xxx666yyy777
Junior Member
Location:
Posts: 2
Encoding Guide - bfprocedit
Quick question to the Guide itself:
You (EoH?) mention to rotate the sound field by 36 degrees counterclockwise using the bfprocedit VST. Why is
that? I did not encounter any reference in any of the descriptions/papers about Ambisonics/Ambiphonics I
could find. Is it "just" based on listening experience?
Thanks!

kempfand
Senior Member
Posts: 194
The guide describes B-format-decoding to Pentagon.
If you check the picture in Emigrator for the Pentagon-decode, you'll see that it has L-R-speakers "in front",
and SL-SC-SR-speakers "in the back" (all with reference to the WXY-soundimage).
Since most of the listener's living-room configuration is such that L-C-F-speakers are "in front", and SL-SR-
speakers are "in the back", you have to correct for this, i.e. rotate the sound-image as described.
If you don't, you'll hear it.
Cheers,
ndreas

xxx666yyy777
Junior Member
Location:
Posts: 2
quote:
The guide describes B-format-decoding to Pentagon.
If you check the picture in Emigrator for the Pentagon-decode, you'll see that it has L-R-
speakers "in front", and SL-SC-SR-speakers "in the back" (all with reference to the
WXY-soundimage).
Since most of the listener's living-room configuration is such that L-C-F-speakers are
"in front", and SL-SR-speakers are "in the back", you have to correct for this, i.e. rotate
the sound-image as described.
If you don't, you'll hear it.
Cheers,
ndreas
Thanks to the answer...makes sense! One more question however:
Isn't is possible (and maybe more exact) to mirror the soundfield at the X and then at the Y axis and then
assigning speaker 3 (based on the Enigmator picture) as the center, skeaker 2 ad FR, speaker 4 as FL, and so
on...? Just asking out of curiosity...not trying to be a pain in the behind...I just think that mirroring on the X
and Y axis should be faster and more precise than doing some rotation around a "non-standard" (i.e. not pi,
or pi/2, ...) angle...

kempfand
Senior Member
Posts: 194
quote:
Isn't is possible (and maybe more exact) to mirror the soundfield at the X and then at the Y axis
and then assigning speaker 3 (based on the Enigmator picture) as the center, skeaker 2 ad FR,
speaker 4 as FL, and so on...?
Interesting idea I think (but correct me if I'm wrong) that mirroring the X-axis would already do it. Speaker
3 would be C, speaker 2 ad FL, speaker 4 ad FR etc.
quote:
I just think that mirroring on the X and Y axis should be faster and more precise than doing some
rotation around a "non-standard" (i.e. not pi, or pi/2, ...) angle...
It's a very long time since I had my math courses on quantum mechanics. I think in theory you are right. In
practice, it really depends on how things with the VST's are implemented. Also, today's power of the PC is so
good that I think it doesn't really matter (but as I said above, I like your idea).
There is a B-Plane Mirror First Order Ambisonic B-format Mirror Control VST, which you might want to try or
play around with.
Regards,
Andreas

Mug Funky
Posts: 611
has anybody seen this? it this exact guide with a different layout (it links to it as well, so i suppose it's not
complete plagiarism).
i was looking for help on emigrator and found it. i suppose in a way it's allright because it gives more people
awareness of ambisonics, and as the guide is lifted from here it's not spreading mis-information about it, but
there's nothing there to say that Eye of Horus came up with the guide this guy is taking credit for.
[edit]
i just posted in this guy's thread on the quake 3 forums...
__________________
Last edited by Mug Funky on 6th February 2004 at 04:12

daphy
Miles Freak
Location: Germany
Posts: 175

nice idea - shurely makes a big boom on the bass (if correct wired - LFE should be channel 4 in the emigrator,
please compare your settings ->
1 > 3 = C
2 > 1 = FL
3 > 5 = SL
4 > 6 = SR -> your HNM-Filter boosts here
5 > 2 = FR
6 > 4 = LFE)
Please share your bidule on our FTP-Server
CYA Daphy
PS: EOH doesnt mean 'Eye Of Horas'
__________________
CYA Daphy
________________________________________

kempfand
Senior Member
Posts: 194
Daphy: The channel-mapping is correct in that "copied guide",
as that guide goes for ac3 using ac3enc.dll.
quote:
-ac3enc/-6chogg
Channel 1->"FL"
Channel 2->"C"
Channel 3->"FR"
Channel 4->"SL"
Channel 5->"SR"
Channel 6->"LFE"
Regards,
Andreas

Eye of Horus
Senior Member
Posts: 255
quote:
Daphy: The channel-mapping is correct in that "copied guide",
as that guide goes for ac3 using ac3enc.dll.
Regards,
Andreas
AC3Enc.DLL ? Yuck !!!
Eye of HorUs !
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by Mug Funky
has anybody seen this? it this exact guide with a different layout (it links to it as well,
so i suppose it's not complete plagiarism).
i was looking for help on emigrator and found it. i suppose in a way it's allright because
it gives more people awareness of ambisonics, and as the guide is lifted from here it's
not spreading mis-information about it, but there's nothing there to say that Eye of
Horus came up with the guide this guy is taking credit for.
[edit]
i just posted in this guy's thread on the quake 3 forums...
Just read what you wrote on that forum. Thanks !
Indeed it would be nice to receive some credits. Especially when you see where my very first guide lead to !
We've gone a long way from the Cooledit method to the ones we use now and we're still not there
New methods are developed and the writing of a new guide is a thing to do the coming weeks.
Also I would like to point out the great improvement on the Fireverb package. I always liked the Fireverb
method to built your own (clean !) pulses, but didn't like the (only) 16 bits method in their Multivolver. Now
you can use 32 bits and up to 96 Khz (although that's still upsampled 48 !). But using 32 bits 48 Khz
roompulses and convolve to 5.1 now gives excellent results without the dull sound the 16 bit version gave.
Highly recommended , if only for 30 days because of the high pricetag.
http://www.catt.se Where you can download a 30 days trial.
grtz,
EoH
__________________

Eye of Horus
Senior Member
Posts: 255
Illness !
I would like to thank everyone here on the forum who knew about my illness and who shared their best
wishes. It seems like it helped, because last Monday I had my first check after the radiation and the tumor is
completely gone ! Of course I need to come back to hospital the coming months for more checks, but
according to the doctor it looks good and she expects it will stay that way !
Now I need to build up some condition again and finally paying more attention to what's going on in 5.1 land

Thanks again m8es !
Eye of Horus
__________________

bitsnbytes
Member
Location:
Posts: 56
Openly Upfront
quote:
Just read what you wrote on that forum. Thanks !
Indeed it would be nice to receive some credits. Especially when you see where my very
first guide lead to !
EoH
Lo All,
Sorry to hear from you in this manner :-(
But hey now you have looked at my site hah!
Sry but not many have taken advantage of your fine findings for quake3 trick movies. and I havnt had the
time to continue the page.
but just so you know I didnt sneeky steal,I gave intentions
posted when they came about.
I also gave direct link to Eye of Horas guide
but if you feel disrespected I will add any comments to the page you like.
reference low numbered pages for my intentions
and here is page where i announced it http://forum.doom9.org/showthread.p...20&pagenumber=8
peace audio/bitsnbytes quake3 arena/G-Man
home of the filterfactory for quake3 arena mod servers for gamespy arcade
ps i hear surcode bronze will be like 99 bucks? for dvd_audio that would be sweet
__________________
[System] AVP-Rotel RSP-1066; DVD Audio/Video-Rotel RDV-1060; AMP-Rotel RMB-1075; Speakers JBL, Rock
Solid And Polk Audio

Mug Funky
Posts: 611
ah, okay. the direct link there made me think it wasn't "sneaky-steal", so i withheld total flamage on the
quake forum... hope that wasn't too blunt there.
still, that was the only place on the page that mentioned EoH which kinda worried me. that's easily fixed i
imagine
hehe.. to be honest, i totally suck at quake. can't move an inch without being blown-up/shredded/chopped/
telefragged...
__________________

Eye of Horus
Senior Member
Posts: 255
Re: Openly Upfront
quote:
Lo All,
Sorry to hear from you in this manner :-(
But hey now you have looked at my site hah!
Sry but not many have taken advantage of your fine findings for quake3 trick movies.
and I havnt had the time to continue the page.
but just so you know I didnt sneeky steal,I gave intentions
posted when they came about.
I also gave direct link to Eye of Horas guide
but if you feel disrespected I will add any comments to the page you like.
reference low numbered pages for my intentions
and here is page where i announced it http://forum.doom9.org/showthread.
p...20&pagenumber=8
peace audio/bitsnbytes quake3 arena/G-Man
home of the filterfactory for quake3 arena mod servers for gamespy arcade
ps i hear surcode bronze will be like 99 bucks? for dvd_audio that would be sweet
Come on ! I am not offended at all. But the way you quote, it looks that way !!
The more people have advantage from my guide, the better !!!
I am honored that it is on other sites too !!
kind regards,
EoH
__________________

kempfand
Senior Member
Posts: 194
FYI : LCR Upmix Bidules
Just for your information:
I've created a few LCR-Upmix Bidule Groups, which you might find useful: LCR Upmix Bidules
Cheers,
Andreas

fateman
Junior Member
Location:
Posts: 27
i dunno what i'm doing wrong, but i followed the guide to a T and the surround file created from Bidule is blank
should i hear the file as i'm playing (and recording) it?
Last edited by fateman on 14th February 2004 at 23:10

Umma
Member
Location:
Posts: 57
Don't take offense at any questions, cuz sometimes the simplest thing is sometimes overlooked. I should
know...done that many times...
Did you make sure the processing button on the was on and the light was green before you started
processing?
What kind of file are you using for a source?
You should hear something while processing, provided you don't have the offline processing enabled.

fateman
Junior Member
Location:
Posts: 27
Yes, green light on.
I'm using a 16 bit 48000 khz stereo wav.
Offline processing is not enabled and I don't hear anything.

fateman
Junior Member
Location:
Posts: 27
ok, I tried it again and this time did get files that contained the audio, but it didn't seem to do much. the files
sound basically the same and one is blank, assuming that is the LFE.

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by fateman
ok, I tried it again and this time did get files that contained the audio, but it didn't seem
to do much. the files sound basically the same and one is blank, assuming that is the
LFE.
Listening to one of the files, can give indeed the impression nothing changed. But....... load all of your 5 files
into SF and you will see the differences.
Better..... make a DTS from them :-) and listen then !
grtz,
Eye of Horus
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
Don't take offense at any questions, cuz sometimes the simplest thing is sometimes
overlooked. I should know...done that many times...
Did you make sure the processing button on the was on and the light was green before
you started processing?
What kind of file are you using for a source?
You should hear something while processing, provided you don't have the offline
processing enabled.
And....... I always do the process without the out-connection to my soundcard ! I never hear anything.... on
purpose !
After the whole process I burn a CDRW and go downstairs to my HTS and that's the first time I hear what I've
been doing
Nothing better than a big surprise
EoH
__________________

fateman
Junior Member
Location:
Posts: 27
quote:
Listening to one of the files, can give indeed the impression nothing changed. But.......
load all of your 5 files into SF and you will see the differences.
Better..... make a DTS from them :-) and listen then !
grtz,
Eye of Horus
what about the subwoofer? just add the center channel there when encoding or what?

Eye of Horus
Senior Member
Posts: 255
Re: FYI : LCR Upmix Bidules
quote:
Just for your information:
I've created a few LCR-Upmix Bidule Groups, which you might find useful: LCR Upmix
Bidules
Cheers,
Andreas
This also will go into test-mode !!
Within a few days, I will finalize my big comparison.
Accessible for everyone interested !
This is what I will do :
I took 10 pieces of different music and will process it with all available published methods.
(A quick count in my head : 10)
This will give 10*10 soundfiles.... That's too much with full songs, so from every song 2 minutes......
This must lead to a final conclusion which method is the best for a certain kind of music...
These are the titles I chose :
1. The Blue Danube - J. Strauss
A classic waltz with a lot of violins
2. When You Got A Friend - Peter Green
A blues song.
3. Mixed Emotions - The Rolling Stones
Uptempo rock.
4. Tainted Love - The Flying Pickets
An acapella song with clean voices
5. Unomathemba - Ladysmith Black Mambazo
Acapella with dark voices
6. Spente Le Stelle - Emma Shapplin
Opera voice on modern synths music.
7. Them heavy People - Kate Bush
A complete different voice !
8. The Eve Of War - Jeff Wayne
Rock with a lot of effects between left and right speakers
9. Oxygene 4 - Jean Michel Jarre
Synths
10. Hotel California - The Eagles
Softpop.
Of course it's easy to add more and more styles..... house, dance, hiphop, rap, big choirs...etc. etc.
I just took some titles from my catalog
I know it is an ambitious project and it will keep me from the streets for a few days. But I think this is the
only way to go, now that the amount of different methods to get from 2.0 to 5.1 is getting out of hand.....
Remember.... from every song 2 minutes encoded with at least 10 different methods.......
Let's see which one gives the best results.....
I will yell here when it's ready and available for download in alt.binaries.sounds and alt.binaries.sounds.dts
Of course I'm open for suggestions the methods.
If there is any method nobody uses (anymore), I'll appreciate feedback, because that saves me quite some
time !!
I also like to hear feedback on my choices of music...... (but please don't let this thread go into a general "I
would chose another song : xxxxxx for song nr. xx" one !!!!!!!!!!)
grtz,
Eye of Horus
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
Originally posted by fateman
what about the subwoofer? just add the center channel there when encoding or what?
You have 5 files, no LFE ? Just leave it blank !
I chose for 5 mono files and no LFE, because it sounds better to let the receiver (or non dedicated subwoofer)
handle the filtering. Also when people have an amplified LFE, it sometimes gives nasty side-effects (a lot of
unwanted reverb and resonance).
(I have an amplified on my HTS and one between the 2 front-speakers on the PC. It is a big difference and I
prefer the sound in 5.0 ! And if I remember correctly that's also the opinion of the Ambisonics experts .....)
Good luck and have fun !!
grtz,
EoH
__________________

fateman
Junior Member
Location:
Posts: 27
alright, thanks man. i'll be interested to hear your results on that time consuming test you have coming up as
well

Tantulus
Junior Member
Posts: 11
Thanks
I just want to give a big thanks to everyone that has contributed bidules, testing etc. I have learned a lot and
I am enjoying my albums as never before. Incidentally, I tried Kempfand's UJH to B layout and it resulted in
the gain being too low. Is there a way to increase this in the Bidule. Otherwise I liked the result

kempfand
Senior Member
Posts: 194
quote:
I tried Kempfand's UJH to B layout and it resulted in the gain being too low
You can increase the Wet gain in SIR if things sounds still ok. I go up to -3dB sometimes.
Important (for this method):
- For the 3 instances of SIR, put the wet-gain at exactly the same values for all 3 of them (or else the sound-
image is destroyed)
- Turn off the "Auto Gain"
- For the pulses used here, keep the "Dry Gain" off ("-oo dB", i.e. slider at the very bottom)
quote:
Is there a way to increase this in the Bidule ?
Yes, you can add 3 instances of Gain (In Bidule: Palette -> Mixing -> Gain), and put them between Emigrator
and 'Audio File Recorder'. As you have 6 speaker-channels, you need 3 of them, because Gain only has 2 in/
out-pins.
Important: Put the gain-values (Amplitude) to exactly the same value for all 3 of them.
quote:
Otherwise I liked the result
I agree the result is very good. Even better results can be achieved by using David Malham's Third Order VST
Ambisonic Tools . As confirmed by the experts and also my listening tests, every 2nd/3rd-order Ambisonic
panning beats 1st order. If you want, I can post a small bidule for Pentagon-decode.
Cheers,
Andreas

SallyDog
Posts: 50
quote:
If you want, I can post a small bidule for Pentagon-decode.
Cheers,
Andreas
Please post
thanks
SallyDog

kempfand
Senior Member
Posts: 194
3rd Order Bidule
I've uploaded a 3rd order bidule (incl. 2 screenshots) to the FTP (currently in /Incoming/3rd_Order/ ). Some
remarks:
- I have set it up such that it decodes to Pentagon-rig.
- You'll need 2 VST's: BFDecFHH.dll & BFPanFHH.dll. They can be found on with Hochschule fr Musik
Winterthur Zrich VST Ambisonic Tools in the York Plugins Windows Version . Just copy the 2 dll's to your VST-
directory.
- I have added gains for all 5 speakers. IMPORTANT: If you change the gain, change it to the same Amplitude
for all 3 instances of Gain (or else the sound-image will not work).
- Some limited documentation is available with the old version of the VST's: Third Order VST Ambisonic Tools
Good luck and happy testing,
Andreas

daphy
Miles Freak
Location: Germany
Posts: 175
Hi Andreas,
those two VST-plugins are already uploaded on the FTP (since 230104 ) and part of the plugins-complete-
0400219.rar.
I would suggest to move the bidule to the ambisonic section?!?
CYA Daphy
__________________
CYA Daphy
________________________________________

puzio
Junior Member
Location:
Posts: 20
Thank You Kempfand.
Can You tell me what is the channel mapping of the multichannel output? Please.
THX
22nd February 2004 07:47

Eye of Horus
Senior Member
Posts: 255
quote:
Thank You Kempfand.
Can You tell me what is the channel mapping of the multichannel output? Please.
THX
Standard ! FL,FR,C,LFE,RL,RR
grtz,
Eye of Horus
__________________
22nd February 2004 12:18

Pages (11): First ... 7 8 [9] 10 11 Last Thread Next Thread
conversion.
Before you start posting please read the forum rules. By posting to this forum you agree
to abide by the rules.

Doom9's Forum > General > Audio encoding > GUIDE: Stereo to 5-Channel
Surround
Pages (11): First ... 8 9 [10] 11 Last Thread Next Thread
Author Thread
Umma
Member
Location:
Posts: 57
I'm curious...with Minnetonka's Discwelder coming out soon,
cheaply, will their be any application of these conversion
methods to DVD-A? Will it sound any better, equal, whatever?
I am planning on getting a DVD-A player, soon. Best Buy has
one for $149...a Pioneer DVD player capable of all the
surround modes.
(thrown in in retrospect)
I would imagine that we would encode UHJ to B, process it,
then go back from B to a multichannel UHJ file. Then run it
through the DVD-A program (Discwelder). Losslessly. DVD-A
is supposed to be lossless, though it will hold dts and DD, and
some video. Just thinkin' aloud, here...
I find it really interesting that, after the explosion of interest in
conversion to dts and DD over here at this forum that
companies such as Minnetonka are now hawking their wares
to the home-user so the home user can upmix their music.
NICE!
But they gotta come down on their prices some more!
Last edited by Umma on 24th February 2004 at 18:07

Van the man
Junior Member
Location:
Posts: 22
this looks great
Gonna try it out, but the emigrator link doesnt work
And how can I mux all the .wav 's back to one .wav file?
Any ideas?
Van

Umma
Member
Location:
Posts: 57
quote:
Originally posted by Van the man
this looks great
Gonna try it out, but the emigrator link
doesnt work
And how can I mux all the .wav 's back to
one .wav file?
Any ideas?
Van
You mean after Besweet? You need Surcode dts encoder or a
Dolby Digital encoder to make either a dts/wav file or AC3 file.
You mean this Emigrator link?

specise_8472
Newbie
Posts: 101
Have just uploaded to the server under
Ambiophonics/3rd_order two new vst's.
3_B-Pan is my version of the york bpanfhh 3rd order
ambisonic encoder.
3rd_order is version 2 of my ambiosonic decoder that allows
you to decode into either 1,2,3 order. 1 being the usual WXYZ,
2 adding the 2nd order UV and 3 adding 3rd order PQ.
To use the associated bidule, you will need to download my
crosstalk plugin from the server under crosstalk.
As you will notice, the 3-b-pan is linked to the associated SIR
settings. So whatever of my impulse files you use in SIR, just
use the same settings in 3-b-pan. (only front at the moment)
But you can use the 3-b-pan as a standalone, jsut choose one
of the settings to enable decode properly. The outputs are the
same as bpanfhh ie: WXYZUVPQ.
And the inputs on 3rd order ambiophonic are LRWXYZUVPQ.
Outputs are the same. Namely L,R,C,SL,SR with SC muxed
into the two rear surrounds for DTSes use.
EDIT: NO LFE OUTPUT
Edit: If there is enouth interest, I can make a reverse VST that
reverses existing 5.1 into 1st 2nd 3rd order files. The same
way as my 1st order reverse VST.
__________________
Last edited by specise_8472 on 29th February 2004 at 20:19

Umma
Member
Location:
Posts: 57
Thanks. I will try the crosstalk plugin again. I couldn't get it to
work the last time I tried.
I don't understand what you mean by the reverse VST. Just
what are the benefits of a VST such as this? If I take a pre-
existing, non-ambi/ambiophonic dts/DD 5.1 mix, run it
through the reverse VST, it will calculate and assign the WXYZ
calculations to a resulting multichannel file? And those WXY
assignments can then be adjusted differently as the file is re-
encoded back to a 5.1 dts or DD5.1?
I'm trying to relate to this in real-world terms. The majority of
what I've fooled with lately are the live boots. Since the
ambience is already there, using room impulses usually serves
to muddle the lower-midranges and lows. Drives me nuts
sometimes and I lose track of what I've tried and what I
haven't tried. Room impulses work great with soundboard
recordings, though. Maybe the WZYZ settings in a newly
calculated, post-reverse VST file can adjust the ambient
muddle that happens in a 5.1 upmix with live recordings? Am I
making any sense?
I guess an experiment would be to take a file and do it the old
way, i.e. combine the stereo for a mono center, split the left
and right for FL, FR, and SR and SL...make a 5.1 file, reverse
it into a 1,2,or 3 order file, adjust the new WXYZ assignments,
re-encode. Would there be any benefits to doing this? Or
maybe I'm missing a crucial concept, here?

specise_8472
Newbie
Posts: 101
quote:
Thanks. I will try the crosstalk plugin again.
I couldn't get it to work the last time I tried.
I don't understand what you mean by the
reverse VST. Just what are the benefits of a
VST such as this? If I take a pre-existing,
non-ambi/ambiophonic dts/DD 5.1 mix, run
it through the reverse VST, it will calculate
and assign the WXYZ calculations to a
resulting multichannel file? And those WXY
assignments can then be adjusted differently
as the file is re-encoded back to a 5.1 dts or
DD5.1?
I have had good results with this Reverse method. One
of the best was Saving Private Ryan scene 2 where they
storm the beach at Omaha. The original has good
bullets whizzing around etc. After reversing then
converting straight back into 5.1 from the WXYZ (no
playing around), there seemed to be a distinct
difference in the surrounds. If anything the overall
effect should be slightly degraded from the original, but
it still sounded better to me. The reason I wrote it in the
first place is that sycho?? (I think) wanted it.
quote:
I'm trying to relate to this in real-world
terms. The majority of what I've fooled with
lately are the live boots. Since the ambience
is already there, using room impulses usually
serves to muddle the lower-midranges and
lows. Drives me nuts sometimes and I lose
track of what I've tried and what I haven't
tried. Room impulses work great with
soundboard recordings, though. Maybe the
WZYZ settings in a newly calculated, post-
reverse VST file can adjust the ambient
muddle that happens in a 5.1 upmix with live
recordings? Am I making any sense?
My crosstalk plugin uses my own special Impulses to
'spread' the soundfield out to the angles in them. And
so my new 3-b-pan (I also have a straight b-pan)uses
these angles to compute the Azimuth for Ambisonic
encoding.
quote:
I guess an experiment would be to take a file
and do it the old way, i.e. combine the stereo
for a mono center, split the left and right for
FL, FR, and SR and SL...make a 5.1 file,
reverse it into a 1,2,or 3 order file, adjust the
new WXYZ assignments, re-encode. Would
there be any benefits to doing this? Or
maybe I'm missing a crucial concept, here?

This method may have merits. Try it and tell us the results.
__________________

Eye of Horus
Senior Member
Posts: 255
quote:
I'm trying to relate to this in real-world
terms. The majority of what I've fooled with
lately are the live boots. Since the ambience
is already there, using room impulses usually
serves to muddle the lower-midranges and
lows. Drives me nuts sometimes and I lose
track of what I've tried and what I haven't
tried.
Hi Umma,
I found a plugin, only DX, that can improve your live boots in
an incredible way. I understand that a lot of studios use this in
a hardware version, but the plugin does the same.
Try to download it somwhere, but if you can't find it, I will
upload it on usenet in the group where we always are :-)
The name of the plugin : BBE Sonic Maximizer .
I used it on an Alan Parsons Live bootleg with a very dull
sound.
After using the plug...... words fail to describe the
improvement !
Use it on the stereo before converting to 5.1.....
kind regards,
EoH
__________________

Eye of Horus
Senior Member
Posts: 255
Stereo to Surround comparison
quote:
This also will go into test-mode !!
Within a few days, I will finalize my big
comparison.
Accessible for everyone interested !
This is what I will do :
I took 10 pieces of different music and will
process it with all available published
methods.
(A quick count in my head : 10)
This will give 10*10 soundfiles.... That's too
much with full songs, so from every song 2
minutes......
This must lead to a final conclusion which
method is the best for a certain kind of
music...
These are the titles I chose :
1. The Blue Danube - J. Strauss
A classic waltz with a lot of violins
2. When You Got A Friend - Peter Green
A blues song.
3. Mixed Emotions - The Rolling Stones
Uptempo rock.
4. Tainted Love - The Flying Pickets
An acapella song with clean voices
5. Unomathemba - Ladysmith Black Mambazo
Acapella with dark voices
6. Spente Le Stelle - Emma Shapplin
Opera voice on modern synths music.
7. Them heavy People - Kate Bush
A complete different voice !
8. The Eve Of War - Jeff Wayne
Rock with a lot of effects between left and
right speakers
9. Oxygene 4 - Jean Michel Jarre
Synths
10. Hotel California - The Eagles
Softpop.
Of course it's easy to add more and more
styles..... house, dance, hiphop, rap, big
choirs...etc. etc.
I just took some titles from my catalog
I know it is an ambitious project and it will
keep me from the streets for a few days. But
I think this is the only way to go, now that
the amount of different methods to get from
2.0 to 5.1 is getting out of hand.....
Remember.... from every song 2 minutes
encoded with at least 10 different
methods.......
Let's see which one gives the best results.....
I will yell here when it's ready and available
for download in alt.binaries.sounds and alt.
binaries.sounds.dts
Of course I'm open for suggestions the
methods.
If there is any method nobody uses
(anymore), I'll appreciate feedback, because
that saves me quite some time !!
I also like to hear feedback on my choices of
music...... (but please don't let this thread go
into a general "I would chose another song :
xxxxxx for song nr. xx" one !!!!!!!!!!)
grtz,
Eye of Horus
It took more time than I thought
And with new methods every week..... well, I just choose 10
and already the differences are remarkable ! (And this was the
first time ever I wrote a website in PHP, which also took its
time !)
I will start uploading tomorrow to alt.binaries.sounds and to
alt.binaries.sounds.dts .
Monday : CD 1
Contents : the original stereo files, song1 and song2
Tuesday : CD 2
Contents : song3, song4, song5
Wednesday : CD 3
Contents : song6, song7, song8
Thursday : CD 4
Contents : song9 and song10
The website where you can vote for the best sounding version
of a song, will be published after the first CD is posted.
You can only vote once for every song. After you have voted,
you can view the results. To see the used methods you have
to vote on every song and after that you will see which
methods were used.
The combination of the style of music and the used method,
delivers different results. Don't think that after hearing one
song, that method will give the same good results for the 9
other songs !!
I hope you will all have a lot of fun with this test.
kind regards,
Eye of Horus
__________________
Last edited by Eye of Horus on 2nd March 2004 at 14:16

Eye of Horus
Senior Member
Posts: 255
quote:
And the inputs on 3rd order ambiophonic are
LRWXYZUVPQ. Outputs are the same.
Namely L,R,C,LFE,SL,SR with SC muxed into
the two rear surrounds for DTSes use.
File recorder outputs FL,FR,C,LFE,SL,SR.
In your layout from 3th-order to the file recorder, you don't
use pin 6 of the recorder, but do use pin4. When decoding the
6 channel wav to 6 separate channels with besweet, pin 4 is
the LFE.
This is your layout :
1>1
2>2
3>3
4>4
5>5
nothing to 6.
IMHO it should be...
1>1
2>2
3>3
4>5
5>6
nothing to pin4
Kind regards,
EoH
__________________

Umma
Member
Location:
Posts: 57
specise - Thanks, I'll try it all over the next couple of days.
EoH - Come to find out, I have BBE Sonic Maximizer! I
went digging around on my storage HDD and found it.
Installed it just now and experimentation phase will begin
after I clean the house... Anybody got a wife to spare??

specise_8472
Newbie
Posts: 101
EOH you are right about the pinouts.
I have just 'hooked' up the pins to the audio recorder wrong.
I knew this at the time, but didn't chaange them for the
release bidule.
The outputs from the VST are the same as before L,R,C,SL,SR.
NO LFE.
I typed wrong above. There is no LFE output. Just too quick on
the fingers
__________________
Last edited by specise_8472 on 29th February 2004 at 20:20

Eye of Horus
Senior Member
Posts: 255
quote:
specise - Thanks, I'll try it all over the next
couple of days.
EoH - Come to find out, I have BBE Sonic
Maximizer! I went digging around on my
storage HDD and found it. Installed it just
now and experimentation phase will begin
after I clean the house... Anybody got a
wife to spare??
Hey man,
This is 2004 !!! FCOL, a real man cleans his own mess
OTOH : did you look on Ebay ?
grtz,
EoH
__________________

Van the man
Junior Member
Location:
Posts: 22
How can I add the line in besweet? I have the non-gui version.
But when I click on the file it only shuts down again

daphy
Miles Freak
Location: Germany
Posts: 175
@ Van the man
try CommandBar with 'CTRL+M' you get a seperate window
inside of the explorer.
Download located in the 'other apps' section
(BTW needs MS Frameworks)
CYA Daphy
__________________
CYA Daphy
________________________________________
The New Ambisonic Server (needfulthings) can be
found here!,
further info thread; upload folder ; (sometimes down please
send PM in this case)
1st March 2004 10:16

Umma
Member
Location:
Posts: 57
quote:
OTOH : did you look on Ebay ?
Nah, they don't have a "try before you buy" option..."
BTW, specise, no file is being created using the crosstalk
bidule. I don't know what I'm doing wrong. I name a file to be
created, I picked a wav song file, I picked one of your SIR
impulse files to use...and the directory is still lacking the file I
was supposed to create. Processing button is on, so is offline
rendering.
Any ideas?
1st March 2004 17:06

specise_8472
Newbie
Posts: 101
Have you checked that Player playing and Recorder recording
are Linked?
__________________
1st March 2004 19:46

Umma
Member
Location:
Posts: 57
Thanks! That is exactly what it was. I just took it for granted
that it was already linked.
2nd March 2004 05:17

Eye of Horus
Senior Member
Posts: 255
Re: Stereo to Surround comparison
quote:
It took more time than I thought
But......... the first CD is uploading right now to alt.binaries.
sounds and alt.binaries.sounds.dts !
This is the website where you can vote for the in your opinion
best sounded conversion of each song :
POLL
The methods used are (in random order) :
1. Surroundboy's
2. Ambisonics (EoH+Kempfand)
3. Synthedit-Gerzon-SIR (Kempfand)
4. CS II (commercial product)
5. SAD51 (Kpex+EoH+Kempfand)
6. WXYZ26 (Specise-8472)
7. Acid Pro (DrPaulNg)
8. VVMic (commercial product)
9. 3th Order (Specise-8472)
10. HRTF-XTC-Panorama-WXYZ (Kempfand)
I used the adviced settings in every methods, some will have
a dedicated LFE channel and some don't.
Have fun !
kind regards,
Eye of Horus
__________________
Last edited by Eye of Horus on 2nd March 2004 at 14:24
2nd March 2004 14:21

Tantulus
Junior Member
Posts: 11
OK , I give up
quote:
Have just uploaded to the server under
Ambiophonics/3rd_order two new vst's.
I can't seem to find the 3rd order bidule on the server. Is it
given another name or do I have to put it together from the
VST's etc. I'd really like to give it a try.
3rd March 2004 16:16

Tantulus
Junior Member
Posts: 11
Now I'm embarresed
I was up all last night nursing my PC back from near disaster
and I guess I'm not paying attention. I've found the 3rd order
in incoming.
Sorry.
Incidentally, 3rd_order.bidule seems to be temporarily
inacessible.
Last edited by Tantulus on 3rd March 2004 at 17:03
3rd March 2004 16:23

Eye of Horus
Senior Member
Posts: 255
Re: Re: Stereo to Surround comparison
quote:
But......... the first CD is uploading right now
to alt.binaries.sounds and alt.binaries.
sounds.dts !
This is the website where you can vote for
the in your opinion best sounded conversion
of each song :
POLL
The methods used are (in random order) :
1. Surroundboy's
2. Ambisonics (EoH+Kempfand)
3. Synthedit-Gerzon-SIR (Kempfand)
4. CS II (commercial product)
5. SAD51 (Kpex+EoH+Kempfand)
6. WXYZ26 (Specise-8472)
7. Acid Pro (DrPaulNg)
8. VVMic (commercial product)
9. 3th Order (Specise-8472)
10. HRTF-XTC-Panorama-WXYZ (Kempfand)
I used the adviced settings in every
methods, some will have a dedicated LFE
channel and some don't.
Have fun !
kind regards,
Eye of Horus
For those of you who don't have access to newsservers : the 4
CD's are also on EDonkey !
Download this filesharing program at http://www.
edonkey2000.com , install and do a search on "stereo2DTS".
Don't forget to vote !!!
kind regards,
EoH
__________________
4th March 2004 21:07

Tantulus
Junior Member
Posts: 11
Re: Re: Re: Stereo to Surround comparison
quote:
For those of you who don't have access to
newsservers : the 4 CD's are also on
EDonkey !
Download this filesharing program at http://
www.edonkey2000.com , install and do a
search on "stereo2DTS".
Don't forget to vote !!!
kind regards,
EoH
Thanks I tried the news groups but I wasn't able to open the
files.
4th March 2004 22:10

@ndy
Junior Member
Location: Germany
Posts: 12
Re: Now I'm embarresed
quote:
I was up all last night nursing my PC back
from near disaster and I guess I'm not
paying attention. I've found the 3rd order in
incoming.
Sorry.
Incidentally, 3rd_order.bidule seems to be
temporarily inacessible.
Well Im sorry, but my homenet is a little bit messy. I just
tried some things out last days so it is possible that you just
picked up the time the server was not reachable.
But now it should run again. If you still have problems
reaching a file, just send me a PM or leave a note in the
incomming folder of our ftp .
For those who mention that the server is too slow:
Im looking for a better solution (webbased or anything like
that and a faster connection) but keep in mind that this is a
private interest/project.
To speek in specise`s words:
"Don`t blame me, I only work here"
CU
@ndy
__________________
If you`re not a part of the solution, you`re a part of the
problem.
4th March 2004 22:53

Shayne
Senior Member
Posts: 193
e= Ambisonics second method -LFE(EOH, Kempfand)
Could you elaborate to what exactly this method entitles.
Thanks
Peace
9th March 2004 02:58

kempfand
Senior Member
Posts: 194
- "Ambisonics second method" = Page one of this threat
- "-LFE" means no LFE (as in the guide)
The 'Ambisonics 1st method' (not used in the test) was the
one using convolution with the Impulse Responses in CoolEdit
& Aurora PlugIns.
Cheers,
Andreas
9th March 2004 09:14

Shayne
Senior Member
Posts: 193
Thank you for your response.
Personally still favor +LFE we just need to find the perfect
normalization process. Minus 5 to 6 db gain to the lfe inputs
works on almost all but not all. Would be nice to isolate the lfe
channel and normalize independently.
Peace
11th March 2004 12:35

kempfand
Senior Member
Posts: 194
Normalisation should be fine if you use it for the pure
Ambisonic method as outlined on page 1 of this threat.
Careful however with the newer methods around
AmbioPhonics (specie's VST's, as well as all bidules using
Stereo Dipoles). Normalising all channels for these will kill the
sound image. You can instead use indiviual gain's for L-R, C-
SL-SR, and eventually LFE. I mentioned this before for the SIR-
convolution: AutoGain Off !
Andreas
11th March 2004 14:48

Shayne
Senior Member
Posts: 193
quote:
Normalisation should be fine if you use it for
the pure Ambisonic method as outlined on
page 1 of this threat.
Andreas
What do you mean fine? I prefer +LFE and i have been using
page 1. I find through the HMN_filter at cut freq of 80 hz and
in/out at 1/2 that bidule will many times (if no gains
introduced) output this lfe channel as the maximum volume
wav therefore it becomes the governing channel in
normalization. We need to isolate this channel and normalize it
to 95% independently of the other 5 channel normalization
process.
Peace
12th March 2004 00:38

daphy
Miles Freak
Location: Germany
Posts: 175
New Server, more performance
Hi folks,
to get rid off the trouble with provider, traffic, performance we
decided to move to another webspace with unlimited traffic
and better performence (thx to Romania )
One little issue is left: UPLOAD
We tried to make uploads via php-script possible but the
provider 'wouldnt like that'. So if you want to upload files,
please contact @ndy or me (daphy) via PM or use the old FTP
(which is still in use - but not all the time online)
The New Ambisonic Server can be found here!
CYA Daphy
__________________
CYA Daphy
________________________________________
The New Ambisonic Server (needfulthings) can be
found here!,
further info thread; upload folder ; (sometimes down please
send PM in this case)
17th March 2004 15:40

@ndy
Junior Member
Location: Germany
Posts: 12
FTP Server down for a while
Hello guys,
the FTP- Server will be down until the 2nd april due to a
practice in my new job. You still can download all the needful
things from our new webhosting. Uploads are possible in the
old manner from the beginning of april on (or send a PM to
Daphy if it is urgent )
Keep surrounding
Yours
@ndy
__________________
problem.
Last edited by @ndy on 22nd March 2004 at 01:46
22nd March 2004 01:41

desertrat
Junior Member
Location: Germany
Posts: 8
Hi folks,
I've got a couple things on my mind...
1. I've been looking both in the newsgroups and on e-donkey
for the tests, couldn't find them on either one. Maybe I'm a bit
too late for the e-donkey... don't know.... only got around to
installing it this week.
2. Also I would like to know if there are guides someplace for
these other methods that are used. I've done the Ambisonic a
few times, the I,J & K methods and the CS II as well. Would
like to try the others out as well. If someone could point me in
the right direction, I'd appreciate it.
3. I've been doing some thinking about the Ambisonic vs J
method mixes. There are things I like about both of them - the
Ambisonic is very good for ambient music or other music that
doesn't required much of a center, it creates a very nice
spacious soundfield which is missing from the other type of
5.1 mixes I have tried. On the other hand the J method is
good for music that needs to have a center.. but misses that
nice soundfield that the Ambisonic gives you. So I've been
thinking about a way to combine the two types mixes and was
able to do it quite easily. I came up with 2 different bidules
yeserday that I am currently testing. So far the results are
promising. I will u/l the bidules as soon as the server is back
up, I am interested to have some other opinions as well.
cheers,
desertrat
Last edited by desertrat on 30th March 2004 at 09:11
28th March 2004 08:27

desertrat
Junior Member
Location: Germany
Posts: 8
The FTP server is up...
I have u/l my 2 bidules in the incoming dir
the file is desertrat_stereo2ambisonic_ambi_1.5_combo.rar
which contains
desertrat_stereo2ambisonic_ambi_1.5_combo (ver 2).bidule
desertrat_stereo2ambisonic_ambi_1.5_combo (ver 3).bidule
cheers,
desertrat
28th March 2004 12:11

@ndy
Junior Member
Location: Germany
Posts: 12
quote:
Originally posted by desertrat
The FTP server is up...
I have u/l my 2 bidules in the incoming dir
the file is
desertrat_stereo2ambisonic_ambi_1.5_combo.
rar
which contains
desertrat_stereo2ambisonic_ambi_1.5_combo
(ver 2).bidule
desertrat_stereo2ambisonic_ambi_1.5_combo
(ver 3).bidule
cheers,
desertrat
I`ve just uploaded the file to our new server . You`ll find them
in the folders "ambisonic" and in "ambiophonics".
Bye Bye
@ndy
__________________
problem.
28th March 2004 13:05

desertrat
Junior Member
Location: Germany
Posts: 8
Thanks @ndy
A couple of comments...
ver 2 seems to have more ambisonic content than ver 3
I'm not sure... but I suspect that the rear channels in ver 3
need to have the gain reduced a bit... still need to do some
more testing and then think about how to go about it.
I used the standard ambisonic settings for the ambisonic side
of the chain and the J setting for the other side.
cheers,
desertrat
28th March 2004 14:02

Umma
Member
Location:
Posts: 57
Thanks for the uploads. I will try them out ASAP!
I have found that I usually need to decrease the rear gains
about 2 dB. It feels unbalanced if I don't do this on some stuff.
I still can't put my finger on just what it is that requires the
gain reduction, though, because I don't have to do it all the
time. It might just be the size of my room and system,
though...Music that has a lot of lead guitar licks usually
require the reduction (but not always), whereas music with
harmonies in the singing (CSN&Y, Peter, Paul and Mary) do
just fine without the reduction. The reduction can usually be
done within Bidule, too, but sometimes it sounds like it's
getting normalized somewhere (BeSweet?). I'm on the run so
much I can't properly analyze what's what and what's
happening where.
28th March 2004 17:28

desertrat
Junior Member
Location: Germany
Posts: 8
I have found a couple of problems with ver 3 (ver 2 still seems
to be ok).
The center channnel is too low in volume and the rear
channels do need to have the gain reduced... looking into it,
but don't know if it can be fixed yet....
cheers,
desertrat
28th March 2004 17:43

kempfand
Senior Member
Posts: 194
quote:
On the other hand the Ambiophonics is good for
music that needs to have a center.. but misses
that nice soundfield that the Ambisonic give you.
I don't won't to be picky, but this statement (as well as the
usage of the word Ambiophonic in the last 6 posts of this
threat) is just completely wrong, full stop.
Ambiophonic always involves crosstalk-free speaker pairs
(Ambiopole or Stereo Dipole) and uses the B-Format approach
(Ambisonic) for the surrounds (either by convolution or by
panning tools).
To quote from one of Farina's papers:
quote:
In fact, a modern technique was recently
developed by Ralph Glasgal, called Ambiophonics,
which aims to the reproduction of a realistic sound
field by simultaneous usage of the binaural
approach (from which a pair of closely-spaced
loudspeakers, with cross-talk canceling filters, can
be driven) and of the B-format approach (driving,
by convolution with the B-format impulse
response, a suitable 3D array of loudspeakers,
employing an Ambisonics decoder or other
decoding schemes).
Also, Ambiophonics comes in different flavours, but as said,
always involves the x-talk cancelled frontal speakers (Stereo
Dipole).
Cheers,
Andreas
Last edited by kempfand on 29th March 2004 at 14:10
29th March 2004 14:07

desertrat
Junior Member
Location: Germany
Posts: 8
Sorry if I got something wrong here... just trying to figure out
things and obviously don't quite have it right. Thanks for
pointing this out.
But that still leaves the question... do we have an
Ambiophonic mixing method using plogue bidule and if so
which one is it? I thought that the ambi 1.5 and the wxyz26
fell under this category (they are both in the ftp under that
dir). Guess I was mistaken.
And could someone help with the first 2 parts... if the
stereo2dts tests are still available someplace and if there is
some information on the other up mixing methods? I've been
able to find information on 1. Surroundboy's, 2. Ambisonics
(EoH+Kempfand), 4. CS II (commercial product), 6. WXYZ26
(Specise-8472) 9. 3th Order (Specise-8472), and HRTF-XTC-
Panorama-WXYZ (Kempfand)(I think this is the one on the ftp)
which of course will keep me busy for awhile... now I got more
testing to do
thanks,
desertrat
30th March 2004 05:52

kempfand
Senior Member
Posts: 194
No problem, desertrat. I was just trying to clarify on what
Ambiophonics is. There is also some very good reading on this
on Ralph Glasgal's Ambiophonics site.
quote:
do we have an Ambiophonic mixing method using
plogue bidule and if so which one is it?
See http://forum.doom9.org/showthread.
p...1435#post451435 , where it mentions specise_8472 x-talk
plugin and the set of filters (impulse responses) he created.
This will create your frontal Stereo Dipole (L & R speakers). As
for the surrounds, you can use the B-Format panner (3_B-Pan
VST) and corresponding decoder (3rd_order VST).
I created a different bidule which uses different filters (actually
the ones available on Farina's public server). I didn't publish
it, but can put it to the server.
In my ears, both specise_8472's and Farina's filters work very
well (i.e. there is no "one is better than the other"), yet they
use different methods to achieve the x-talk cancel. Up to you
to decide which ones you like best.
I am still hoping that someone will write a good guide
---
quote:
HRTF-XTC-Panorama-WXYZ (Kempfand)(I think
this is the one on the ftp)
I think EoH used a bidule which I sent him, which is not on the
server, as it uses a commercial product to achieve the x-talk
(Spinaudio's 3D Panner Studio). This is yet another method to
create the x-talk, but it doesn't give better results than
specie's or Farina's filters.
Hope this wasn't too confusing ...
Kind regards,
Andreas
30th March 2004 08:53

Eye of Horus
Senior Member
Posts: 255
quote:
No problem, desertrat. I was just trying to
clarify on what Ambiophonics is. There is
also some very good reading on this on Ralph
Glasgal's Ambiophonics site.
See http://forum.doom9.org/showthread.
p...1435#post451435 , where it mentions
specise_8472 x-talk plugin and the set of
filters (impulse responses) he created. This
will create your frontal Stereo Dipole (L & R
speakers). As for the surrounds, you can use
the B-Format panner (3_B-Pan VST) and
corresponding decoder (3rd_order VST).
I created a different bidule which uses
different filters (actually the ones available
on Farina's public server). I didn't publish it,
but can put it to the server.
In my ears, both specise_8472's and Farina's
filters work very well (i.e. there is no "one is
better than the other"), yet they use
different methods to achieve the x-talk
cancel. Up to you to decide which ones you
like best.
I am still hoping that someone will write a
good guide
---
I think EoH used a bidule which I sent him,
which is not on the server, as it uses a
commercial product to achieve the x-talk
(Spinaudio's 3D Panner Studio). This is yet
another method to create the x-talk, but it
doesn't give better results than specie's or
Farina's filters.
Hope this wasn't too confusing ...
Kind regards,
Andreas
Hi Andreas,
Can you put together a bidule which fits this discussion ? And
upload it to the FTP server ?
I was thinking of 2 bidules :
1. Stereo Dipole + Ambiophonics for the rears
2. LCR (Gerzon's ?) and Ambiophonics for the rears
back to work......
our thanks will be great :-))
EoH
__________________
30th March 2004 13:04

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kempfand
Senior Member
Posts: 194
Ambiophonics Bidule Prototype
Here is the outline of a true, complete Ambiophonics bidule, which implements:
- A frontal Ambiopole (Stereo Dipole)
- Ambisonics for the surround speakers, 2 in this specific case (LS, RS)
You can download the bidule, filters, and well as 2 pictures for illustration of the settings here: http://www.app.demon.nl/20040330_-_XTC_01.rar (thanks for the hosting w direct link) or from web-based
FTP interface.
Please note:
- This is a prototype in order to get you started. I personally use different set-ups, as I use Center Surround speaker(s), but you can easily adjust the posted bidule.
- The left tree of the bidule is for the Stereo Dipole, the right tree is for the B-Format (2nd order Ambisonic) surrounds.
- The Stereo Dipole is created by x-talk cancelling filters (impulse responses) which are from Angelo Farina's public web server. They were employed in the original Stereo Dipole setup by David
Wareing (one of the originators of this methodology). These are optimised for +/- 10 deg spacing of the frontal speakers (totalling up to 20 deg), and for "normal" (not binaural) recordings (no HRTF
inverse filtering here). Again: If you use them on "regular" speaker setup's (such as +/- 30 deg, they will not work).
- As mentioned before, be aware that a Stereo Dipole is the basis of the Ambiophonics method (look at Ralph Glasgal's great site).
- specise_8472 implemented another methodology for the x-talk cancellation (look at
his posting on this and the following posts). His most recent VST's and filters have been posted on the FTP by Daphy & @ndy, and are now available on the web-based interface. My personal opinion is that
species' VST's (x-talk and others) as well as filters work very well, so you might want to also try his setup.
- The Ambiophonic prototye bidule uses all free VST's (i.e. SIR for the convolution). This makes things a bit tricky, as SIR has a 8960 samples latency, which makes it necessary to use 'dummy'
instances of SIR in order to keep the Stereo Dipole and the Surrounds in-sync.
Things are much easier if you use commercial convolver VST's, such as the new state-of-the-art Voxengo Pristine Space convolver, which solved all these problems. Voxengo can convolve 8 impulse
responses simultaneously, and much more efficiently than SIR.
- Usage of the SIR convolvers: An important point is to disable the autoranging feature of SIR, setting manually the gain, and ensuring that the two instances have exactly the same gain.
-- Always use 0 dB for Wet, and -6dB for Dry.
-- As for the dummy instances of SIR, don't load any filter, and use 'Off' for Wet, and 0 dB for Dry.
-- See enclosed pictures for details.
- Surrounds: I use Gerzonic's Panorama and Emigrator for the surrounds. They both apply 2nd order Ambisonic.
As mentioned above, the prototype bidule only decodes to 2 surrounds (SL, SR), but you can adjust this if you have 1-2 Center Surround speakers. Be aware that for a proper Ambisonic system
feeding the surrounds, you will need more speakers (sweet-spot problem with less than 5-6 speakers at the min), but since Ambisonic in this case is 'just' used to 'support' the Stereo
Dipole, this might be ok.
- Credits: This 'poor man's Ambiophonic' would not be possible without the pioneering research of Gerzon, Farina, Glasgal, Miller, Kirkeby, Nelson, Hamada, Wareing, Dalenbck, Leese, just to name a few.
Needless to say that I have enormous respect for these people, as well as for the creators of great software tools such as Plogue Bidule, SIR, Gerzonic, CATT.
- Final remarks:
-- I think that you will never find filters that x-talk cancel well on all types music, that is free of coloration on all music, and works for all people. Up to each of us to experiment and see where it works well.
-- You need good (!), matched speakers for the Stereo Dipole to work well (i.e. balanced frequency response curve).
- Suggested Music: I've found that the Stereo Dipole works extremely well on Pink Floyd's The Final Cut, as well as on Roger Waters' Pros And Cons Of Hitchhiking (and possibly also on Amused to
Death). Give it a try if you have these handy.
Hope this is helpful,
Andreas
Last edited by kempfand on 2nd April 2004 at 15:19
30th March 2004 23:47

daphy
Miles Freak
Location: Germany
Posts: 175
Hi folks,
I have uploaded the file to www.needfulthings.webhop.org , you will find it in folder ambisonic>latest!
kempfand & me are thinking about a change in structure of this website - mixing the methodes leads to sorting problems
-> So this target folder could be moved very soon on an other place
Thx to EOH for quick help!
CYA Daphy
__________________
CYA Daphy
________________________________________
1st April 2004 06:59

PeterC
Junior Member
Location:
Posts: 3
Hi all
I seem to have completely lost the ball with all these
new bidules flying around, haven't read this thread for a while
. I tried the "complete Ambiophonics bidule" with Rocket Man by Elton John, I don't know if I did something wrong, I had trouble locating
the vocal when listening, the rears sounded very loud and it was like Elton's voice was floating around me. Perhaps somebody slipped something in my tea ;-)
Anyway, could anybody please tell me what is the current so-called "best" or preferred method that utilizes the center channel?
I also tested Desertrat's V2 and 3, but is the Besweet
commandline output the usual one for this? this confused me.
Surely I did something wrong since both output a file with
..well one surround was like a phantom effect - very very quiet
, the other surround was loud as the front. Rest were normal
sounding. Might this have something to do with also my program
not having a recognizable 5.01 to WXYZ_O plugin . The bidule still didn't recognize it after installing and I chose to replace it from the list (I have it installed), and I tested each of the three settings and
always got the same oddly quiet rear.
What am I doing wrong :S ?
EDIT: Addition, on desertrat's it's the right-surround
that always comes out as a total fuzzy quiet piece :S
Last edited by PeterC on 1st April 2004 at 17:03
1st April 2004 17:00

Shayne
Senior Member
Posts: 193
Quote "the vocal when listening, the rears sounded very loud" = new amp
all is here and you should add help. Please read and look for linkz and all methods are there ........... the first page and 25 after is all you have to read here.
All sounds so good at the farm
Peace
4th April 2004 04:49

Lex55
Junior Member
Location:
Posts: 19
kempfand new method
Nice results
Tested with Eric Clapton Layla unpluged
Trying to Add LFE AS Well Thinking on crossover off 60hz.
5th April 2004 13:58

kempfand
Senior Member
Posts: 194
quote:
thinking on crossover off 60 Hz
Actually, if you use the bidule for the Stereo Dipole I posted in the first place, you should put things to 0 Hz (i.e. no cross-over at all).
Reason: Someone kindly made me aware that I made a mistake in the x-talk bidule I posted.
Actually, x-talk cancellation is needed at low frequencies but not so much at the high frequencies. At high frequencies the head itself inhibits the x-talk since it is larger than the wavelength. If at all, the
cross-over filters should be used at high frequencies, not at low frequencies as outlined in my original bidule. Guess I had too much Wisky when I did it ...
Anyhow: I took the cross-over filters out for now (post corrected, as well as bidule in the zip-file).
Kind regards,
Andreas
5th April 2004 14:43

Lex55
Junior Member
Location:
Posts: 19
Nice Results
Got some nice results using your new enlightments
My next test will be combining First method with last one
Finale objective is crating good 5.1 DD from my Produced films.
6th April 2004 13:23

specise_8472
Newbie
Posts: 101
Have just uploaded into FTP server under Incoming\Ambiophonics
Allinone.dll
My latest offering.
Basically it is as stated, All in one package.
Stereo in 5 channel out.(Usual ordering L,R,C,SL,SR)
It uses my own custom method to get true? stereo surrounds. Not the same sounding as is the case now with the Ambi methods.
It still uses Ambisonics to compute the surrounds and Center Front. But with my own twist.
BTW for the Neo6 fans, the SC is mixed into the surrounds as I normally do.
__________________
7th April 2004 12:54

daphy
Miles Freak
Location: Germany
Posts: 175
Hi folks,
thx specise_8472 for the new methode -> the old FTP is unbelieveable slow - stay tuned - I will upload the file to new webspace within a couple of hours
CYA Daphy
__________________
CYA Daphy
________________________________________
7th April 2004 13:53

Tantulus
Junior Member
Posts: 11
I'm completely at a loss!
I have been following this thread for several months now and I'm afraid that I'm becoming hopelessly confused.
Part of the problem was that I was unable to download the test files from the newsgroup (I never used one before) nor could I obtain them from donkey2000.
I would appreciate a little hand-holding in my attempts to master converting stereo into surround sound.
Essentially, I am interested in encoding my stereo orchestral CD's into dts. I wish to emulate a symphony or chamber hall with a bit of ambience.
What I have been doing is making an image file of my CD, run it through Bidule and use wav2wav6 to convert the resultant file into 6 mono files and then use softencode to encode into dts. I use
Wave2Wave6 because the CD images are usually to large for besweet to handle. I understand that the ultimate result is a matter of taste but I would appreciate some suggestions as to which method will
give me a somewhat broad space with a hint of reverb but not to much gain in the rear surround.
I've noticed that most tests were done on vocal or band tracks and I find that with symphonic music I'm swamped with sound from all directions. I've developed quite a coaster collection and I would like to
narrow the possibilites a little.
If anyone has experience with orchestral music and has some recommondations as to which methods would be most useful I will be greatly appreciative.
Thanks again and keep up the good work.
8th April 2004 03:39

kempfand
Senior Member
Posts: 194
First of all, if you don't know (yet) which is a good set-up for you, I'd recommend that you use small extracts of the music (1 minute max; 30 secs is usually enough). You can collate some snippets (i.e. 10
of them coming from various CD's, having 30 secs each) and use these for testing purposes until you have a setup which is good for you. This will save you lot's of time, plus you will learn to get a feeling
what different bidules do.
As for ochestral music, I don't want to discourage you, but you might be out of luck here ... Reason is that the music you use is not anechoic (dry) and already has the characteristics (reverb, ambience) of
the listening space / concert hall where it was recorded. This is especially the case for orchestral music.
Another thing to keep in mind is, that some people will not like the room-sound that Ambisonics creates (even if it is correct!) ... This is because many people are too much used to what what the
"Tonmeisters" (German expression for soundmasters) usually produce on current multichannel dolby mixes.
Also keep in mind that most people are so used to stereo, such that hearing too much of the room won't sell (even if it is right). Just my opinion ...
You might want to try the VST created by specise_8472 (either the original one called 'WXYZ_to_5.0', or the latest one just announced a few posts above), as it keeps the stereo image for L & R speakers,
yet adds some nice "room" through C, SL, SR.
Good luck,
Andreas
8th April 2004 09:52

Lex55
Junior Member
Location:
Posts: 19
Advice from newbie
Few weeks ago i was in the same feeling you are now.
As my recommendation is just start at the first method as guided in the first page by Eye of Horus
Play with it so you will learn the pros and cons of that guide.
Only after you familiar with that you might think of starting upgrading your results following expert's remarks.
Good Luck
9th April 2004 10:04

Shayne
Senior Member
Posts: 193
Re: I'm completely at a loss!
quote:
I've developed quite a coaster collection and I would like to narrow the possibilites a little.
Well i would buy a 60 buck Hercules sound card for my computer and optical out to my amp. ie use my computer as the dvd player. This would produce absolutely no coasters and give you 100 gig dvd's
which makes changing a lot less frequent.
9th April 2004 14:34

rc
Junior Member
Registered: Feb 2004
Location:
Posts: 1
RE: specise_8472 new method
Hi Specise,
I dl your new method All In One.Rar and it came out only a
file "allinone.dll". Pls let me know how to use it. Is it just
a plug in for Winamp????
Thank you
rc
15th April 2004 17:33

Tantulus
Junior Member
Posts: 11
Ambiophonics Bidule Prototype
First of all, thanks for the earlier advice, Andreas, I will start experimenting with the other VST's. I guess what I'm trying to do is take away the stereo sweet spot and translate the recorded ambience to
give the recording a fuller feel. The first thing I did was to adjust the initial stero2ambi bidule with LFE and 6 channel gain to decrease the surround. Still not satisfied.
BTW, I read Ralph Glasgal's article and decided I am asking too much out of 5.1. However, I did try your Ambiophonics method even though my room couldn't accomodate a 20 degree separation of my
speakers. I used your first bidule (the server was down so I couldn't use your second bidule). Anyway, the results still allowed me to localize a solo instrument (although the instrument seemed somewhat
broader than life) and the surround effects were wonderfully warm, giving a decent illusion of ambience.
If possible could you please PM me your bidule with the corrected X-talk while the server is down. Also, any more info on this method would be very interesting. I am especially interested in the use of
impulse responses in your method. I want to apologize in advance but I'd like a little more hand-holding until I can get better familiarized with ambisonics/ambiphonics.
Sincerly,
Scott
Last edited by Tantulus on 15th April 2004 at 22:33
15th April 2004 22:26

kempfand
Senior Member
Posts: 194
quote:
If possible could you please PM me your bidule with the corrected X-talk while the server is down
The hosting with EoH now contains the corrected bidule and pictures: http://www.app.demon.nl/20040330_-_XTC_01.rar. But it will only work well for +/- 10 deg speaker setup.
On howto create the B-Format:
quote:
I am especially interested in the use of impulse responses in your method.
I often use the ones outlined on Angelo Farina's public server. The link is in his guide: Conversion between UHJ and B-format, and it uses the SIR convolver.
State of the art these days is the Voxengo's Pristine Space 1.1, which can convolve simultaneous 8 indepdendent filters, and even has a "zero latency mode". There is a nice outline for AudioMulch, also on
Farina's public server: Voxengo-Bformat.
quote:
I want to apologize in advance but I'd like a little more hand-holding until I can get better familiarized with ambisonics/ambiphonics.
No problem, but I suggest you start with EoH's excellent guide on the 1st page of this threat. Having done so will help you to understand the basics (i.e. what is done why), and you will later be able to
compare (by listening) where more advanced methods differ.
If you really have time (now or later), I highly recommend tha CATT-F (FIReVerb) package. For pure Ambisonics, it currently delivers the best results (at the expense of slower speed, and no real-time
processing as with Bidule). You can get a fully-working test-license, which is valid for 1 months (I believe). The package also contains an excellent Help-file and even a small Ambisonics tutorial (of which I
learnt alot). Link: CATT-F / FIReVerb
Good luck & in a hurry,
Andreas
19th April 2004 16:01

desertrat
Junior Member
Location: Germany
Posts: 8
quote:
Originally posted by PeterC
I also tested Desertrat's V2 and 3, but is the Besweet
commandline output the usual one for this? this confused me.
Surely I did something wrong since both output a file with
..well one surround was like a phantom effect - very very quiet
, the other surround was loud as the front. Rest were normal
sounding. Might this have something to do with also my program
not having a recognizable 5.01 to WXYZ_O plugin . The bidule still didn't recognize it after installing and I chose to replace it from the list (I have it installed), and I tested each of the three
settings and always got the same oddly quiet rear.
What am I doing wrong :S ?
[/B]
Sorry to take so long getting back to you on your question concerning the bidules I posted. First of all I would like to comment that you can forget V3, after additional testing and modification I still haven't
got it to work to my satisfaction.
Now back to the other question....if your progam doesn't recognize the 5.01 to WXYZ_O plugin, you will not the proper result, since the majority of the imformation comes from that side of the chain. And
yes the besweet command line is the usual one. I would suggest testing the ambi_1.5.bidule first. If you get this one working... mine should work as well.
Hope this helps a bit.
Cheers,
desertrat
Last edited by desertrat on 20th April 2004 at 07:56
20th April 2004 07:18

puzio
Junior Member
Location:
Posts: 20
Hello Kempfand
Please help me.
I use Voxengo Pristine Space convolver. But Voxengo Pristine Space convolver has only 2 in pins and 2 out pins. So I don't know how I can convolve strereo file with W,X,Y impulses at one time ( I need 6
out pins!!!!!!!!!).
Thank You.
25th April 2004 17:08

Eye of Horus
Senior Member
Posts: 255
quote:
Hello Kempfand
Please help me.
I use Voxengo Pristine Space convolver. But Voxengo Pristine Space convolver has only 2 in pins and 2 out pins. So I don't know how I can convolve strereo file with W,X,Y
impulses at one time ( I need 6 out pins!!!!!!!!!).
Thank You.
Above the quality tab (left bottom), you can set the channels ....
eoH
__________________
25th April 2004 17:14

puzio
Junior Member
Location:
Posts: 20
Thank You EoH
I did. I enable 8 channel convolution but on the bidule's screen there are only 2 out pins!!!!
Thanks.
25th April 2004 17:20

kempfand
Senior Member
Posts: 194
quote:
But Voxengo Pristine Space convolver has only 2 in pins and 2 out pins
In Pristine Space, use the comfiguration settings to set for 2 ins / 4 outs (or whatever you need). You have to click on the "?" on the upper-right part of the VST-screen.
Just in case, there is also a manual which describes this
As for the settings, you can check the picture on Farina's public server:

Cheers,
Andreas
25th April 2004 17:20

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-- Audio encoding
Doom9's Forum - LCR Upmix Bidules
conversion.

Doom9's Forum > General > Audio encoding > LCR Upmix Bidules

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kempfand
Senior Member
Posts: 194
LCR Upmix Bidules
I have created 4 new Bidule groups to upmix L-R into
L-C-R.
They are currently uploaded on the Ambisonic FTP
server in the "LCR Upmix" directory.
The Bidule Groups use slightly different methods, but
all preserve the Stereo image pretty well.
Pins: L, R = inputs, L', C', R' = outputs.
LCR Gerzon.bgrp
The Gerzon approach which mathematically-correctly
recreates the sound of the original two-channel mix:
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).
LCR MultiSonic 0.5.bgrp
The Multisonic(R) approach which is:
L' = L - 0.5 * R
C' = (L + R) * 0.5
R' = R - 0.5 * L
http://forum.doom9.org/showthread.php?threadid=70678 (1 von 5)01.05.2004 02:34:02
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.
LCR MultiSonic - Var M.bgrp
A more flexible approach than MultiSonic 0.5 above.
Note that you can enter the variable "M" if you double-
click on the group. Important: After you entered the
value for M, press <enter> to make it work.
From the Technical Description:
quote:
M is an adjustable matrix factor (changed
with the Multisonic Balance control) which
can go from -1 to 1 but in general is set
in the range of 0.4 to 0.7 with
conventional program material. The
"ideal" setting is 0.5, which works well
with most stereo signals. The adjustment
allows compensation for unusual
recordings which are either too narrow
(resulting in a stronger center), or too
wide (resulting in a weaker center).
And finally, what I call the Farina approach:
LCR Farina - var c.bgrp
Recently described in the Sursound Mailing List.
quote:
(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.
As before, variable "c" can be specified by double-click
on the group, with the same procedure as before:
After you entered the value for M, press <enter> to
make it work.
Note
- Credit for all these methods goes to the cited
authors. I just translated to Bidule Groups and hope
you find it useful
- As mentioned above, all these methods more or less
preserve the Stereo image in the front (similar to
Ambiophonics). This is often desirable.
- If you like to add surround speakers (such as SL,
SR), you can easily drive them by Ambisonic or by
convolution with suitable impulse responses (using
the SIR VST).
- I encourage you to try them all. I have found that
the Gerzonic approach often perfectly isolates the
singer to the Center-speaker, leaving "music only" to
L & R speakers (e.g. Eagles: Hotel California).
Cheers,
Andreas

daphy
Miles Freak
Location: Germany
Posts: 175
Hi kempfand
one thing we spoke before via PM. Does it make sens
to seperate the part SL/SR from the standart
ambisonic bidule and make the thing in one go.
I am so sorry, but I didnt find the time to make
some tests on species crosstalk bidules (also can be
found on the FTP) but which of your 4 groups would
you prefer
-> this could be the basis for combination of both
methodes
link to LCR Upmix Bidules
__________________
CYA Daphy
________________________________________
The New Ambisonic Server (needfulthings)
can be found here!,
further info thread; upload folder ; (sometimes down
please send PM in this case)
Last edited by daphy on 30th April 2004 at 07:29

kempfand
Senior Member
Posts: 194
Hi Daphy,
quote:
Does it make sens to seperate the part SL/
SR from the standart ambisonic bidule
and make the thing in one go.
This does not make sense at all. Ambisonic panning
creates a 2D (3D ) -soundfield, which then is decoded
to a speaker-rig (a Pentagon in the case of the guide).
The more speakers you decode to, the better the
Ambisonic reproduction. If I rememebr correctly,
something like 6 speakers (Hexagon) is the minimum
to have full 2D-Ambisonic coverage.
However, if you do not "kill" the stereo image,
separating the 'surround' speakers (i.e. SL/SR) is a
different story. This is the case with:
- LCR-upmix (using the Bidule groups I posted)
- Stereo Dipole using x-talk cancellation
For these scenarios, you can drive the surrounds
using various methods, such as Ambisonic,
convolution with suitable Impulse Responses, or using
the methods of the CATT FireVerb Suite (for speakers
4-5 = SL-SR).
Cheers,
Andreas


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-- Audio encoding
Doom9's Forum - 6ch Wave to dts with BeSweet (thread split)
conversion.

Doom9's Forum > General > Audio encoding > 6ch Wave to dts with
BeSweet (thread split)
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bitsnbytes
Member
Location:
Posts: 56
I'm pretty sure I have followed to the "t" but I end up
with a clicking thumping file.
help?
29th August 2003 23:59

kempfand
Senior Member
Posts: 194
My guess is that you didn't use the stable BeSweet
1.4. The command-line only works with the v1.4, not
with the latest beta.
Other suggestions: Try to connect the "Microsoft
Sound Mapper" (see section 8: Tips & Tricks). Doing
so, do you hear Audio ?
Further on: If you WinAmp play the multichannel-
audio (called result.wav in the guide), do you get
sound ?
Andreas
30th August 2003 00:10

http://forum.doom9.org/showthread.php?s=&threadid=60907 (1 von 26)01.05.2004 02:47:41
bitsnbytes
Member
Location:
Posts: 56
yes im using latest beta :-( damnit i goofed
so much for the "t"
30th August 2003 00:13

Eye of Horus
Senior Member
Posts: 257
version 1.4
I will give the letters version 1.4 another color and
size, as I foresee that more people will overlook it :-)
kind regards,
EoH
__________________
30th August 2003 00:35

bitsnbytes
Member
Location:
Posts: 56
ok Lets see if I got this right first off my .bat
BeSweet.exe -core( -input C:\BeSweetv1.4\Rush.wav
-output C:\BeSweetv1.4\surround waves\temp- -type
Now my files are named as follows
surroundC.wav : Is really Surround Left Channel?
surroundFL.wav : Is Really Center Channel?
surroundFR.wav : Is Really Front Left Channel?
surroundLFE.wav : Is Really Surround Right Channel?
surroundSL.wav : Is Really Front Right Channel?
surroundSR.wav : Is A Empty wav file put in dts
encoder but mute it?
If this is correct My listening experience as follows,
front left was almost silent, most of the sound was
generated from rear speakers and front right. Center
was there but less than the 3.
Oh yeah the Subwoofer was thumpin even without
LFE?
Say it aint so Joe
30th August 2003 00:54

Eye of Horus
Senior Member
Posts: 257
quote:
ok Lets see if I got this right first off
my .bat
BeSweet.exe -core( -input C:
\BeSweetv1.4\Rush.wav -output C:
\BeSweetv1.4\surround waves
\temp- -type wav -6ch ) -ota( -G
Now my files are named as follows
surroundC.wav : Is really Surround
Left Channel?
surroundFL.wav : Is Really Center
Channel?
surroundFR.wav : Is Really Front Left
Channel?
surroundLFE.wav : Is Really Surround
Right Channel?
surroundSL.wav : Is Really Front
Right Channel?
surroundSR.wav : Is A Empty wav file
put in dts encoder but mute it?
If this is correct My listening
experience as follows, front left was
almost silent, most of the sound was
generated from rear speakers and
front right. Center was there but less
than the 3.
Oh yeah the Subwoofer was thumpin
even without LFE?
Say it aint so Joe
This doesn't look good.....
@Kempfand, I got the idea we used the renaming
routine for Besplit in the guide but now we use
Besweet.... (Is that possible)
@bitsnbytes : was your listening experience from the
original names or from the renamed ones. That's not
clear to me....
EoH
__________________
30th August 2003 01:06

Eye of Horus
Senior Member
Posts: 257
quote:
This doesn't look good.....
@Kempfand, I got the idea we used
the renaming routine for Besplit in
the guide but now we use Besweet....
(Is that possible)
@bitsnbytes : was your listening
experience from the original names
or from the renamed ones. That's not
clear to me....
EoH
Quick look : this is indeed looking like the output files
from Besweet are ind ifferent order because in Besplit
the empty channel was the SR channel and here it is
the SL ! (after renaming it must become the LFE
channel)
So, what's the correct channel-mapping that comes
out of Besweet ???
EoH
__________________
Last edited by Eye of Horus on 30th August 2003 at 01:18
30th August 2003 01:14

kempfand
Senior Member
Posts: 194
Your BeSweet line will produce files called temp-FL.
wav, temp-FR.wav, ... , temp-SR.wav (all normalzed).
And yes, the channel-remapping you listed is correct
('is really').
If you go via SurCode CD, you can leave out the wav
you call surroundSR.wav. It is empty.
In order for the thing to work, please re-check your
settings for B-Pan , B-Proc , and Emigrator
Good luck,
Andreas
30th August 2003 01:15

bitsnbytes
Member
Location:
Posts: 56
From the .bat the names generated are the ones you
see on the left. I put them into the encoder as u see
on the right.
surroundC.wav : Surround Left Channel
surroundFL.wav : Center Channel
surroundFR.wav : Front Left Channel
surroundLFE.wav : Surround Right Channel
surroundSL.wav : Front Right Channel?
surroundSR.wav : LFE Is muted
30th August 2003 01:15

kempfand
Senior Member
Posts: 194
@ EoH:
quote:
is it correct as said in the guide ?
Yes
@ bitsnbytes:
1) You mention 'dts encoder', so your BeSweet line
should leave out the -ssrc( ) resampling
2) I cannot see how you get filenames called
'surround...wav.
BeSweet.exe -core( -input C:\BeSweetv1.4\Rush.wav
-output C:\BeSweetv1.4\surround waves\temp- -type
should create files calles temp-...wav.
Can you post your logfile (i.e. after -6ch , add "-logfile
C:\BeSweet.log" ).
Andreas
30th August 2003 01:33

bitsnbytes
Member
Location:
Posts: 56
ok just made a new one and here is the log...
also i keep it at 48000.
Output does not work whats up with that?
files go into C:\BeSweetv1.4
--------------------------
Using Shibatch.dll v0.2 by Naoki Shibata & DSPguru
(shibatch.sourceforge.net).
Logging start : 08/29/03 , 16:52:38.
BeSweet.exe -core( -input C:\BeSweetv1.4\Rush3.
wav -output C:\BeSweetv1.4\surround waves\temp- -
type wav -6ch -logfile C:\BeSweet.log ) -ota( -G
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : C:\BeSweetv1.4\Rush3.wav
[00:00:00:000] +------ Shibatch -----
[00:00:00:000] | Dest. Sample-Rate: 48.0KHz
[00:00:00:000] | Attenuation : 0.0db
[00:00:00:000] +---------------------
Logging ends : 08/29/03 , 16:54:00.
made a command line change to -output C:
\BeSweetv1.4\temp-
now they are named like your example
guess i cant send them to a diff dir or 2 dir deep?
Last edited by bitsnbytes on 30th August 2003 at 01:58
30th August 2003 01:51

kempfand
Senior Member
Posts: 194
quote:
Output does not work whats up with that?
Hope DSPGuru see's this and can advise. Output
should go into C:\BeSweetv1.4\surround waves\ ...
Andreas
30th August 2003 01:57

DSPguru
BeSweet Author
Posts: 3446
quote:
B. BeSweet -core( -input In-6ch.wav -
output Out_ -type wav -6ch ) -ota( -G
For BeSweet v1.5b20, this produces
garbled (clicking) mono-wav's on 2
XP machines I tested.
fixed!
__________________
31st August 2003 00:18

kempfand
Senior Member
Posts: 194
quote:
fixed!
Confirmed i.e. working now Unbelievable ...
Thanks !
31st August 2003 00:43

daphy
Miles Freak
Location: Germany
Posts: 175
@ dspguru
is there still that 2 GB input file size limit in the latest
beta? These multichannel wav produced from Bidule
could be very very
big
CYA Daphy
31st August 2003 18:59

DSPguru
BeSweet Author
Posts: 3446
quote:
This noise only takes 1~2 seconds
and then the normal sound starts -
but it could be enough for your
speakers
you can add -split( -start 2 ) to your commandline.
BeSweet will skip that noise.
quote:
is there still that 2 GB input file size
limit in the latest beta?
yes
__________________
31st August 2003 20:40

bitsnbytes
Member
Location:
Posts: 56
ok I see it in preferences... I thought recorder heh
A little help with A new besweet command line for this
mode
this is proly it
BeSweet -core( -input In-6ch.wav -output Out_ -type
wav -6ch ) -ota( -G max )
__________________
Last edited by bitsnbytes on 1st September 2003 at 21:44

DSPguru
BeSweet Author
Posts: 3446
quote:
questions:I thought one of the goals
was to work in 32bit mode.. I take it
besweet and besplit only handle 16
bit?
BeSplit works with any wave format.
BeSweet v1.5b21 supports 32bit-float input and 32bit-
float output ("-core( -6chfloat )").
i got the Bidule some days ago, but only tested it
today :
i had created a 32bit 6chwave with Bidule and
normalized+resampled+encoded to dts (in 32bit) with
v1.5b21.
works fine.
__________________

bitsnbytes
Member
Location:
Posts: 56
DSP Guru
BeSweet v1.5b21 supports 32bit-float input and 32bit-
float output ("-core( -6chfloat )").
i got the Bidule some days ago, but only tested it
today :
i had created a 32bit 6chwave with Bidule and
normalized+resampled+encoded to dts (in 32bit) with
v1.5b21.
works fine.
the only problem i noticed is the spike on the start of
the file, so i skipped it by using "-split( -start 1 )".
might I have a copy of ur commandline im using
surcode dvd... and do you use the gain or just go into
besweet? I never get the spike only once on first few
tries.
thanks
__________________

bitsnbytes
Member
Location:
Posts: 56
Grrrrrrrrrrrr
BeSweet v1.5b21 by DSPguru.
--------------------------
Logging start : 09/05/03 , 17:07:01.
BeSweet.exe -core( -input track01.wav -output C:
\BeSweetv1.5b20\mono\temp- -logfile c:\dts.txt -split
( -start 1 ) -6chfloat ) -ota( -g peak ) -surdvd( -b
1536 ) -output C:\DTS Rips\Besweet\track01.dts -
path C:\Program Files\Minnetonka Audio Software
\SurCode DVD DTS )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : track01.wav
[00:00:00:000] | Output: (null)
[00:00:00:000] | Floating-Point Process: Yes
Error 83: Surcode's output files not set!
Quiting...
Logging ends : 09/05/03 , 17:07:01.
I set surcode and saved the .sff to destination? does it
go in besweet dir?
how do we use -split( -start 1 ) ?
it starts lame encoding when i use it?
i removed a few things then besweet just crashed
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bitsnbytes
Member
Location:
Posts: 56
Thanks DSP Guru.... I knew I blew the commandline.
Now she's A thing of beauty
Quate: btw, multichannel, 32bit floating-point waves
bigger than 2gb are supported as well, so i don't see
any reason why not to create 32bit waves. (unless
you're short in H.D. space)
So now I can rip the audio from A Divx movie and
rencode to dts mwhahahah um wait the -split seems
to skip the first second of a file... thats not good for
sync... are there any other options?
On this matter I dont get spikes very often... more
like rarely least I dont here a loud glitch sound.
__________________

DSPguru
BeSweet Author
Posts: 3446
quote:
Thanks DSP Guru.... I knew I blew
the commandline.
Now she's A thing of beauty
would be nice to see your logfile.
btw, if you feel messy with the commandline, you
should use BeSweetGUI (latest is b82), there's a
ready made DTS-CD predefined profile.
quote:
the -split seems to skip the first
second of a file... thats not good for
sync... are there any other options?
this is a solved problem, since you can assert a
1000ms of delay to compensate the skip of 1sec.
commandline is -ota( -d 1000 ) -split( -start 1 ),
but there again - use the GUI !
__________________

bitsnbytes
Member
Location:
Posts: 56
As Requested
--------------------------
Logging start : 09/06/03 , 07:29:05.
BeSweet.exe -core( -input track01.wav -output C:
\BeSweetv1.5b20\mono\temp- -logfile c:\dts.txt ) -ota
( -g peak ) -surdvd( -b 1536 -output C:\DTS Rips
\Besweet\track01.dts -path C:\Program Files
\Minnetonka Audio Software\SurCode DVD DTS ) -split
( -start 1 )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : track01.wav
[00:00:00:000] | Output: C:\DTS Rips\Besweet
\track01.dts
[00:00:00:000] | Overall Track Gain: 1.755dB
[00:00:00:000] +---------------------
[00:11:59:911] Actual Avg. Bitrate : 1512kbps
Logging ends : 09/06/03 , 07:38:00.
and thanks for the delay info...
I would prefer to continue to know and understand
the many switches of your most excellent tool
can I use .lst file and convert 16-44.1 audio files to
32-48 float
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bitsnbytes
Member
Location:
Posts: 56
Besweet gui DTS Preset
Can we implement our own presets?
Can you add A preset for DTS-DVD 24-48?
back to other question.
can besweet convert from a .lst file cd audio files to
32-48 float stereo .wavs?
If so help on the switches.
Thanks
__________________

kempfand
Senior Member
Posts: 194
quote:
for confirmation, i guess we'll have to
wait to kempfand's logfiles
Strange : I don't have the spike most of the times,
so I currently go with
BeSweet -core( -input 6ch.wav -output out_ -type
wav -6chfloat ) -ota( -G max ), plus the documented
command-line for dts.
BTW: I noticed that Shibatch also offers normalisation
( ssrc --normalize. If I read this correctly, this does
the same as -ota( -G max, with the understanding
that the azid-options offers more finetuning (% od db
values).
quote:
That said, this thread has waken up the
need for a Offline (render) mode which
would allow to process much faster than
real time
The idea of having the 2 power-horses (BeSweet &
Bidule) work at full CPU load in parallel sounds great.

Andreas

DSPguru
BeSweet Author
Posts: 3446
Re: Besweet gui DTS Preset
quote:
Can we implement our own presets?
Can you add A preset for DTS-DVD 24-
48?
come on, man, play with the GUI, i really don't have
the time to teach you anything from the start.
think of how many users are in here.
quote:
can besweet convert from a .lst file
cd audio files to 32-48 float stereo .
wavs?
floating-point output is only offered for creating
splitted mono files. -core( -6chfloat )
@kempfand
ssrc's normalization & ota's maxgain are EXACTLY the
same.
"-type wav" has no meaning in BeSweet, only in
BeSplit.
"-ota( -G max )" is actually useful for hybridgain,
what you had in mind is "-ota( -g max )".
(don't worry, since hybridgain is only applicable for
mp2,mp3&ogg while you encode Waves, BeSweet
interpretates your "-G max" as "-g max").
anyway, i don't see any reason to go for two-pass
normalization ("-g max"), when you can use the PEAK
info that can be found in the waves created by Bidule
to normalize in a single pass ("-g peak").
btw,
all this commandlines discussions are really tiring.
it's much easier to learn the commandlines by playing
with BeSweetGUI.
Andreas,
this is my suggestion :
download BeSweetGUI v0.6b82 and extract it to your
BeSweet folder,
download BeSweet v1.5b21 and extract it to the same
folder,
in the newest BeSweet zip, you'll find an updated 'DD-
Profiles.ini' file with an updated "DTS-CD" preset.
try it out!
it should work very good, and it's much much easier
than trying to track all different commandline structs.
imHo,
it could be better to replace the BeSweet
commandlines in your guide with single screenshot of
BeSweetGUI ("AC3/OGG/dts" form), and simple
instructions :
1. click on the "AC3/OGG/dts" button (link to
screenshot)
2. set BeSweet.exe path
3. set input file to be the one you had created with
Bidule (test.wav?)
4. set output filename
5. set Surcode executable path & temp folder
6. select "dts_for_DTS-CD" profile and click on the
green "WAV to WAV" button.
__________________

kempfand
Senior Member
Posts: 194
DG: The "DTS-CD" preset with v0.6b82 in the 'DD-
Profiles.ini' file works very well (as far as I can see
from the .ini file, it has the same dts-preset as with
the previous v0.6b81 version). I already confirmed
this in dts encoding via commandline
Regarding GUI vs. command-line: In order to make
use of the latest enhancements of BeSweet,
command-line is a must, as GUI v0.6b82 does not
(yet) handle "-g peak" and "-6chfloat".
Regards,
Andreas

DSPguru
BeSweet Author
Posts: 3446
beta82 does handle "-g peak".
"-6chfloat" is automaticly used in dts encoding.
the new "dd-profiles.ini" can be found in latest
BeSweet zip, as stated above.
quote:
in the newest BeSweet zip, you'll find
an updated 'DD-Profiles.ini' file with
an updated "DTS-CD" preset.
Cheers ,
Dg.
__________________

kempfand
Senior Member
Posts: 194
My mistake (had the BeSweetv1.5b21 as of Sep 5,
not the latest as of Sep 7) & apologies.
Kind regards,
Andreas

bitsnbytes
Member
Location:
Posts: 56
The Click
I found the click finally, though its not at the
beginning of the track.. it is placed at the end
thus needing a 1 sec or preferably less skipping of the
end.
I would not call it so much a spike but a small click.
damn no attachements
__________________

bitsnbytes
Member
Location:
Posts: 56
I back tracked it to EAC seems it does a peak level
and I presume it puts that information in the track... I
played the orig... no click i extracted uncompressed i
get click
at the end of the track.
__________________

TRILIGHT
Moderator
Location:
Posts: 1217
Where are my output WAV's?
Ok, I just jumped through all the many hoops to get
this up and running so I could test it out. However,
when I run the command line for BeSweet, it writes
the files out quickly and they are all 1KB! What?!?! Do
I have to have 1.5? I am using 1.4 right now.
__________________
Das UberGuide lives! Check it out! --> HTTP:/ /
DVDGUI DES.TRI LI GHT.NET

bitsnbytes
Member
Location:
Posts: 56
Re: Where are my output WAV's?
quote:
it writes the files out quickly and they
are all 1KB! What?!?! Do I have to
have 1.5? I am using 1.4 right now.
I would use latest beta. its fixed.
Dont know what you are doing with it...end result
wise but here is commandline to 6 wavs at 32/48
BeSweet.exe -core( -input track01.wav -output "C:
\BeSweetv1.5b21\temp waves\temp-" -logfile c:
\besweet.txt ) -6chfloat -ota( -g peak )
so start by converting ur stereo wav to 32/48 set 48
in pref of bidule.. restart bidule, set recorder to 32
float, then run this commandline... well u know set ur
paths.
adjust instructions to fit ur application... ac3
dts cd etc
i record to besweet dir and put sub dir in besweet so i
can launch and delete temp files farly quickly
__________________

TRILIGHT
Moderator
Location:
Posts: 1217
Thanks for the reply. I should have been more specific
as to what I was doing. The command line I am using
is that found in the guide in the first post of this
thread (adjusted to my file locations, of course). I am
using a 48kHz stereo original WAV and attempting to
get the same in 6 WAV's for use in encoding to AC3
for DVD. Everything has gone fine except for the
BeSweet part. I'll download 1.5 and try it out.
EDIT: I just tried 1.5 and it appears to be writing the
files.
EDIT2: Spoke too soon. It seems to scan the whole
file and create the fileset but nothing ever gets
written to them. When it hits the end of the file,
BeSweet just crashes. This is the command line I am
using...
BeSweet.exe -core( -input E:\TOP20.wav -output E:
\temp- -type wav -6ch ) -ota( -G max )
__________________
Das UberGuide lives! Check it out! --> HTTP:/ /
DVDGUI DES.TRI LI GHT.NET
Last edited by TRILIGHT on 16th September 2003 at 07:21

kempfand
Senior Member
Posts: 194
TRILIGHT: See also 6ch Wave to dts with BeSweet
(thread split) about this.
Suggestion:
1) Re-download the latest BeSweet (there were
versions from different dates)
2) Use "-ota( -g max )" (not -G)
Good luck,
Andreas

ilmanu
Junior Member
Location:
Posts: 14
where is the first part of "6ch to dts" 3d?

Umma
Member
Location:
Posts: 57
This thread IS the first part.

ilmanu
Junior Member
Location:
Posts: 14
sure?
the 1st reply is:
quote:
I'm pretty sure I have followed to the "t"
but I end up with a clicking thumping file.
help?
i'm not sure but i think this isn't the 1st part...

Umma
Member
Location:
Posts: 57

SORRY!! You are correct. The first part is the stereo to
ambisonics surround by EoH. The discussion was
veering off into a thread on BeSweet command lines
so it was split. I thought this was THAT thread when I
responded!
I'm 43 and going senile...

bitsnbytes
Member
Location:
Posts: 56
Besweet Question
Maybe A little guidance DSP Guru
When connecting Emigrator to 6 channel Recorder
emigrator pins 1 - 6 left to right
c fl sl sr fr lfe
besweet commandline to 6 waves output log shows
1 > 5 = C
2 > 1 = FL
3 > 3 = SL
4 > 6 = SR
5 > 2 = FR
6 > 4 = LFE
Besweet GUI via ogg 6ch settings gave me the
impression that its settings are as follows.
1 > 2 = C
2 > 1 = FL
3 > 4 = SL
4 > 5 = SR
5 > 3 = FR
6 > 6 = LFE
And the guide shows
1 > 3 = C
2 > 1 = FL
3 > 5 = SL
4 > 6 = SR
5 > 2 = FR
6 > 4 = LFE
whats the right and wrong way here?
Or have I been starring at audio files too long?
__________________

Pages (3): [1] 2 3 Last Thread Next Thread

Doom9's Forum > General > Audio encoding > 6ch Wave to dts with BeSweet (thread split)
Pages (3): 1 [2] 3 Last Thread Next Thread
Author Thread
DSPguru
BeSweet Author
Posts: 3446
quote:
BeSweet's naming convention is :
Channel 1->"FL"
Channel 2->"FR"
Channel 3->"C"
Channel 4->"LFE"
Channel 5->"SL"
Channel 6->"SR"
__________________

bitsnbytes
Member
Location:
Posts: 56
I guess when wierd things happen you ask stupid questions.
again I say when I use Besweet GUI to encode to .ac3 it sends pin4 LFE to SL as Im not getting any sound out of
that speaker.
I tried setiing ogg to to that format then checked back to ac3 and still no sound.
but if I change the recorder to match the initial grayed out settings in Besweet GUI
it appears to work.
file is 32/48 should I be using 16/48?
I see that theres no floating opperation also.
rewired for a 3rd time I am 99% sure that this is the case. Maybe its ac3enc.dll using different format?
__________________
And Polk Audio

DSPguru
BeSweet Author
Posts: 3446
i'm sorry, i reread your post and saw my reply wasn't complete. here's a more complete answer :
-6chwav/-6ch/-6chfloat
Channel 1->"FL"
Channel 2->"FR"
Channel 3->"C"
Channel 4->"LFE"
Channel 5->"SL"
Channel 6->"SR"
-ac3enc/-6chogg
Channel 1->"FL"
Channel 2->"C"
Channel 3->"FR"
Channel 4->"SL"
Channel 5->"SR"
Channel 6->"LFE"
floating-point processing iZ involved in ogg encoding, but not in ac3 encoding.
anyway, i would really advise you to avoid encoding with ac3enc, since it's an amature ac3 encoder and suffers from
several flaws.
__________________

bitsnbytes
Member
Location:
Posts: 56
Thanks Guru I am doing this test so Gamers who make in game video with soundtrack may add a bit of flair to there
film making ;-) Not A archival piece of audio work heh.
Much Appreciation for your help.
__________________
And Polk Audio

Calabuig
Junior Member
Location:
Posts: 22
@DSPguru
I'm trying to follow step by step the Eye of Horus guide to make surround from stereo.
But I've this problem: after Bidule has created the 6ch wave (3,85 GB), I open BesweetGUI and I demux the file
with this parameters:
"C:\Applicazioni\Gordian Knot\BeSweet.exe" -core( -input "f:\Temp\Garth_48KHz.WAV" -output "f:\Temp
\Garth_48KHz-" -6ch ) -ota( -g max )
The all resulted files are 658 Mb but only FL and FR are ok. The others are empty, no sound.
But if I try to open the 6ch wave in CoolEdit (multitrack view)... all the 6 waves are there and good.
Can you help me?
P.S. I'm using BeSweet 1.5b22
10th October 2003 14:51

DSPguru
BeSweet Author
Posts: 3446
as mentioned in q30 at the BeSweet FAQ, for multichannel waves bigger than 2gb, you need to create a 32bit
floating-point wave with bidule (i believe the guide suggests 16bit).
btw : next time, please create & post full logfile.
__________________
Last edited by DSPguru on 10th October 2003 at 17:45
10th October 2003 17:41

Eye of Horus
Senior Member
Posts: 257
@DSPGuru
There isn't much to report !
This is the log :
--------------------------
Logging start : 10/10/03 , 20:47:53.
E:\BeSweetv1.5\BeSweet.exe -core( -input L:\test\def3.wav -logfile l:\dts.txt -output L:\def3test- -type wav -6ch ) -
ota( -g max )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : L:\test\def3.wav
After the first walkthrough it stops with this errormessage :
Besweet.exe has generated errors and will be closed by Windows.
You will need to restart the program.
An errorlog is being created.
On Windows 2000 it does that with all version I have and tried on files larger than 2 GB. On smaller there are no
problems......
Here is also the log from dr. Watson :
Application exception occurred:
App: (pid=828)
When: 10/10/2003 @ 20:48:52.638
Exception number: c0000005 (access violation)
*----> System Information <----*
Computer Name: AART-CB5DBD25CF
User Name: Administrator
Number of Processors: 1
Processor Type: x86 Family 15 Model 2 Stepping 4
Windows 2000 Version: 5.0
Current Build: 2195
Service Pack: 4
Current Type: Uniprocessor Free
Registered Organization:
Registered Owner: aart
*----> Task List <----*
0 Idle.exe
8 System.exe
192 SMSS.exe
220 CSRSS.exe
240 WINLOGON.exe
268 SERVICES.exe
280 LSASS.exe
456 svchost.exe
488 spoolsv.exe
520 AVGUARD.exe
532 ati2evxx.exe
560 AVWUPSRV.exe
584 svchost.exe
612 gearsec.exe
636 hidserv.exe
672 regsvc.exe
696 mstask.exe
724 stisvc.exe
776 spnsrvnt.exe
808 vsmon.exe
824 WinMgmt.exe
860 MsPMSPSv.exe
872 svchost.exe
1296 atiptaxx.exe
1320 daemon.exe
1340 CloneCDTray.exe
1312 rundll32.exe
308 winampa.exe
1352 rmctrl.exe
1360 AVGNT.exe
1372 ATIX10.exe
1396 msnmsgr.exe
1424 QuickDCF.exe
1432 WinCinemaMgr.ex.exe
1468 rundll32.exe
1476 zonealarm.exe
1616 wuauclt.exe
624 explorer.exe
828 BeSweet.exe
1528 IEXPLORE.exe
1328 DRWTSN32.exe
0 _Total.exe
(00400000 - 0048B000)
(77F80000 - 77FFB000)
(7C4E0000 - 7C599000)
(7C2D0000 - 7C332000)
(77D30000 - 77D9E000)
(78000000 - 78045000)
(782F0000 - 78538000)
(77F40000 - 77F7C000)
(77E10000 - 77E75000)
(70BD0000 - 70C35000)
(71710000 - 71794000)
State Dump for Thread Id 0x45c
eax=00000000 ebx=7fffffff ecx=fffed180 edx=7fffffff esi=0048b000 edi=00000000
eip=00410766 esp=0012fcd4 ebp=0012fcf4 iopl=0 nv up ei ng nz ac pe cy
cs=001b ss=0023 ds=0023 es=0023 fs=0038 gs=0000 efl=00000293
function: <nosymbols>
00410741 3bcf cmp ecx,edi
00410743 7ce4 jl 0041e129
00410745 d91c8dfcf54300 fstp dword ptr [0043f5fc+ecx*4] ds:fffed180=????????
0041074c ddd8 fstp st(0)
0041074e 894dfc mov [ebp+0xfc],ecx ss:00ba9bda=????????
00410751 8b4dfc mov ecx,[ebp+0xfc] ss:00ba9bda=????????
00410754 bbffffff7f mov ebx,0x7fffffff
00410759 8d3500f64300 lea esi,[0043f600] ds:0043f600=38000000
0041075f fc cld
00410760 8b15702c4200 mov edx,[00422c70] ds:00422c70=3f7ffe00
FAULT ->00410766 ad lodsd ds:0048b000=????????
00410767 23c3 and eax,ebx
00410769 3bc2 cmp eax,edx
0041076b 0f4f ???
0041076d d0e2 shl dl,1
0041076f f68915702c42 ??? ds:422b4195=??
00410775 008b5d083b3d add [ebx+0x3d3b085d],cl ds:bd3b085c=??
0041077b c8324200 enter 0x4232,0x0
0041077f 897de8 mov [ebp+0xe8],edi ss:00ba9bda=????????
00410782 db45e8 fild dword ptr [ebp+0xe8] ss:00ba9bda=????????
00410785 d80da8a94100 fmul dword ptr [0041a9a8] ds:0041a9a8=3b77adf5
0041078b d805b4a94100 fadd dword ptr [0041a9b4] ds:0041a9b4=4a7ea9a1
*----> Stack Back Trace <----*
FramePtr ReturnAd Param#1 Param#2 Param#3 Param#4 Function Name
0012FCF4 00402D71 7FFFFFFF 00000000 40240000 00000000 !<nosymbols>
0000AC44 00000000 00000000 00000000 00000000 00000000 !<nosymbols>
*----> Raw Stack Dump <----*
0012fcd4 01 00 00 00 2c 00 00 00 - 00 0c 00 00 b0 ff 12 00 ....,...........
0012fce4 4c ff 12 00 00 00 00 00 - 00 00 00 00 00 00 00 00 L...............
0012fcf4 44 ac 00 00 71 2d 40 00 - ff ff ff 7f 00 00 00 00 D...q-@.........
0012fd04 00 00 24 40 00 00 00 00 - 00 00 00 00 1b 00 00 00 ..$@............
0012fd14 00 00 00 00 08 fd 12 00 - 00 02 00 00 00 00 2f 00 ............../.
0012fd24 20 24 2f 00 1b 00 00 00 - d4 fe 12 00 51 c6 fc 77 $/.........Q..w
0012fd34 9d c5 fc 77 b4 c5 fc 77 - 18 2b 13 00 38 07 04 78 ...w...w.+..8..x
0012fd44 00 f0 fd 7f 79 fe 12 00 - b8 26 2f 00 00 02 00 00 ....y....&/.....
0012fd54 f8 fe 12 00 91 81 f9 77 - 98 2a f9 77 ff ff ff ff .......w.*.w....
0012fd64 78 01 2f 00 98 3a 2f 00 - e8 12 13 00 04 01 00 00 x./..:/.........
0012fd74 38 07 04 78 00 f0 fd 7f - 7d 00 00 00 a8 57 13 00 8..x....}....W..
0012fd84 a0 57 13 00 68 b2 f8 77 - 00 b2 f8 77 8f 00 00 00 .W..h..w...w....
0012fd94 81 b1 f8 77 00 16 54 7c - 00 00 00 00 18 5c 13 00 ...w..T|.....\..
0012fda4 e4 03 e6 03 78 01 13 00 - a0 57 13 00 a8 57 13 00 ....x....W...W..
0012fdb4 78 04 00 00 04 54 13 00 - 6c 04 00 00 6c 04 00 00 x....T..l...l...
0012fdc4 60 05 2f 00 60 05 2f 00 - 0f 00 00 00 20 00 00 00 `./.`./..... ...
0012fdd4 60 05 2f 00 ef cb 4f 7c - ca 57 13 00 a0 03 54 7c `./...O|.W....T|
0012fde4 46 04 00 00 00 00 00 00 - 30 4d 13 00 00 00 00 00 F.......0M......
0012fdf4 fc ff 12 00 23 00 00 00 - 00 00 00 00 00 e0 fd 7f ....#...........
0012fe04 00 00 00 00 18 5c 13 00 - 28 4d 13 00 00 00 13 00 .....\..(M......
__________________
10th October 2003 20:56

DSPguru
BeSweet Author
Posts: 3446
1. what happens if you omit the "-g max" and process with only single pass ?
2. either way, i sugggest you to replace "-g max" with "-g peak".
3. what's the playback time of the track, what size it ?
4. are you sure that this wave is 32bit floating-point wave ?
5. please use latest BeSweet builds, namely, v1.5b22
__________________
10th October 2003 21:57

Eye of Horus
Senior Member
Posts: 257
quote:
1. what happens if you omit the "-g max" and process with only single pass ?
2. either way, i sugggest you to replace "-g max" with "-g peak".
3. what's the playback time of the track, what size it ?
4. are you sure that this wave is 32bit floating-point wave ?
5. please use latest BeSweet builds, namely, v1.5b22
#1 makes no difference. Whatever option, the program crashes after completion of the transcoding.
#2 same as #1
#3 2.209.929.260 bytes
#4 no.... 16 bits ! (NB Win2000 NTFS filesystem)
#5 no difference
regards,
EoH
__________________
20th October 2003 20:00

DSPguru
BeSweet Author
Posts: 3446
32bit floating-point is the way to go
__________________
20th October 2003 22:11

kempfand
Senior Member
Posts: 194
Does -6chfloat handle multichannel waves > 4 GB ?
I'm having some premature termination with a 4.64 GB multichannel wav (32bit floating-point output from Bidule),
which contains 78 mins of Audio.
Demuxed (mono) wavs are only 11 minutes.
code:
--------------------------
Logging start : 12/14/03 , 04:39:01.
C:\BeSweet\BeSweet.exe -core( -input m:\CDImage_BiDule_32.wav -output i:\Out_32_- -6chfloat -
logfile C:\BeSweet\BeSweet.log )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : m:\CDImage_BiDule_32.wav
[00:00:00:000] +---------------------
Logging ends : 12/14/03 , 04:39:33.
Kind regards,
Andreas

Shayne
Senior Member
Posts: 193
I would like to re-hash the wave clipping problem again in a hope that DSPguru can shine a little light on it for me.
When i produce 32 bit 6 channel wave file and then demux it into 6 waves with
D:\Video\AAC_Encode\BeSweet.exe -core( -input "D:\Surround\Temp\Temp6Ch.wav" -output "D:\Surround\temp
\temp_" -type wav -6chfloat )
i get .002 to .004 seconds at the beginning of the waves that is clipped right out. This of course is not a good sound
for spdif out and my speakers.
if i load the 6 ch wave into soft encoder this clip does not exist.
Can anyone see something i am doing wrong. Possible wrong besweet version? maybe a format c: could help?
Peace

kempfand
Senior Member
Posts: 194
Do you stil have the clipped issue with the latest version of Bidule ?
If yes, I suggest you currently go for 16bit output of Bidule. I can barely hear a difference vs. output in 32 bit and
subsequent -6chfloat in BeSweet (maybe it's just my "average" speakers). Keep in mind that feeding 32-bit into
Bidule (for the VST-processing) is more important IMHO.
A format c: wont help
Kind regards,
Andreas

Shayne
Senior Member
Posts: 193
yes i do still have this problem with the new and the .1 versions and at 16 bit i do not have this problem but do hear
the difference. 32bit is the way i would like to go.
Thank you for your quick response
Peace

kempfand
Senior Member
Posts: 194
// off-topic on //
Guess we have to live with such things that are not perfect, and patiently wait for improvements.
I.e. on my side, now that Bidule 6.501+ is out, I wanted to go to convert some of my favourite CD's, but (surprise
surprise) I found out that they are all such that they result in >2 GB (with 16-bit) or >4 GB respectively (for 32-
bit) ...
So I either have to split or use wav2wav6 (without the nice BeSweet -g max for normalisation).
Disclaimer: This is in no way ment to 'demand' or 'request' immediate action from the side of the tool-wizzards
(DSPGuru, David&team), as I fully realize that such behavior is dis-respectful and kills additional work (as happened
with DVD Shrink).
// off-topic off //
Regards,
Andreas
Edit 12/16: changed from -g max to -g peak
Last edited by kempfand on 16th December 2003 at 12:58

DSPguru
BeSweet Author
Posts: 3446
don't use "-g max" but use "-g peak"! this is faster and supposed to avoid the click at start.
sorry kempfand, haven't found the time to look for your bugreport.
__________________

Shayne
Senior Member
Posts: 193
Thank you for your help it is always great to have the people that know the software aid in your problems.
no normalization i have click
g peak i have click
g max i have click
g 14db i have click
What is weird if i open the bidule file in soft encoder and encode it as is and load the ac3 file into besweet and set to
five waves there is no click of the waves after demux.
so i think i have a problem with the dd Wave that bidule is creating or with the demux process of besweet (1.5 b23)
of 32 bit dd wave files.
thanks again
peace

daphy
Miles Freak
Location: Germany
Posts: 175
Ive never used softencode with the bidulefile (6channelWAV) -
some questions on this:
are you able to split with softencode (bidule WAV into 6 mono WAVs)?
is the channel mapping right if you open/split the file with softencode?
quote:
problem with the dd Wave that bidule is creating
it surprises me to read that bidule makes dd files
CYA Daphy

Shayne
Senior Member
Posts: 193
Soft encoder is a Wave ====> ac3 program used first here by the dark avenger with DVD audio encoding to down
sample dd to 2 channel. I takes 6 waves or 1 6 ch wave as input. It allows you to pick what channel is what channel
and shows you 5 Wave forms similar to cool edit. you can demux 1 at a time and do some .... but no it is not as
sweet as besweet when it comes to audio manipulation.
You are right and i do apologize for calling them dd files when actually they are wave.
Peace

daphy
Miles Freak
Location: Germany
Posts: 175
quote:
I takes 6 waves or 1 6 ch wave as input. It allows you to pick what channel is what channel and shows
you 5 Wave forms similar to cool edit. you can demux 1 at a time
fine, maybe we dont have to care about 2GB/4GB barrieres anymore!
One things left in theorie (sorry for only theorie -> x-mas stress, I havent found any time to test this myself).
Shayne, how do you manage the channel mapping? This might be a real problem, can you please post a screen shot
with an opened 6wav file in softencode and the correct mapping . I ve spent so much time on this subject
'correct mapping' (in the two postings there do was a little confusion -> I always had a signal on the LFE but this can
t be possible at 5.0 ) and wouldnt want to start again right at the beginning! thx
@ kempfand
one easy but comfortable solution could be to split the 32/44.1 source file (2.0) in parts of about 40 min and merge
the splitted monofiles again before encoding to dts/ac3. I know, the works is 6 time more but the new bidule should
compensate this
CYA Daphy

DSPguru
BeSweet Author
Posts: 3446
quote:
I would like to re-hash the wave clipping problem again in a hope that DSPguru can shine a
little light on it for me.
please attach a short sample of the beginning of bidule's wave output (10k would suffice).
quote:
Does -6chfloat handle multichannel waves > 4 GB ?
i believe it works for files larger than 2gb (right?), but maybe not larger the 4gb.
please check attached binary, using your huge file and the "-g peak" switch.
i'll look for it.
quote:
Edit 12/16: changed from -g max to -g peak
good!
__________________
Last edited by DSPguru on 19th December 2003 at 07:40

Shayne
Senior Member
Posts: 193
I think with respect we should honor this thread split and remember its name. If you would care to discuss this in
the original thread this would be the appropriate place to do so and i would be more than happy to.
mapping is not a problem and the gain filter of bidule is helpful if you notice besweet average db increase is about
14 db. This is however an ac3 encoder n not a dts encoder.
DTS i have not got anything out of my spdif or my 6.1 analog as of yet except hiss ........ still searching? Q24 and B
doesn't say a whole lot.
Peace
Edit ...... we must have posted at the same time "you may not post attachments" seems to be my nickname
Last edited by Shayne on 17th December 2003 at 19:11

kempfand
Senior Member
Posts: 194
quote:
please check attached binary, using your huge file and the "-g peak" switch
Did so. My conclusion is, that neither BeSweet v1.5b24pre nor v1.5b23 support demuxing of >2GB 16-bit wav (using -
6ch) nor >4 GB 32-bit wavs (using -6chfloat).
As test, I used a stereo-wav of 78 min 32 sec, which was run in offline-mode through Bidule, and saved as
- 32 bit wav : "IMG_32.wav" and
- 16 bit wav : "IMG_16.wav" respectively.
Both produce prematurly terminated mono-wav's.
Logfile 16 bit
code:
BeSweet v1.5b24pre by DSPguru.
--------------------------
Logging start : 12/18/03 , 21:08:34.
C:\BeSweet\BeSweet.exe -core( -input m:\IMG_32.wav -output i:\DeMux_32- -6chfloat -logfile C:
\BeSweet\BeSweet.log ) -ota( -g peak )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : m:\IMG_32.wav
[00:00:00:000] +---------------------
Logging ends : 12/18/03 , 21:09:08.
Demuxing starts, but the resulting 6 mono-wav's are prematurly terminated (at 10 min 54 sec).
Logfile 32 bit
code:
--------------------------
Logging start : 12/18/03 , 20:56:17.
C:\BeSweet\BeSweet.exe -core( -input m:\IMG_16.wav -output i:\DeMux_16- -6ch -logfile C:
\BeSweet\BeSweet.log ) -ota( -g peak )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : m:\IMG_16.wav
[00:00:00:000] +---------------------
This is the complete log-file. Demuxing starts, but the resulting 6 mono-wav's are prematurly terminated (at 6 min 45
sec).
Notes:
- This also applies for "-g max" instead of "-g peak" (logfiles not attached).
- On 16-bit multichannel wav's >2GB, wav2wav6.exe works fine (except, of course, there is no pregain option).
I think I found anther bug, but need confirmation:
Using the channel-mapping as outlined in the Guide, the 16-bit demux (-6ch) correctly maps the channels. I.e.
code:
--------------------------
Logging start : 12/19/03 , 00:52:55.
C:\BeSweet\BeSweet.exe -core( -input i:\BiD_16.wav -output i:\16-DeMux- -6ch -logfile C:
\BeSweet\BeSweet.log )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : i:\BiD_16.wav
[00:00:00:000] +---------------------
Logging ends : 12/19/03 , 00:52:59.
In my example, when running Bidule, the LFE is "empty" (i.e. silent).
Based on information earlier in this threat, I am assuming that -6chfloat gives the same channel mapping:
code:
-6chwav/-6ch/-6chfloat
Channel 1->"FL"
Channel 2->"FR"
Channel 3->"C"
Channel 4->"LFE"
Channel 5->"SL"
Channel 6->"SR"
However, using Bidule to save a 32-bit-wav, results in an "empty" (sllent) SR-channel, and an LFE-channel with
audio:
code:
--------------------------
Logging start : 12/19/03 , 00:53:37.
C:\BeSweet\BeSweet.exe -core( -input i:\BiD_32.wav -output i:\32_DeMux- -6chfloat -logfile C:
\BeSweet\BeSweet.log )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : i:\BiD_32.wav
[00:00:00:000] +---------------------
Logging ends : 12/19/03 , 00:53:46.
Could be that something is screwed up here, or that I mis-understand, or that Bidule wrongly handles 32-bit wav's....
So I would appreciate confirmation of this finding.
Thanks for helping. Hope I have more time over Xmas to follow-up more quickly.
Kind regards,
Andreas

puzio
Junior Member
Location:
Posts: 20
Hello Kempfand.
I have the same problem like You have. I am sure that it's a bug of Besweet because when I use Channelex of
Mctools everything is OK(channel maping). I checked Channelex with 16,24,32, 32float 6channel input. Besweet
works OK only with 16bits input.
I thing that structure of 16bits 6channel and 32bits float 6channel are simillar but not same. So DSPguru do
mistake. I wrote a program for myself. My program also works OK with 16,24bits 6channel input,
but with 32bits float 6channel input my program works like Besweet - the same problem with channel mapping.
THX.
@DSPguru.
Sorry for my words.

DSPguru
BeSweet Author
Posts: 3446
first off,
i value your time, kempfand! plese don't waste it on 16bit waves or "-g max". those modules hadn't been updated
for a very long time, and needn't to be tested.
as for 4gb 32bit float,
just ran a test on 4.2gb float file under win2k and it worked fine for me.
the only problem i can suspect is that bidule's output is wrongly recognized as 16bit wave instead of 32bit wave,
hence the wrong module is handling the file.
please attach the first 10kb of your 32bit fp wave, so i can analyze it.
regarding channel mapping,
my previous statemenat still applies, eg, channel mapping is the same for -6ch/-6chfloat and for 16/32bit inputs.
happy xmas !
Dg.
__________________

Shayne
Senior Member
Posts: 193
@ kempfand
If channel mapping has you confused add a gain in bidule and lower/raise the volume of a channel between your
guide picture.
The pentagon is a graphical picture of the surround pin outs. Then split the waves with no normalization -g(..). The
result should be clear what channel you have added the gain to. Do this with each channel till you are clear.
I have to say DSPguru is correct at least for 32 bit i have not played with 16 too much.
Peace

kempfand
Senior Member
Posts: 194
Hi Shayne,
Thanks for the suggestions. Actually, I developed the method and the guide together with EoH, so I know the
channel mapping very well (up to the point where 2/5/1/3/4 = FL/FR/C/SL/SR made my head turning like a
Dodecahedron ).
If DSPGuru is correct that Besweet's demuxing works same for -6ch and -6chfloat (and I assume he is correct, plus
you just confirmed), this leaves 2 options: either I got sth screwed up, or Bidule/Emigrator did a strange thing.
I'll re-check later (when back home) and attach a small part of the 32-bit wav.
Kind regards,
Andreas

puzio
Junior Member
Location:
Posts: 20
Hello Kempfand
Please demux bidule's 32bits float 6channel wave file with channelex from MCtools ( I think You know). Channelex
supports 16,24,32 bits interger and float.You will see that Bidule is OK. Then You can Wavegain to loudness
normalize 6 mono file as a album.
Please check it with me.
THX

kempfand
Senior Member
Posts: 194
@puzio: This would imply the "black sheep" with BeSweet. However, it still could be that Channelex is more 'sloppy'
and accepts the 32-bit wav as 32-bit, whereas BeSweet recognises the 32-bit wav as 16-bit.
I'll check this evening.
quote:
Then You can Wavegain to loudness normalize 6 mono file as a album
I'll 1st need a new harddisc then (just kidding).
Kind regards,
Andreas

kempfand
Senior Member
Posts: 194
@DSPguru: So far, I've not reliably managed to split the 32-bit wav. Here is the log-file as far as it goes:
code:
M:\>Besplit -core( -input IMG_32.wav -prefix Check- -type wav -a ) -split( 0 20 40 )
BeSplit v0.9b6 by DSPguru.
--------------------------
[00:00:00:000] +------- BeSplit -----
[00:00:00:000] | Input : IMG_32.wav
[00:00:00:000] | Channels Count: 6, Bitrate: 8467kbps
[00:00:00:000] | Output Prefix : Check-
[00:00:00:000] +---------------------
[00:00:00:000] | Starting address = 0x2c.
[00:00:20:000] | Writing Check-01.wav
M:\>
This produces an 11 kb "Check-01.wav", which I cannot however attach (..."you may not post attachments"...).
When I tried to open the file in CoolEdit Pro, it only reads up to 10 min 54 sec (same time as in the logfile posted
further above). The same story when opening in Sonic Foundry Soft Encode: It opens 6 mono-wav's of 10 min 54
sec.
===
On the channel-mapping front: This is really strange. I did a small example (3 min 40 sec) and Bidule-saved as 32-
bit wav.
When I open this in SF Soft Encode, the 4th channel (labelled "channel 3", as numbering starts with 0), which
should be the LFE, is empty. This is correct, and implies to me that Bidule writes the file correctly.
However, running
code:
--------------------------
Logging start : 12/19/03 , 21:49:30.
C:\BeSweet\BeSweet.exe -core( -input m:\01_BiD_32.wav -output m:\01_BiD_32-New- -6chfloat -
logfile C:\BeSweet\BeSweet.log ) -ota( -g peak )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : m:\01_BiD_32.wav
[00:00:00:000] +---------------------
Logging ends : 12/19/03 , 21:49:46.
results in 01_BiD_32-New-LFE.wav with sound, and in 01_BiD_32-New-SR.wav without sound.
Don't know what's going on here
Kind regards and happy Xmas to you and the ones close to you,
Andreas

Shayne
Senior Member
Posts: 193
is the gain function working in the latest beta?
clipping still at the beginning of center channel and some clipping in others (not all) not at start but at .0004
seconds?it appears to be getting better.
getting funny pin outs, has something changed?
thank you again for your great work
peace

kempfand
Senior Member
Posts: 194
Just a "wild" idea: Try un- & re-installing your soundcard SW and drivers... Maybe this helps.
Good luck,
Andreas

puzio
Junior Member
Location:
Posts: 20
Hello All.
Sorry but I want to tell You that BESWEET DOESN'T SUPPORTS 32BITS (INTERGER OR FLOAT) INPUT OR OUTPUT
FILES (2CHANNEL OR 6CHANNEL). It's the result that Output mapping and Output quallity is no good.
The second, don't use BESWEET to gain output files because it's only peak normalization. You can not be sure that
every songs will have the same loudness.
Please remember as EoH says, when We convert stereo2dts5.1 We need quallity and only quallity. We are happy
that Bidule has offline now and I thing is enough for us.
THX
@DSPguru sorry.

kempfand
Senior Member
Posts: 194
@puzio: BeSweet & 32bit worked fine with Bidule_0.6, using -6chfloat and -ota( -g peak ). It looks like the problem
here is not BeSweet. As mentioned above, it could easily be a change in Bidule output of 32-wav. We'll see
@ All: Going with BeSweet is still my current recommendation. For the moment, you can safely Bidule-write 16-bit
wav's, and process these as outlined in the guide. BeSweet has the advange of
- directly converting into dts_wav
- ssrc-upmix to 48 kHz for feeding an AC3-encoder
- optimized usage of disc-resources and speed
Regards,
Andreas

Shayne
Senior Member
Posts: 193
quote:
Just a "wild" idea: Try un- & re-installing your soundcard SW and drivers... Maybe this helps.
Good luck,
Andreas
Well it was a good day my new amp came in, i am getting dts spdif no problem with my sound card and it appears
that DSPguru is working on 32 bit float support.
How would reinstall soundcard drivers increase the peak of a wave form viewed through cool edit? and whats is SW?
Thanks
Peace

daphy
Miles Freak
Location: Germany
Posts: 175
again softencode for splitting
@Shayne
Hi Shayne,
you had mentioned that SoftEncode could split the 6WAV file
now back to the questions:
-channel mapping
how do I find the right order out of this?
Is this the right setting?
1=FL
2=FR
3=C
4=LFE
5=SL
6=SR
-saving as single 6 monoWAV
I havent found the necessary option to save those single files, maybe you can give me/us a hint!
THX
CYA Daphy
Last edited by daphy on 20th December 2003 at 16:23

kempfand
Senior Member
Posts: 194
@Daphy:
quote:
Is this the right setting?
This depends on how you set up your Bidule wireing. If you did as
for BeSweet in the Guide , then yes, this is the correct setting.
@Shayne:
quote:
i am getting dts spdif no problem with my sound card
I guess this applies to files with *.dts-extension, not for dts_wav (with *.wav-extension).
quote:
How would reinstall soundcard drivers increase the peak of a wave form viewed through cool edit? and
whats is SW?
As I said: It was just a "wild" suggestion, not a sulotion. But I've experienced this kind of procedure to help (even if
it wasn't "logical").
Kind regards,
Andreas

Shayne
Senior Member
Posts: 193
@ Daphy
When you open an ac3 file it creates 5 pcm Waves in the temp directory. I said it not sweet.
ADD gains 3 (doubles > 10 db) to your bidule graph. Pin your fronts Through 1 gain, Your rears through the 2nd and
the center through the 3rd.
look at your Pentagon for the correct pin outs of the Emigrator into the gains. reduce one gain c,r or f and create a
wave. Split it with besweet and open besweet split waves in soft encoder and view the wave forms. Ones you have
reduced (if enough) should be noticeable. Repeat to see the trend, shifting gain out around on your recorder till you
get besweets format.
@ kempfand
I will try your guide thank you, is gains (- g ..) working for you at 6ch float? its not here.
Well we reinstalled the system here which included the sound card and same problems exist.
dts created with sur dvd pro and feed through the spdif of power dvd .... yes i have typed in a dts extension, amp
reads dts through the optical in.
Peace

DSPguru
BeSweet Author
Posts: 3446
kempfand had sent me header & footer samples of latest bidule's output.
from some reason, the new waves doesn't hold a peak chunk, and it's disappointing. i hope this can be fixed.
regarding large waves - it works for me, and i can't find a clue for why it doesn't work for others.
regarding the channels drift - it's due to the change in latest bidule's wave output, it's header is different from old
one's waves.
should be fixed in attached build.
__________________
Last edited by DSPguru on 21st December 2003 at 18:07

kempfand
Senior Member
Posts: 194
quote:
gains (- g ..) working for you at 6ch float? its not here
Not working here either. Currently not
It worked with the early 0.6 version of Bidule though. DSPGuru is aware of the problem and I'm sure he will help for
a solution.
Let's not forget that all this is beta-versins, work-in-progress. "Where no men has gone before" (so to speak in
Trekkie-language).
I realise you prefer 32-bit-out, and I fully respect this. Still I argue that the key thing is to have the intermediate
steps (Bidule, VST) do their work in 32-bit. This is based on extensice listening-test EoH and I did, and you can also
read this guidance from audio-experts (i.e. A. Farina) in other boards.
About the noise problem in the header: Sorry I can't currently help further Only "quick & dirty" suggestion is to
add a 2-sec-wav to the beginning of your track, and cut it out after all processings (i.e. before you park into dts or
ac3).
Regards,
Andreas

Pages (3): 1 [2] 3 Last Thread Next Thread

Doom9's Forum > General > Audio encoding > 6ch Wave to dts with BeSweet (thread split)
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Author Thread
kempfand
Senior Member
Posts: 194
DSPGuru: Doesn't work. Tested on 2 independent tracks. Logfile for one:
code:
--------------------------
Logging start : 12/21/03 , 01:11:57.
C:\BeSweet_T\BeSweet.exe -core( -input h:\01_BiD_32.wav -output p:\32-DeMux- -6chfloat -
logfile C:\BeSweet\BeSweet.log )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : h:\01_BiD_32.wav
[00:00:00:000] +---------------------
Logging ends : 12/21/03 , 01:12:07.
Resulting LFE has audio (and shouldn't), resulting SR has no audio (and should).
Just to be on the safe side: I Bidul'ed with the latest version posted by David here in the forum (dated 12/14/2003, in
the RAR modified 8:30:40 PM). Not that I imply it matters, but one never knows.
I could upload a 2 sec Bidule 32-bit-wav (ca. 2 MB, start-to-end) to the newsgroups (i.e. alt.binaries.test). Let me
know.
Kind regards,
Andreas

Shayne
Senior Member
Posts: 193
well i went through the above procedure and here are the results
pin 1 ====> pin 1 C
pin 2 ====> pin 5 FL
pin 3 ====> pin 3 SL
pin 4 ====> pin 4 SR
pin 5 ====> pin 6 FR
pin 6 ====> pin 2 LFE
This is a for sure based on the process of elimination

kempfand
Senior Member
Posts: 194
DSPGuru: Great job I think we are getting there.
As far as I can judge, pure de-muxing is working, both for 16/32-multichannel wav's. I.e.
code:
--------------------------
Logging start : 12/21/03 , 21:14:18.
C:\BeSweet_T\BeSweet.exe -core( -input i:\01_32.wav -output i:\01_32-DeMux- -6chfloat -logfile
C:\BeSweet_T\BeSweet.log )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : i:\01_32.wav
[00:00:00:000] +---------------------
Logging ends : 12/21/03 , 21:14:42.
Resulting LFE.wav is empty, and resulting SR.wav has audio, both as they should.
Remarks & Additional Findings to Gaining:
(1) Peak chunk data: As I would expect, adding -ota( -g peak ) doesn't change gain (the current Bidule-out-wav's
simply lack the info : Red-mark for Plogue ). This applies for both 32 & 16-bit-multichannel wav's.
(2) Demuxing a 32-bit-multichannel into 16-bit-out (using "-6ch") seems to work fine, both with "-g max" and "-g
peak".
(3) Demuxing a 16-bit multichannel into 16-bit-out (using "-6ch") also seems to work fine, both with "-g max" and
"-g peak".
(4) Further on: Demuxing into 32-wav-out ("-6chfloat") and gaining with "-ota( -g peak )" works fine . This applies to
both 32- & 16-bit-multichannel-wav's to start with.
(5) @all: Note that my report on (5) was wrong. It probably works fine on other systems, and only produced 'scratchy'
sound on my system when played through WinAmp.
Playing through CEP or Zoomplayer is perfect without any 'scratches'.
Lastly, the following 2 combinations produce a really 'scratchy' sound:
--> 32-bit-in demuxed to 32-bit-out with -ota( -g max )
[ i.e. C:\BeSweet_T\BeSweet.exe -core( -input i:\01_32.wav -output i:\01_32-DeMux- -6chfloat -logfile C:\BeSweet_T
\BeSweet.log ) -ota( -g max ) ]
--> 16-bit-in demuxed to 32-bit-out with -ota( -g max )
[ i.e. C:\BeSweet_T\BeSweet.exe -core( -input i:\01_16.wav -output i:\01_32-DeMux- -6chfloat -logfile C:\BeSweet_T
\BeSweet.log ) -ota( -g max ) ]
In other words: Whenever demuxing into 32-wav-out ("-6chfloat") and gaining with "-ota( -g max )", the
resulting mono-wav's are 'scratchy'.
question on next steps deleted
Hope this little feedback was not too confusing and hope I got it right.
Kind regrads,
Andreas
Note: later edits above marked in Green
Last edited by kempfand on 22nd December 2003 at 00:05

DSPguru
BeSweet Author
Posts: 3446
4) "-g peak" works fine and this is my advised commandline, so why don't you use it ?
5) may i see the logfiles ?
__________________

kempfand
Senior Member
Posts: 194
quote:
4) "-g peak" works fine and this is my advised commandline, so why don't you use it ?
It's the preferred commandline here too, but it (currently) doesn't normalise/gain-adjust (if I understand correctly). Not
that it really matters, as it's just a question of turning up the volume a bit when playing.
quote:
5) may i see the logfiles ?
Logfiles follow below, but 1st let me state, that I made a mistake here, and reported an error too quickly. The sound is
'scratchy' when I play through winamp v2.9. This is what I usually do.
However, I now openend the resulting mono-wav's in CEP as well as ZoomPlayer. Surprise surprise: It
plays perfectly. I don't know what's going on here, but it means for me avoiding WinAmp in the future. Sorry for that
(I'll change my post above).
Logile 5a:
code:
--------------------------
Logging start : 12/21/03 , 23:42:28.
C:\BeSweet_T\BeSweet.log ) -ota( -g max ) -profile( ~~~~~ Default Profile ~~~~~ )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : i:\01_32.wav
[00:00:00:000] +---------------------
Logging ends : 12/21/03 , 23:42:42.
Logile 5b:
code:
--------------------------
Logging start : 12/21/03 , 23:45:46.
C:\BeSweet_T\BeSweet.log ) -ota( -g max ) -profile( ~~~~~ Default Profile ~~~~~ )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : i:\01_16.wav
[00:00:00:000] +---------------------
Logging ends : 12/21/03 , 23:46:01.
In summary, it works as it should. I didn't test test the 2GB/4GB-thing, but this is no big deal, as slighly over 66 mins
will fit into a biduled 16/32-output (when respecting a 2GB/4GB-barrier).
Again many thanks for the quick adjustments in this busy time.
Take care,
Andreas
22nd December 2003
00:00

DSPguru
BeSweet Author
Posts: 3446
this is good news. cheers, kemp!
note : i updated the b24 archive.
it is now safe to always use the "-g peak". if BeSweet will find that the PEAK chunk is missing, it will use "-g max".
(so the sound will be normalized anyhow)
btw,
please encourage david to re-enable the "peak" chunk feature.
__________________

kempfand
Senior Member
Posts: 194
Cheers, DSPGuru. It's really a pleasure to work with you.
quote:
if BeSweet will find that the PEAK chunk is missing, it will use "-g max".

Take care for now,
Andreas

daphy
Miles Freak
Location: Germany
Posts: 175
confirm
quote:
well i went through the above procedure and here are the results
pin 1 ====> pin 1 C
pin 2 ====> pin 5 FL
pin 3 ====> pin 3 SL
pin 4 ====> pin 4 SR
pin 5 ====> pin 6 FR
pin 6 ====> pin 2 LFE
This is a for sure based on the process of elimination
I can confirm at least pin 2 is definitivly LFE

Shayne
Senior Member
Posts: 193
my results on the new beta 24B (above)
Clipping at the beginning of waves is gone here
g peak normalization is working
Pin out back to old (daphy check kempfand guide jpg previous page)
Thanks for the update ...... U the Man DSPguru
same beta number is a bit confusing with such a change
Peace

DSPguru
BeSweet Author
Posts: 3446
Cheers for the confirmation daphy & Shayne !
a new version number would be applied whenever more bugfixes/features are introduced
__________________

Shayne
Senior Member
Posts: 193
quote:
[B]Hello All.
Sorry but I want to tell You that BESWEET DOESN'T SUPPORTS 32BITS (INTERGER OR FLOAT) INPUT OR
OUTPUT FILES (2CHANNEL OR 6CHANNEL). It's the result that Output mapping and Output quallity is no
good.
I think we need to 5hit this man down beside my denon and ask if its 32 bit or not.
92 ripped since the latest beta2, Norm OK here. spot checks and 100% listen lfe gain -5 DB .... purrrfect ? 1010 to
go ....... batchen.
Thinken cd sales r going to pick up here again. KooL
Peace and Happiness

daphy
Miles Freak
Location: Germany
Posts: 175
besweet 1.5 b25
@DSPGURU
we tried several times to split a 6WAV with correct mapping (without success) - then I found the latest besweet
version (1.5b25) on the beta section - and all of our problems are gone
genius work!
Happy New Year
CYA Daphy

Shayne
Senior Member
Posts: 193
Have you noticed that besweet will crash every time when processing a file with "(" or possibility ")" in its title.
Gui ===> Batch dts (Sur dvd) ===> multiple to multiple
I presume it is the nature of the dos command and not to much you can do about it except avoid () in titles. Would " "
work?
Funny how it goes .... about the same number of DTS fit on one DVD as did MP3 of the CD's in 95.
Peace
1st March 2004 01:27

Shayne
Senior Member
Posts: 193
Since i got such a loud response from my last post here maybe i will ramble on some more. It appears to me that with
Ambisonics 2ch to 5.1 the tricky thing is the LFE and its amplitude during the normalization process. I have a question.
Would it be possible to break the waves into 5.1 mono files and then normalize the 5 as they are dependent (equal
amplitude increase) and the .1 (lfe) independent? This would for sure be the fix for the lfe channel and give the best
result.
Peace
Edit typo
Last edited by Shayne on 11th March 2004 at 12:38
10th March 2004 01:42

kempfand
Senior Member
Posts: 194
A quick (but dirty) solution would be to demux your 6-ch-wav twice (into different directories).
-> once without normalisation : just keep the LFE, and delete the other channels
-> once with normalisation : delete the LFE, and keep the 5 main channels
Again this only applies to the pure Ambisonic, not AmbioPhonic (Stereo Dipole, HRTF etc) stuff.
Andreas
11th March 2004 14:51

Shayne
Senior Member
Posts: 193
This should work fine!
The dirty would be more than twice as long to do. You would have to run all waves through both programs twice and
could not use .bat file pass through in besweet.
The quick? i really don't see anything in the process you mention as being quick.
I guess i was kinda wondering if its possible to implement into besweet and if anyone else thinks it would be valuable
switch.
Peace
14th March 2004 23:54

kempfand
Senior Member
Posts: 194
Have you tried to use the "gain" VSZ inside bidule (you will need more than one, because it only has 2 pins in & out).
Then setting the main channels as they "sound" OK on the PC speakers, using a gain for the LFE you knoe is working
from your experience.
Then runninh the bidule in offline mode, and BeSweeting w/o gain.
Sorry I cannot help more, as I don't use an LFE.
Cheers,
Andreas
15th March 2004 00:12

Shayne
Senior Member
Posts: 193
Yes that is how i am currently controlling the lfe channel but gain does not equal normalization.
Peace
15th March 2004 02:23

Shayne
Senior Member
Posts: 193
40 gigs n counten
Golden Earing_Big Tree, Blue Sea.dts
Golden Earing_Vanilla Queen.dts
I find ring interesting
Bump GUI batch "file.name" Quote avoid batch crash plz
Norm .1 independent 5. Dependant ??? DSPguru U the man
Peace its time we would all be friends
Ramble on
4th April 2004 04:26

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