GUIDE: Stereo to 5-Channel Surround

[Eye of Horus]

removed

[Sycho]

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

[bitsnbytes]

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

[kempfand]

@ 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 2 = Front Left
Channel 3 = Surround Left
Channel 4 = Surround Right
Channel 5 = Front Right
Channel 6 = LFE -empty

Andreas

[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 (http://www.niederma.de/freeware/Audio/VST/HNM_Filter.zip).

i integrated it into the processing chain as shown on this picture (http://downloada.in-berlin.de/hnmfilter.gif). 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 (http://downloada.in-berlin.de/hnm_lowpass80.fxb). 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

[downloada]

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

[kempfand]

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 (http://www.muse.demon.co.uk/ref/speakers.html) suggest a LFE.

As said: We are very keen to learn how it compares.

Kind regards,

Andreas

[KpeX]

@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 :sly:. I will definitely have to add an LFE channel :D for my musical taste; haven't experimented with that yet.

Cheers,

[DSPguru]

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 ?

[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?

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!

[kempfand]

@ DSPGuru:

A. you can remap the channels in the graph That is indeed the most easy & elegant solution :) Why didn't we think at that before :devil: :o :devil: 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.

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

[downloada]

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

[downloada]

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

[daphy]

Hi,

I´ve done so much testings and comparisons between the SAD5.1, Kpex and this method (->EOH whats the new name for this :D) and for me ambisonic system wins the match :rolleyes:
The addon of the LFE is definitivly a must have because it´s really the bass what misses.

For those who want to save their time I´ve uploaded the complete VSTPlugins (the same as in the guide + lfe patch) (http://needfulthings.webhop.org/) 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 :D ;) :D
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 :cool:

[kempfand]

whats the new name for this [method]
- Visual Ambisonics :D :cool:
- Hitchhiker's Guide to Ambisonics :p

Andreas

[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.

[daphy]

Hi folks,

found a bug in this routine, don´t 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 don´t 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 :scared:

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 -> I´m shure this can´t be the reason , also Surcode isn´t guilty because the 6 mono files already have this damage

last but not least it must be Bidule itself! I´ve got this error two times EOH shurely not, so maybe it depends on the lfe transcoding :confused:

I´ll 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

[kempfand]

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

[daphy]

thx for the suggestion

I added this profile to the layouts
(see the connections) (http://needfulthings.webhop.org/)
If this is the only problem - nevermind :D

Fine toy!
I´m 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

[SallyDog]

Originally posted by daphy
Hi folks,

found a bug in this routine, don´t 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 don´t 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 :scared:

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 (http://www.daphy.de/ambisonic/error.gif)

Bug search:

I tried the new Besweet 1.5 B20 and the old 1.4 -> I´m shure this can´t be the reason , also Surcode isn´t guilty because the 6 mono files already have this damage

last but not least it must be Bidule itself! I´ve got this error two times EOH shurely not, so maybe it depends on the lfe transcoding :confused:

I´ll 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

[erymax]

Can you post a link for the latest version ?

thanks



erymax

[daphy]

@ erymax

read the posting right from the start
;)

CYA Daphy

[erymax]

@daphy

thanks, I've got it now, sorry for my mistake.

erymax

[Sycho]

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

[kempfand]

@ 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 (http://www.dmalham.freeserve.co.uk/bpan_help.html) . 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

[bitsnbytes]

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 :D

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 :sly:

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

[bitsnbytes]

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 :-)

[kempfand]

@ 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

[Eye of Horus]

Originally posted by daphy
thx for the suggestion

I added this profile to the layouts (http://www.daphy.de/ambisonic/2.0to5.1_soundmapper.bidule) (use save as.. option).
(see the connections) (http://www.daphy.de/ambisonic/verbindun.gif)
If this is the only problem - nevermind :D

Fine toy!
I´m 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

[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.:D

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.

[Eye of Horus]

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.:D

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]

@EoH:
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 (http://www.muse.demon.co.uk/utils/ambidec.html) does (where the -g switch is disabled by default).

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]

Today I did so many experiments...... I lost count :D

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.asp?fTitle=&fAuthor=&fType=&fMoreLess=0&fPrice=1&fVST=true&fPC=true&fDX=&fMac=&fSort=Author&fPe=50&fFreeCom=&fPlugin=&Submit2=Submit

For those who are real ambitious : make your own VST !
http://www.steinberg.net/en/ps/support/3rdparty/vst_sdk/download/index.php


EoH

[bitsnbytes]

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.

[kempfand]

@ 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 :devil: :scared: LOL

Andreas

[Eye of Horus]

Originally posted by bitsnbytes
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]

Originally posted by kempfand
@ 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 :devil: :scared: LOL

Andreas

Nope :rolleyes:

Author of Bidule said : 48 Khz before using it ! But..... don't forget you also have to change in the settings-menu !!

EoH

[bitsnbytes]

ok got this sucker workin now:cool:

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 2 = Front Left
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 ;-)

[kempfand]

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

[kempfand]

@ EoH / all: Today I did so many experiments...... I lost count

More food: :D

B-Plane Mirror (http://www.york.ac.uk/inst/mustech/3d_audio/vst/bfpmirror_help.html)

B-Mic (http://www.york.ac.uk/inst/mustech/3d_audio/vst/bfmic_help.html)

Bpan3 - Third Order Ambisonic Encoder (http://www.york.ac.uk/inst/mustech/3d_audio/vst/bpan3_help.html)

Let's test :cool:

Andreas

[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?

[TerraForce1]

Hi, well as the post subjects says it all:

I love the way your mind works. Meaning you are right.

Thx.

[daphy]

@ 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 :D

CYA Daphy

[TerraForce1]

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

[kempfand]

@ 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 :eek:

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 (http://www.muse.demon.co.uk/ref/speakers.html) . Keep us posted on your preference.

Regards,

Andreas

[specise_8472]

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:confused:

[bitsnbytes]

Originally posted by specise_8472
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:confused:


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]

Originally posted by specise_8472
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:confused:

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]

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:cool:

[specise_8472]

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.

[kempfand]

TapeIt (http://www.silverspike.com/Download/TapeIt.zip)

[bitsnbytes]

tapeit worked worse than the on board recorder

rendering 1kb files when configured 6 times

[bitsnbytes]

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]

its microsoft acm waveform mpeg layer 3?

I always thought bidule works only with non compressed files?
Use for example cooledit´s save as option to save into a non compressed WAV (PCM 16bit 48/44.1khz)

CYA Daphy

[TerraForce1]

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

[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

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]

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]

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]

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 ??


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.)








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 !!!!


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....



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 ! :D


kind regards,

EoH

[davidv@plogue]

>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.

[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.

[Eye of Horus]

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]

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]

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

[kempfand]

@ 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: 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.

Regards,

Andreas

[bitsnbytes]

Originally posted by TerraForce1
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

[DSPguru]

Hello David, Welcome,

Originally posted by davidv@plogue
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 :)

Originally posted by davidv@plogue
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.


Originally posted by davidv@plogue
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.

Originally posted by davidv@plogue
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 :D


Cheers,
Dg.

[bitsnbytes]

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 :cool:

[specise_8472]

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]

Originally posted by bitsnbytes
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 (http://forum.doom9.org/showthread.php?s=&threadid=15738).
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]

@ 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]

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.

Originally posted by Eye of Horus
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):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]

/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]

Originally posted by DSPguru
/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 ?

Please don't forget about us 48 Khz AC3 fans !!!! :D

How about the Besweet line for encoding with the commandline of Softencode ????????????

EoH

[DSPguru]

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]

Originally posted by DSPguru
/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 ?

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]

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]

Originally posted by DSPguru
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]

Originally posted by DSPguru
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

[bitsnbytes]

:D

[specise_8472]

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????:rolleyes: (I only attack this project when in the mood as it does not seem to be as easy as first thought:( )

[DSPguru]

what's going on with your brackets :o ?
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)

Originally posted by Eye of Horus
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]

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 :D )

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]

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 :mad: !) 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]

Originally posted by Eye of Horus
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 (http://forum.doom9.org/showthread.php?s=&threadid=60907).

Cheers :),
Dg.

[bitsnbytes]

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/download_freeware.html


A filter info page from above site:

http://www.arboretum.com/support/manuals/manual_hvst/Files/hppc_proc_filters.html

[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 )

[Eye of Horus]

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

[ilmanu]

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]

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]

Originally posted by ilmanu
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

[ilmanu]

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]

Originally posted by ilmanu
yes after....but i have found the problem.....i made a 16bit wav 2ch and try to convert in 5.1 24bit...

OKAY :D

EoH

[ilmanu]

:D
well, my english is good like your italian :D :D :D

[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...)!

:cool:

[Eye of Horus]

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...)!

:cool:

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

[TRILIGHT]

Thanks. I finally got it running with this command line...
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!!

[kempfand]

@ 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 (http://www.ambisonic.net/)

Martin Leese's FAQ page (http://members.tripod.com/martin_leese/Ambisonic/)

Ambiophonics (http://www.ambiophonics.org/index.htm)

Technical Papers (http://www.ambiophonics.org/TechnicalPapers.html)

Kind regards,

Andreas

[bitsnbytes]

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 :D

[Eye of Horus]

@bitsnbytes

Any new bidules ???

please post them......


TIA.....


EoH

[bitsnbytes]

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 :sly: . 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 :D

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]

Originally posted by bitsnbytes
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 :sly: . 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 :D

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 ! :D

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

[TRILIGHT]

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]

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 :D

For sure, the 'Mirror type' with B_Pan_m is sth to experiment with, as is B-Plane Mirror (http://www.york.ac.uk/inst/mustech/3d_audio/vst/bfpmirror_help.html) , as well as 'improvement' of Quad-recordings.


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

[joshbm]

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 (http://www.videocdpro.com/josh/download.aspx?file=5.1 Surround.bgrp)
6 Channel Gain (http://www.videocdpro.com/josh/download.aspx?file=6ch Gain.bgrp)

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 (http://www.videocdpro.com/josh/setup1.gif), you get a nice group like this (http://www.videocdpro.com/josh/setup2.gif) 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

[davidv@plogue]

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]

Originally posted by davidv@plogue
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]

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
6 > 4 = LFE (an empty wav)
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

[Eye of Horus]

Originally posted by TRILIGHT
Thanks. I finally got it running with this command line...
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.... :rolleyes:

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

[Eye of Horus]

Originally posted by bitsnbytes
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
6 > 4 = LFE (an empty wav)
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]

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]

Originally posted by joshbm
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 (http://www.videocdpro.com/josh/download.aspx?file=5.1 Surround.bgrp)
6 Channel Gain (http://www.videocdpro.com/josh/download.aspx?file=6ch Gain.bgrp)

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 (http://www.videocdpro.com/josh/setup1.gif), you get a nice group like this (http://www.videocdpro.com/josh/setup2.gif) 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 :D ) and the LFE was completely clipped..... really terrible !

Just FYI....

Any idea ?

EoH

[Eye of Horus]

Originally posted by TRILIGHT
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.



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......


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 :confused:

kind regards,

EoH

[Eye of Horus]

Originally posted by Eye of Horus
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 :D ) 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]

Originally posted by Eye of Horus
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]

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]

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]

The updated version can be redownloaded from the first link I gave you:

5.1 Surround (http://www.videocdpro.com/josh/download.aspx?file=5.1 Surround.bgrp)

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]

Originally posted by Umma
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]

"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]

Originally posted by Umma
"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]

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]

:rolleyes:

@Umma

that´s easy:

1. use Exact Audio Copy V0.95 prebeta 3 (http://www.exactaudiocopy.de/eac6.html)(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 (http://www.dcsoft.com/prod03.htm) or a simple Texteditor to make your own CUE in the EAC stil - you will find out, it´s not that difficult ;)

4. burn the CUE as CDIMAGE with NERO/CDRWIN... (howto and further informations (http://www.dcsoft.com/cue_mastering_progs.htm))


-> 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

[Calabuig]

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]

EXCEPTION_ACCESS_VIOLATION

Could you send the full Bidule.log to support@plogue.com?

Cheers

[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

[Eye of Horus]

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 !!!! :D
(Read The F.....g Thread !)

But I won't.........

Joshbm posted a 5.1 group !

kind regards,

EoH

[bitsnbytes]

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

[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 )
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]

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 )
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.

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]

@ EOH

I did much encoding last week -> it now works perfect for me without LFE (you´re right, no need of this hn-filter) :D :D

I also managed to integrate the results in DTS to VOBs - this wasn´t a nice job, ´cause programs like IFOEDIT doesn´t do the expected actions (wasting sooo much time on it) :angry: so I had to switch to a commercial program (maybe I will write a guide for this :rolleyes: ).

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 won´t accept them. Until now I don´t 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 :o this means less than ~5GB

CYA Daphy

[daphy]

I've had the same problems lately with files larger than 2 GB.
I used WAV2WAV6.EXE (http://needfulthings.webhop.org/) 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



like:

c:\wav2wav6.exe 6wav.wav :confused:

I guess somethings missing

I haven´t tried this so far, but I remember this as part of the former ambi-guide :rolleyes:

CYA Daphy

[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. :D :D

SallyDog

[Eye of Horus]

Originally posted by daphy
I guess somethings missing

I haven´t tried this so far, but I remember this as part of the former ambi-guide :rolleyes:

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]

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. :D :D

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]

Originally posted by daphy
@ EOH

I did much encoding last week -> it now works perfect for me without LFE (you´re right, no need of this hn-filter) :D :D

I also managed to integrate the results in DTS to VOBs - this wasn´t a nice job, ´cause programs like IFOEDIT doesn´t do the expected actions (wasting sooo much time on it) :angry: so I had to switch to a commercial program (maybe I will write a guide for this :rolleyes: ).


@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]

@EoH

See the BeSweet FAQ thread in this forum, question #30.

Regards,

SallyDog

[rasta21]

Originally posted by Eye of Horus
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

...i think you have to change some front to center, too.

fr > c
c > fr

...or fl...if you prefer this way

[TerraForce1]

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.

[daphy]

Originally posted by daphy
I guess somethings missing

I haven´t tried this so far, but I remember this as part of the former ambi-guide

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 can´t use this with greater file size, what means we stick at besweet and the 2GB barrier. :(

[daphy]

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)
-> I´m 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 :angry:

1. open the app and drag´n´drop 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 :rolleyes:

2. Now drag´n´drop 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, don´t forget to use a nice splitting graphic :D

That´s it

For the first take calculate a weekend on a two hour film ;)

Any suggestions - or ideas to explain it easier?

CYA Daphy

[DSPguru]

Originally posted by daphy
I did some testing with wav2wav6.exe -> result 32bit/48KHz splitting ended always in 6 empty wavs :( so we can´t use this with greater file size, what means we stick at besweet and the 2GB barrier. :( your attention to q30 (http://forum.doom9.org/showthread.php?s=&threadid=15738).. ;)


@EoH
any chance that you update your guide according to this (http://forum.doom9.org/showthread.php?s=&threadid=60907&perpage=5&pagenumber=6) ?

[Eye of Horus]

Originally posted by DSPguru
your attention to q30 (http://forum.doom9.org/showthread.php?s=&threadid=15738).. ;)


@EoH
any chance that you update your guide according to this (http://forum.doom9.org/showthread.php?s=&threadid=60907&perpage=5&pagenumber=6) ?

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

[daphy]

@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, you´ll 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 :rolleyes: 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 wouldn´t 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 :confused: :confused: :confused:


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 wouldn´t 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 :cool:

[Calabuig]

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]

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]

16bit - 48Khz

Wav2wav6 is ok, but I'm in a mess with the resulted files' names...

[Eye of Horus]

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]

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]

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.

[Eye of Horus]

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

[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?

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?

[Eye of Horus]

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 ....


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 !

[24hourloop]

Great, thanx.

The DVD Architect method looks promising ;-)

[Umma]

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.

[Eye of Horus]

Originally posted by Umma
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

[vtwin0001]

Can somebody please point me out where can I find Wav2wav6.exe ???

I couldn't find it here nor in google.

Thanks.

[davidv@plogue]

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?file=PlogueBiduleInstaller6500test1.exe

Please try it out if you experienced problems.

Davidv

[Eye of Horus]

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 :
http://akson.sgh.waw.pl/~wj23277/wav2wav6.zip

EoH

[vtwin0001]

Thank you, I guess I didnt searched that well...

:D

[Boulder]

Originally posted by davidv@plogue
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?file=PlogueBiduleInstaller6500test1.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?

[Eye of Horus]

@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

[davidv@plogue]

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

[davidv@plogue]

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

[Boulder]

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.

[Vidi_It]

Originally posted by davidv@plogue
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.:)

[davidv@plogue]

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/rim456/Quad/Quad_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...

[daphy]

sounds interesting :rolleyes:

hope you find the necessary time ;)

[Tantulus]

Originally posted by Eye of Horus
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?

[Eye of Horus]

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 :D

EoH

[echooff]

Has anyone run DSOTM thru bidule. Can anyone reccomend other albums that compare with sterotomy after bidule.

[Eye of Horus]

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

[Tantulus]

Originally posted by Eye of Horus
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 :D

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.

[TerraForce1]

Hi, is there a freeware .mpa vst-plugin avalaible. I would like to run some mp2 audio files trough bidule.

[davidv@plogue]

Originally posted by TerraForce1
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.

[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

[Umma]

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...

[echooff]

Thanx, I'm going to play with my new toys tonite! Anybody try tubular bells or any van halen?

[kempfand]

Gerzonic (http://www.gerzonic.net/) has published 1 update (Emigrator) and 2 new toys:

Panorama
Panorama is a 2nd Order Ambisonic Panner with a very simple user interface in the tradition of bPan.
InterGate8 Demo
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)
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

[ChristianHJW]

Please explain a bit more in detail ? What exactly can i do with those plugins ? Could it help us using Vorbis surround some day ?

[echooff]

Do a search on ambisonics in the forum. Several extensive threads on the subject.

[kempfand]

Btw: I meant to add this to the sticky-post / Ambisonic guide, so I made a mistake here :devil:

No big deal, but if one of the Moderators can move / end-add it there, the better.

Regards,
Andreas

[daphy]

Hi Andreas,

have you made some testings with the new emigrator? ;)

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 didn´t apear again since Bidule 0.6500test1 :rolleyes:

CYA Daphy

[kempfand]

No tests yet from my side on the latest built of emigrator :(
Too much at professional work-place currently ongoing for me :o

I'll wait with further tests until the final Bidule 6.5 is out, which is rumored to offer a warp-speed offline-mode :D

Andreas

[Tantulus]

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.:)

[kempfand]

You might want to try your preference from the following list. They have been reported in the quadraphonicquad forum section (http://www.quadraphonicquad.com/) 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) :sly:

Albums:

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

[echooff]

Thank you for the input. Gonna have fun tonite.

[Gradius]

I'm trying to do a 5.1, but I just get an empty wave on LFE channel.

Check my config here (http://www.digitalanime.com.br/config.jpg).

Please, someone post a clean LFE guide.

Gradius

[PeterC]

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.

:D

[iSpy]

I was just wondering where I could find the program wav2wav6.exe that is spoken of in this thread. Thanks, iSpy

[jorel]

link from Maxiuca posted here:
http://forum.doom9.org/showthread.php?s=&threadid=52692&perpage=20&highlight=wav2wav6.exe&pagenumber=2

maybe is working:
http://tommers.w.interia.pl/wav2wav6.zip

http://akson.sgh.waw.pl/~wj23277/wav2wav6.zip

:)

[Umma]

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. :D

[Eye of Horus]

Originally posted by Umma
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. :D

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]

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]

Bidule's home page says that it is mostly for Win2K though some people have gotten it to work with Win98. :(

[Umma]

So, uh...when is the new Bidule coming out?

[davidv@plogue]

Originally posted by Umma
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]

Originally posted by davidv@plogue
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 !!!!! :devil: :D :D

Eye of Horus

[Umma]

Thanks for the reply, David. I can't WAIT for that offline rendering mode!

Yup. EoH already got the guide written up... :D

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.

[Eye of Horus]

Originally posted by Umma
Thanks for the reply, David. I can't WAIT for that offline rendering mode!

Yup. EoH already got the guide written up... :D

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]

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]

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 !:D

grtz,

EoH

[Shayne]

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]

Originally posted by Shayne
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]

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]

Originally posted by Shayne
<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 :D
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]

Originally posted by Eye of Horus
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]

Originally posted by Shayne
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

[Shayne]

@ 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

[Umma]

There is a thread at the Bidule site (http://www.plogue.com/phorum/read.php?f=1&i=1295&t=1295#reply_1295) that mentions the click at the beginning, but I'm not sure it applies here.

Might...

Might not... :)

[joshbm]

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

[joshbm]

Originally posted by davidv@plogue
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]

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 voilà a player that would definitely out play anything we have now. I am sure its thought of and in the works.

Peace

[joshbm]

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 (http://www.videocdpro.com/josh/dloads/surround_grp.zip)

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 (http://www.kellyindustries.com/products.html) 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 (http://www.york.ac.uk/inst/mustech/3d_audio/vst/yorkvst.zip)

3. Emigrator
Link : http://www.gerzonic.net/zips/Emigrator_1.0_win32.zip

4. Bass Manager
Link: http://www.kellyindustries.com/downloads/bass_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]

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]

The latest Bidule is out! :)

[davidv@plogue]

Originally posted by Umma
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]

Originally posted by davidv@plogue
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

Excellent David !
Can't wait to give it a try......
Thanks !

kind regards,

EoH

[Umma]

Much faster...MUCH faster!

I had stopped doing the conversions because of the time involved, but now...well, it begins anew! :D

Thank you very much, David! To you AND your team... :thanks:

[joshbm]

Originally posted by davidv@plogue
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

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]

Originally posted by joshbm
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:scared:

[Eye of Horus]

Originally posted by specise_8472
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:scared:

Sounds exciting !
Please keep us posted !

kind regards,

EoH

[joshbm]

Originally posted by specise_8472
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:scared:

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]

Originally posted by joshbm
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.

[joshbm]

Originally posted by specise_8472
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]

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]

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]

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]

Just tried it... No go :-\. Still had to do a cold restart.

Regards,
-joshbm

[Eye of Horus]

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]

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]

Originally posted by davidv@plogue
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]

@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]

Sorry but with the new version the offline process is slower than online.

[davidv@plogue]

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]

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]

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]

Originally posted by puzio
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 ! :D

(no pun intended !)

EoH

[puzio]

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]

Originally posted by puzio
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]

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]

Originally posted by davidv@plogue
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.

:D I understand.

Thanks for listening to my suggestion,
- joshbm

[daphy]

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.
That´s why I also did some further experiments with 5.1.

CYA Daphy

[Shayne]

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]

@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]

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]

@ joshbm

I´ve sent you a PM ;)

CYA Daphy

[kempfand]

@ joshbm:

I'd apprciate if you share your *.bidule(s).

Kind regards,

Andreas

[PeterC]

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]

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]

@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

[daphy]

THX :D

[Eye of Horus]

Originally posted by joshbm
@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 !! :D

Happy Christmas everyone !!

EoH

[daphy]

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) (http://needfulthings.webhop.org)

CYA Daphy & @ndy

PS: if the server is down, please send me PM

[kempfand]

@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: 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]

Some notes to the settings (see 2 attached JPG's for

Um...what two attached jpgs? :)

[kempfand]

The jpg's are within the _Panorama_-_Pentagon_.RAR I uploaded.

Andreas

[Shayne]

Just a follow up since i rambled on about this problem for so long in this thread. :o

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.php?s=&postid=417052#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

[kempfand]

From Shayne Dec 11: 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 voilà 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) (http://www.knufinke.de/sir/index_en.html). :D

A few (selected) details on the SIR-VST: - 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 (http://www.ramsete.com/aurora/conversion_between_uhj_and_b.htm#Audiomulch) 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 (http://forum.doom9.org/showthread.php?threadid=29277) , 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 (http://pcfarina.eng.unipr.it/public/B-format/UHJ-conversion)
- 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]

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]

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]

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]

Hi puzio,
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.

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 (http://home.wanadoo.nl/appyhappy/19.jpg)
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]

Has anyone every contacted Angelo Farina about us using his guidelines over here?

[kempfand]

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 ! (http://forum.doom9.org/showthread.php?threadid=29277) .

As I said: My native language is not English, so I might mis-read.

Andreas

[puzio]

.

[Umma]

Originally posted by kempfand
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...


- 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 ! (http://forum.doom9.org/showthread.php?threadid=29277) .


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]

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.

[Eye of Horus]

Originally posted by kempfand
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]

Originally posted by Umma
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]

Great news: I've just got mail from Angelo Farina :), which I share with his permission. 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 (http://www.angelofarina.it/aurora/conversion_between_uhj_and_b.htm#Audiomulch)

I also saved the Bidule patch on my Public server area: Link (http://www.angelofarina.it/public/B-format/UHJ-conversion/)
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 (http://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]

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. :confused:

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]

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]

specise_8472: Thanks for sharing this. This looks really :cool:

It's funny: Yesterday, I was just reading the corresponding paper (http://www.ambiophonics.org/AESJune2003/AES_June_2003_3.htm) 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]

Update: Specise's ambiophonics-VST ROCKS ! I just did a testing series with one song (Armik: Rubia; guitar music).

Created 9 tracks as follows:
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 (http://www.filmaker.com/) , especially
- Transforming Ambiophonic + Ambisonic 3D Surround Sound to & from ITU 5.1/6.1 (http://www.filmaker.com/papers/AES114_Transform.pdf)
- Scalable Tri-play Recording for Stereo, ITU 5.1/6.1 2D, and Periphonic 3D (with Height) Compatible Surround Sound Reproduction (http://www.filmaker.com/papers/AES115_Scalable%20Tri-play%20Recording.pdf)

@specise_8472:

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]

Originally posted by kempfand
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:D

[Ace_V]

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

[kempfand]

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 (http://www.dtsonline.com/media/uploads/pdfs/DTS-ES%20Overview.pdf).

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 (http://www.tronika.no/support/files/dts-es.pdf)

Update / Edit added below

Now: 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 :mad: (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.

---
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 (http://www.filmaker.com/2dsurround.htm) mentions this as 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 :D

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)

[specise_8472]

Okay, one at a time

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:(

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:D). I am still testing/playing around. A lot of this centers around crosstalk cancellation, which I am investigating.

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.

[specise_8472]

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:angry:

[@ndy]

@specise_8472
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 :D

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 :cool:

BTW: except of daphy and me has no one the (official) right to delete anything :D

Last note for this posting:

please upload your files with your nickname in the file file like ²Ambi-rc1_specise_8472.dll".

CU
Andy

[specise_8472]

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:D) 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.:D

[kempfand]

@ 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 (http://www.muse.demon.co.uk/ref/speakers.html) 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. :confused: Would appreciate if you clarify a bit.

(2) 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 :D

(3) Regarding DTS ES Matrix and 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 (http://forum.doom9.org/showthread.php?threadid=61203) 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 (http://forum.doom9.org/showthread.php?threadid=39044), where member 'S_O' talked about a small proggie 'ac3ex'. Might be useful here.

Kind regards,

Andreas

[specise_8472]

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.

[Umma]

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]

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:eek:
(Now I suppose everyone who wants will have to redo all their hard work:D - assuming anyone wants 6.1 audio cd's;)

Btw I am using a Marantz 7300ose amp for testing. Works perfectly everytime.

[@ndy]

Originally posted by specise_8472
[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 (ftp://daphy.mine.nu) : \ambiophonics and \incomming\ambiophonics.

Have fun

@ndy

[kempfand]

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"). :D

There also is a 6.1 dtswav at SR 5.1 MultiKanal (http://www.sr.se/multikanal/english/e_index.stm) 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).

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

[Umma]

(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... :sly:

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]

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]

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. :confused:

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]

@ All: If you want to get the latest Swiss Center / York Ambisonic VST's, go here: Please note - this is a temporary download page (http://www.dmalham.freeserve.co.uk/alpha/vst_ambisonics_alphas.html) . 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 :D . 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]

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]

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

[specise_8472]

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]

Originally posted by specise_8472
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 :D :D

grtz,

EoH

[DSP8000]

.

[Eye of Horus]

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]

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:devil:

[bobn4burton]

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]

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]

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

[specise_8472]

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:p

[daphy]

I have just uploaded a bidule to do reverse engineering of 5.1 sounds.
Read the Post 5.1 to WXYZ and Back (http://forum.doom9.org/showthread.php?s=&threadid=68646) to get a better idea what this is about.
I think with this link it´s 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

[dannyv]

Originally posted by specise_8472
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

[@ndy]

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 (ftp://daphy.mine.nu/ambiophonics)

@ndy

[kempfand]

It's always good to have the latest updates ;)

- Bidule v0.6502 : Bidule Site (http://www.plogue.com/bidule/)

- Swiss & York Ambisonic VST's January 13th, 2004: this is a temporary download page (http://www.dmalham.freeserve.co.uk/alpha/vst_ambisonics_alphas.html)

Cheers,
Andreas

[daphy]

- 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 (http://www332.pair.com/plogue/download/PlogueBiduleInstaller.exe) (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 ;)

[desertrat]

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

[specise_8472]

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.

[desertrat]

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

[trooper11]

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.

[Shayne]

@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

[xxx666yyy777]

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]

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,
Ândreas

[xxx666yyy777]

Originally posted by kempfand
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,
Ândreas

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]

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.
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 (http://www.york.ac.uk/inst/mustech/3d_audio/vst/bfpmirror_help.html) VST, which you might want to try or play around with.

Regards,
Andreas

[Mug Funky]

http://www.planetquake.com/filterfactory/3daudio.html

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...

[daphy]

http://www.planetquake.com/filterfactory/images/bidule.jpg
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 (http://needfulthings.webhop.org/)

CYA Daphy

PS: EOH doesn´t mean 'Eye Of Horas' ;)

[kempfand]

Daphy: The channel-mapping is correct in that "copied guide",
as that guide goes for ac3 using ac3enc.dll.

-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]

Originally posted by kempfand
Daphy: The channel-mapping is correct in that "copied guide",
as that guide goes for ac3 using ac3enc.dll.

Regards,
Andreas

AC3Enc.DLL ? Yuck !!! :D

Eye of HorUs !

[Eye of Horus]

Originally posted by Mug Funky
http://www.planetquake.com/filterfactory/3daudio.html

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 :D
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]

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 :D

Thanks again m8es !

Eye of Horus

[bitsnbytes]

Originally posted by Eye of Horus
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.php?s=&threadid=60137&perpage=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

[Mug Funky]

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]

Originally posted by bitsnbytes
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.php?s=&threadid=60137&perpage=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]

Just for your information:

I've created a few LCR-Upmix Bidule Groups, which you might find useful: LCR Upmix Bidules (http://forum.doom9.org/showthread.php?threadid=70678)

Cheers,
Andreas

[fateman]

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?

[Umma]

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]

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]

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]

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]

Originally posted by Umma
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 :D

Nothing better than a big surprise ;)

EoH

[fateman]

Originally posted by Eye of Horus
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]

Originally posted by kempfand
Just for your information:

I've created a few LCR-Upmix Bidule Groups, which you might find useful: LCR Upmix Bidules (http://forum.doom9.org/showthread.php?threadid=70678)

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..... :D

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]

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]

alright, thanks man. i'll be interested to hear your results on that time consuming test you have coming up as well ;)

[Tantulus]

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]

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)


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.


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 (http://www.york.ac.uk/inst/mustech/3d_audio/vst/vst3_ambisonics.html) . 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]

Originally posted by kempfand
If you want, I can post a small bidule for Pentagon-decode.

Cheers,
Andreas


Please post :D

thanks
SallyDog

[kempfand]

I've uploaded a 3rd order bidule (incl. 2 screenshots) to the FTP (ftp://daphy.mine.nu/) (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 für Musik Winterthur Zürich VST Ambisonic Tools (http://www.dmalham.freeserve.co.uk/alpha/vst_ambisonics_alphas.html) in the York Plugins Windows Version (http://www.dmalham.freeserve.co.uk/alpha/v2yorkambisonic.zip) . 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 (http://www.york.ac.uk/inst/mustech/3d_audio/vst/vst3_ambisonics.html)

Good luck and happy testing,
Andreas

[daphy]

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

[puzio]

Thank You Kempfand.

Can You tell me what is the channel mapping of the multichannel output? Please.

THX

[Eye of Horus]

Originally posted by puzio
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

[Umma]

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!

[Van the man]

this looks great :)
Gonna try it out, but the emigrator link doesnt work :confused:
And how can I mux all the .wav 's back to one .wav file?

Any ideas?

Van

[Umma]

Originally posted by Van the man
this looks great :)
Gonna try it out, but the emigrator link doesnt work :confused:
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 (http://www.gerzonic.com/?rub=3) Emigrator link?

:)

[specise_8472]

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.

[Umma]

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? :confused:

[specise_8472]

Originally posted by Umma
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.


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.


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? :confused:
This method may have merits. Try it and tell us the results.

[Eye of Horus]

Originally posted by Umma


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]

Originally posted by Eye of Horus
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..... :D

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

[Eye of Horus]

Originally posted by specise_8472
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]

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]

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:D

[Eye of Horus]

Originally posted by Umma
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 :D :D

OTOH : did you look on Ebay ?

grtz,

EoH

[Van the man]

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 :confused:

[daphy]

@ Van the man
try CommandBar with 'CTRL+M' you get a seperate window inside of the explorer.
Download located in the 'other apps' section (http://needfulthings.webhop.org/)
(BTW needs MS Frameworks)

CYA Daphy

[Umma]

OTOH : did you look on Ebay ?

Nah, they don't have a "try before you buy" option..." :D


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?

[specise_8472]

Have you checked that Player playing and Recorder recording are Linked?

[Umma]

Thanks! That is exactly what it was. I just took it for granted that it was already linked. :)

[Eye of Horus]

Originally posted by Eye of Horus
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 (http://members.lycos.nl/dtsac3forum)

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

[Tantulus]

Originally posted by specise_8472
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.

[Tantulus]

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.

[Eye of Horus]

Originally posted by Eye of Horus
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 (http://members.lycos.nl/dtsac3forum)

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

[Tantulus]

Originally posted by 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

Thanks I tried the news groups but I wasn't able to open the files.

[@ndy]

Originally posted by Tantulus
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 I´m 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 (ftp://daphy.mine.nu) 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 (ftp://daphy.mine.nu) .

For those who mention that the server is too slow:
I´m 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" :D

CU
@ndy

[Shayne]

e= Ambisonics second method -LFE(EOH, Kempfand)


Could you elaborate to what exactly this method entitles.

Thanks

Peace

[kempfand]

- "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

[Shayne]

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

[kempfand]

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

[Shayne]

Originally posted by kempfand
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

[daphy]

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 'wouldn´t 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! (http://needfulthings.webhop.org/)

CYA Daphy

[@ndy]

Hello guys,
the FTP- Server (ftp://daphy.mine.nu) will be down until the 2nd april due to a practice in my new job. You still can download all the needful things (http://needfulthings.webhop.org/) from our new webhosting. Uploads are possible in the old manner from the beginning of april on (or send a PM to Daphy (http://forum.doom9.org/private.php?s=&action=newmessage&userid=33004) if it is urgent :D )

Keep surrounding
Yours
@ndy

[desertrat]

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

[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

[@ndy]

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 (http://www.daphy.3x.ro) . You`ll find them in the folders "ambisonic" and in "ambiophonics".:D

Bye Bye
@ndy

[desertrat]

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

[Umma]

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. :(

[desertrat]

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

[kempfand]

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: 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 (http://www.isvr.soton.ac.uk/FDAG/vap/html/sd.html)).

Cheers,
Andreas

[desertrat]

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

[kempfand]

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 (http://www.ambiophonics.org/).

do we have an Ambiophonic mixing method using plogue bidule and if so which one is it? See http://forum.doom9.org/showthread.php?postid=451435#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 ;)

---

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

[Eye of Horus]

Originally posted by kempfand
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 (http://www.ambiophonics.org/).

See http://forum.doom9.org/showthread.php?postid=451435#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

[kempfand]

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 (http://needfulthings.webhop.org/).

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 (http://www.ambiophonics.org/)).

- specise_8472 implemented another methodology for the x-talk cancellation (look at
his posting on this (http://forum.doom9.org/showthread.php?postid=451435#post451435) and the following posts). His most recent VST's and filters have been posted on the FTP by Daphy & @ndy (ftp://daphy.mine.nu/), and are now available on the web-based interface (http://needfulthings.webhop.org/). 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, Dalenbäck, 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

[daphy]

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 :confused: - mixing the methodes leads to sorting problems :D

-> So this target folder could be moved very soon on an other place :o

Thx to EOH for quick help!

CYA Daphy

[PeterC]

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

[Shayne]

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

[Lex55]

Nice results

Tested with Eric Clapton Layla unpluged

Trying to Add LFE AS Well Thinking on crossover off 60hz.

[kempfand]

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

[Lex55]

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.

[specise_8472]

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.

[daphy]

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

[Tantulus]

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.

[kempfand]

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

[Lex55]

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

[Shayne]

Originally posted by Tantulus
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.

[rc]

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:D

[Tantulus]

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

[kempfand]

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: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 (http://www.angelofarina.it/aurora/conversion_between_uhj_and_b.htm#Audiomulch), 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 (http://pcfarina.eng.unipr.it/public/Voxengo-Bformat/).

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 (http://www.catt.se/the_suite.htm)

Good luck & in a hurry,
Andreas

[desertrat]

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

[puzio]

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.

[Eye of Horus]

Originally posted by puzio
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

[puzio]

Thank You EoH
I did. I enable 8 channel convolution but on the bidule's screen there are only 2 out pins!!!!

Thanks.

[kempfand]

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 :p

As for the settings, you can check the picture on Farina's public server: http://www.ramsete.com/Public/Voxengo-Bformat/AudioMulch_Image.jpg

Cheers,
Andreas

[Eye of Horus]

Kempfand and me developed a new bidule based on the old Kpex method.
The results (especially on synths music are amazing !)

You need of course Bidule from Plogue and the HNM filter which can be found on Daphy's site.

The new bidule can be downloaded from :
http://www.app.demon.nl/SAD5.1inBidule.rar

It uses freeware only !!

Please give it a try and share your thoughts !!

kind regards,

Eye of Horus

[Eye of Horus]

Daphy,

Can you download it and put it on your server too ?

Thanks !!

EoH

[daphy]

@EoH

done :D

[Eye of Horus]

Hi,

thanks !!!

2 remarks :

a. It was a co-production of Kempfand AND EoH ! (credits should go , where credits should go ;) )
b. It is not an Ambisonics plugin !! It's a total different method, nothing to do with Ambisonics or Ambiophonics !! (You've put it in the Ambisonics dir.....)

The bidule is actually based on this method :
http://forum.doom9.org/showthread.php?s=&threadid=57179
which we improved .

kind regards,

Eye of Horus

[daphy]

2 remarks :

a. It was a co-production of Kempfand AND EoH ! (credits should go , where credits should go )
b. It is not an Ambisonics plugin !! It's a total different method, nothing to do with Ambisonics or Ambiophonics !! (You've put it in the Ambisonics dir...

sorry, my mistake ;)

corrected!

[Tantulus]

Well Andreas, I managed to rewire my speakers so that I can place them at a 10 degree angle. The results were incredible. I tested it agaist the other methods using CD-RWs and I feel this method is the best for orchestral music. Congratulations on work well done. Also thanks again for all the others who have worked hard to create bidules for surround sound.

Regards,
Scott

[Eye of Horus]

Originally posted by Tantulus
Well Andreas, I managed to rewire my speakers so that I can place them at a 10 degree angle. The results were incredible. I tested it agaist the other methods using CD-RWs and I feel this method is the best for orchestral music. Congratulations on work well done. Also thanks again for all the others who have worked hard to create bidules for surround sound.

Regards,
Scott

Please try our new bidule too :D
Perhaps we can surprise you again !

EoH

[Tantulus]

I actually tested the new bidule and compared it to the ambiphonics using Brian Eno's Music for Movies CD. I found the surround was impressive but it didn't have the breadth and ambience that the ambiphonics had. When I get around to it I'll compare some of my rock CD's. Currently I'm testing some other impluse responses in SIR.

I do have a request. I find the bidule rather complex and I would like to take you up on experimenting with it. At some point could you expain some of the components, such as audio matrix and the 500.0000 module (or provide me with a link). I was thinking of substituting the pristine convolver but I think that I'm getting enough sense of ambience for my ears.

Regards,
Scott

[kempfand]

Tantulus: I am glad you had good success with the Stereo Dipole. As you say correctly: "results can be incredible".

In fact, I would expect it to work very very well with orchestral music you use, as this type of music often already contains "wet" information (i.e. reverb from the recording space, ambience etc). All you then can do is to give it a better left-right localization of instruments, and that's exactly what the Stereo Dipole does.

In case you haven't already done so, I also recommend you download the test-tracks from Virtual Acoustics Project @ ISVR (http://www.isvr.soton.ac.uk/FDAG/VAP/), burn them on CDR, and listen on your STereo Dipole. There are the 6 Stereo Dipole wave files in the Download section (http://www.isvr.soton.ac.uk/FDAG/VAP/html/download.html).

Cheers,

Andreas

[Tantulus]

Well Andreas, I took your suggestion and downloaded the tracks fromVirtual Acoustics Project@ISVR (http://www.isvr.soton.ac.uk/FDAG/VAP/) It made me really appreciate the stereo dipole. Incidentaly, I also found out that my speakers were out of phase when I first listened to the tracks and I couldn't hear the effect. Looks like I'll have to do my bidule comparisons all over again.

Do you think it would be possible to produce a stereo dipole for the surround speakers or would that not be an advantage?

Thanks again

Scott :cool:

[Eye of Horus]

Hi all,

A simple guide for our new Bidule is on this page :

http://www.dtsac3forum.digitalzones.com/SAD51Bidule.htm

kind regards,

EoH

[SallyDog]

Originally posted by Eye of Horus
Hi all,

A simple guide for our new Bidule is on this page :



@ EoH and Kempfand

Thanks for the guide. While I'm still most satisfied with the standard ambisonic bidule, your new one gave excellent results.

@EoH
On a side note, are the results of your testing of different encoding methods as performed in alt.bin.... still available? Could you (will you) post the poll results in this thread so everyone can see them?

Thanks alot guys, and keep up the good work.

Regards,

SallyDog

[Shayne]

@EoH

Thank you for your continued great work

Tried the new method on some APP and i have to say I am too still happy with page 1 plus .1

Peace

[Eye of Horus]

Hi all,

Sometimes the most simple solution is the most effective :-)

Thanks to Alastor (also member here) who made the soltion to the large file problem.

You can download the new Bidule here :

http://www.app.demon.nl/SAD51inBidulemod.rar

grtz,

EoH

[Eye of Horus]

Originally posted by Eye of Horus
Hi all,

Sometimes the most simple solution is the most effective :-)

Thanks to Alastor (also member here) who made the soltion to the large file problem.

You can download the new Bidule here :

http://www.app.demon.nl/SAD51inBidulemod.rar

grtz,

EoH

I didn't test it, but Kempfand replied with advice not to do it this way. The beginning of the files can be slightly out of sync.
He has however two alternaticves, which you can read here :

http://forum.doom9.org/showthread.php?s=&threadid=75613

EoH

[daphy]

sorry, this link leeds to your post again

http://forum.doom9.org/showthread.p...8181#post488181

[Eye of Horus]

Originally posted by daphy
sorry, this link leeds to your post again

http://forum.doom9.org/showthread.p...8181#post488181

Oops....

http://forum.doom9.org/showthread.php?s=&threadid=75613

kind regards,

EoH

[alastor]

Sorry but i d'ont speake fluently english.

The solution d'ont work with small files like mp3 songs, but work very well with big files like films sountrack.

I test tree different films 2 chapters in a serie 40 minutes each and a regular film. 105 minutes. 2,5 GB input wav file.

In all cases the bidule's layout work.

May be differences between song wav file and film wave file? The frames indicators? I d'ont know,

Alastor

[Eye of Horus]

Originally posted by alastor
Sorry but i d'ont speake fluently english.

The solution d'ont work with small files like mp3 songs, but work very well with big files like films sountrack.

I test tree different films 2 chapters in a serie 40 minutes each and a regular film. 105 minutes. 2,5 GB input wav file.

In all cases the bidule's layout work.

May be differences between song wav file and film wave file? The frames indicators? I d'ont know,

Alastor

When you use NTFS the limit is 4 GB !!
On Fat32 it's 2 GB.....


Away now for a 2.5 week vacation ! Yeah !!!

EoH

[alastor]

I finish to encode a very large film 2:09:04:412
Efectively i have small differences in time lenght of files center channel and lfe are ...04:064, surround 04:053 and front channels 04:405.

These differences are not significatives for a film, and the sound resultant is greather.

The proces is Bidule layout mod ( 6 recorders ) 16 bit deep, offline processing on.

Tomorrow test with offline processing off.

Alastor

[Tantulus]

I know that when I work with analogue to digital it is always advantagious to have the extra headroom with 32 bit. Why is there a need to process CD data in 32 bits when it ultimately will remain as 16 bit?

Also How does one label the files for Alaster's six recorder method E.G. Audio 001, 002 etc and how does the besweet batch file accomodate the 6 separate files. Does one run a batch file for the original batch file? I'm confused at this point!

However, as in the guide I thing multiple instances of the besweet batch should do the trick.

I haven't tried it yet but I think I could do this in Audition without using besweet. But I would like to save the work.

Regards,

Scott

[kempfand]

Why is there a need to process CD data in 32 bits when it ultimately will remain as 16 bit?
This was discussed multiple times all over.
Main reasons:
(a) Advised by all audio professionals and experts
(b) Most VST's work internally using 32 or 64 bit, and driving them with more data depth will just make 'work' them better
(c) Most important: Make a listening test yourself. We did many blind ones, and could tell the difference most of the time

But: If (after you tested yourself) you are happy with 16-bit, go for it. After all, this is just a guide, not a law :)

Also How does one label the files for Alaster's six recorder method As written elsewhere, don't use 6 mono-recorders because that will lead do un-synched things in the beginning (wouldn't be an issue if it was at the end). Use 6-channel File Recorder (32-bit files), and the BeSweet v1.5b27+ to demux to 32-bit with option "-6chfloat".

Regards,
Andreas

[Eye of Horus]

Originally posted by kempfand
This was discussed multiple times all over.
Main reasons:
(a) Advised by all audio professionals and experts
(b) Most VST's work internally using 32 or 64 bit, and driving them with more data depth will just make 'work' them better
(c) Most important: Make a listening test yourself. We did many blind ones, and could tell the difference most of the time

But: If (after you tested yourself) you are happy with 16-bit, go for it. After all, this is just a guide, not a law :)

As written elsewhere, don't use 6 mono-recorders because that will lead do un-synched things in the beginning (wouldn't be an issue if it was at the end). Use 6-channel File Recorder (32-bit files), and the BeSweet v1.5b27+ to demux to 32-bit with option "-6chfloat".

Regards,
Andreas

With a max length of the CD of 682 MB.......

I am away ;)

TTYL guys !

EoH

[alastor]

A question please.

I undestand the convenience of using 32 bytes deep.

Usualy the sountrack film is in 16 bytes deep, is possible to convert for process at 32 bytes float, but at end of process it is neccesary re-convert a 16 bit, or 24 for sonic foundry, because the dvd reproducers not support 32 bytes.

All convert reconvert not suppose a quality loss?

Headac3 has these conversion, is a good program?

Alastor

[kempfand]

All convert reconvert not suppose a quality loss?
Headac3 has these conversion, is a good program?

My understanding is that it depends which SW you use to convert/reconvert. If done properly, you won't get too much quality loss. I use CoolEdit/Audition with the "Convert Sample Type". Slow but good results.

I don't know HeadAC3 too well, so I cannot comment on this.

Andreas

[Tantulus]

This was discussed multiple times all over.

Sorry Andreas! I found the discussion on the merits of 32 bit processing early in this thread a bit confusing. (pardon the pun)

As to the multiple recorders, I am nervous about using besweet with files greater then 2 megs. I'll give it a try with the suggests mentioned in the thread. However I did try the SAD method on a shorter file with one recorder and I did find the results "amazing".

Thanks for the patience and the advice from others

Scott :o

[Umma]

I used Besweet tonight on a 2.58 GB 48/32-bit file tonight. Worked fine. :)

[Tantulus]

I'm trying a 2.9 Gig file and I'm getting different file lengths (at least it transcode all the way). Looks like I'm going to have to split the file.

Scott

P.S. Never mind! I didn't have enough disk space. After clearing things up besweet worked fine.

Once again, my naivety is exposed. 2.9 GB works fine.

[desertrat]

Originally posted by Eye of Horus
Kempfand and me developed a new bidule based on the old Kpex method.
The results (especially on synths music are amazing !)

You need of course Bidule from Plogue and the HNM filter which can be found on Daphy's site.

The new bidule can be downloaded from :
http://www.app.demon.nl/SAD5.1inBidule.rar

It uses freeware only !!

Please give it a try and share your thoughts !!

kind regards,

Eye of Horus

Thanks very much EOH and Kempfand. I have tried this new SAD5.1 bidule and am very pleased with the results. My test was done using The Best Of The Art Of Noise (1988 Version) and it came out so good, I had to do another AON right away. I have used ambisonic and wxyz methods with this album and was never quite happy with how they came out. For this type of music this bidule is excellent.

Cheers,
desertrat

[ursamtl]

Hi,

As I mentioned in another thread earlier today, I've been playing around with the various stereo-to-5.1 methods presented in the forum here. I've tried the Ambisonics guide several times, along with some of the variations presented throughout the thread. So far, I've not been at all impressed with the results. I've read the statements that the method presented here may not be to everyone's liking, but I've also read other comments from people who describe the ambisonic approach as providing a nice soundstage that gives a sense of a defined space, etc. All I've been able to get so far is a greatly narrowed stereo field that borders on mono with the same thing duplicated in the rear surround speakers. I don't think it's a case of bad connections as I've retried it several times according to the instructions. Plus, if I play with the controls in b_pan and bprocedit, I can hear the soundfield move around from one speaker to the other, but I don't get a sense of width to the sound. It's almost like panning a mono signal around four or five channels.

I'm not looking for instruments coming out of all speakers. Ideally I want as close as possible to the same big, immersive sound I get from a nice 5.1 DVD sountrack. I want the surround channels to reinforce the front and give me the sense that the instrument is playing on a soundstage in front of me, as if I'm in the room. The positive comments I've read about ambisonics seem to indicate that this is what people are getting, but I'm not!

I've gotten a far more pleasing result by creating a bidule that more or less duplicates the calculations in the Matrixmixer DirectX filter (from sourceforge) and then adding SIR to the back (with another dummy SIR on the front channels) and loading it with an impulse recorded from a Lexicon 960 that I found at www.noisevault.com. The impulse called "big wooden room" seems to work very well with the right levels. I'm at work now on my lunch break so I have no access to my files, but I'll try to post the bidule when I'm home.

In closing let me echo some other comments I've read here in saying that all the work you people are doing is truly amazing. I've learned a lot and am thoroughly enjoying all this experimenting! Keep it up!

[Tantulus]

Ursamtl:
Kempfand has a "true ambiphonics" with ambisonics for the surround. I found this method to truely broaden the sound with depth added by the rear speakers. The only problem is getting the front speakers to form a 10 degree angle. I rewired my front speakers to acheive this angle and I was impressed. However, it's a nusience to move the speakers when switching from music to movies.

I have experimented with reverb but my inexperience only produced an echoing sound but no ambience. I've thought of using Audtion (formally Cool Edit Pro) and then feeding the file to the bidule. Again, I'm too inexperienced to work out the myriad of settings for the Reverb effect. Short of the ambiphonic method I've found the SAD5.1 method very satisfactory. I can hear solo instruments coming from the center speaker and I can locate sections of a symphonic orchestra. I agree with you that I still don't feel true ambience but I compared some of the bidules with a DVD of a live recording of Beethoven's 9th and I got some ambience but the soundscape wasn't that broad either. Someday, I'd like to sneak into Symphony Hall and record an impulse for convolution but I don't have the equipment.

That said, I'm looking forward to your bidule. Thanks for your contribution.

Scott

[specise_8472]

Originally posted by Tantulus
Ursamtl:
Kempfand has a "true ambiphonics" with ambisonics for the surround. I found this method to truely broaden the sound with depth added by the rear speakers. The only problem is getting the front speakers to form a 10 degree angle. I rewired my front speakers to acheive this angle and I was impressed. However, it's a nusience to move the speakers when switching from music to movies.

What I think Kempfand has done is the same as me.
Most modern Amps have an A and B output for the front speakers.
Just put a second set of speakers on the B channel for Ambipole recordings. (10 deg.). Then just use remote to switch between music and movies.

[ursamtl]

Originally posted by Tantulus
Ursamtl:

I have experimented with reverb but my inexperience only produced an echoing sound but no ambience....Someday, I'd like to sneak into Symphony Hall and record an impulse for convolution but I don't have the equipment.



Well hey, go to www.noisevault.com and check out the impulses there that are free to download. Load them into SIR and try it out. I haven't had the chance to play around much with this stuff yet, but the potential is amazing.


That said, I'm looking forward to your bidule. Thanks for your contribution.


I haven't had time to put it all together yet. Last night when I sat down and looked at my bidule, I realized it was a messy mass of connections. It sounds good, but it looks like hell! :) I also realized that, although I did some initial experiments with encoding 5.1 files and burning to CD, most of my time has been spent monitoring the different bidules through a Soundblaster Live using the amazing kx drivers from www.kxproject.dom. I add a kx ASIO out device in plogue (15 channels available!) and then route them to the appropriate speakers using the kx DSP applet, which also has a similar modular connection setup.

Anyway, I'll clean up my bidule, add a 6-chan File Recorder in place of the kx ASIO outs and upload it on the weekend.

[kempfand]

Originally posted by specise_8472
What I think Kempfand has done is the same as me.
Most modern Amps have an A and B output for the front speakers.
Just put a second set of speakers on the B channel for Ambipole recordings. (10 deg.). Then just use remote to switch between music and movies. This is exactly how I do it. It actually was specise_8472 who triggered the idea of doing it that way.

Originally posted by ursamtl
Well hey, go to www.noisevault.com and check out the impulses there that are free to download. Load them into SIR and try it out. I haven't had the chance to play around much with this stuff yet, but the potential is amazing. The readings and filters @ NoiseVault are TRUELY amazing, and it is so much fun to play with all these impulse responses.

using the amazing kx drivers from www.kxproject.com Didn't know this drivers, but am very keen to play with it. Thanks for hinting to this.

Kind regards,
Andreas

[specise_8472]

Have just uploaded to the FTP server some new toys.
Daphy will put on Web server when he can.

It is a very good Mono to stereo conversion utility. And feeding the result into one of the above 2 to 5 methods produces astounding results. Better than I had hoped for. I myself personally use my Allinone process.

Uploaded are 5 groups to put in the group directory of Plogue.
And 1 VST .dll file. This file is just the same x-talk cancel that I posted a while ago. But all of these files are needed to make it work.

They are
allpass 44.bgrp
allpass 48.bgrp
mono to stereo 44.bgrp :edit
mono to stereo 48.bgrp :edit
x-talk.bgrp
x-talk 15.dll

USEAGE:
After everything installed, just drag the mono to stero 44 or 48 group onto the screen and input mono, output to stereo recorder. As simple as that. At the moment the group only uses the allpass 44.bgrp. I am now uploading two new groups to do inputs of 44.1 and 48.

EDIT: have just uploaded the new bgps as mono to stereo 44 and 48.
Just choose the right one to match your input frequency.

For those interested - it is basically feeding the mono signal through two banks of cascaded filters. Each bank is a 19 pole Allpass filter. Then feeding the resultant signals through my x-talk filter to produce better channel seperation.

[Umma]

Each bank is a 19 pole Allpass filter. Then feeding the resultant signals through my x-talk filter to produce better channel seperation.


Can you explain, or point me to someplace that explains what is meant by "pole" in the context of audio?

Thanks!

[Tantulus]

I wish I could afford more speakers so that I don't have to move mine for the ambiphonic affect; perhaps someday...

So far my favorit method is SAD5.1inbidule and I would like to experiment with the noisevault SIRS. Could someone suggest or give me a hint how to add them to the bidule and how to use the multiple impluses.(16 in one set).

Finally, hurrah for the mono to stereo bidule. I've bee doing it with Audition using delays, mixing and echo effects with mediocre results. I really envy all of you that construct these bidules.

Regards,
Scott

[kempfand]

Could someone suggest or give me a hint how to add them to the bidule and how to use the multiple impluses The answer was already given some posts above on this page by ursamtl:

"Adding SIR to the back (with another dummy SIR on the front channels) and loading it with an impulse recorded from a Lexicon 960 that I found at www.noisevault.com."

Note the importance of adding dummy SIR's (you'll need 2 to cover L, R, C), or else things will be out of sync due to SIR's latency.

As to which impulses: This really depends on your taste and the music, but if you read the posts at Noisevault, you'll get an idea of what people rank best (such as the Sony DRE777 or Lexicon 960 ones :cool: ).

For artificial reverbs, you can try
- Classic Reverb (http://www.kjaerhusaudio.com/classic-reverb.php)
- Ambience (http://www.smartelectronix.com/~magnus/)
- FreeVerb (http://www.hitsquad.com/smm/programs/Freeverb/)
or the various commercial ones from Wave Arts, SpinAudio, PSP, Waves, etc.)

Regards,
Andreas

[Shayne]

Hi,

Made me update bidule

Is anyone else having trouble with the new bidule 6601?

Or does anyone have it working with the guide?

Thank you

Peace

[Umma]

My files keep getting saved as "Raw PCM or Unsupported Format." Driving me bloody insane, it is! Can't figure it out!

[kempfand]

Look slike BeSweet currently only handles no WAVE Format Extensible and no 24-bit.

So you must set 'never' for the WAVE Format Extensible (with Preference), and use 16 or 32 bit for the Audio File Writer.

Regards,

Andreas

[Umma]

That did it, Andreas. Thanks. :)

[PeterC]

I get the following error from one of the mono to stereo groups in Bidule.

Unable to parse C:\program files\plogue\bidule\groups\mono to stereo 48.bgrp
An exception occurred! Type:UnexpectedEOFException, Message: The end of input was
not expected

What might be the cause of this ? :S

[ursamtl]

Hi folks,

As promised, I just uploaded a zip file containing the bidule I created in an attempt to emulate the equations for 5.1 I found in the MatrixMixer DirectX filter available on SourceForge.net. I've also included a couple of screen grabs to show the basic layout I use (it includes the kX ASIO outputs, which you may not have) and my rear SIR settings. It's simply a first attempt so my apologies if it duplicates something someone else here has tried and I've overlooked.

Note that the two front channels by themselves sound almost like surround channels. Adding the center channel gives them the body they need. You can use the four level controls to play around with the mix, providing you can monitor. For the LFE level, if your input file has a lot of bass, back off on this level a bit.


One thing I may try to implement for next version is a crossover routing lower octaves from the center back to the individual front left and right channels (since a lot of center speakers don't have as full a range).

This was all done @ 44.1k, 16-bits. I haven't had time to experiment with higher levels. One thing I am curious about is that no one has mentioned dithering if you convert to 32 bits for processing and then write 16-bit files.

Anyway, enjoy. This gives a huge sound to some recordings with a lot of ambient pads, etc. I tried Seal's Love's Divine last night and it sounded gigantic! It also sounded great with the remastered tracks from Deep Purple Machine Head or Elton's Tiny Dancer.

[Eye of Horus]

Originally posted by ursamtl
One thing I am curious about is that no one has mentioned dithering if you convert to 32 bits for processing and then write 16-bit files.

When you convert in Soundforge from 16 bits to 32 bits, there is no dither option.
The 6 mono 32 bits files are directly used in Surcode.
So there is no dithering involved at all.......
Of course it would be different if you do the 32 to 16 bits in SF or CE !

I certainly will try out your method and compare it with the one we're currently working on (and the others)!

Good job and perhaps you can consider a newbie manual ?

kind regards,

EoH

[specise_8472]

About my Mono to Stereo conversion groups.

Had nothing but trouble with the new version of plogue.

So reverted back to the old version 6502. Works fine in this version. Will have to look into why. But I think all the fancy DSP preference settings etc are going to be nothing but trouble. So will stick with a known good version (6502).

[kempfand]

@ ursamtl: Thanks for sharing your bidule, I am keen to try it.

This was all done @ 44.1k, 16-bits. I haven't had time to experiment with higher levels. One thing I am curious about is that no one has mentioned dithering if you convert to 32 bits for processing and then write 16-bit files.

I am just looking at this topic too. There are various commercial options (such as Wave's IDR which uses the Gerzon approach, or Wave Art's FinalPlug).

There is also a nice free version from maxim digital audio (mda) (http://www.abel.net.uk/~maxim/vst/help/mdaplugs.htm) called mda dither (http://www.abel.net.uk/~maxim/vst/help/dither.htm) .

Cheers,

Andreas

[Eye of Horus]

Originally posted by specise_8472
About my Mono to Stereo conversion groups.

Had nothing but trouble with the new version of plogue.

So reverted back to the old version 6502. Works fine in this version. Will have to look into why. But I think all the fancy DSP preference settings etc are going to be nothing but trouble. So will stick with a known good version (6502).

My old version said : trial version ended, new version available for download.
I tried the new version for only a brief moment with some new ideas, but didn't find any problem...... (I must admit however that I didn't use the filerecorder.....)

EoH

[Tantulus]

Thanks Andreas for your help. I was starting to add some SIR to the SAD5.1 bidule and forgot about the dummy SIRs. You saved me some frustration.

Ursamtl: Can't wait to try your bidule!
Species_8472: Your's too. It will be a while though.

There are so many bidules out there that I am becoming overwhelmed. I've been so busy testing what you all construct that I haven't had time to work on some myself. Once again, I am idebted!

Regards,
Scott

[ursamtl]

Originally posted by Eye of Horus
When you convert in Soundforge from 16 bits to 32 bits, there is no dither option.
The 6 mono 32 bits files are directly used in Surcode.
So there is no dithering involved at all.......
Of course it would be different if you do the 32 to 16 bits in SF or CE !

No I meant dithering when converting back to 16-bit files. My understanding is that to do surround audio CDs, the files have to be 44.1khz and 16 bits. I know the 44.1 part is right because when I tried one at 48KHz, my DVD player just played digital noise. I haven't had time yet to try 32-bit files.

Good job and perhaps you can consider a newbie manual ?
Thanks. Actually I wouldn't mind but it will take some time as I'm very busy at work. Plus, I want to investigate this whole field more. I'm fascinated by the potential.

[Tantulus]

As promised, I just uploaded a zip file containing the bidule I created in an attempt to emulate the equations for 5.1 I found in the MatrixMixer DirectX filter available on SourceForge.net. I've also included a couple of screen grabs to show the basic layout I use (it includes the kX ASIO outputs, which you may not have)

IS KX ASIO necessary for your bidule and if so what does it do? I've downloaded the drivers but I'll wait for your respose before adding it.

Regards,
Scott

[ursamtl]

Originally posted by Tantulus
IS KX ASIO necessary for your bidule and if so what does it do? I've downloaded the drivers but I'll wait for your respose before adding it.

Regards,
Scott

No not at all. Sorry I should have made that clearer. As I mentioned in a previous post, I use the kX drivers with my Soundblaster Live. These are free drivers available at www.kxproject.com and they provide an incredible increase in usability over the standard Creative drivers. For instance, you can have 15 channels of ASIO with almost no latency.

The bidule I uploaded to the ftp site has no kX output in it.

[Eye of Horus]

Originally posted by ursamtl
[B]No I meant dithering when converting back to 16-bit files. My understanding is that to do surround audio CDs, the files have to be 44.1khz and 16 bits. I know the 44.1 part is right because when I tried one at 48KHz, my DVD player just played digital noise. I haven't had time yet to try 32-bit files.

But I did mention that :D !!
The 6 mono 32 bits files can be used in Surcode to create an audio CD. There is no need to convert them to 16 bits first ! Surcode itself handles the conversion back to 16 bits.

EoH

[Eye of Horus]

BTW under which name did you upload them ?
I cannot find the bidule on needful things........

EoH

[@ndy]

Originally posted by Eye of Horus
BTW under which name did you upload them ?
I cannot find the bidule on needful things........

EoH

Hello EoH,
I guess you wouldn`t find the *.zip file on needfulthings cause daphy is on holidays. So this site isn`t uptodate also unless the last provider killed our webspace.

You should find this file on my FTP- Server (ftp://daphy.mine.nu) in the incomming folder. The original Post from ursamtl is dated 23.05. and I didn`t moved any files ;)

BTW: I played with creating myself a website to have a second mirror of the FTP- Server which is really slow :(

I`ll post the url this evening when I`m finished

CU
@ndy

[ursamtl]

Originally posted by Eye of Horus
BTW under which name did you upload them ?
I cannot find the bidule on needful things........

EoH

I uploaded a zip file called MatrixMixer Emulator.zip to the incoming folder of daphy's ftp server.

[ursamtl]

Originally posted by Eye of Horus
But I did mention that :D !!
The 6 mono 32 bits files can be used in Surcode to create an audio CD. There is no need to convert them to 16 bits first ! Surcode itself handles the conversion back to 16 bits.

EoH

Ok, sorry I didn't see that. Does Surcode do any dithering or does it simply truncate the extra bits? The reading I've done on dithering says that simple truncation introduces digital artifacts and distortion.

If this is the case, then it would be better to convert back to 16 bites with dithering before loading the 6 files into Surcode.

I'll see what I can find out.

[ursamtl]

Originally posted by ursamtl
I uploaded a zip file called MatrixMixer Emulator.zip to the incoming folder of daphy's ftp server.

Since the ftp server doesn't seem to be active this morning, I've also thrown together a little web page on Yahoo. You can download the bidule there.

http://www.geocities.com/ursamtl/audio.html

[ursamtl]

By the way, for those of you looking for impulses to experiment with for your bidules, www.noisevault.com has just put up a new complete set of 44.1k, 32-bit impulses recorded from a Lexicon 960L! It's 38MB download, but it seems worth it.

These should fit in well with the discussion of 32-bit files we're having!

[kempfand]

@ursamtl: Just had a quick look at your bidule and MatrixMixer group. 2 questions regarding Kelly's Bass Manager which you use inside the MatrixMixer Group:

1) The manual says: This plug-in is to be used for monitoring only!
You need to turn off the bass manager before exporting your mix otherwise all the low-end will be removed from L, C, R, Ls, and Rs files, which is not what you want. The most common problem seems to be that people don't realize that the LFE and SUB are two different things Wouldn't that mean that your current bidule sould only be used for listening, but not for Audio File Recording ?


2) The manual says (and I can confirm this by a quick test I made): output assignment is L, R, C, SUB, Ls, Rs! I.e. Bass Manager's Pin-Out #4 is the SUB ... But: Your Matrix mixer assumes Pin-Out #4 as Surround Right (SUB on #6).

Not sure, but I think something is not correctly linked.

Kind regards,
Andreas

[@ndy]

Originally posted by ursamtl
Since the ftp server doesn't seem to be active this morning, I've also thrown together a little web page on Yahoo. You can download the bidule there.

http://www.geocities.com/ursamtl/audio.html

Finally I get along with a new mirror for my FTP- Server (ftp://daphy.mine.nu) :cool:

The new website (http://www.freewebtown.com/andiz/) is still under construction (panta rei (http://www.archaeonia.com/philosophy/presocratics/heraclitus.htm) ; for the greeks between us) but you have full access to every file you`ll find at my ftp- server.

Use my FTP- Server (ftp://daphy.mine.nu) to upload your files :D


Don`t forget: needfulthings (http://www.nedfulthings.webhop.org) is still alive ;)

[ursamtl]

Originally posted by kempfand
@ursamtl: Just had a quick look at your bidule and MatrixMixer group. 2 questions regarding Kelly's Bass Manager which you use inside the MatrixMixer Group:

1) The manual says: Wouldn't that mean that your current bidule sould only be used for listening, but not for Audio File Recording ?


kempfand, it depends what you want. I read the manual as well, and while I understand what the author is getting at, I want the low-end removed from the L, C, R, Ls, and Rs and sent through the SUB output. You'll notice I didn't connect anything to the LFE input because I want Bass Manager to trim the bass under the chosen crossover frequency from each channel, sum it and route it through the SUB output. In effect, I'm using this to generate an LFE channel. Obviously, people who don't want this should remove the bass manager altogether.

2) The manual says (and I can confirm this by a quick test I made): I.e. Bass Manager's Pin-Out #4 is the SUB ... But: Your Matrix mixer assumes Pin-Out #4 as Surround Right (SUB on #6).

Not sure, but I think something is not correctly linked.


Thanks for pointing this out. I did my bidule using the Bass Manager beta 0.3 that I downloaded from Needful Things as part of the zipped VST plugins. I see now that the latest version switches to the ITU standard of L, R, C, SUB, Ls, Rs. I did update to SIR_1005 from the old SIR_086 that was in the zip file but I didn't realize that Kelly had updated their VST. Just goes to show that it pays to keep an eye out for updates. :)

Thanks,
Ursa

[Tantulus]

I did download the Lexicon impules and did a quick test by adding SIRs to the SAD5.1 bidule of the one called ambience. Sounded great so far.

BTW what are the SFK files for?

Still planning to use the matrixemulator after I finish a two disc set I've already started. I plan to experiment with the Lexicon impulses in your bidule.

Regards,
Scott

[kempfand]

kempfand, it depends what you want. I read the manual as well, and while I understand what the author is getting at, I want the low-end removed from the L, C, R, Ls, and Rs and sent through the SUB output. You'll notice I didn't connect anything to the LFE input because I want Bass Manager to trim the bass under the chosen crossover frequency from each channel, sum it and route it through the SUB output. In effect, I'm using this to generate an LFE channel. Obviously, people who don't want this should remove the bass manager altogether. Thanks for the info. A small write-up such as this really helps, and I now see what you try to achieve :)

Will do some extensive tests coming long weekend here, with various types of music.

Kind regards,

Andreas

[ursamtl]

I've uploaded version 2 of MatrixMixerEmulator to the FTP server's incoming folder and also posted it at http://www.geocities.com/ursamtl.

This updates the entire signal path to L, R, C, LFE, Ls, Rs as per ITU standards. It's designed to work with Kelly Industries Bass Manager v0.5 and SIR 1.005.

I've also replaced the dummy SIR with a native Plogue delay line set to 8960 samples as per SIR 1.005 specs. I haven't tested it, but I imagine this will be less CPU and/or memory intensive than creating an extra instance of SIR just to compensate for the delay.

Thanks for the feedback.

Ursa

[Eye of Horus]

Originally posted by ursamtl
I've uploaded version 2 of MatrixMixerEmulator to the FTP server's incoming folder and also posted it at http://www.geocities.com/ursamtl.

This updates the entire signal path to L, R, C, LFE, Ls, Rs as per ITU standards. It's designed to work with Kelly Industries Bass Manager v0.5 and SIR 1.005.

I've also replaced the dummy SIR with a native Plogue delay line set to 8960 samples as per SIR 1.005 specs. I haven't tested it, but I imagine this will be less CPU and/or memory intensive than creating an extra instance of SIR just to compensate for the delay.

Thanks for the feedback.

Ursa

I did some testing today and I'm afraid I don't have good news.
The rears sounded terrible ! There was a definite delay, so bad that it sounded like echo !!
I will recheck all the settings, but I'm sure I got them right this morning.

In one of your previous mails here, (the long one) you made quite some comments about Ambisonics and that you want to achieve more separation. That's what we're working on too. SAD51inBidule gives at this moment better results than Matrixemulator !
Of course all IMHO, because I only tested with two tracks.

I will do some more tests ASAP to see if I can get rid of the echo in the rears. Perhaps it's just an overseen setting in your bidule or in the new Plogue version, but for now I was not impressed......

I take for granted that you tested it thoroughly, so maybe you can point me to what can be wrong ?

kind regards,

EoH

[ursamtl]

Originally posted by Eye of Horus
I did some testing today and I'm afraid I don't have good news.
The rears sounded terrible ! There was a definite delay, so bad that it sounded like echo !!
I will recheck all the settings, but I'm sure I got them right this morning.

It may be a problem with your levels or your SIR settings. The rear level should be low enough so that the sound appears to come from the front but if you shut off the rear, the whole soundfield collapses to more or less the original stereo field.

Also, make sure you have the center level high enough because what appears in the front left and right channels is basically what is unique to each channel. The center needs to be high enough to re-construct the original soundstage.

This also reveals the one downside with my approach in that one really needs to be able to monitor the mix in surround and not simply monitor everything to the two Microsoft Mapper channels as in your original guide. It's not necessary to have a proper 5.1 computer setup. I`m just using a 4.0 setup and mixing the center and LFE channels into the front. So far, the tests I've done when burning DTS CDs and playing them on my true 5.1 system have shown that I`m getting close with the monitoring.

As for ambisonics, I was playing around with a 3rd-order setup last night and while it still was terrible in terms of separation, it does some interesting things to the music, bringing out instruments that were buried in the stereo mix. I`m thinking of trying a combination of the MatrixMixer Emulator with the ambisonics.

By the way, have you tried the real MatrixMixer DirectX filter from SourceForge? I'm very impressed with its sound. If we could just fit it into a bidule! It provides exactly the kind of big, expansive sound I'm looking for in an upmix.

[specise_8472]

By the way, have you tried the real MatrixMixer DirectX filter from SourceForge? I'm very impressed with its sound. If we could just fit it into a bidule! It provides exactly the kind of big, expansive sound I'm looking for in an upmix. [/B]

My Ambiophonics methods are based upon this. I have kicking around somewhere in my backups, my VST version of MM. I originally was going the MM method, but found the needed paremeters for Ambiophonics. So went that way.
Don't forget that there are two versions of Ambiophonics. The version I use that takes WXYZ and computes speaker feeds.
Or the Stereo Dipole method. Which is why I created the X-talk VST. Which I am overhauling to produce binaural and transaural mixes and conversions.

Try my Allinone VST for rear channel seperation. It is 3rd order and computes the rears independant of each other. What I have done to really enhance the seperation all round is to use my x-talk impulses.

Just use three instances of SIR.
Left and Right Fronts using one of the front impulses.
Left and Right Surround using one of the Rear Impulses.
Center using the Zero impulse.

Really spreads out the sound image. And by feeding all signals through SIR's there is no delay.

At the moment, among other things, I am working on a better V3 ambiophonic mix using some enhanced methods to extract better channel seperation.

[ursamtl]

Originally posted by specise_8472
My Ambiophonics methods are based upon this. I have kicking around somewhere in my backups, my VST version of MM.

Hey, if you find it, please upload it! That would be very cool. By the way, MM is now part of the new AC3Filter release candidate. I noticed last night when playing with it that adding a 20ms delay to the left rear and 30ms to the right (in the filter's own delay pane) really gave me that big spread I hear in commercial 5.1 releases. The delay isn't really big enough to produce discreet echoes, but it does work well at giving the sound a sense of space. The 10ms difference between channels is based on the "TISDU" method used by traders of live mono recordings who use the delay to simulate stereo.

Try my Allinone VST for rear channel seperation. It is 3rd order and computes the rears independant of each other. What I have done to really enhance the seperation all round is to use my x-talk impulses.

Just use three instances of SIR.
Left and Right Fronts using one of the front impulses.
Left and Right Surround using one of the Rear Impulses.
Center using the Zero impulse.

Really spreads out the sound image. And by feeding all signals through SIR's there is no delay.

Actually I tried the AllinOne VST again last night and I'm afraid I had the same problem with it as with the ambisonics methods (no separation, rears sound like copies of the fronts), but I forgot about the impulses. I had read about them in the thread a few weeks ago and downloaded them but I forgot about them last night. I look forward to trying them with the Allinone.

By the way, instead of loading a SIR with no impulse, why not try what I did in v2 of my MM Emulator? Add a Plogue Delay Line and connect a Plogue Constant to its right pin entering the SIR delay value (8960 for v1.005, or 16384 for SIR 0.86). This does the same thing without having to load in an entire instance of SIR.

At the moment, among other things, I am working on a better V3 ambiophonic mix using some enhanced methods to extract better channel seperation.

Sounds cool. Getting the separation into it makes the sound come alive and gives the sense of realistic space.

[Eye of Horus]

Originally posted by ursamtl
Sounds cool. Getting the separation into it makes the sound come alive and gives the sense of realistic space.

The sense of realistic space ???????????

I have some old quad conversions here, where the level of separation is unbeatable. A guitar from FL, a singer from FR, drums from RL and another guitar from RR. What I miss is the band :D

Realistic space can only be achieved by Ambisonics and certainly not by separation. There is nothing realistic about separation at all, actually it will only be "realistic" when we have 5.1 ears !

IMHO separation is artificial !
Just simply because our ears are not build that way. We will hear from all directions at once, thus a left channel also influences what we hear with our right ear ! In other words, even if you can completely separate 5 sounds, there is still the fact that we will not hear it completely separated. So why this search for complete separation ?
Are we that much indoctrinated by Dolby ??

You mentioned the excellent results on DVD's. But...... are you comparing methods like Ambisonics with a music DVD or with a DVDA ?
Or were you talking about movie DVD's ?
I am just wondering because you will never get that separation from stereo to 5.1, whatever method ! You can come close, but not with the bidule you presented us ! I have tested it to it's limit this morning and it's not what I hoped it would be !
But I have to confess that I used a very difficult piece of music.
When I use DPL II, this happens : a guitar starts in the front right channel. The guitarplayer starts to sing. His voice is in the Center channel and the guitar remains in the right channel. The separation between C and R is perfect here with DPL.
Playing it in regular stereo you get the guitar from the right channel and the singer from both. And a very small bit of the guitar from the left, but you really have to play it vey loud to hear that.
With all methods I used on this piece of music, not one gave the same results. They just don't do a good job on this separation !
The only thing that came close was to use Gerzon LCR, and process the L=R from there and leave the Center as is.
All others made a mess of it ! Including yours.

For Ambisonics this piece is also difficult, but although no perfect separation, you still can sense where the singer comes from (Center) and where the guitar comes from (C+R). It's sounds more natural in Ambi because it's more fitted to our 2 ears. It makes a complete soundfield, which actually resembles what we hear with our 2 ears.

Also I would like to point out the zillion of music DVD's where the rears are nothing more than some weak reverb from the fronts. I even have this on a few DMP demo SACD's !
We can do better :-)

So there's still work to do !

kind regards,

EoH

[puzio]

Hello All

I tested ursamtl's method. The result is very bad. I think
E O H is right.

sorry for my english

[ursamtl]

Originally posted by Eye of Horus
The sense of realistic space ???????????

I have some old quad conversions here, where the level of separation is unbeatable. A guitar from FL, a singer from FR, drums from RL and another guitar from RR. What I miss is the band

Realistic space can only be achieved by Ambisonics and certainly not by separation. There is noth-ing realistic about separation at all, actually it will only be "realistic" when we have 5.1 ears !

Since everyone's reaction to sensory stimulation is subjective, there are no absolutes. What sounds "realistic" to one person sounds like crap to the next! Yes, in theory, Ambisonics looks to be a wonderful solution, but obviously some people don't agree or else this thread would be much shorter. Instead, it's full of alternative bidules, different approaches, changes, modifications, debates, etc.

All I'm saying is that I'm extremely disappointed with the results from Ambisonics so far. I came upon your guide quite by accident and read with enthusiasm its promise. I kept redoing the connections thinking surely I must be doing something wrong because the resulting 5.1 recording sounded small and boxy. I took Floyd's Comfortably Numb, Sarah Brightman's Captain Nemo, Marillion's Lavender, Zeppelin's Gallows Pole, Deep Purple's Lazy, Elton's Tiny Dancer, plus a couple of classical recordings, and tried over and over again to get some sense of the acoustic space you say Ambisonics gives. They all sounded like I remixed the tracks to mono and then routed the same mix to the rears. There was no sense of a realistic soundstage as if the real instruments were being played in front of me. The closest I achieved with Ambisonics was with the Surround configuration in Emigrator and only after tweaking the bpan controls (and obviously without the 36° rotation). This was on the classical guitar intro to Rush's A Farewell to Kings. It almost sounded as if the guitar were there right in front of me.

Once I took this same track and played it through the MatrixMixer filter, I got a much better sense of realism. I put together my bidule with a similar approach and with similar positive results. Whereas Comfortably Numb sounded really small and dull when rendered in Ambisonics, it became massive through my bidule. I dug out the remastered version of Deep Purple's Lazy and reveled with the sound and realism through my bidule. Blackmore's guitar sounded as if it were emanating from the amp in that Swiss hotel corridor so long ago, with gigantic Hammond groans coming from the right and sweeping through the room. In Ambisonics, this same song sounded like a 60's mono single mix (albeit with improved fidelity).


IMHO separation is artificial !
Just simply because our ears are not build that way. We will hear from all directions at once, thus a left channel also influences what we hear with our right ear ! In other words, even if you can completely separate 5 sounds, there is still the fact that we will not hear it completely separated. So why this search for complete separation ?
Are we that much indoctrinated by Dolby ??

I think you misunderstood what I meant by separation. I don't mean isolating instruments on one channel, either as a musical approach in the late 60s or as a sound effect in a modern movie soundtrack. I mean using all available channels to give a sense of the original sound of the instrument in an acoustic space, be it the original recording space or some other artificial space. You might complain that the rears of some music DVDs are nothing more than a weak reverb of the front, but if that's what it takes to make the sound seem to come from a stage in the front, then it's much more realistic than some artificial routing of sound to the back.

You mentioned the excellent results on DVD's. But...... are you comparing methods like Ambisonics with a music DVD or with a DVDA ?
Or were you talking about movie DVD's ?
I am just wondering because you will never get that separation from stereo to 5.1, whatever method ! You can come close, but not with the bidule you presented us ! I have tested it to it's limit this morning and it's not what I hoped it would be !

EoH, I never presented my bidule as the ultimate approach or even a "good" approach. If you check the first post I made to this thread, you'll see that I was extremely disappointed by the results of Ambisonics and I mentioned getting better results from a bidule I'd put together. Someone asked me to post it, so I cleaned it up a bit and did so. If you "hoped" it would be so good and if there is "still work to do" then obviously you see a deficiency with the Ambisonics approach. :)

Did it ever occur to you that perhaps you're expecting too much from Ambisonics? The theory seems good and probably sounds quite stunning if one is listening to material recorded with the proper miking, and correctly encoded. However, most music that was recorded for 2-channel stereo reproduction was not properly recorded for Ambisonic reproduction. Moreover, many modern recording studios record in rooms specially treated to add no coloration to the sound whatsoever. Therefore, any soundstage produced by its decoding method is going to be artificial. Perhaps Ambisonics does work wonders on a properly recorded sound that was recorded in an acoustic space capable of being reproduced, but even with proper recording, if an sound is recorded in a space that has little or no reflective capability--as is the case with many recording studios--there is no natural spatial information present in the recorded sound.

So there's still work to do !
Yes indeed. Obviously none of us has a perfect solution. Let's try to enjoy this experimenting with sound. I'm sure you didn't mean it, but the tone of your message seemed to me as if you felt any criticism of Ambisonics was almost criminal. Again, I'm sure this isn't intentional but even with some smiles, grins and "IMHO"s thrown in, it comes across as such. We're all just having fun exploring sound and technology and we all have valid contributions to make.

Regards,
Ursa

[kempfand]

@ puzio: I agree with a BUT: It's not easy to come up with a method that works well on many pieces of music. I still appreciate that ursamtl shared his method, which allows us others to learn.

Ambisonics / AmbioPhonics is truely exceptional and hard to beat :) Just to repeat that Ambisonic's goal is not to bring the orchestra into your living room, but bring you into the concert hall, so no need to look or search for single instruments coming from defined speakers.

For non-Ambisonics, I recently posted LCR Upmix Bidules (http://forum.doom9.org/showthread.php?threadid=70678). Amongsts these is the Gerzon LCR, but the others are also worth checking :sly: as they produce some reasonable LCR from LR.

Regards,
Andreas

[ursamtl]

Originally posted by puzio
Hello All

I tested ursamtl's method. The result is very bad. I think
E O H is right.

sorry for my english

I'm sorry to hear that. Did you check your levels and monitor them?Could you describe what was wrong with the result? If the rears echoed too much, try backing off on the reverb level in SIR. If the front seemed too wide and indistinct, try increasing the center channel. If the LFE was either too loud or too soft, try adjusting accordingly. With all of these methods, I've found that each piece of music requires its own adjustments. None of these will give you "plug and play" satisfaction, even the almighty Ambisonics!;)

Ursa

[kempfand]

All I'm saying is that I'm extremely disappointed with the results from Ambisonics so far.
...
I kept redoing the connections thinking surely I must be doing something wrong because the resulting 5.1 recording sounded small and boxy. Just keen: What speakers do you use for listening the DTS-coded end product ? Did you take a DTS CD and listened in a store over different speakers. Doesn't have to be $$$ ones, but outcome for Ambisonics can be very different. Most speakers these days are designed to sound good with Dolby stuff ...

Regards,
Andreas

[ursamtl]

Originally posted by kempfand
Ambisonics / AmbioPhonics is truely exceptional and hard to beat :)
Unfortunately, my experience so far is that everything else beats them, even the kpex method that you guys seem to ridicule so much!
Just to repeat that Ambisonic's goal is not to bring the orchestra into your living room, but bring you into the concert hall, so no need to look or search for single instruments coming from defined speakers.
But you see I don't get this result from either Ambisonics / AmbioPhonics (to be fair, I haven't properly tested the latter). I don't want single instruments coming from defined speakers, but neither do I want everything mashed into a single sound with no definition. To me these approaches so far have sounded like a mono mix piped through 5.1 speakers. There is no sense of any concert hall known to man!


For non-Ambisonics, I recently posted LCR Upmix Bidules (http://forum.doom9.org/showthread.php?threadid=70678). Amongsts these is the Gerzon LCR, but the others are also worth checking :sly: as they produce some reasonable LCR from LR.
Yes, I heartily agree. These produced a really nice front sound.

[Eye of Horus]

Originally posted by ursamtl
[B]Unfortunately, my experience so far is that everything else beats them, even the kpex method that you guys seem to ridicule so much!

English is not my native language, perhaps that's why I don't understand you.
We rewrote his method in a bidule and we haven't had any negative comments, except for one. We wouldn't have done that if we didn't see the potential of Kpex' method !!!!
So I cannot see where you get that statement from...... or I must misunderstand you !

EoH

[kempfand]

@ ursamtl / All: But you see I don't get this result from either Ambisonics / AmbioPhonics (to be fair, I haven't properly tested the latter). I don't want single instruments coming from defined speakers, but neither do I want everything mashed into a single sound with no definition You might want to try some real (i.e. 3D) B-Files. Some short demos are available:
- Soundfield B-Format (http://www.soundfield.co.uk/downloads/b-format.php)
- Farina's OSER-recordings (public) (http://www.ramsete.com/Public/OSER-recordings/Reharsal-Room/B-format/)

You might have to resample to 32/44 or 16/44 (using CoolEdit or similar), but you then can use Gerzonic's B-Player (http://www.gerzonic.net/?prod=1&rub=2) and decode using Emigrator as Pentagon or Square.

This will give you an idea of what you are can expect from Ambisonics. Sure these samples have been properly recorded (Soundfield Mic), but you can get similar (good) results from Stereo (but 2D only).

Good luck,

Andreas

[ursamtl]

Originally posted by kempfand
Just keen: What speakers do you use for listening the DTS-coded end product ? Did you take a DTS CD and listened in a store over different speakers. Doesn't have to be $$$ ones, but outcome for Ambisonics can be very different. Most speakers these days are designed to sound good with Dolby stuff ...

Regards,
Andreas

My speaker system sounds quite good. It's a combination Infinity/Polk Audio system. While it's not state-of-the-art, it provides a nice balanced sound with crisp, transparent highs, a smooth midrange and tight, solid bottom end. My PC speaker system is also nicely balanced for music without the overwhelming super subwoofer sound that's so prevalent for gaming fans.

Your point is valid but only somewhat. After all, if my system were skewing my impressions of the different bidule results, it would do likewise to my impressions of movie soundtracks, DVD-As, etc. I don't get what you and a couple of others mean when you talk about the "Dolby stuff." Do you mean movie sound effects? If so, then that's not at all what I'm aiming for. I want something like Kevin Shirley's wondeful mix on the Led Zeppelin DVD last year. He mixed the surround to sound as if the listener were sitting about five rows back from the front center of a Led Zeppelin show. The rears mainly provide the ambience of the hall and the audience and serve to reinforce the front, which is presented as an accurate soundstage of the band from the vantage point I just mentioned. If this were done with the Ambisonics bidule as I've heard them so far, the entire band would be cramped into a thick middle that's repeated in the rears. It would sound something like the old Motown remasters that have been released in mono. They might sound punchier than a stereo mix, but only 1 speaker is necessary for such reproduction and 4.1 speakers are being completely wasted on such material.

I'll retry some Ambiophonics testing this evening to see if the results are better, but certainly Ambisonics by itself seems to be an utter disappointment.

Ursa

[ursamtl]

Originally posted by kempfand
@ ursamtl / All: You might want to try some real (i.e. 3D) B-Files. Some short demos are available:
- Soundfield B-Format (http://www.soundfield.co.uk/downloads/b-format.php)
- Farina's OSER-recordings (public) (http://www.ramsete.com/Public/OSER-recordings/Reharsal-Room/B-format/)

You might have to resample to 32/44 or 16/44 (using CoolEdit or similar), but you then can use Gerzonic's B-Player (http://www.gerzonic.net/?prod=1&rub=2) and decode using Emigrator as Pentagon or Square.

This will give you an idea of what you are can expect from Ambisonics. Sure these samples have been properly recorded (Soundfield Mic), but you can get similar (good) results from Stereo (but 2D only).

Good luck,

Andreas

Thanks for the links. I tried a couple of these. While they certainly are more impressive than the results from the ambisonics bidules, I don't see them lending a convincing sound to a 5.1 mix of music. The vantage point would be somewhat like sitting on stage in the middle of a performance instead of sitting in the audience of a concert hall. I still got no sense of an acoustic space.

I'll try some more tests tomorrow. I have some domestic stuff to take care of now.

[Eye of Horus]

Originally posted by ursamtl
My speaker system sounds quite good. It's a combination Infinity/Polk Audio system. While it's not state-of-the-art, it provides a nice balanced sound with crisp, transparent highs, a smooth midrange and tight, solid bottom end. My PC speaker system is also nicely balanced for music without the overwhelming super subwoofer sound that's so prevalent for gaming fans.

Your point is valid but only somewhat. After all, if my system were skewing my impressions of the different bidule results, it would do likewise to my impressions of movie soundtracks, DVD-As, etc. I don't get what you and a couple of others mean when you talk about the "Dolby stuff." Do you mean movie sound effects? If so, then that's not at all what I'm aiming for. I want something like Kevin Shirley's wondeful mix on the Led Zeppelin DVD last year. He mixed the surround to sound as if the listener were sitting about five rows back from the front center of a Led Zeppelin show. The rears mainly provide the ambience of the hall and the audience and serve to reinforce the front, which is presented as an accurate soundstage of the band from the vantage point I just mentioned. If this were done with the Ambisonics bidule as I've heard them so far, the entire band would be cramped into a thick middle that's repeated in the rears. It would sound something like the old Motown remasters that have been released in mono. They might sound punchier than a stereo mix, but only 1 speaker is necessary for such reproduction and 4.1 speakers are being completely wasted on such material.

I'll retry some Ambiophonics testing this evening to see if the results are better, but certainly Ambisonics by itself seems to be an utter disappointment.

Ursa

A few IMHO's wouldn't hurt !!

Ever thought of that it can be yourself ???

I have had friends listening here to the Ambi mix of War of the Worlds. 6 out of 7 : I didn't know this was available in surround.
The 7th reacted like you : I don't hear it !

OK, as you probably know : I did a listening test. I took 10 stereo songs with all kind of styles of music. I converted these 10 songs with 10 different methods to surround. I made a poll online where people could fill in which version they liked the most. I didn't say before the poll what the used methods were.
End result : 70 % preferred the conversions with Ambisonics !

So obviously a. you're doing something wrong (BTW did you convert the stereo first to 32 bits ?) or b. you just don't (want to ?) hear it.

But keeping on saying how thrash Ambisonics is without adding a few "IMHO" gives me indeed the impression you're behaving like a troll. Eespecially since your own published bidule didn't give 3 people a good result. And believe me : if it was better than anyting already published, people sure would let it know here !

So come up with something better (your MM bidule was definitely worse). If not, than please stop all this whining and groaning !
Thanks.

EoH

[ursamtl]

Originally posted by Eye of Horus
A few IMHO's wouldn't hurt !!

Ever thought of that it can be yourself ???

I have had friends listening here to the Ambi mix of War of the Worlds. 6 out of 7 : I didn't know this was available in surround.
The 7th reacted like you : I don't hear it !

OK, as you probably know : I did a listening test. I took 10 stereo songs with all kind of styles of music. I converted these 10 songs with 10 different methods to surround. I made a poll online where people could fill in which version they liked the most. I didn't say before the poll what the used methods were.
End result : 70 % preferred the conversions with Ambisonics !

So obviously a. you're doing something wrong (BTW did you convert the stereo first to 32 bits ?) or b. you just don't (want to ?) hear it.

But keeping on saying how thrash Ambisonics is without adding a few "IMHO" gives me indeed the impression you're behaving like a troll. Eespecially since your own published bidule didn't give 3 people a good result. And believe me : if it was better than anyting already published, people sure would let it know here !

So come up with something better (your MM bidule was definitely worse). If not, than please stop all this whining and groaning !
Thanks.

EoH

EoH, I don't care if your first language isn't English, the tone of this message is extremely offensive. I have never attacked you personally, I've simply expressed disappointment with the method of 5.1 upmixing that you advocate. Your response is to attack me personally and characterize my comments as "whining and groaning." This is extremely childish.

I repeat what I said in my earlier post, if Ambisonics is so great, then why are there so many alternatives presented in this thread? Why did you yourself say there's lots of work to do? If Ambisonics is the great method you pretend, then surely your work is done. Why did you say you were working on ways to improve separation? Surely if Ambisonics were so wonderful, then there would be no need to "improve" anything!

Obviously, if my bidule is so bad, then so are the AC3Filter and Matrix Mixer on which it is based. Those seem to be quite popular. As well, my bidule's front generation is a greatly simplified version of the LCR Gerzon method that kempfand posted. His method is actually much better than mine in generating a solid soundstage (and far better than encoding traditional stereo signals with Ambisonics), but I didn't include it in my bidule because I wanted to honestly present it as what I said it was, an interpretation of the MatrixMixer DirectX filter. I repeat what I said in an earlier post: I never said it was the answer, just that I got better results for my listening taste with it than with your method. Someone asked me to post it, so I did. You seem to treat this whole thread as if it were some kind of competition and if someone comes along and criticizes Ambisonics, you pounce on them as if they were attacking you personally.

I don't think this is the first time I've seen you berate others in threads on the Doom9 forum. It's extremely unpleasant to read and it certainly adds nothing to the spirit of community present here. Please restrict yourself to a discussion of the methods and stop the name calling and personal attacks.

[Eye of Horus]

Originally posted by ursamtl
EoH, I don't care if your first language isn't English, the tone of this message is extremely offensive. I have never attacked you personally, I've simply expressed disappointment with the method of 5.1 upmixing that you advocate. Your response is to attack me personally and characterize my comments as "whining and groaning." This is extremely childish.

When you write in almost every post how bad the results are from Ambisonics and seldom add IMHO, yes, this is whining and groaning.
This has nothing to do with a personal attack, but simply with the fact that you are constantly saying how bad Ambisonics is for you in this originally Ambisonics thread. I consider your comments about Ambisonics offensive, because 7 out of 10 have another opinion, so you are insulting 7 out of 10 !


I repeat what I said in my earlier post, if Ambisonics is so great, then why are there so many alternatives presented in this thread? Why did you yourself say there's lots of work to do? If Ambisonics is the great method you pretend, then surely your work is done. Why did you say you were working on ways to improve separation? Surely if Ambisonics were so wonderful, then there would be no need to "improve" anything!


Talking about childish behaviour ! Stereo was so damn good, why search for anything better ??
All those remarks like "Ambisonics is so wonderful" in the above and other contexts, are suggestive and I think you meant it that way. So who is offensive here ?


Obviously, if my bidule is so bad, then so are the AC3Filter and Matrix Mixer on which it is based. Those seem to be quite popular. As well, my bidule's front generation is a greatly simplified version of the LCR Gerzon method that kempfand posted. His method is actually much better than mine in generating a solid soundstage (and far better than encoding traditional stereo signals with Ambisonics), but I didn't include it in my bidule because I wanted to honestly present it as what I said it was, an interpretation of the MatrixMixer DirectX filter. I repeat what I said in an earlier post: I never said it was the answer, just that I got better results for my listening taste with it than with your method. Someone asked me to post it, so I did. You seem to treat this whole thread as if it were some kind of competition and if someone comes along and criticizes Ambisonics, you pounce on them as if they were attacking you personally.

There is a difference between criticizing and putting down !
I can take good criticism without any problem. Otherwise Kempfand and me wouldn't have added so many extensions to the original bidule. I never criticized others bidules too. But...... when presenting your bidule with the attitude you spread out on Ambisonics, is IMHO not the way to discuss things here.
You can present your bidule without that constant nagging about Ambi !! Thanks !


I don't think this is the first time I've seen you berate others in threads on the Doom9 forum. It's extremely unpleasant to read and it certainly adds nothing to the spirit of community present here. Please restrict yourself to a discussion of the methods and stop the name calling and personal attacks.

You said something before about us ridiculizing Kpex methods ?
Now you make another suggestive remark.
Come with the facts !

The only one with his constant nagging about Ambisonics and who seems not to get the spirit of this forum is not me : it's you !
When I reacted to your remark about Kpex, you didn't come back with a "sorry". Not that I am waiting for that, but you set the tone ! Not me, I only react.

You stated clearly that your method gave you a better result. The first one you posted was completely wrong. The second should be fine. Well perhaps for your ears, but till now I have heard 3 negative experiences and not one positive one (except your own of course). When I say something about that you feel attacked ?
I would consider it a remark to stimulize you to come up with something working :)

Till now I sense more pretentions than inventions !
You put down Ambi, every time, you would come up with something better, but didn't.

On this forum we respect everyone's effort, but not by first putting down on the hard work of others first !

EoH

[ursamtl]

Originally posted by Eye of Horus

When you write in almost every post how bad the results are from Ambisonics and seldom add IMHO, yes, this is whining and groaning.

EoH, I've been involved with public forum messaging for well over a decade in many contexts, and most people I know consider using "IMHO" on a regular basis to be a sign of arrogance, as if those who use it really considers their opinion to be superior to opposing ones but want to make themselves appear more humble and avoid criticism for being arrogant. NOTE: I am not accusing you of this, just telling you why I avoid using such acronyms because, in my experience, they color the message. It is of course understood that if someone is expressing an idea on a forum and does not supply some sort of reference to another source or some sort of factual data as proof then he or she is obviously expressing his or her opinion, be it humble or otherwise. :)


This has nothing to do with a personal attack, but simply with the fact that you are constantly saying how bad Ambisonics is for you in this originally Ambisonics thread. I consider your comments about Ambisonics offensive, because 7 out of 10 have another opinion, so you are insulting 7 out of 10 !
This statement proves what I've been saying. If you consider criticism of the Ambisonics method to be "offensive" then you are taking it personally, as an attack on you as a person. I have never intended this! When I discovered your thread, I read it with a great deal of enthusiasm, as if I'd found some sort of holy grail. I wanted Ambisonics to work, but it didn't work for me. The more I read your comments about how superior it was, the more I tried redoing it to see if I'd overlooked anything. Eventually I came to the realization that it represents one approach to surround sound and not the ultimate solution. It is an approach or definition of surround that I personally find disappointing. I like surround sound to reveal something new and hidden in the music, some nuance that stereo couldn't deliver. To my ears, Ambisonics sounded like a collapsed stereo image across the fronts with the same image repeated in the rear speakers. Yes, there is a slight bit of separation, but it sounds to me like someone took a stereo signal and collapsed it to only about 20% of it's original width. I'm sure on a good sound system in a store, it sounds nice, but there's no magic. When I experimented with my bidule, the Gerzon LCR bidules, the AC3filter/MatrixMixer, and simply playing with the many stereo widener plugins, etc., out there, I discovered new sounds within my favorite music. Have you ever tried some of the QSound DirectX plugins? They take instruments that seemed buried in a mix and present them in a new light. Take a simple old Jimi Hendrix recording and run it through one of these and you get some stunning revelatory results. I believe it was you who said the Circle Surround II demo produced the best upmix surround you've yet heard (or perhaps kempfand, I don't have time to go searching through the forum to see who said it). These methods produce what I'm looking for, Ambisonics does not. If you take that statement as an attack on you or consider it offensive, then that's your problem, not mine. Do not make it mine by calling me a "troll" or dismiss my comments as "whining and groaning."


You said something before about us ridiculizing Kpex methods ?
Now you make another suggestive remark.
Come with the facts !

The only one with his constant nagging about Ambisonics and who seems not to get the spirit of this forum is not me : it's you !
When I reacted to your remark about Kpex, you didn't come back with a "sorry". Not that I am waiting for that, but you set the tone ! Not me, I only react.

Look, I don't have the time or desire to go searching back through every thread I've read over the last couple of months to quote examples of where I felt you were berating people. All I know is that I got that impression. If I'm wrong, I apologize, but that was how it seemed to me. If I remember correctly, you implied that kpex's SAD5.1 method was sort of a lowest possible benchmark, when you said that even it was better than mine. I seem to recall reading other comments where you implied it wasn't very good. Given my disappointing experiences with your Ambisonics method, can you not understand why I felt it unfair that you would berate someone else's method?

You stated clearly that your method gave you a better result. The first one you posted was completely wrong.

No, this statement is completely wrong! I already explained to you that my first bidule worked perfectly on the software I downloaded from needfulthings. The version of Kelly Industries BassManager in the needfulthings zipped file was 0.3. Check the Kelly Industries web site. It states very clearly that they changed the pinouts to the ITU standard in their v0.5. Still, they haven't changed their interface to match.

The second should be fine. Well perhaps for your ears, but till now I have heard 3 negative experiences and not one positive one (except your own of course). When I say something about that you feel attacked ? I would consider it a remark to stimulize you to come up with something working

Yes it did sound fine to my ears. I've also explained some possible reasons why there have been two other negative responses, the lack of adequate monitoring to achieve the levels required being the most probable. A professional recording studio monitors the mix of a recording while it is being mixed and using some sort of reproduction matching the mixing format. Look at section 0.8 your ambisonic guide. It suggests monitoring the front and rear lefts through the same left channel and the front and rear rights through the right channel. All that can give is a notion that the sound is present and at a certain loudness level. It cannot possibly give any sense of a surround mix. I have already said that the bidule I presented requires at least a 4.0 setup for monitoring to be effective. Every individual song requires such monitoring to bring out the best combination of factors. Perhaps a classical recording or a live performance recording could be set once and then left for the duration of the entire recording, but a modern studio album of popular music is full of many variations and nuances that require different approaches. For example, if the vocals were recorded with a lot of stereo reverb, this will dominate the rear dry level in my bidule and thus the wet level in SIR should be brought down. If the rears are without much reverb then the opposite is true. If the bass is tight and punchy already, then the LFE level needs to be brought down or it will overwhelm the final mix. If the original stereo mix was quite wide, the center level needs to be higher to avoid a washed out sound. If it was overly narrow, then bringing the front lefts and rights up with provide a nicer sense of a soundstage. But, if the fronts are too high, then a good vocal performance will lose its distinctiveness. I prefer this kind of approach because it allows me to tailor my mix to the music. Even if I put a separate gain on each channel in your Ambisonics bidule (which I've done), I don't have this kind of control to tailor the final mix. It just changes the balance of the overall sound the same way a balance control on a traditional stereo system works.

Till now I sense more pretentions than inventions !
You put down Ambi, every time, you would come up with something better, but didn't.
Surely if you read the detailed description of mixing levels I just wrote with any sense of objectivity, you must see that I didn't just throw my bidule together for fun, I did put some thought into it. You may not agree with the methodology or the results, but you cannot dismiss it as "pretention." That's an arrogant word that I consider offensive. No I don't consider my bidule an "invention" by any means and I do give you credit for presenting your application of Ambisonics as an approach to upmixing. I do not agree with you when you make a sweeping, absolute statment such as "Realistic space can only be achieved by Ambisonics and certainly not by separation." Unfortunately, I get the sense that just because I disagree with this, you feel I'm some sort of competitor or enemy of yours and so you dismiss me as a "acting like a troll" or "whining and groaning." EoH, if you feel offended that I criticized your Ambisonics method, I'M SORRY. Now, get over it!

[kempfand]

Thanks for the links. I tried a couple of these. While they certainly are more impressive than the results from the ambisonics bidules, I don't see them lending a convincing sound to a 5.1 mix of music. The vantage point would be somewhat like sitting on stage in the middle of a performance instead of sitting in the audience of a concert hall. I still got no sense of an acoustic space.
ursamtl: I have no problems that you are personally disappointed with Ambisonics/Ambiophonics. But for the rest of the readers, I would like to make a few (also subjective) comments from my side.

1) I think you (and others) are "victims" of what you (others) are used to in terms of 5.1 music. Note that this is not meant as a 'personal attack' or criticism. It's just how I see things.

I think the discussion in the previous few posts of the threat are not so much about "Ambisonic or Other Method". It's more about how the recording was done (mics used etc.) and how it was mixed before it came to the stores.

In detail: We are used to what we are used to, and I think the enormous (marketing) power of commercial recording companies and patent holders (e.g. Dolby & DTS just to name two) is often not considered enough as influencing factor on our behaviours. After all, if we are made to believe that what we buy is good, this means $$$ for the seller.

2) The same applies to how music is recorded and mixed before it ends on 5.1 DVD's or DTS CD's. Most 'modern' "Tonmeisters" (German expression for "mixing masters") have their style, which is often similar, i.e. music soundfield in the Fronts (incl. C), Ambience in the Surrounds. That's it. Full stop. In the worst case, there is but some echo and some audience applauding in the Surrounds. Horrible for my taste ...

More so: Professional listening tests (Verdie -> Listening Test (http://www.stud.tu-ilmenau.de/~proverdi/indexen.html)) raised the criticism, that Ambisonics gives too much room sound, and that isn't what the "Tonmeisters" want (even if it is correct!). It is also so that most people are so used to Stereo so hearing to much of the room won't sell (even if it is right !).

3) I agree with your point about "coloration" with Ambisonics, but this can be avoided by doing it right. Note that the most of the Ambisonics methods here (such as the guide at the beginning of this threat) are suited for the masses. Real Ambisonics uses signed filters (real-world impulse repsonses) to creat the B-format, as well as creating the decode to xyz speakers. See Conversion between UHJ and B-format (http://www.ramsete.com/aurora/conversion_between_uhj_and_b.htm#Audiomulch) for an outline on howto use signed filters for the B-format.

4) The vantage point would be somewhat like sitting on stage in the middle of a performance instead of sitting in the audience of a concert hall. That's how Ambisonics works. It brings you to the concert hall (restricted by how it was recorded and mixed, as outlined above). If you want to experience different positions in the audience, the only way to go is to use Impulse Responses (such as the ones from Noisevault) you mentioned and used.

Regards,
Andreas

[ursamtl]

Originally posted by kempfand
1) I think you (and others) are "victims" of what you (others) are used to in terms of 5.1 music. Note that this is not meant as a 'personal attack' or criticism. It's just how I see things.

No, I don't see this as a personal attack at all. It's a valid point, but.... what if we like being victims? In other words, mean what if we like the sound we are used to from 5.1 music? Give us some credit for being discerning listeners. When I first picked up my 5.1 system and started listening critically to the sound on the DVDs I rented or bought, I was astounded at the fidelity and clarity of reproduction. I'm not some newbie. I've been an audio fanatic since the mid-70s and have heard some pretty fine systems over the past 30 years! Properly mixed and presented 5.1 sound can be beautiful, providing a dimension and depth I've never heard otherwise. It's like the difference between looking out a small window at a lush garden and actually stepping out into the midst of it. I can't speak for anyone else, but if I'm a victim then I am a damned happy one!! :)

I think the discussion in the previous few posts of the threat are not so much about "Ambisonic or Other Method". It's more about how the recording was done (mics used etc.) and how it was mixed before it came to the stores.

This leads to the point I made earlier to EoH. Ambisonics might be an incredible approach given the right recordings but to take tracks that were not recorded properly for Ambisonics and attempt to encode and decode them as such doesn't provide the same effect. Call it Ambisonics for the masses if you like, but even following the instructions for what you call Real Ambisonics with signed filters, I found the result lacking life, or depth. It was certainly better than the original guide here, but the effect produced sounded similar to others I've heard where some work is done on phases at certain frequency. It didn't give me that "stepping into the garden" feeling I mentioned above.

Don't take my word for it, check out Ralph Glasgal's site (which you mentioned yourself when you presented your Ambiophonics bidule):

from Ambiophonics by Ralph Glasgal at http://www.ambiophonics.org/2ndEd_Preface.htm
There are two basic theoretical technologies that are prime candidates to replace stereo in elite high-end music reproduction, where mass marketing and complex technical concepts should not be (but of course are) major stumbling blocks. One is the wavefront reconstruction method often employing hundreds of microphones and speaker walls or, where recording is involved, Ambisonics. The Ambisonic wavefront reconstruction method generates the correct sound pressure and sound direction in a region that at least encompasses one listener's head. The other is the binaural technology method that more directly duplicates the live experience, independently, at each ear. Of course, both tehnologies aim to deliver to the entrance of your ear canal an accurate replica of the original sound field. The Ambisonic method does have the advantage that it can reproduce direct sound sources from any angle and so is quite well suited to non-concert events or movies. But since the Ambisonic wavefront reconstruction method requires a special microphone, a minimum of three (or better four recording) channels and a very complex decoder, is not as accurate as the binaural technology methods, and does nothing for the existing library of LPs and CDs it will not be considered further here.

I have yet to explore his approach to the point where I can form a valid opinion on its effectiveness for upmixing 2.0->5.1, but Glasgal obviously knows his stuff.

In detail: We are used to what we are used to, and I think the enormous (marketing) power of commercial recording companies and patent holders (e.g. Dolby & DTS just to name two) is often not considered enough as influencing factor on our behaviours. After all, if we are made to believe that what we buy is good, this means $$$ for the seller.

2) The same applies to how music is recorded and mixed before it ends on 5.1 DVD's or DTS CD's. Most 'modern' "Tonmeisters" (German expression for "mixing masters") have their style, which is often similar, i.e. music soundfield in the Fronts (incl. C), Ambience in the Surrounds. That's it. Full stop. In the worst case, there is but some echo and some audience applauding in the Surrounds. Horrible for my taste ...

That last word says it all, this is a matter of taste. We all have a right to our individual taste. I personally am looking for the big, multi-dimensional pristine sound I hear on 5.1 mixes. I like the soundstage in front of me. Since I play guitar, I know what it's like to be on stage in the midst of a band in full flight. That has its appeal and its unique sound as well. I don't want that kind of surround sound on my home system when I sit down to appreciate a piece of music.

4) That's how Ambisonics works. It brings you to the concert hall (restricted by how it was recorded and mixed, as outlined above). If you want to experience different positions in the audience, the only way to go is to use Impulse Responses (such as the ones from Noisevault) you mentioned and used.

Actually, that brings me to another point that's only somewhat related. While I'm enthralled by the quality of reverb impulse responses give as well as the ability to cpature the sound of vintage amps, mics, etc., I have yet to discover convincing impulses for giving me that "being there" feeling. For example, at work I can listen to music through a nice set of headphones, but the plugins or impulses suggested when I search on the net for "out of head" sounds just don't seem to work. In theory, with the right impulses I should be able to sit there and have the music sound as if I had a PA sitting in front of me, but it hasn't worked so far. But, I digress...

Anyway, time to do some more testing.

Regards

[pelmen]

first off a big thanks to all the people here who have worked hard on this whole 2.0 to 5.1 issue. i've been playing around with several of the posted bidules and have settled on the SAD5.1 for now as being the better of the bunch for my tastes.

1)
i'm certainly no expert when it comes to audio though but i was wondering if it was possible somehow to integrate a way of making use of dolby surround (and i think pro logic) if it exists in a 2.0 stream? from my understanding both dolby surround and prologic were multichannel signals mixed into a 2.0 format and re-expanded by the amp at the playback stage. i have a dolby surround test CD which i use to test the bidules (mostly using the channel identifier to make sure i ended up with all 6 channels going to the correct speakers :) and by far the result after processing isn't as good as the original. so i don't know if its even possible or not to detect if any such surround encoding is already existing in the 2.0 WAV file and once identified be able to make use of that knowledge to get an even cleaner 5.1 seperation?

2)
also it seems the main use for people in this thread is for turning music into 5.1. do people feel the same bidules are just as good for tv/movie audio or would a bidule layout specific for non-music sources provide a better end result?

3)
my setup is a computer with sound audigy and cambridge soundworks 5.1 amp/speakers and i connect my dvd players audio to the same amp. with the audigy card it plays 2.0 sources to all 5.1 channels already (but i'd like to put things onto DVDR so i can do other work with my computer). does anyone know how the bidule approach compares to the audigy? or even making use of the audigy hardware somehow if it is better to do hardware processing into 6 channel wav files?

4)
ideally what is the end goal people here are aiming to achieve? complete sound seperation with for example voices staying in the centre channel instead of "leakage" into all channels.

its a great thread so far, the only problems i've ever come across have been related to 2/4GB filesize limitations with various apps. i'm certainly happy with the current results when compared to boring old stereo :) but i hope it'll be possible to improve and get results closer to proper 5.1 source results :)

thx

[Eye of Horus]

Originally posted by ursamtl
EoH, I've been involved with public forum messaging for well over a decade in many contexts, and most people I know consider using "IMHO" on a regular basis to be a sign of arrogance, as if those who use it really considers their opinion to be superior to opposing ones but want to make themselves appear more humble and avoid criticism for being arrogant. NOTE: I am not accusing you of this, just telling you why I avoid using such acronyms because, in my experience, they color the message. It is of course understood that if someone is expressing an idea on a forum and does not supply some sort of reference to another source or some sort of factual data as proof then he or she is obviously expressing his or her opinion, be it humble or otherwise. :)

Although I can understand your point here, you should be aware that this is an international forum ! Every now and then making clear that it is your personal opinion, doesn't harm, but would make it much more clear to users who don't have English as native language.
TO ME it sounded as if you condemn the whole Ambisonics thing and when the majority of users is very satisfied with the results, I feel like you're nagging. A bit more IMHO would lead ME to the conclusion : hey, for this guy Ambi doesn't seem the solution !, instead of : hey, this guy is saying Ambi is thrash !
These are only nuances, which make it easier for non English speakers. I hope you understand that !


This statement proves what I've been saying. If you consider criticism of the Ambisonics method to be "offensive" then you are taking it personally, as an attack on you as a person. I have never intended this! When I discovered your thread, I read it with a great deal of enthusiasm, as if I'd found some sort of holy grail. I wanted Ambisonics to work, but it didn't work for me. The more I read your comments about how superior it was, the more I tried redoing it to see if I'd overlooked anything.


You must take into consideration that : a. it's more than a year ago ! and b. we were the first to use AND publish this way of using Bidule and Ambisonics ! Did you ever try the very first method of Ambi with Cooledit and the Aurora plugins ? The thread is still on this forum. Perhaps you can get better results with that approach....


Eventually I came to the realization that it represents one approach to surround sound and not the ultimate solution.


Again : things develop ! The first method lead to the Bidule method and that method lead to ALL the others ! It's the ease of Bidule that you can experiment so easy with all kinds of filters and layouts. More and more I get the impression you reacted to the first method only, without taking the time ago it was made into your opinions. For instance : you could have said the same about 78 rpm mono recordings :-) It's not very realistic to have a go at Ambi only, based on old material. We improve where we can. (actually our latest development, which is almost ready for publishing will be more for your taste !)


It is an approach or definition of surround that I personally find disappointing. I like surround sound to reveal something new and hidden in the music, some nuance that stereo couldn't deliver. To my ears, Ambisonics sounded like a collapsed stereo image across the fronts with the same image repeated in the rear speakers. Yes, there is a slight bit of separation, but it sounds to me like someone took a stereo signal and collapsed it to only about 20% of it's original width. I'm sure on a good sound system in a store, it sounds nice, but there's no magic. When I experimented with my bidule, the Gerzon LCR bidules, the AC3filter/MatrixMixer, and simply playing with the many stereo widener plugins, etc., out there, I discovered new sounds within my favorite music. Have you ever tried some of the QSound DirectX plugins? They take instruments that seemed buried in a mix and present them in a new light.


I have the same experience with the Ambi mix of War of the Worlds !!


Take a simple old Jimi Hendrix recording and run it through one of these and you get some stunning revelatory results. I believe it was you who said the Circle Surround II demo produced the best upmix surround you've yet heard (or perhaps kempfand, I don't have time to go searching through the forum to see who said it). These methods produce what I'm looking for, Ambisonics does not. If you take that statement as an attack on you or consider it offensive, then that's your problem, not mine. Do not make it mine by calling me a "troll" or dismiss my comments as "whining and groaning."


Understood, but now you explain. You didn't in one of your first posts ! I said this before : a simple : "Ambisonics doesn't produce what I'm looking for....." would have been enough.
Hey, even we have records where we have the same opinion !! That's why we are still developing new methods. I said it many times before and also in this thread : use the method that gives YOU the best results. Listen and do not just base your opinions on theory !
I never said it was the ultimate method !



Look, I don't have the time or desire to go searching back through every thread I've read over the last couple of months to quote examples of where I felt you were berating people. All I know is that I got that impression. If I'm wrong, I apologize, but that was how it seemed to me.


It's always dangerous to suggest based on things you think you remember without quoting the facts. Thus you can get a very strange opinion about people and I don't want to be a subject like this. thanks !


If I remember correctly, you implied that kpex's SAD5.1 method was sort of a lowest possible benchmark, when you said that even it was better than mine. I seem to recall reading other comments where you implied it wasn't very good.


The only comment we ever had on Kpex method was the fact that using 16 bits material (his method didn't work by then with 32 bits) made the music sound dull and that we could (and did) improve it with adding some reverb to the rears. We even made a Bidule to be able to work in 32 bits and : excellent results on especially electronic music ! So again : I don't know where you get that impression from that we condemn his method ! Again : don't be too fast with making YOUR own conclusions about people. Thanks !


Given my disappointing experiences with your Ambisonics method, can you not understand why I felt it unfair that you would berate someone else's method?


Again : where do you get this idea from ? And there is a difference in the tone of the messages...... Whatever I write here or in other threads and how hard I do my best : I'm still not English or American. Things I put in English can sound to you strange, while in my own language it sounds normal. We even had this discussion with Kpex, when I called the added reverb a major improvement, which is a quite normal expression in non English spoken countries, but at the same time I guess it sounded to Kpex as if we invented the wheel :-)
For you I only suggest you ask me how I meant something, instead of take it for granted that your interpretation of what I wrote is exactly how I meant it .It's easier for you to adjust your English than it is for me !


No, this statement is completely wrong! I already explained to you that my first bidule worked perfectly on the software I downloaded from needfulthings. The version of Kelly Industries BassManager in the needfulthings zipped file was 0.3. Check the Kelly Industries web site. It states very clearly that they changed the pinouts to the ITU standard in their v0.5. Still, they haven't changed their interface to match.


OK!


Yes it did sound fine to my ears. I've also explained some possible reasons why there have been two other negative responses, the lack of adequate monitoring to achieve the levels required being the most probable. A professional recording studio monitors the mix of a recording while it is being mixed and using some sort of reproduction matching the mixing format. Look at section 0.8 your ambisonic guide. It suggests monitoring the front and rear lefts through the same left channel and the front and rear rights through the right channel. All that can give is a notion that the sound is present and at a certain loudness level. It cannot possibly give any sense of a surround mix. I have already said that the bidule I presented requires at least a 4.0 setup for monitoring to be effective. Every individual song requires such monitoring to bring out the best combination of factors. Perhaps a classical recording or a live performance recording could be set once and then left for the duration of the entire recording, but a modern studio album of popular music is full of many variations and nuances that require different approaches. For example, if the vocals were recorded with a lot of stereo reverb, this will dominate the rear dry level in my bidule and thus the wet level in SIR should be brought down. If the rears are without much reverb then the opposite is true. If the bass is tight and punchy already, then the LFE level needs to be brought down or it will overwhelm the final mix. If the original stereo mix was quite wide, the center level needs to be higher to avoid a washed out sound. If it was overly narrow, then bringing the front lefts and rights up with provide a nicer sense of a soundstage. But, if the fronts are too high, then a good vocal performance will lose its distinctiveness. I prefer this kind of approach because it allows me to tailor my mix to the music. Even if I put a separate gain on each channel in your Ambisonics bidule (which I've done), I don't have this kind of control to tailor the final mix. It just changes the balance of the overall sound the same way a balance control on a traditional stereo system works.


Fully understood, but we tried to make a method that can be used by everyone. Newbies and pro's. You're more into the pro approach, while we try to make a method where you only have to input stereo and output surround. No adjustments in the bidule to make !!
Of course : gimme a few thousand dollars and I will buy the C3 soundanalyser, buy Wavelab 5 and Nuendo and a few other goodies and I can make a perfect mix with the output from C3.
But that's not our goal ! As soon as you have to taylor anything, we think our bidule is not good enough !!! And that's a very different apporach. Don't you agree ?
And : we will try to keep closer to the original stereo. I think that what we want is : when you play a surround CD and put your receiver into stereo mode, it should sound the same as the original stereo. In other words : We try to change the mix, but not the sound and when possible in an "convert only" bidule. If the stereo is narrow, we will leave it narrow ! That's the intention !
On a sidenote : every receiver will do DPL. Why bother at all ?
Why trying to re-invent the wheel ?
a. because it's just so much fun to experiment
b. because we're not always satisfied with DPL


Surely if you read the detailed description of mixing levels I just wrote with any sense of objectivity, you must see that I didn't just throw my bidule together for fun, I did put some thought into it. You may not agree with the methodology or the results, but you cannot dismiss it as "pretention." That's an arrogant word that I consider offensive. No I don't consider my bidule an "invention" by any means and I do give you credit for presenting your application of Ambisonics as an approach to upmixing. I do not agree with you when you make a sweeping, absolute statment such as "Realistic space can only be achieved by Ambisonics and certainly not by separation."


OK, so we disagree on that. No big deal ! Glad we're not all the same !! But I tried to nuance it, with the statements about hearing with both ears, in one of my previous mails.


Unfortunately, I get the sense that just because I disagree with this, you feel I'm some sort of competitor or enemy of yours and so you dismiss me as a "acting like a troll" or "whining and groaning." EoH, if you feel offended that I criticized your Ambisonics method, I'M SORRY. Now, get over it!

Feeling offended is too strong ! Irritated about how you put it in words in the first place : yes.
But these exchange of posts should clear things up and for me it did ! Did it for you ?

kind regards,

EoH

[Eye of Horus]

Originally posted by pelmen
first off a big thanks to all the people here who have worked hard on this whole 2.0 to 5.1 issue. i've been playing around with several of the posted bidules and have settled on the SAD5.1 for now as being the better of the bunch for my tastes.

1)
i'm certainly no expert when it comes to audio though but i was wondering if it was possible somehow to integrate a way of making use of dolby surround (and i think pro logic) if it exists in a 2.0 stream? from my understanding both dolby surround and prologic were multichannel signals mixed into a 2.0 format and re-expanded by the amp at the playback stage. i have a dolby surround test CD which i use to test the bidules (mostly using the channel identifier to make sure i ended up with all 6 channels going to the correct speakers :) and by far the result after processing isn't as good as the original. so i don't know if its even possible or not to detect if any such surround encoding is already existing in the 2.0 WAV file and once identified be able to make use of that knowledge to get an even cleaner 5.1 seperation?


I am not 100% sure, but wouldn't Softencode give that information when you load in a WAV with integrated Dolby ? Worth to try !



2)
also it seems the main use for people in this thread is for turning music into 5.1. do people feel the same bidules are just as good for tv/movie audio or would a bidule layout specific for non-music sources provide a better end result?


It's just a matter of trying ! Only you can decide what sounds the best for what you want. I can imagine you want more on the LFE when you're dealing with non-music. Just experiment and try to find what you like most. There is no general rule, unfortunately !


3)
my setup is a computer with sound audigy and cambridge soundworks 5.1 amp/speakers and i connect my dvd players audio to the same amp. with the audigy card it plays 2.0 sources to all 5.1 channels already (but i'd like to put things onto DVDR so i can do other work with my computer). does anyone know how the bidule approach compares to the audigy? or even making use of the audigy hardware somehow if it is better to do hardware processing into 6 channel wav files?


See the above answer ! Just try it and compare the results and use what you like most !
I found out that a simple listening test says more than the opinion or theory of others :-)


4)
ideally what is the end goal people here are aiming to achieve? complete sound seperation with for example voices staying in the centre channel instead of "leakage" into all channels.


Simple : the end goal is to get the best surround possible from stereo. How ? That is so difficult to answer, because the original stereo more and more decides which bidule you should use to get the best result. There are recordings where we can make a perfect (almost !) separation 5.1 from with indeed a voice only in the center speaker. But there are also recording that will sound better with Ambisonics or with Ambiophonics. It's too difficult to give a general answer to which bidule to use.
A lot of people are very enthusiastic about the SAD51inBidule we made. I like it very much on electronical music, but not on music with singers ! It's also a matter of taste and personal preferences.



its a great thread so far, the only problems i've ever come across have been related to 2/4GB filesize limitations with various apps. i'm certainly happy with the current results when compared to boring old stereo :) but i hope it'll be possible to improve and get results closer to proper 5.1 source results :)

thx

LOL !!! Define proper !!! LOL !!!

kind regards,

EoH

[pelmen]

EoH,

by "proper 5.1" i basically meant what a professionally produced 5.1 track can sound like. like what you'd expect on a good special effects movie with things audibly moving clearly throughout the room. so while atm we may be able to take a 2.0 copy of say the battle sequence from Master and Commander it might sound "big" when we turn it into 5.1 but it is still a long way from the original 5.1 soundtrack.

i guess there's no such thing as a "surround sound benchmarking" system or something. something that could analyse a 5.1 track and show you a visual "spread" of the soundstage...maybe something like a colourspace graph. it could look at how much each channel has in common at any given instant and graph it over time. so when you put a single mono track into all 6 channels it would show you a dot in the middle of the room because each channel is identical. something along those lines anyway i was hoping might exist, it might be a way of helping to benchmark bidules and fine tune them? or am i just fantasising? would also help people like myself who dont have a very good surround setup nor a good "ear". Also being able to use an app to test a file i'd probably trust more anyway since the audigy spits everything i play out to 5.1 channels anyway so its hard to tell if i'm hearing the results of the upmixing or the audigy's own processing.

so far i've been using a channel seperation dolby surround track for my testing. it is a voice speaking each channel in turn (left, centre, right, surround) followed by a few seconds of (pink? white?) noise. then i load up each of the 6 mono wavs at the end and "measure" the height of the noise sections for each channel mentioned by the audio. from that i can see how much cross channel leaking there is easily. since i can't subjectively trust my hearing for small differences between techniques i tend to use this one track as my benchmark to compare a new bidule layout to what i've used already and go from there. its probably the best testing method i'm going to be able to use.

does anyone know if there are prologic channel seperation tests available free online in a two channel format? my dolby surround test cd rips to a 2 channel wav and that wav clearly plays in each designated speaker as the original cd does. from memory prologic should be the same, just it gives more channels but i dont have a prologic test cd but something like that i'd have thought would be available free online to help developers.

i've been having a lot of fun playing around and exploring this process anyway since i recently got a dvd burner. i originally stumbled across this thread, it was not even something i'd considered possible but its certainly given me some inspiration to put a bit of extra effort into my vhs to dvd conversion so i end up with something that will last a long time to come. can't wait to see what new tricks people come up with! jsut wish i understood it all :)

thx

btw.. SMILE!!! its the weekend!!! :)

[ursamtl]

Originally posted by kempfand
ursamtl: I have no problems that you are personally disappointed with Ambisonics/Ambiophonics. But for the rest of the readers, I would like to make a few (also subjective) comments from my side.


Andreas,

Just to follow up on this, let me say that I retried your Ambiophonics dipole last night and was quite impressed with the results. First of all, for some reason the wave files in the www.app.demon.nl download would not load into SIR. However, the version on Needfulthings did work.

While the separation was still lacking and the rears were disappointing, the biggest thing I noticed about the front sound was a significant enhancement to the bass. There was much more clarity, definition, and punchiness without it sounding over-EQed. On some tracks it was actually quite stunning. In addition, the overall mix seemed to possess the kind of clarity and richness I normally associate with 5.1 mix.

What was missing of course was the sense of width and depth to the rear. First of all, even if theoretically the speakers are supposed to be spaced at +-10°, this just collapsed the width of the soundstage too much for me. Returning my speakers to +-25° (the way my PC system is setup this is the maximum front width I have to work with), gave me a better sense of width. Still, on some music, it seemed too dry. For example, I have a beautifully remastered CD of Elton John's Madman Across the Water. Running "Tiny Dancer" through this bidule resulted in the vocals sounding incredibly dry, virtually devoid of any ambience. While one might argue that this is a more realistic recording of his actual voice, the effect was not as pleasant. I tried inserting a stereo expander VST and this improved the image significantly. There is a sweet spot at approximately 120-130% width. Anything over that improves the sense of ambience but starts to negatively affect that glorious bass response.

There remains the problem of the rears. I don't want to just slap some reverb on it as that can get tedious. But the Ambisonics rears in the original bidule just don't work for me. In fact, I find that they detract from the enhanced fronts.

I have a couple of ideas for both the front and rear that I'll try to work on this weekend. Nonetheless, this is truly an encouraging result and definitely a step in the right direction.

Ursa

[Eye of Horus]

Originally posted by pelmen
EoH,

by "proper 5.1" i basically meant what a professionally produced 5.1 track can sound like. like what you'd expect on a good special effects movie with things audibly moving clearly throughout the room. so while atm we may be able to take a 2.0 copy of say the battle sequence from Master and Commander it might sound "big" when we turn it into 5.1 but it is still a long way from the original 5.1 soundtrack.


Starting with 6 mono wavs recorded separately would be ideal. But we're using stereo and need to extract as much as possible from only 2 channels !
I don't think we can get the same results, but with music we can get close !


i guess there's no such thing as a "surround sound benchmarking" system or something. something that could analyse a 5.1 track and show you a visual "spread" of the soundstage...maybe something like a colourspace graph. it could look at how much each channel has in common at any given instant and graph it over time. so when you put a single mono track into all 6 channels it would show you a dot in the middle of the room because each channel is identical. something along those lines anyway i was hoping might exist, it might be a way of helping to benchmark bidules and fine tune them? or am i just fantasising? would also help people like myself who dont have a very good surround setup nor a good "ear". Also being able to use an app to test a file i'd probably trust more anyway since the audigy spits everything i play out to 5.1 channels anyway so its hard to tell if i'm hearing the results of the upmixing or the audigy's own processing.


I do think there is some software that can analyse surround sounds, but I am not sure. I found that the C3 analyser is the most promising, but I could only find a Chinese (or Japanese ?) site, and..... no english translation there. But seeing the screenshots for stereo, it makes me feel this could be the ideal analyse software !
OTOH what would you exactly do with such software ?
I mean : as you say with a single mono the spot is in the middle, but this doesn't represent the ideal situation for the listener.
Music can be loud sometimes in a song from one speaker only and than with the same benchmarking you're thinking of, that spot won't be in the middle. And that doesn't indicate the mix is wrong then !!


so far i've been using a channel seperation dolby surround track for my testing. it is a voice speaking each channel in turn (left, centre, right, surround) followed by a few seconds of (pink? white?) noise. then i load up each of the 6 mono wavs at the end and "measure" the height of the noise sections for each channel mentioned by the audio. from that i can see how much cross channel leaking there is easily. since i can't subjectively trust my hearing for small differences between techniques i tend to use this one track as my benchmark to compare a new bidule layout to what i've used already and go from there. its probably the best testing method i'm going to be able to use.


Aha ! Yes , this way makes sense !
Can you upload that track to alt.binaries.sounds ?
It would be very interesting to use as test sound for the new bidules we're developing (Kempfand and me).
They are already very nice and give more separation than the things I tried till now, but such a track would be very helpful to finetune even further !


does anyone know if there are prologic channel seperation tests available free online in a two channel format? my dolby surround test cd rips to a 2 channel wav and that wav clearly plays in each designated speaker as the original cd does. from memory prologic should be the same, just it gives more channels but i dont have a prologic test cd but something like that i'd have thought would be available free online to help developers.


I never heard of it, but perhaps some googling can help ?
I will do a search online too and let you know if I could find anything.....


i've been having a lot of fun playing around and exploring this process anyway since i recently got a dvd burner. i originally stumbled across this thread, it was not even something i'd considered possible but its certainly given me some inspiration to put a bit of extra effort into my vhs to dvd conversion so i end up with something that will last a long time to come. can't wait to see what new tricks people come up with! jsut wish i understood it all :)


I wish that too !! :-)
I am not the writer of the bidules made ! I am more the guy with the (sometimes very weird) ideas behind bidules and I try to translate every technobabble into a useable guide. Some of the techniques used to get the results are too difficult for me to understand. The outcome however is for me the only thing important !!



thx

btw.. SMILE!!! its the weekend!!! :)

ALWAYS smiling :-)

kind regards,

EoH

[ursamtl]

Originally posted by pelmen
my setup is a computer with sound audigy and cambridge soundworks 5.1 amp/speakers and i connect my dvd players audio to the same amp. with the audigy card it plays 2.0 sources to all 5.1 channels already (but i'd like to put things onto DVDR so i can do other work with my computer). does anyone know how the bidule approach compares to the audigy? or even making use of the audigy hardware somehow if it is better to do hardware processing into 6 channel wav files?


Pelmen,

If you haven't done so already, run (don't walk) all the way to www.kxproject.com and see if your Audigy model works with the kx drivers. These are free ASIO-compatible drivers for the Creative Labs cards based on either of two EMU DSPs (most internal Soundblaster Live's, Audigy's and Audigy 2's) and they can turn your card into a much more powerful tool for music. Their DSP applet is very similar to Plogue Bidule in that you can customize routing and connections graphically to suit your needs. With your system, you could easily create bidules with multichannel ASIO outputs for realtime monitoring of your surround mixes. In Plogue, if you're sample rate is set to 44.1k, just insert a kX ASIO output device (15 channels), or if at 48k, a kX ASIO Duplex input/output device (15 channels each way). Then check your routings in the kX DSP to make sure you're routing the 5.1 channels properly. After that in Plogue, you can connect each channel in real time while playing back and experimenting with different approaches, settings, etc.

[Eye of Horus]

Originally posted by ursamtl
Andreas,

Just to follow up on this, let me say that I retried your Ambiophonics dipole last night and was quite impressed with the results. First of all, for some reason the wave files in the www.app.demon.nl download would not load into SIR. However, the version on Needfulthings did work.

Hey, the link to my homepage is nice, but realize people clicking on that link won't get what they want :D :D :D

I will have another look into these wavs, because they should work in SIR, because we used them too.....

I also found this method giving a good sound, but my objection :
a. I cannot move my speakers and
b. I am not in the position to buy 2 extra ones :-)

(I'm sure you will like our new ones...... but they need some more work on them before we publish them.)

Just curious : do you convert your stereo to 32 bits first ?

kind regards,

EoH

[ursamtl]

Originally posted by Eye of Horus
Hey, the link to my homepage is nice, but realize people clicking on that link won't get what they want :D :D :D

Oops, sorry. I posted just the text without using the http:// button on the forum interface because I didn't want to repost a link to a file that might potentially be corrupted. I didn't notice that the forum software automatically converted it to a link.

I also found this method giving a good sound, but my objection :
a. I cannot move my speakers and
b. I am not in the position to buy 2 extra ones :-)

Well, did you try the dipole without moving your speakers? I found the results were better with my speakers at +-25°. At +-10° there was very little sense of width to the sound at all.

(I'm sure you will like our new ones...... but they need some more work on them before we publish them.)

Just curious : do you convert your stereo to 32 bits first ?
I look forward to the new bidules. As for the 32 bit conversion, yes, I have done this a couple of time for comparison sake and it does provide a marginally cleaner sound especially in high-frequency details such as cymbals, etc., but I didn't notice any appreciable difference in terms of the Ambisonic effect. I'm lucky in that, for one thing, I can monitor my mixes in real time on my PC so I don't always burn DTS CDs. In addition, my DVD player reads CDRWs so when I do write burn DTS CDs, they're temporary tests to see how the sound is transferring to my 5.1 system. Once I settle upon an upmix method that gives me the results I seek, I'll do some proper full CDs and start with 32-bit files. I still haven't found a satisfactory answer concerning the whole question of dithering. It's generally agreed that going from 32-bit to 16-bit with no dithering reduces sound quality, but so far the Surcode documentation hasn't mentioned whether it downsamples to 16-bit and if it does so, does it use dithering. All it says it that the program accepts 16- or 32-bit files for input. The CD redbook spec is 16-bit, but it's not clear that a DVD player recognizing a CD but playing a DD or DTS wave file does so only if the file is 16-bit.

By the way, yes, your earlier post did clear things up between us. I'm at work and don't have time for a detailed reply, but yes, there are surely cultural nuances to our communication that can cause misunderstandings. For example, I once worked in an office with some Swiss and German personnel who were working on a contract for the Canadian govt. here. They would get into discussions that seemed to me like they were ready to haul out weapons and kill each other, yet walk out of the office at the end of the day smiling at each other! Here I was the poor Canadian fellow shocked into thinking I was about to witness murder! These folks told me this was simply their European temperament and nothing I should be concerned about. Perhaps we Canucks are just too polite! Plus, having worked with them for awhile I found I was adopting some of their attitude.

Have a great weekend!
Ursa

[Eye of Horus]

Originally posted by ursamtl
Oops, sorry. I posted just the text without using the http:// button on the forum interface because I didn't want to repost a link to a file that might potentially be corrupted. I didn't notice that the forum software automatically converted it to a link.

No need to apologize : perhaps it gives some extra visitors :-)



Well, did you try the dipole without moving your speakers? I found the results were better with my speakers at +-25°. At +-10° there was very little sense of width to the sound at all.


Yep ! I try out every bidule :-)


[b]
I look forward to the new bidules. As for the 32 bit conversion, yes, I have done this a couple of time for comparison sake and it does provide a marginally cleaner sound especially in high-frequency details such as cymbals, etc., but I didn't notice any appreciable difference in terms of the Ambisonic effect. I'm lucky in that, for one thing, I can monitor my mixes in real time on my PC so I don't always burn DTS CDs. In addition, my DVD player reads CDRWs so when I do write burn DTS CDs, they're temporary tests to see how the sound is transferring to my 5.1 system. Once I settle upon an upmix method that gives me the results I seek, I'll do some proper full CDs and start with 32-bit files. I still haven't found a satisfactory answer concerning the whole question of dithering. It's generally agreed that going from 32-bit to 16-bit with no dithering reduces sound quality, but so far the Surcode documentation hasn't mentioned whether it downsamples to 16-bit and if it does so, does it use dithering. All it says it that the program accepts 16- or 32-bit files for input. The CD redbook spec is 16-bit, but it's not clear that a DVD player recognizing a CD but playing a DD or DTS wave file does so only if the file is 16-bit.

By the way, yes, your earlier post did clear things up between us. I'm at work and don't have time for a detailed reply, but yes, there are surely cultural nuances to our communication that can cause misunderstandings. For example, I once worked in an office with some Swiss and German personnel who were working on a contract for the Canadian govt. here. They would get into discussions that seemed to me like they were ready to haul out weapons and kill each other, yet walk out of the office at the end of the day smiling at each other! Here I was the poor Canadian fellow shocked into thinking I was about to witness murder! These folks told me this was simply their European temperament and nothing I should be concerned about. Perhaps we Canucks are just too polite! Plus, having worked with them for awhile I found I was adopting some of their attitude.

Have a great weekend!
Ursa


LOL !!! How recognizable !!!

Even some of my friends in the USA (for over 8 years now) still have problems in a discussion with me :-) They always think I'm angry or too sharp.

Have a good (productive !) weekend too !

kind regards,

EoH

[kempfand]

Ursa: Regarding the Stereo Dipole method I was posting here, I wrote it is a prototype, as the 2 "only" speakers for the Ambisonics (placed in the back) are probably not enough to make it a full AmbioPhonics by the books. Do do things correctly, there are at least 4 speakers (square) or even more, as well as convolution with real space impulse responses. Also, the dipole spreads the image in the front onyl (ideally over 180 deg), but there are methods which use a 2nd dipole in the back.

I have also read from two sources (Farina being one of them) that a frontal dipole works best if it comes from above (i.e. two closely spaced speakers at the ceiling to produce a dipole). I can't really report any personal experience on this, as my other half would probably get a nervous breakdown if I changed the living room too much ... :D ... but maybe one day ... :)

On the 32 bit conversion of the feeding Stereo-in, I quote here a Farina post from April 2002 from SurSound, which you/all might find interesting.
Now CEP2 instead defaults to the most usual mode.
In any case, being a float number, no dither is required (nor any special trick) when rescaling. Most availablòe libraries of loops and sound effects are actually 32-bits waveform. All known software smoothly converts betweeen 32 and 16 bit waveforms, so probably You are already using a lot of 32-bits WAV files, and You do not know this.... CEP, as any other program natively supporting 32-bits files (such as Audio Mulch, Cubase, Sound Forge, etc.) plays without problems a 32-bits file over a 16-bits sound board.

This does not pose any conversion problem, because the sampling rate is unaffected, and the bit depth can always be increased without any particular trick.

This is advantageous also when the original sampling was also at 16 bit. A complete mixing chain, which start at 16 bits, makes all intermediate computation and fading at 32 bits-float, and switch back to 16 bits only at the end, gives a sound quality which is definitely much better than a 16-bits integer processing chain.

Apart from the storage space on the HD (which is very cheap now), all the computations made in float-32bits are nowadays very efficiently performed in the FPU of the computer, which can outperform the most powerful DSP chips by a factor 16 or more, if the computer is properly programmed (making extensive use of the SIMD instructions, for example). This means that, in practice, che computational load is lower working in 32-bits float than in 16-bits integer! Regards,
Andreas

[ursamtl]

Hi,

I've just added a zip file containing three SIR delay compensation groups to the FTP server and to my web page at www.geocities.com/ursamtl.

This zip file contains three Plogue Bidule groups I've created as alternatives to loading dummy instances of SIR in your bidules to compensate for its sample delay of 8960 (in SIR v.1005). The savings in CPU usage can be significant, ranging from 5-10%


SIRDelComp2 provides 2 channels in and out with a 8960-sample delay on each. This replaces one dummy instance of SIR
SIRDelComp4 provides 4 channels in and out with a 8960-sample delay on each. This replaces two dummy instances of SIR
SIRDelComp4+2 also provides 4 channels in and out with a 8960-sample delay on each, plus 2 channels directly through (pins 5 & 6) for those who want 5.1 channels on one neat bar. For example, L, R, C, LFE, might have delay compensation with channels 5&6 connected to one instance of SIR elsewhere in the bidule.


When I conducted tests on my PC (Athlon Thunderbird 1.1GHz), the results were as follows:
Average CPU usage
-----------------
1 SIR with stereo impulse + 1 dummy SIR: 28%
1 SIR with stereo impulse + 2 dummy SIRs: 32%
1 SIR with stereo impulse + SIRDelComp2: 20%
1 SIR with stereo impulse + SIRDelComp4: 22%
1 SIR with stereo impulse + SIRDelComp4+2: 22%
2 SIRs with stereo impulse + SIRDelComp2: 27%
2 SIRs with stereo impulse + 1 dummy SIR: 33%


Although I did no testing of memory usage, obviously there should be some saving given that SIR is 1.6MB and the groups are a few KB!

Enjoy,
UrsaMtl

[Shayne]

Originally posted by kempfand


........So you must set 'never' for the WAVE Format Extensible (with Preference)........

Regards,

Andreas

Sorry for the late reply but thank you Sir. All is fine now and my computer no longer needs to live in the past.

I have checked pin outs and this release seems to be consistent with besweet (all is well). I did find myself dropping the gains on lfe down to -6.5 db but we are batching again.

Weird being a forced beta tester. But free is good.

Peace

[kempfand]

Originally posted by ursamtl
I have yet to discover convincing impulses for giving me that "being there" feeling. For example, at work I can listen to music through a nice set of headphones, but the plugins or impulses suggested when I search on the net for "out of head" sounds just don't seem to work. In theory, with the right impulses I should be able to sit there and have the music sound as if I had a PA sitting in front of me, but it hasn't worked so far. But, I digress...Check the samples at the bottom of The FIReverb Suite audio demonstration (http://www.catt.se/suite_music/index.htm) for beautiful "out of head" binaural listening experience.

The FIReverb Suite is a superbe package IMHO (as well as the small version CATT-Acoustic (http://www.catt.se/)

Cheers,
Andreas

[kempfand]

Originally posted by Shayne Weird being a forced beta tester. But free is good.[/B]

Plus it's one of the rare examples where beta is labelled as beta. Far too often, betas is sold for $. Also, I've never seen an example where the company (Plogue) was so responsive to implement enhancements, and help for bug fixing.

I'm not a fan of advertising, but I'll be in what will hopefully be a line when the free-version changes to "fee-version".

Cheers and peace,

Andreas

[Shayne]

Well you would think a batch process routine would need to be implemented to make it truly useful that there would be any sort of line.

But they need not listen.

Peace

[Umma]

Well you would think a batch process routine would need to be implemented to make it truly useful that there would be any sort of line.

I wanted a batch routine, too, when I first started with it, but Plogue crashes on me after 5 or so straight uses without closing it down and restarting it. I just use one big wav file now.

Plogue wasn't even developed with this sort of music manipulation in mind. I may be wrong, but I think it was mostly meant for midi or homegrown electronic music. It would be nice if they took out the midi part and sold a "mini-Plogue" to those of us who like to fool with music conversions such as those we are doing over here. :)

[kempfand]

Originally posted by specise_8472
Have just uploaded to the FTP server some new toys.
Daphy will put on Web server when he can.

It is a very good Mono to stereo conversion utility. And feeding the result into one of the above 2 to 5 methods produces astounding results. Better than I had hoped for. I myself personally use my Allinone process.

/// snip ///

For those interested - it is basically feeding the mono signal through two banks of cascaded filters. Each bank is a 19 pole Allpass filter. Then feeding the resultant signals through my x-talk filter to produce better channel seperation.

Species: I tested it (both the original one, as well as the 44.1 & 48 ones). Setup: Mono Signal -> Mono_To_Stereo -> Audio File Recorder. The levels of the recorded L and R are extremely different. Same results when EoH runs this.

Guess it's just a small wrong routing somewhere, and would appreciate if you could re-check.

Kind regards,
Andreas

[Tantulus]

Well,finally I got around to experimenting with Bidules. I decided to add the "ambient hall" impulse from NoiseVault to SIRs on the SAD5.1 bidule. At that time I also decided to spring for a better center speaker and a powered subwoofer. This gave me a chance to try out my "SIR" SAD5.1 adaptation in a decent listening room. Honestly, I was blown away! My cramped living room never did justice to surround sound. Anyway, I felt the soudscape fill the room and the Impulse Response gave enough ambience that when I shut my eyes I could actually imagine being in a concert hall. My impression is that at least with orchestral music one can get a sense of "broader-than-room-size" with a decent speaker setup. Although, at times I felt that I was just bathed in the music rather than sensing the music orginated from the front as in a concert hall, I was very satisfied. Just as mentioned earlier in the thread, the salesman wanted to know how this was done and called others in the store to hear the disc.

Also, as Kempfand noted, there are discs that capture the ambience of the enviroment where they are recorded in stereo and I found with these recordings I get an added benefit of nice localization of instruments. Ambience was augmented without the need for using impulse responses. I guess the results of encoding into 5.1 ultimately depends on the orginal recording process, the speaker setup, the room acoustics and the bidule used. Once again, hats off Kempfand; my receiver has DPLII, DTS, and Neo6 settings. None of them come close to the bidules produced in this forum. At best I get some broading of music and tighter vocals in the center channel with TV or CD's.

Anyway, that's my two cents!

Finally, please, please, could somebody explain what the AmpIn and AmpOut settings in the HNM filter does? I looked through the whole thread I can only find references to settings but not their meaning.

Also, some recordings tend to be too bassy and no amount of changing the low pass control on the subwoofer corrects this. I know I can add a gain control but can this be adjusted with the HNM filters connected to LFE?

A less timid but greatful,
Scott

[ursamtl]

Learning to like Ambisonics
This forum has several guides or suggestions for converting 2-channel stereo to some variation of surround sound, usually 5.1. Each method is fun to experiment with, and each presents something different. I’ve been playing around with them and trying different combinations for a few weeks now. What follows is the result of my experiments so far. I’m quite pleased with the sound it provides and I hope you’ll enjoy it too.

Basically, I took the equations on Angelo Farina’s web site and created a bidule group that converts 2-channel stereo (UHJ) to B format (WXY, the Z is null because no height info is available). This group is a basic Ambisonic encoder with no controls for azimuth, elevation, distance, or anything. It just creates the basic signals.

Then I used different combinations of the equations on Richard Furse’s web site to put together a similar basic bidule group to serve as an Ambisonic decoder. I tried the calculations for the “Surround” speaker layout, but the sound wasn’t that impressive, and Furse’s site points out that this results in symmetry distortion. I then considered the Pentagon suggested by EoH and kempfand, but when I looked at Furse’s suggested equations, I discovered that they do not form a truly equilateral pentagon. Thus, rotating this by 36° to provide three front channels and two back results in a sort of lopsided listening field. To lay out one's speakers to match the calculated points of the rotated Pentagon, one would have to place the center speaker roughly three degrees to the left of center, the left speaker at 74° from center and right speaker at 71° from center. Even then, the center of the rotation would not be centered between the left and right speakers.

http://stevethomson.ca/audio/guides/Pentagon36.gif

I got better results from the Octagon 2 layout. Since it calls for 8 speakers, and there are only five available in my target 5.1 system, I created phantom speakers by splitting the signals for each of the side left, side right, and center surround speakers and feeding them through the two speakers on either side of each phantom.

http://stevethomson.ca/audio/guides/Oct2layout.gif

I tried the equations for both the strict spherical harmonics layouts and the controller opposites. As Furse points out, the latter results in a more diffuse image. I found the spherical harmonics layout sounded better.

When I put this all together and tested it, the sound was quite good, much more expansive than any other Ambisonics I’d heard to date, yet still providing that sometimes uncanny sense of realism that simple Dolby Surround upmixes just can't convey.

I then played around with various enhancements to the input stereo signal, such stereo widening, brightening the stereo edges with EQ, etc. These provided some interesting results, but it got complicated and CPU intensive.

Then I thought about the nature of the signals, if the Y signal consists of the spherical harmonics representing the width of the soundfield, then boosting it with respect to the others should spread the sound out. I tried it, and it worked! More importantly, nothing else in the sound suffered. Stereo wide effects generally kill tight, accurate bass response, but this widened Ambisonic image did not. I also tried the same trick on the X channel to see if a similar improvement would occur in the depth of the soundfield, but it didn’t provide such a spectacular result. However, applying reverb to the X channel--even simple mono reverb--does provide some amazing results.

So, I put it all together in a bidule consisting of nested groups, and learned how to put together some controls in a bidule. This resulted in the following controls, which are available from a control panel when you double-click on the UpMix Studio group in the center of the bidule:

http://stevethomson.ca/audio/guides/Controls.gif

Input Level A standard bidule Gain. The standard setting is at 0dB.
LFE? Checking this box causes all really deep bass centered around 32Hz or so to be redirected to the LFE channel. If the box is unchecked, the channel will be null. If you have monitoring capabilities with a subwoofer on your PC, try disconnecting all other outputs but the fourth from the left, and you'll hear exactly what's going through the LFE channel. By the way, the LFE channel is not affected by the Bass Boost controls below.

Enable Soundfield and Bass Boost? I put this control there for the purists. If you don’t want any width control or bass boost, and wish to just upmix to 5.1 using the basic Ambisonic implementation in this bidule, uncheck this box. If you do want the enhancements, check it!

Soundfield Width Gives you the chance to boost the Y or width channel, thus spreading out the sound. This also tends to make the highs a little crisper, but there is no EQ involved. This one control provides an incredibly spacious yet accurate soundfield. Of course, it all depends on the source file. Try, some Floyd or Roxy Music's Avalon.

Bass Boost Level Adjusts how much boost there is to the bass frequencies below the crossover frequency.

Bass Boost Crossover Frequency Moves the upper limit of the bass boost. If you’d like a more solid bottom but without much more bass, Move this all the way to zero and use the Bass Boost Level to boost the deepest bottom end a bit. This puts a nice bottom on drums without overpowering anything.

And if you really like the sound of the grand canyon…..
If you have a fairly dry piece of music wish to add some reverb, try adding it only to the X channel. I was astounded by the huge, professional sound image I got when I did this. I also tried running both the X and the Y channels through the two channels on a SIR VST. It was pretty massive, but almost too much so. Have fun, play a bit.

http://stevethomson.ca/audio/guides/AddSIR.gif

Monitoring
To get the most out of this system, it’s best to monitor the sound while you make the adjustments. If you don’t have a surround sound system on your PC, just hook up the standard 2-channel output. I’ve found that this gives you a really clear idea of what the sound of the final mix will be. Do not hook up the surrounds, the center or the LFE to this, however, as the sound will be inaccurate. It amazes me just how nice even a 2-channel mix sounds through this bidule. I think I'll use it to do some new mexies for listening in the car.

Note: You’ll have to remove this monitoring sound output device if you want to process in Plogue Bidule’s offline mode.

http://stevethomson.ca/audio/guides/Monitor2chan.gif

Of course, if you have a surround system and a multichannel ASIO setup, you can monitor the whole mix.

http://stevethomson.ca/audio/guides/MonitorASIO.gif

So, have fun. As I said, this provided me with some very nice surround mixes. Unless you add some reverb, the natural ambience of the original recordings is all you have, and most of the time, that’s all you need.

<b>You can download the resulting bidule from

My website on Geocities (http://www.geocities.com/ursamtl/audio.html), another link

here (http://stevethomson.ca/audio/guides/UUMStudio.zip)

or an offline processing mode version from the Projects\Ursa's UpMix Studio folder on the NeedfulThings server (http://www.needfulthings.webhop.org).
Thanks to daphy for doing the mod for offline processing and linking the player and recorder!

As for what to do with the multichannel wave file, that’s been covered extensively in this thread and others. If you’re encoding to AC3 using SoftEncode, just load the file in and launch the encoder. Otherwise, use Besweet to split the 6-channel file and then encode using your preferred AC3 or DTS encoder. By the way, as mentioned earlier in this thread, converting your input wave files to 32 bits does indeed make a difference. The sound difference may be subtle, but you will notice more detailed high frequencies, etc.

For more reading, check out the following:

This whole fascinating thread! (http://forum.doom9.org/showthread.php?s=&threadid=60137)
Thanks to EoH, kempfand, kpex, species, daphy, andy and all the others, you’ve started something very, very cool.

Conversion between UHJ and B-format (http://www.ramsete.com/aurora/conversion_between_uhj_and_b.htm) by Angelo Farina.

First and Second Order Ambisonic Decoding Equations (http://www.muse.demon.co.uk/ref/speakers.html) by Richard Furse.


UrsaMtl, June 6, 2004

Moderator Edit: Fixed Image links per request of ursamtl

[kempfand]

ursamtl: Thanks for sharing your work. It's always good to see what others do, and playing with the various possible schemes in Bidule is so much fun, as you also mention. A few remarks:

1) In the current uploaded bidule (June 6), I believe that the decoding maatrix for C is broken (in "B-Octo2SH-to-5.1"). It should be C = 0.1768xW + 0.2500xY (but currently is C = 0.1768 x W + 0.1768 x X). The "0.2500"-box is there but not linked, so it probably is just broken.

2) It is true that "Spherical Harmonics" decodes result in a (slightly) loss of directional information, but it should be added that "Controlled Opposites" produce a larger listening area, and guarantee that speakers will never generate an `anti-phase' signal.

I did many tests some time back and personally prefer the "controlled opposites", but it's a question of taste (not right or wrong).

3) I analyzed your modified Octagon2 decode, and it clearly produces a slightly more narrow soundfield as compared to the Pentagon decode. This is also expected, as the different speakers receive different W's, i.e. you would expect some 'distortions' for the soundfield (in the case of the modified Octagon2 some kind of squeezing along the x-axis, especially in the FL and FR areas of the soundfield).

I would suggest you still give the Pentagon decode a try, as it produces a perfectly equilateral pentagon (72 deg between each speaker). You can use 2 options:

i) Using BFprocEdit (rotate the soundfield by 36 deg) and then Emigrator (decode to the Pentagon), as on the guide on the 1st page of this threat or

ii) Modify the decoding matrices and use a bidule scheme similar to the one you did:

For Spherical Harmonics:
speaker________________Weights
__________x_______y_________W_______X_______Y
C ___1.0000__0.0000____0.2828__0.4000__0.0000
FL___0.3090__0.9511____0.2828__0.1236__0.3804
SL__-0.8090__0.5878____0.2828_-0.3236__0.2351
SR__-0.8090_-0.5878____0.2828_-0.3236_-0.2351
FR___0.3090_-0.9511____0.2828__0.1236_-0.3804

For Controlled Opposites:
speaker________________Weights
__________x_______y_________W_______X_______Y
C____1.0000__0.0000____0.2828__0.2000__0.0000
FL___0.3090__0.9511____0.2828__0.0618__0.1902
SL__-0.8090__0.5878____0.2828_-0.1618__0.1176
FR___0.3090_-0.9511____0.2828__0.0618_-0.1902
SR__-0.8090_-0.5878____0.2828_-0.1618_-0.1176

In summary, I think you did a great job, and I'm sure it was great fun an a good deal of deep understanding of how Ambisonics works, and what it does and does not.

Kind regards,

Andreas

[kempfand]

Originally posted by Tantulus I guess the results of encoding into 5.1 ultimately depends on the orginal recording process, the speaker setup, the room acoustics and the bidule used.
Tantulus: Think you made a key statement here, so I am re-quoting it. A big deal depends in fact on the original recording and how it was mixed . As a matter of fact, I know some repurposing experts, who use LP's as source, just because they contain the 'right' mix for what they want to do. Now I'm not suggesting to use LP's as source, but I think it makes the point.

As to the AmpIn and AmpOut settings with the HNM filter, I am sorry I cannot really help, as I never use an LFE, and if it is part of a bidules, I just rely on the settings published. I simply think there is too much danger of "things going wrong", unless of course one knows exactly what one does.

To quote from recent posting from the QuadrophonicQuad forum, from a guy I highly respect for his knowledge:

The biggest problem with nearly all of the DTS-CD's is that they are home cooked, and essentially qualify as bootlegs.
If the person who did the original transfer was working on, say, PC speakers or HiFi speakers then they used an inaccurate monitoring environment, and as the studio saying goes "If you can't hear it you can't mix it".
Another big area for error is the extensive use of the LFE channel as a sub channel, when it is by definition a completely different thing.
The Sub channel exists to extend the bass response in systems that do not have 5 full range monitors/speakers, and NOT as a subwoofer.
When you start encoding music with an LFE channel, you run the following risks.

A/. The end user has not set things up correctly, and that pumpimg bass just isn't there.

B/. You can never know all the crossover frequencies that end user sub/sattelite systems use, so phase cancellation is a big problem. There is also the reverse of this, and what is piped to the LFE may well add to what is already in the bass end of the main speakers, giving the "bass heavy" problem you describe

C/. (The most common) the system for playback is simply not properly calibrated, the sub is too loud, the balances between all the other 5 channels are phase delayed & comb filtered and the result is an unpredictable bloody mess.

D/. There is a bass management setting on the AV amp, and it is conflicting with what is piped to the LFE Channel.

E/. Incorrect crossover frequency set in the DTS conversion. In an ideal world, you do not use the LFE but instead use an LFE splitter to emulate the effects of Bass Management. Set the X-Over to 80Hz and the slope to 24dB/Octave, or even steeper if possible. Never, ever encode with the LFE active unless you know that your monitoring environment is 100% accurate & properly calibrated. And only then with extreme caution.

In short, do not use the LFE channel unless you are doing film scores or the 1812 overture, and trust to the Bass Management in the AV amps to sort everything out. It is what it is there for.


Kind regards,

Andreas

[ursamtl]

@kempfand,

Thanks for the feedback. Just a couple of brief comments for now as I have some balcony gardening to take care of.

Originally posted by kempfand
1) In the current uploaded bidule (June 6), I believe that the decoding maatrix for C is broken (in "B-Octo2SH-to-5.1"). It should be C = 0.1768xW + 0.2500xY (but currently is C = 0.1768 x W + 0.1768 x X). The "0.2500"-box is there but not linked, so it probably is just broken.

Actually the numbers I chose are intentional. I did forget to delete the unused 0.2500 constant. You see the Octagon is designed for a soundfield where the center speaker is at the same distance from the listening point as the others. Since most surround systems situate the center speaker on the same plane as the fronts, I felt it would be more accurate to attenuate the center somewhat. After doing a scale drawing in Illustrator (which I will share once I resolve some problems linking to graphics), I determined that since the center speaker on the same plane is 0.7071 the distance (interesting how that number keeps coming up :), I attenuated the center by that amount. As for there being different values going to different speakers, of course there are. That's how the phantom speakers are created. For example, the front left speaker receives 0.1768 for it's own signal, plus 884 (half of 1768) for the phantom left side speaker. This totals 0.2652. For the rear left speaker, I took this amount representing the rear left plus half the rear side, plus an additional 884 for half of the rear center phantom speaker. In fact, all the virtual speakers in my implementation of Furse's Octagon 2 receive the same amount of sound pressure with the exception of the center, the reason for which I just explained.


2) It is true that "Spherical Harmonics" decodes result in a (slightly) loss of directional information, but it should be added that "Controlled Opposites" produce a larger listening area, and guarantee that speakers will never generate an `anti-phase' signal..

Yes I read that too, but surprisingly, to me controlled opposites don't sound bigger. They actually remind me of the sound of audio that is out of phase, in that they produce an overly diffuse soundfield with nothing in focus. I personally found the spherical harmonics sounded much more appealing and more like the kind of professional 5.1 sound I so like.

3) I analyzed your modified Octagon2 decode, and it clearly produces a slightly more narrow soundfield as compared to the Pentagon decode. This is also expected, as the different speakers receive different W's, i.e. you would expect some 'distortions' for the soundfield (in the case of the modified Octagon2 some kind of squeezing along the x-axis, especially in the FL and FR areas of the soundfield)..

Your statement really surprised me. In all the tests I've done on all the bidules over the past few weeks, the Pentagon decoded bidules sounded much more narrow to me. Until I actually tried the Octagon 2 layout (which I believe species 8472 came up with first in this thread. I simply added the phantom speakers). I dismissed Ambisonics and completely unacceptable for producing surround sound.

I would suggest you still give the Pentagon decode a try, as it produces a perfectly equilateral pentagon (72 deg between each speaker)..

I did try a Pentagon layout and it sounded really compressed and lifeless compared to Octagon. Even a 4-channel cube layout is much better. Once took the calculations on Furse's web site and plotted them in Illustrator, I realized tht it does not seem to produce an equilateral pentagon (assuming the point of rotation is the geometrical center at -0.0489, ambisonic X). Check the diagram I just posted in my original message. Worse, once rotated 36° to the left, the soundfield is completely distorted. I already edited my message this morning once I had time to measure the angles in my drawing (which I'd left at work).

Even if the rotation is centered on the 0,0 listening position, there is still a problem with playing back a file with harmonics calculated for a wide soundfield, but on sound systems with real speakers placed differently. The ideal position for the front left speaker in the rotated Pentagon is at 74° from center, a full 29-44° from the suggested placement in a typical 5.1 system. the Right speaker is at 71°, or 26-41° from the typical position. Therefore, if you play back a file decoded through the Pentagon, the width of your soundfield is being compressed between 45-85°!! No wonder I found the Pentagon bidules lacking in width! I'm really surprised that you actually ended up making such a conclusion about the Octagon 2 layout. Sit down and draw it out, you'll see. I prepared some graphics for my post last night but there were some problems with linking to my geocities site. I'm going to place them on another site and edit my post. I think if you look at them and consider the geometry of it all, there's no way you can conclude that the Octagon 2 soundfield is compressed , certainly nowhere nearly as much as the rotated Pentagon.

Anyway, time to tackle the garden. By the way, did you try the effect of widening the soundfield with the slider I implemented? It sounds amazing!

Regards,
Steve.

[kempfand]

@ursamtl/Steve:

Thanks for the explanations. Few comments from my side.
did you try the effect of widening the soundfield with the slider I implemented? It sounds amazing! I tried, and the effect IS truely amazing. As indicated before, I also like the way you implemented this finding using bidule, as it makes testing so easy.
I felt it would be more accurate to attenuate the center somewhat. After doing a scale drawing in Illustrator (which I will share once I resolve some problems linking to graphics), I determined that since the center speaker on the same plane is 0.7071 the distance (interesting how that number keeps coming up , I attenuated the center by that amount. That makes it clear to me now.
As for there being different values going to different speakers, of course there are. That's how the phantom speakers are created. For example, the front left speaker receives 0.1768 for it's own signal, plus 884 (half of 1768) for the phantom left side speaker. This totals 0.2652. For the rear left speaker, I took this amount representing the rear left plus half the rear side, plus an additional 884 for half of the rear center phantom speaker.Understood. But:
In fact, all the virtual speakers in my implementation of Furse's Octagon 2 receive the same amount of sound pressure with the exception of the center, the reason for which I just explained. I disagree :) : Pressure (W) going into C: 0.1768, each of FL & FR: 0.2652, each of SL & SR: 0.3536.

Now: This is just a factual statement (i.e. observation) I made, i.e. not saying 'right' or 'wrong'. In fact, I have seen unpublished decodes from well respecetd experts in this area who also apply different pressures to the decodings to the speakers.

3) I analyzed your modified Octagon2 decode, and it clearly produces a slightly more narrow soundfield as compared to the Pentagon decode. This is also expected, as the different speakers receive different W's, i.e. you would expect some 'distortions' for the soundfield (in the case of the modified Octagon2 some kind of squeezing along the x-axis, especially in the FL and FR areas of the soundfield).. Your statement really surprised me. In all the tests I've done on all the bidules over the past few weeks, the Pentagon decoded bidules sounded much more narrow to me. Until I actually tried the Octagon 2 layout (which I believe species 8472 came up with first in this thread. I simply added the phantom speakers). I dismissed Ambisonics and completely unacceptable for producing surround sound. You clearly see the mentioned deviation when you plot your layout using the CATT-FIReVerb Suite, whereas there are no deviations for the rV and rE deviations when using a Pentagon decode.

But same remark as above: Just an observation :) As a side-remark, it's interesting to see how people have have & are putting patent protection to 'their' decoding specs (IMhO one of the reasons why Ambisonics never really made it to the masses). My personal opinion is summarized well in Anegelo Farina's presentation from May 2004 (at the Berlin AES convention): "We are still learning what is the best way to render ... over a standard 5.0 (or 5.1) setup".

I did try a Pentagon layout and it sounded really compressed and lifeless compared to Octagon. Even a 4-channel cube layout is much better. Once took the calculations on Furse's web site and plotted them in Illustrator, I realized tht it does not seem to produce an equilateral pentagon (assuming the point of rotation is the geometrical center at -0.0489, ambisonic X). Check the diagram I just posted in my original message. Worse, once rotated 36° to the left, the soundfield is completely distorted. I already edited my message this morning once I had time to measure the angles in my drawing (which I'd left at work). This is actually the point I am most keen to understand better. Maybe I just have 'bricks' in front of my eyes, but plotting the Pentagon (rotated by 36 deg) gives a perfect equilateral Pentagon in my plots (see the x,y coordinates I gave above). In fact, I was and am using this to construct signed filters to decode B-format to 5 speakers (using 15 filters in total, 3 for each of the Pentagon-rig speakers, using 2 instances of Prinstine Space to do the 16 concurrent convolutions).

I think if you look at them and consider the geometry of it all, there's no way you can conclude that the Octagon 2 soundfield is compressed , certainly nowhere nearly as much as the rotated Pentagon. In addition to my remark above (i.e. Octagon2_MOD is compressed a bit on FL & FR, as per CATT-FIReVerb analysis), I think a 'complication' comes by the fact that there are 2 aspects:
a) decoding for a specific speaker configurarion. If this was the only criteria, we should all go for the Surround-decoce (= ITU-5.1), which is what we actually do not.
b) actual speaker-set-up in our listening room, which delivers the created output.
I other words: A decode for a specific speaker configuration (be it Pentagon or Octagon2_MOD or xyz) CAN sound give good results, as a function of listener's preference and actual speaker set-up (which most often is not the standard ITU-5.1).

Hope the gardeing was a succes. Time to get some sleep here.

Regards,

Andreas

[Tantulus]

Thanks Kemfand for your reply:

Part of my problem with the LFE channel is that I have too damn many controls on my AV receiver (Onkyo TX-DS797) not to mention my subwoofer. Since I use my system for DVD's and music I decided to take the bidules as is and do the adjustments on the receiver so that I can switch from one to the other.

B/. You can never know all the crossover frequencies that end user sub/sattelite systems use, so phase cancellation is a big problem. There is also the reverse of this, and what is piped to the LFE may well add to what is already in the bass end of the main speakers, giving the "bass heavy" problem you describe

I have a switch on the subwoofer that allows my to adjust the phase which seems to be set correctly for my ears.


I have a dial that allows my to adjust the low pass from 200htz to 60. The manual for the subwoofer recommends setting the low pass at 200hts because I'm using the decoder's SUB OUT jack. However, this takes on to much of the bass and results in the bass heavy problem. Although its a pain to keep changing the Lowpass it does allow me the flexibity of how much bass I want to capture from the other speakers depending on the format.

C/. (The most common) the system for playback is simply not properly calibrated, the sub is too loud, the balances between all the other 5 channels are phase delayed & comb filtered and the result is an unpredictable bloody mess.

I went to "Radio Shaft" and bought a SPL meter to set the levels of all the speakers to 75 decibel sound pressure. However, again by rereading the manual I found that I can vary the levels while monitoring the music to get the reponse I'm looking for.

Set the X-Over to 80Hz and the slope to 24dB/Octave, or even steeper if possible

I'll check out this setting on the subwoofer.

I feel I do need the LFE. It brings out the lower register instuments such as cellos, bass's and bassons. I'm curious why you do not use the LFE. Do you have speakers with good bass response or do you use the receiver?

Anyway, your respones has spurred my on to find the ideal settings. Thanks again.

Now to Ursmtl:

I noticed in your diagramns that you have the listener placed in a central location relative to the speakers. However the manuals suggest that the surround speakers should be placed in line with the listener about 3 feet above the ears. Are you implying there is a sweet spot for ambisonic 5.1? Also, I recently purchased surround speakers that can switch from dipole to monople to bipole. Would any of these settings be better for the surround effect or do I need back surround speakers?

Thanks everyone for your assistance!

Scott

[Tantulus]

Sorry I forgot one more thing. For those of you with Windows 2000 Pro or XP. I'm trying to write a script using Windows Script Host to automate the encodeing process. However, it's going to take some time. Is anybody trying to do this?

[Eye of Horus]

Hi all,

After 2 months of hard work, discussions all over, listening, rewriting, discussions again, adjustments, etc. etc.,
Kempfand and EoH proudly present our new bidules.
These bidules are completely different from everything published till now and in our opinion so are the results.
Our goal was to come as close as possible to the HardWare Repurpose guys and..... in our opinion on some tracks we even did better.
If you agree with us remains the question
We would love to hear your feedback !

NEEDED
--------

1. Besweet b28
2. MDA VST
3. Voxengo stereo VST (freeware !)
4. SSRC.exe
5. bidules and groups

DOWNLOAD
---------

The whole package can be downloaded here :
www.app.demon.nl/Kempfand-EoH-06-09-2004.rar

or on Needfull Things......

SETUP
------

1. Install Besweet, start.bat and SSRC.exe in the same folder
2. Install MDA and Voxengo Stereo into the VST folder of Plogue
3. Install the 3 bidules in the layout folder of Plogue
4. Start Plogue Bidule and rescan for plugins and groups
5. Make sure the filerecorder is saying : 16 bits !!!!

USAGE
------

There are 3 bidules which handle different kind of music.

1. Voice-Center.bidule
This bidule handles music with a singer in the center of the stereo. This is checked quite easily by putting your receiver into DPL mode and listen very carefully to the voice. If it's coming from the Center speaker only, then you should use this one.
After conversion, the voice comes out of the Center only.

2. Voice-Non Center_or_instrumental.bidule
This one is useful for songs where the voice in not (or not only) in the center and for instrumentals. After the conversion you will hear the voice from all speakers, but the accent is on the Center

3. Instrumental.bidule
Use this one or the previous for tracks without voice.

4. For a first run, don't change a thing in the bidules, but do make sure the filerecorder is using 16 bits. Of course your source need to be converted to 32 bits floating first for the best results !!

5. Save your output to the Besweet folder

6. Run start.bat, but first edit (right-click and chose "edit") it to suit your settings. Here is the batch routine :

BeSweet.exe -core( -input 01.wav -output f:\01- -6ch )

ssrc.exe --twopass --normalize f:\01-FL.wav f:\01-New-FL.wav
del f:\01-FL.wav

ssrc.exe --twopass --normalize f:\01-FR.wav f:\01-New-FR.wav
del f:\01-FR.wav

ssrc.exe --twopass --normalize f:\01-C.wav f:\01-New-C.wav
del f:\01-C.wav

ssrc.exe --twopass --normalize f:\01-SL.wav f:\01-New-SL.wav
del f:\01-SL.wav

ssrc.exe --twopass --normalize f:\01-SR.wav f:\01-New-SR.wav
del f:\01-SR.wav

copy f:\01-LFE.wav f:\01-New-LFE.wav
del f:\01-LFE.wav

As you see I use 01.wav as name for the 6channel WAV file in the Besweet folder (in my case on e and I do the processing to drive f:\ .
Change these values to the ones you wish to use. (Don't use spaces in the filename !) If you want the input WAV in the same directory, leave out the harddisc letter , but make sure it has another name after "-output" in the Besweet line !

In my case it will output 6 mono 16 bits 44.1 Khz WAV's to harddisc f: .

As usual you can use these to make a DTS cd......

EXTRA INFORMATION
--------------------

Voxengo stereo VST uses presets and we got a very good result with a preset of "pretty wide" . If you wish to experiment with these settings, it's very easy. Just doubleclick on the VST in Plogue and adjust to another preset or make your own. However : our results are all based on the "pretty wide" setting and we don't take responsibility for a worse conversion when using other settings

The same goes for the normalize routine. With these bidules we found that making all 5 mono files at the same peak level, gives the best result. It can be that you wish to have less sound in the rears. Just edit the batch and replace the setting for SSRC to your own preferences.

Regarding the messages about the LFE : all three routines don't have a separate LFE output ! The LFE file you see in the routines is empty !


Well, this is about it..... happy testing and please share your thoughts here !

kind regards,

Kempfand and Eye of Horus

[MaroonMike]

EOH,

Are you suggesting that this new method is an improvement over the Ambisonics method? Can you explain how this method is different (from a sound perspective?)

Thanks for your work on this. Mike

[AllTimeSToneD]

once i start the "start.bat" i get following error:

Error 58: Error : Unknown Input-File Format : "01.wav".
Quiting...

i named my wave file also 01.wav i also tryed to fix the wave file using besliced but still getting the same error :(

[kempfand]

The 3 new bidules make usage of Gerzon technology (Gerzon 1997 Groups & Bidules (http://forum.doom9.org/showthread.php?s=&postid=508560#post508560)), but are much more elaborated than the basic groups.

We are actually keen on your feedback & opinion as to how they do, but I wouldn't see them as or call them "competing" to Ambisonics, as they try to achieve a different result.

Cheers,
Andreas

[kempfand]

http://forum.doom9.org/showthread.php?s=&postid=499028#post499028

Course assuming you are using adjusted paths to the 01.wav too (f:\)

[AllTimeSToneD]

Hehe its weird but yes i use "f:\" for all my wave processings too :D
Thanks for the link! Haven't seen that option at all :)

[ursamtl]

EoH & kempfand,

Just some initial comments since I don't have time tonight to do more thorough testing.

It's obvious you put some thought and experimentation into your new bidules and I congratulate you for this. My initial impression is that the Voice-Non Center or instrumental bidule has quite a bit of potential. This first version produces a nice large soundfield, but it's a bit too "reverby" for my taste. What I mean by this is that it seems to "color" the sound as if it were running through a multichannel reverb unit with the wet level a bit too high. Perhaps it's because of the delays in Stereo Touch. For example, I just took the song "Avalon" by Roxy Music and ran it through the bidule and then did a DTS CD. The percussion in this song really seemed doubled, at points where the highs were flammed by the doubling and the bass suffered from lack of focus. For comparison, I ran the same song through the UpMix Studio bidule with the wdith control on full and put it as the second track on the DTS CD. I still got the big soundfield, but without the doubling on the percussion.

For another test, I took Pink Floyd's High Hopes. Processed through the Voice-Non Center or instrumental bidule, the bells at the beginning sounded like they were in an enclosed space, say a huge arena. The birds chirping and the fly buzzing through seemed overly loud and reverberated. The UpMix Studio bidule version sounded as if the bells were outside and the fly and birds sounded as if they they were right there.

Then I tried some classical guitar music, and I found that the delay and reverb seemed to overwhelm the details in the guitar sound.

Having said all that, I think with some adjustment, this bidule has some potential. As for the other two, I didn't test them very much because after listening to them, I found that their sound is basically not much different from a good, old-fashioned stereo difference signal (L-R) with a bit of bass thrown in, plus a single mono center on the voice-center. I've been playing around with difference signals since the mid-70s when I found a circuit in an electronics magazine for hooking a speaker up across the two positive terminals of an amp. This can produce a very ambient sound from properly recorded stereo material, but it can also sound really weird if instruments are hard panned. An out of phase version is basically the surround channel in Dolby Surround encoding.

Anyway, I'm sorry if this sounds overly negative, but to my ears, the Voice-Non Center or instrumental bidule is the only one with potential. The other two actually sound far worse than my first MatrixMixerEmulator bidule. Could it be that some connections are missing from the two that sound like just a difference channel?

Please don't take this the wrong way. I appreciate that you both put a lot of effort into this, but you did ask for feedback. Perhaps you could explain some of the inner design of the bidule and we can all work together to improve them to the point where they meet our needs better.

Regards,
Ursa.

[ursamtl]

Originally posted by Tantulus
Now to Ursmtl:

I noticed in your diagramns that you have the listener placed in a central location relative to the speakers. However the manuals suggest that the surround speakers should be placed in line with the listener about 3 feet above the ears. Are you implying there is a sweet spot for ambisonic 5.1? Also, I recently purchased surround speakers that can switch from dipole to monople to bipole. Would any of these settings be better for the surround effect or do I need back surround speakers?

Thanks everyone for your assistance!

Scott

Hi Scott,

Actually, if you do a search around the net for "ITU-5.1", you'll find some diagrams showing the surround speakers at rear angles of about 45° from the center with the fronts at about 30° from center. I've seen some manuals suggest putting the surround speakers point towards each other, others suggesting pointing them towards the listener, and still others suggesting that one point them directly towards the front. The problem with some of this is that it comes from the days of the original Dolby Surround, when both surround speakers were fed with the same signal, an out-of-phase difference signal (front L-front R) with the highs rolled off at 7kHz. The intention was not to create any accurate directionality in the rears, but simply to give the impression of being surrounded by ambience. By shifting the signal out of phase, the ambience is not really directional. The actual ITU-5.1 diagrams I've seen point the speakers towards the listener. As for the fronts, the ideal listening spot is an equilateral triangle with the listener at one point and the L and R at the other points. Of course the center is right in the middle.

As for there being an Ambisonic 5.1 sweet spot, well, yes by definition, the listening point is in the center. I'm starting to come to the conclusion that what we're doing is not "real" ambisonics. If you check Angelo Farina's web site, where I got the formulas for converting stereo to B-format, he states that they are for converting "UHJ" to WXYZ. He states that "UHJ is a 'standard' stereo waveform, which includes information capable of driving a complete horizontal surround system (for example, equipped with 5, 6 or even 8 loudspeakers)." I've written him for confirmation on this, but what he's basically saying is that UHJ is a "stereo" waveform in that it is two channels, but it is unlike regular stereo waveforms in that it also contains encoded extra surround information that, when decoded, allows one to recontruct a complete horizontal surround soundfield. I did some more searching around the net for UHJ info, and the info I found seems to confirm this.

Therefore, taking a regular commercial stereo recording and running it through any of the Ambisonic bidules and/or VSTs we've been discussing is not going to produce true ambisonics. Obviously it does do something. The observation I made last weekend that increasing the Y channel with respect to the W and X increases the perceived width of the soundfield, seems to confirm that the encoding formulas create combinations of sum and difference signals plus phase information in each signal. If one listens to the Y channel isolated from the others, it sounds a lot like a standard difference signal (L-R) but with better frequency response. This was confirmed in the UpMix Studio bidule when I increased the width, the bass remained strong and focused. Doing that with a normal stereo wide effect reduces bass response dramatically. Therefore, encoding a regular stereo signal and processing it as real Ambisonic material certainly seems to have a lot of potential. Of course if part or all of the mix is recorded properly, it [i]is[/] really Ambisonic, but most off-the shelf recordings nowadays feature a whole hodge-podge of recording techniques.

Anyway, I have to get some sleep. I didn't get a chance yet to address the whole issue of LFE use and Kevin Shirley's (mixed the Led Zeppelin DVD) response to my email concerning LFE use. Briefly, he does use it, but makes sure the bass on a system without LFE sounds good and strong enough for those who do not have subwoofers. One thing I found interesting in his response and also in his musings as revealed in the diary on his web site (http://www.cavemanproductions.com) is that he does completely separate mixes for stereo and 5.1 on a DVD. Some engineers would probably just do the stereo and then recycle it as the fronts for a 5.1 mix, but Kevin approaches the two formats as separate entities and mixes them as such.

My feeling as well is that a properly adjusted system with a bass management system renders the whole point moot. When I first set up my system, I had to tweak the sub settings and phase switching to get the sound right, but once I did, the system sounds great. I recommend a test DVD or one of those THX test tracks that are found on some commercial DVDs one rents. The whole notion of avoiding the LFE channel seems ridiculous and a waste of a potential resource.

Ok, the last of a huge multipart download is complete and my better half is going to leave me if I don't come to bed, so goodnight!

If you have any further questions, don't hesitate to ask!

Regards,
Ursa

[kempfand]

ursamtl/Ursa,

Speaking for myself, I do not take your comments as overly negative, and I value the initial feedback which we were looking for. I and a few others like the 3 new bidules very much, and I think they beat the existing bidule methods, but they still can be improved (hopefully), and yes, you are right: maybe a joint effort gets us there. Maybe the bidules do not beat the outcome of a professional audio engineer who sits for 2 weeks to do an upmix, but this wouldn't be a fair comparison, as the bidules do a number of CD's in just an hour.

I would suggest (to all readers also) that you try the new bidules on a large number of various pieces of music. You might be disappointed if you just shoot for the most difficult pieces of music (did that myself in the first place). You might also be disappointed if you shoot for music you know well & love very much, as you might have too harse expectations to the outcome. You might be very happy if you compare what they do to standard DPL2 upmixes.

The overall idea of the new bidules ("inner design" as you say) is simply as follows: Reproduce using Software (SW) what some of the Hardware (HW) Repurposing wizzards did & do.
As crazy as this might sound, I believe that SW Repurposing these days can do it, maybe even better. The same applies for Ambisonics, where the increased precision of internal processing of the SW tools just beats the established old HW units.

The idea of HW repurposing "per se" is IMHO not really a huge secret: "All" it does (i.e. what the HW people do) is to to apply (as an example) a little bit of SRS II and use that for the C, apply DPL 2 and use it for L & R, and finally apply a Sansui decode for the Surrounds. Some of the best HW Repurpose works I know of are done that way or similar (maybe even applying small variations to different parts of the album, or even to different parts within the same track).

As far as I can see, the real difficulty for Repurposing is to select the 'right' music, and often the right mix of the same album. As mentioned earlier, some even take "old" LP's, just because they contain the right mix. After all, isn't it surprising that there are not too many works around which were HW repurposed ? I think it is because of the reasons I outlined (not all music suitable; often various mixes around, some working well, some not).

"Reverby" thing: This is something I noted too. I peronally like it, but you can play with the settings of Voxengo's Stereo VST to change this. Plus there is quite a number of other mono_2_stereo schemes or VST's (i.e. using a simple 'pseudo stereo' scheme: S->L', S->Inverse->R', or specise_8472's more elaborated mono_to_stereo groups mentioned and published earlier in this threat)

"Color": Again try changing the settings of Voxengo's Stereo VST to suit your taste. Speaking about "coloration" (not sure if you mean that one), this also can happen with Ambisonic decodes (esp. on the ITU-5.1).

Finally, a brief comment to your reply to Tantulus/Steve: The methods described here do proper Ambisonics, but the feeding material is not true UHJ. Most importantly, it lacks the 3D information (Z-axis), but who cares, as most of us don't have a 3D speaker setup.

The key thing is that feeding Stereo through an Ambisonics upmix still can produce surprisingly results on "Stereo". In fact, it often does ! For very dry recordings, Ambiophonics is probably a better way to do things (i.e. a Stereo Dipole plus Ambisonics as 'supportive' means 'only'). See also: Farina's comments regarding the concern about the usage of UHJ-to-Surround conversion for not-UHJ stereo recordings (http://forum.doom9.org/showthread.php?s=&postid=419102#post419102)

Regards,

Andreas

[brock101]

Hi EOH And kempfand

I have been following your Adventures with regards to stereo manipulation :D
Sadly every guide which i have tried, i have had the same problem occur many many times, when importing the mono wav files into surcode, some wavs cause surcode to warn me that the wavs are shorter than the others :angry:

Has anyone else had the same problems
as i have seen no one else mention this problem

I had given up on converting some of my cd collection into dts... due to the frustration caused... but i decided to give it one last shot due to the new way ... sadly no luck :(

may i add out of the 20+ attempts i had 1 success
i did war of the worlds sadly disk 1 worked but not disk 2 -_-

Regards
Brock101

[ursamtl]

Originally posted by brock101
Sadly every guide which i have tried, i have had the same problem occur many many times, when importing the mono wav files into surcode, some wavs cause surcode to warn me that the wavs are shorter than the others :angry:

Has anyone else had the same problems
as i have seen no one else mention this problem
[/B]

Hi Brock,

I haven't had this problem per se, but I have noticed that you have to clear Surcode from the first set of files you were working with before starting a new project. It retains the files loaded into the previous session when you first open it. If you then try to load individual files without first starting a new project (Ctrl+N) or clicking the C beside the channels you want to empty, you'll run into problems. Also, are you following the advice from the original guide and exporting to one 6-channel wave file? I recall reading some talk of just connecting six individual 1-channel file recorders in a bidule as a way of skipping the splitting with Besweet step. I haven't tried this but if I remember correctly, it causes problems as well.

Don't give up! all this experimentation can be fun.

Regards,
Steve.

[kempfand]

Hi brock101,

Welcome to the forum ! :)

I am having this "error" from time to time as well, and so do others I am aware of. I am also assuming you use the 6-channel Audio File Recorder (i.e. not 3 x 2-channel recorders).

As we found out, this "error" (i.e. warning upon import into SurCode about the mono-wav's having different lengths) does not matter, i.e. has no negative impact on the DTS-packaging. I have tested this on a very long file (nearly one hour), and didn't hear anything out of sync even near the end of the playing time.

It is really difficult to say where this happens^. Could be with Bidule (which I doubt, because Plogue looked into this I think), could be BeSweet, could even be SurCode, could be slow drives or too many other processes running during the demux.

In any case, I would encourage you to try some and see if you really hear something 'out of sync' at the very end.

Good luck,

Andreas

[Eye of Horus]

Originally posted by brock101
Hi EOH And kempfand

I have been following your Adventures with regards to stereo manipulation :D
Sadly every guide which i have tried, i have had the same problem occur many many times, when importing the mono wav files into surcode, some wavs cause surcode to warn me that the wavs are shorter than the others :angry:

Has anyone else had the same problems
as i have seen no one else mention this problem

I had given up on converting some of my cd collection into dts... due to the frustration caused... but i decided to give it one last shot due to the new way ... sadly no luck :(

may i add out of the 20+ attempts i had 1 success
i did war of the worlds sadly disk 1 worked but not disk 2 -_-

Regards
Brock101

Hi Brock,

I have had these problems too, but as kempfand said : nothing to worry about. Did you try the different I/O and DSP settings for the buffers in Bidule ? You could try increasing these to see if that solves the problem.....

Anyway, glad you enjoy working with our stuff, but it would be better to get the results you want !
I have done WotW myself several times. Actually it was the very first one I ever did, with the Cooledit/Aurora method. I also used it on several other bidules to compare. And..... the voice-center bidule gives an excellent result !
BTW I never had problems with the second disc of WotW !

I hope you will have better results in future.

Another tip : try to run on a clean machine (no other programs or tools or virussoftware running....

kind regards,

EoH

[ursamtl]

Originally posted by kempfand
ursamtl/Ursa,

Speaking for myself, I do not take your comments as overly negative, and I value the initial feedback which we were looking for. I and a few others like the 3 new bidules very much, and I think they beat the existing bidule methods, but they still can be improved (hopefully), and yes, you are right: maybe a joint effort gets us there. Maybe the bidules do not beat the outcome of a professional audio engineer who sits for 2 weeks to do an upmix, but this wouldn't be a fair comparison, as the bidules do a number of CD's in just an hour.

I would suggest (to all readers also) that you try the new bidules on a large number of various pieces of music. You might be disappointed if you just shoot for the most difficult pieces of music (did that myself in the first place). You might also be disappointed if you shoot for music you know well & love very much, as you might have too harse expectations to the outcome. You might be very happy if you compare what they do to standard DPL2 upmixes.


I think your comments might underline where we differ on this whole thing. If I'm going to invest the time to put into this experimentation, I don't want to do some rush job and neither do I want to spend my time on music that isn't my favorite or that isn't too complicated. As I said in an earlier post, I can do tests on CDRW so I'm not ending up with a bunch of written CDs wasted because I'll never listen to them again. It's just that I believe that quality results are possible, otherwise, I could use my time much more wisely by slapping on the Pro Logic II, cranking up some good music and getting on with the other tasks that fill my life outside of work.

I understand that's it's great to be able to automate this process as much as possible, and maybe that's what most people are looking for, but since each piece of music is unique and presents different demands, I think it's worth taking the time to tailor the upmix for each one. Hence, the ability to adjust and monitor the bidule's effect is important to me. I was happy to discover that my last bidule processed the music in such a way that the adjustments I provided are still easily heard if someone doesn't have a surround setup on his or her PC. It's so rewarding to be able to do this kind of adjustment to make the music the best it can be. I know it's not everyone's goal, but I don't think I should compromise my goal just because a bidule might not do justice to my favorite music or to a particular piece of music that's a bit more complex. Let's strive for excellence here and if whatever level we do attain will be higher than if we simply aimed for something that's a bit more convenient. :)


The overall idea of the new bidules ("inner design" as you say) is simply as follows: Reproduce using Software (SW) what some of the Hardware (HW) Repurposing wizzards did & do.
As crazy as this might sound, I believe that SW Repurposing these days can do it, maybe even better. The same applies for Ambisonics, where the increased precision of internal processing of the SW tools just beats the established old HW units.

The idea of HW repurposing "per se" is IMHO not really a huge secret: "All" it does (i.e. what the HW people do) is to to apply (as an example) a little bit of SRS II and use that for the C, apply DPL 2 and use it for L & R, and finally apply a Sansui decode for the Surrounds. Some of the best HW Repurpose works I know of are done that way or similar (maybe even applying small variations to different parts of the album, or even to different parts within the same track).

This is interesting info. Where did you find this?

Finally, a brief comment to your reply to Tantulus/Steve: The methods described here do proper Ambisonics, but the feeding material is not true UHJ. Most importantly, it lacks the 3D information (Z-axis), but who cares, as most of us don't have a 3D speaker setup.

The key thing is that feeding Stereo through an Ambisonics upmix still can produce surprisingly results on "Stereo". In fact, it often does ! For very dry recordings, Ambiophonics is probably a better way to do things (i.e. a Stereo Dipole plus Ambisonics as 'supportive' means 'only'). See also: Farina's comments regarding the concern about the usage of UHJ-to-Surround conversion for not-UHJ stereo recordings (http://forum.doom9.org/showthread.php?s=&postid=419102#post419102)

Actually, I wrote Angelo Farina yesterday on this subject and he was kind enough to respond today. I didn't think to ask him for permission to quote his answer, but basically he confirmed my suspicion that we've been doing is not capable of producing the proper Ambisonic result. I don't think he'd mind my quoting the following from his email to me:
Written by Angelo Farina to UrsaMtl on June 10, 2004
In a hiearchy scale, I would rate the surround recovered from a not UHJ recording, converted to B.format and finally decoded to 5.1 as 1 over 5. The same score for the original Dolby Surround decoder.
He did suggest that taking B-format impulse responses and convoluting the original stereo track with them, combining the results to one file and treating it as a B-format file by decoding it to derive the 5.1 feeds is quite a bit better. He rated it 4/5 as apposed to a true 5.1 recording, which he rated as 5/5.

Anyway, you are right in saying that processing a non-UHJ stereo file as Ambisonics does in fact produce an interesting result. I've got a couple of ideas I want to test out on a version 2 of UpMix Studio and then I'll post it.

Have a good weekend!
Steve (Ursa)

[Shayne]

Originally posted by Eye of Horus
Hi Brock,

I have had these problems too, ....

kind regards,

EoH
I have never seen this problem in 1934 encodings and therefore would wonder if it has something to do with 44.1 hz (i always use 48).

Peace

[alastor]

HI kempf and eoh.

I tested your new bidules, voice center, and my conclusion....

! Great Works, sounds wonderful ¡

Only a comments :

In my computer the modication for besweet d'ont work, may be because my processor ia AMD, i d'ont Know.

Usualy i use the six mono recorders without a problem, i use the old bidule for more of 25 films without incidents.

Best Regards

[ursamtl]

Originally posted by alastor
In my computer the modication for besweet d'ont work, may be because my processor ia AMD, i d'ont Know.


Hi Alastor,

Your computer being an AMD shouldn't make any difference. I've been using a 3-year-old Athlon Thunderbird 1.1 GHz for all this surround processing without any problems at all.

Regards,
Steve

[alastor]

Hi all

if any is interested :

I use these procedure for six mono recorders, six beacuse i added two filters hnm for lfe, power off in bidule, open input. indicates names of all outputs, play in input. Power On.

In my case all the files ever all the same lenght. Remember i only works with film soundtracks, i d'ont know if this procedure is also valid for small files.

Bye

[trooper11]

ok i have a question, i was trying out the SAD5.1 Bidule using the guide that was provided here:

http://www.dtsac3forum.digitalzones.com/SAD51Bidule.htm

i thought i did everything correctly, i finished using bidule and i saw the out put of the one large wav file. i went to try and use besweet and i created the start.bat file, but everytime i try to run it, the screen just pops up quicklt then closes. i belive its saying error, unknown input file format. here is the line i pasted into the bat file:

BeSweet.exe -core( -input 01.wav -output 01- -type wav -6chfloat ) -ssrc( --rate 48000)

what am i doing wrong here? after i get the mono files, im going to be using them to convert to ac3 dd files, if i can just get this step to work. thanks.

[ursamtl]

Originally posted by trooper11
ok i have a question, i was trying out the SAD5.1 Bidule using the guide that was provided here:

http://www.dtsac3forum.digitalzones.com/SAD51Bidule.htm

i thought i did everything correctly, i finished using bidule and i saw the out put of the one large wav file. i went to try and use besweet and i created the start.bat file, but everytime i try to run it, the screen just pops up quicklt then closes. i belive its saying error, unknown input file format. here is the line i pasted into the bat file:

BeSweet.exe -core( -input 01.wav -output 01- -type wav -6chfloat ) -ssrc( --rate 48000)

what am i doing wrong here? after i get the mono files, im going to be using them to convert to ac3 dd files, if i can just get this step to work. thanks.

I ran into the same problem when I tried it. I got very wrapped up in trying different Ambisonic stuff so put the SAD51 on the back burner. I think it might have to do with long filenames in Windows and having to enclose them in quotation marks. As I said I didn't have time to test this theory, but I seem to recall reading something to that effect. I just found another source for ambisonic formulas so I want to try that out a bit more first, but I do intend to get back ot SAD51 for comparison purposes.

Regards,
Steve

[Eye of Horus]

Originally posted by trooper11
ok i have a question, i was trying out the SAD5.1 Bidule using the guide that was provided here:

http://www.dtsac3forum.digitalzones.com/SAD51Bidule.htm

i thought i did everything correctly, i finished using bidule and i saw the out put of the one large wav file. i went to try and use besweet and i created the start.bat file, but everytime i try to run it, the screen just pops up quicklt then closes. i belive its saying error, unknown input file format. here is the line i pasted into the bat file:

BeSweet.exe -core( -input 01.wav -output 01- -type wav -6chfloat ) -ssrc( --rate 48000)

what am i doing wrong here? after i get the mono files, im going to be using them to convert to ac3 dd files, if i can just get this step to work. thanks.


Everything in the line is correct, but....... Do you have the latest Plogue Bidule version ? (0.6601)
If so : goto the edit menu
Chose preferences
Goto the disk I/O tab
Make sure use WAVEFORMATEXTENSIBLE is set to "never"

I bet you have set it on "always" (default setting) or "adaptive".

Good luck and please let me know if this was the solution.

BTW when you like the output of SAD51inBidule, I recommend you to try our latest 3 bidules too !

Kind regards,

EoH

[trooper11]

hmm ok, well i ownder why long filenames would be giving me this proble, my file name is 01.wav. could it be the location of besweet? it is ina folder that has spaces, which i know has caused problems in the past with other things, but this time the error was for the file type. i wish someone would take a look at this.

[ursamtl]

Originally posted by trooper11
hmm ok, well i ownder why long filenames would be giving me this proble, my file name is 01.wav. could it be the location of besweet? it is ina folder that has spaces, which i know has caused problems in the past with other things, but this time the error was for the file type. i wish someone would take a look at this.

Yes, I think the filename problem has to do with the Besweet location, but EoH is probably on track with the setting in Bidule. I know that has caused quite a few problems from what I've read.

Steve.

[kempfand]

Ladies & Gentlemen,

I think the discussion over the previous few posts is going into the wrong direction, as these technical problems have null and nothing to do with the discussion about the Audio Methods discussed in this threat (which reads "Stereo to 5-Channel Surround").

If you have problems with BeSweet & GUI, SSRC, recording & splitting multichannel WAV's, short/long filenames etc. etc. etc.:

1) Use the technical threat(s) on this as agreed earlier: 6ch Wave to dts with BeSweet (thread split) (http://forum.doom9.org/showthread.php?s=&threadid=60907)
2) Always post your logfile :logfile:
3) Use the Search function. 99% of the problems have been described before, and a solution was given.



@ ursamtl:

This is interesting info. Where did you find this? From discussions which took place on the Usenet Newsgroup in middle 2002.



Actually, I wrote Angelo Farina yesterday on this subject and he was kind enough to respond today. I didn't think to ask him for permission to quote his answer, but basically he confirmed my suspicion that we've been doing is not capable of producing the proper Ambisonic result. Wrong interpretation from your side on Angelo Farina's email (based on what you quoted).
The methods described here do propoper Ambisonics.
If we feed the methods with 'Stereo only', that's a point which has nothing to do with the method per se.



Written by Angelo Farina to UrsaMtl on June 10, 2004
In a hiearchy scale, I would rate the surround recovered from a not UHJ recording, converted to B.format and finally decoded to 5.1 as 1 over 5. The same score for the original Dolby Surround decoder.He did suggest that taking B-format impulse responses and convoluting the original stereo track with them, combining the results to one file and treating it as a B-format file by decoding it to derive the 5.1 feeds is quite a bit better. He rated it 4/5 as apposed to a true 5.1 recording, which he rated as 5/5.

:devil: :devil: :devil:

That's exactly what many people here do as of mid 2002, with excellent & proven results: GUIDE: Stereo to 5-Channel Surround (as of 07/2002) (http://forum.doom9.org/showthread.php?s=&threadid=29277).
Note that this exactly does what Farina describes on his public site.

The new guides on Ambisonics GUIDE: Stereo to 5-Channel Surround (as of 08/2003) (http://forum.doom9.org/showthread.php?s=&threadid=60137) just make usage of newer tools (VST's, Bidule or Audiomulch) and expand the topic into Ambiophonics.

Cheers,
Andreas

[trooper11]

ok id love to post a logfile, if it gave one! using besweet, no log file was outputted, how do you get a log file to output?

also, i searched for long file name , to see if there was anyhting in the forum, and no there were no solutions that i found for my problem. i also tried input file type unknown, also no results to help. so if oyu could point me to the proper thread to ask questions, ill move there.

[ursamtl]

Originally posted by kempfand
Ladies & Gentlemen,

I think the discussion over the previous few posts is going into the wrong direction, as these technical problems have null and nothing to do with the discussion about the Audio Methods discussed in this threat (which reads "Stereo to 5-Channel Surround").

Andreas, if people have problems with instructions you give in this thread, then it's ridiculous to tell them they should not post their problems here! After all, the people reading this thread who may have a helpful response for someone with a problem may not read some other thread, and thus the person with the problem misses a chance to get help. That's rather unfair, don't you think? You can't control a discussion. You can perhaps remind people to stay on topic, but if you bring information into a discussion, you cannot control how others will react to that information.

Wrong interpretation from your side on Angelo Farina's email (based on what you quoted). The methods described here do propoper Ambisonics. If we feed the methods with 'Stereo only', that's a point which has nothing to do with the method per se.

Please don't try to tell me how to think or how to interpret someone's response to an email I sent him. I asked him if I was correct in assuming that feeding a non-UHJ signal through the Ambisonic encoding and decoding process is not true Ambisonics. I believe his answer as I quoted it was quite clear. He said it was no better than the original Dolby Surround!

Look, if Ambisonics by definition is the science of manipulating sound information in three steps: 1. recording sounds a certain way, or convolving sounds to give them missing information, 2. encoding the recorded information a certain way, and finally 3. decoding the information to present it a certain way. If one of these steps is missing, it is not truly Ambisonics. What Mr. Farina suggested was to convolve the stereo signals with true B-format impulses. This would in fact create a true B-format file by sort of grafting the soundfield information from the IRs onto the original stereo file. The resulting file would be a substitute for the recorded file, and thus fulfill the requirements for the first step of the process, thereby making it truly Ambisonic. Seems pretty clear to me!

Originally posted by ursamtl
He did suggest that taking B-format impulse responses and convoluting the original stereo track with them, combining the results to one file and treating it as a B-format file by decoding it to derive the 5.1 feeds is quite a bit better. He rated it 4/5 as apposed to a true 5.1 recording, which he rated as 5/5.

Originally posted by kempfand
:devil: :devil: :devil:

[QUOTE]That's exactly what many people here do as of mid 2002, with excellent & proven results: GUIDE: Stereo to 5-Channel Surround (as of 07/2002) (http://forum.doom9.org/showthread.php?s=&threadid=29277).
Note that this [B]exactly does what Farina describes on his public site.

The new guides on Ambisonics GUIDE: Stereo to 5-Channel Surround (as of 08/2003) (http://forum.doom9.org/showthread.php?s=&threadid=60137) just make usage of newer tools (VST's, Bidule or Audiomulch) and expand the topic into Ambiophonics.

Hmmm, who are you calling a devil, me or Angelo Farina? He was the one who gave me the ratings! Now as for the rest of your quote, yes, the original guide uses convolution, but if you read the beginning of this thread, which appears at the top of the Audio Encoding forum, you present it as a newer way of doing Ambisonics through VSTs (in fact finding the old guide takes a bit of reading for someone who comes across this forum for the first time). The Ambiophonics and other methods do not come up until later on.

In fact now that I think about it, you yourself said in response to me in this thread in a message dated May 27, 2004:

3) I agree with your point about "coloration" with Ambisonics, but this can be avoided by doing it right. Note that the most of the Ambisonics methods here (such as the guide at the beginning of this threat) are suited for the masses. Real Ambisonics uses signed filters (real-world impulse repsonses) to creat the B-format, as well as creating the decode to xyz speakers. See Conversion between UHJ and B-format for an outline on howto use signed filters for the B-format.

Sorry to throw your words back at you, but you did say this.

Now why don't you stop telling me what I understand from an answer I received from one of the foremost experts in the field, and stop telling people that they shouldn't complain here about problems they're having with information presented here. Perhaps I'm mistaken, but it almost seems as if you're frustrated because you want people to talk about your new bidules, but they keep talking about other things such as problems with information presented earlier in the thread, or debating the pros and cons of LFE.

I'm sure others are like me and haven't had the time to thoroughly test the new ones. I gave you my preliminary impression the other night, but I'm going to do some more testing on the weekend and I'll have more to say about the new ones as well. Since the title of the thread is GUIDE: Stereo to 5-Channel Surround, any discussion of any method to achieve this result should be fair game, as well as problems encountered by those following the instructions given here. If you don't want this, then perhaps you should ask the Doom9 management to make this a regular thread because it appears as if it's an overall thread for discussing all methods for converting 2-channel to surround.

Have a nice weekend!
Steve

[kempfand]

Originally posted by trooper11
ok id love to post a logfile, if it gave one! using besweet, no log file was outputted, how do you get a log file to output?
trooper11: In Bidule, did you set 'never' for the WAVE Format Extensible (with Preference), and use 16 or 32 bit for the Audio File Writer ?

Andreas

[trooper11]

ok well i have a question for a problem im having using besweet. i have a wav file i made using bidule. its a 32bit wav file and im wanting to split it into 6 mono files. here is the command line i tried using and placing it in a text file called start.bat:


BeSweet.exe -core( -input 01.wav -output 01- -type wav -6chfloat -logfilea "G:\program_setup\Sound_tools\Stereo_to_Surround\BeSweet_1.5b28\BeSweet.log" -ssrc( --rate 48000)

but i get the error 58 saying unknown input file format, pointing to "01.wav". here is the log:


BeSweet v1.5b28 by DSPguru.
--------------------------
Error 58: Error : Unknown Input-File Format : "01.wav".
Quiting...
[00:00:00:000] Conversion Completed !

Logging ends : 06/11/04 , 16:42:49.



would this mean the file i produced in bidule is bad or do i need to change soemthing in the command line?

[trooper11]

ok see there you go, that helped alot, no i had the wave extensible format on adpative, since there was no mention of it in the guide for the SAD5.1 Bidule. ill try it on never and see if that helps. i did use 32bit floating on output though. it may be soemthing that should be added to the SAD guide

also, in the guide it shows using 44100 as the sample rate, but the wav file im inputting is 48000, so shouldnt i set the sample rate at 48000?

[kempfand]

Steve: I think your comments are not appropriate. Asking that people use a search function or scroll a few pages up to see if a problem was reported before is a senesful request.

You might call this unfair, but that's how things are in life. Again read the title of this threat: "GUIDE: Stereo to 5-Channel Surround". Information I bring is mostly is on that topic. I repsect your helping hand to people's technical problems, but again: This should not the threat for it.


Please don't try to tell me how to think or how to interpret someone's response to an email I sent him. If you mention it here, partially quote it, and make your conclusions which you post here, I can comment on it, and I will do if I think it helps to clarify. If you don't like it, don't post it in a forum.


What Mr. Farina suggested was to convolve the stereo signals with true B-format impulses. This would in fact create a true B-format file by sort of grafting the soundfield information from the IRs onto the original stereo file. The resulting file would be a substitute for the recorded file, and thus fulfill the requirements for the first step of the process, thereby making it truly Ambisonic. Seems pretty clear to me! Seems pretty clear to me too. Again: That's exactly what Angelo Farina describes on his public server, and what is described in some of the guides here. So I don't see us (you and me) having a difference here.



Hmmm, who are you calling a devil, me or Angelo Farina? My native language is not English, but I think you turn around words here in a misleading way. Please stop this.

It should be clear that the 'devil' is not calling you or anyone devil (I would write that out if that was my intention, which it isn't). The :devil: is meant to express my non-understanding when you praise a method as "rating 4/5", and people here actually use this method for 2 years now.


Now why don't you stop telling me what I understand from an answer I received from one of the foremost experts in the field, and stop telling people that they shouldn't complain here about problems they're having with information presented here. See my comment above. As this is not your personal web-page but a public forum, I feel free to comment, and I believe I have always done in an appropriate and helpful way. Asking that people take reasonable effort to use the Search function or read if problems were solved before doesn't seem wrong to me (maybe a Moderator can comment ?).


Perhaps I'm mistaken, but it almost seems as if you're frustrated because you want people to talk about your new bidules, but they keep talking about other things such as problems with information presented earlier in the thread, or debating the pros and cons of LFE. Could also be the other way round. You didn't seem to like the comments that many people are happy with a Pentagon decode, and that your modified Octagon2 decode has some compressions in the soundfield.


I gave you my preliminary impression the other night, but I'm going to do some more testing on the weekend and I'll have more to say about the new ones as well. As I said, I am grateful for any feedback, and I really credit you gave quick and honest feedback. It is well understood that it is a fair game, and the only implication with the 3 new bidules was that we felt they do better than the previous on (basically SAD51). I always made it clear (I hope) that I don't see it in competition with the Ambio methods, as the intention is somewhat different (i.e. soundfield overall for Ambi, illusion of discrete speakers for the repurposing methods).

Regards,
Andreas

[trooper11]

ok well your help did fix my problem. the only reason i posted ym quesiton here is becuase this is the thread about the guides that have been posted, so my question was about a problem whne using your guide, so that seems reasonable. and i had used the search function before i posted.

i wanted to ask about other bidules you mentioned, you said oyu had 3 new ones, and that they are updates to SADbidule5.1? or are they just other options? are these other bidules avialable on the ftp or on a site? thanks

[kempfand]

The 3 new bidules I was mentioning are here: 3 new exciting bidules !! (http://forum.doom9.org/showthread.php?s=&postid=508505#post508505)

They are on the FTP, but you can also download directly from www.app.demon.nl/Kempfand-EoH-06-09-2004.rar (http://www.app.demon.nl/Kempfand-EoH-06-09-2004.rar)

I personally see them in comparison to the SAD5.1 method, because they try to create an illusion of discrete speakers, where as the Ambimethods are more about the overall soundfield. Just my reading of course.

Regards,
Andreas

[kempfand]

Add-on:

Originally posted by ursamtl Since the title of the thread is GUIDE: Stereo to 5-Channel Surround, any discussion of any method to achieve this result should be fair game, as well as problems encountered by those following the instructions given here. If you don't want this, then perhaps you should ask the Doom9 management to make this a regular thread because it appears as if it's an overall thread for discussing all methods for converting 2-channel to surround. Again please stop turning around words.

If you want Doom9's management to change things, you do it, not me, because this threat did run very well until a couple of weeks.

Have a nice day.
Andreas

[Eye of Horus]

Originally posted by ursamtl
<snip>
Have a nice weekend!
Steve

I started conversions back in 1999 and I learned a lot in all these years. When I "met" Kempfand, I finally had a partner who could code my sometimes weird ideas. My ideas, his coding and the discussions about how to improve our efforts, lead to the second guide for 2 channels to surround. (This thread). In all these years I loved working with Kempfand, as he is willing to take criticism from me, bad or good. We became good friends ! And..... very productive :-)

Steve, after re-reading everything you placed here, I can only come to the conclusion that you won't achieve this with any of us because of your attitude. You are still a newbie here and you are not the one to tell us how to handle things or how to behave. A bit more modesty would surely suit you.
I would also like to add, that it's pretty rude to answer postings here that are not addressed to you. We can speak for ourselves. We don't need you to answer things addressed to us. And also when you answer, please give a sensible answer and not rubbish, that's useless to the one with the question. (long filenames : 01.wav ???)

Now you are also harrassing Kempfand and I sure don't like it at all ! First throwing all kinds of nasty remarks on Ambisonics and now obviously you changed your mind. One or two tests and immediately the conclusion that our new bidules are not good. Every expert says to leave out the LFE in a conversion, according to you it's ridiculous. And I can go on like this for another 15 minutes........

If you don't get it by now : we don't need this behaviour here !
So I suggest the following :
a. keep this thread (FWIW : it's MY thread) clean of your negative attitude
b. or just leave to another forum

Thanks.

EoH

[ursamtl]

Originally posted by kempfand
Steve: I think your comments are not appropriate. Asking that people use a search function or scroll a few pages up to see if a problem was reported before is a senesful request.

You might call this unfair, but that's how things are in life. Again read the title of this threat: "GUIDE: Stereo to 5-Channel Surround". Information I bring is mostly is on that topic. I repsect your helping hand to people's technical problems, but again: This should not the threat for it.

Andreas, you and EoH present this as a "guide" and give instructions on how to do something. Then if people have a problem and ask for help, or if someone else tries to help them, you tell them to post it in a different thread. This is a "guide" not just some theoretical discussion! If you feel my pointing that out is inappropriate, I'm sorry but I believe in free speech.

If you mention it here, partially quote it, and make your conclusions which you post here, I can comment on it, and I will do if I think it helps to clarify. If you don't like it, don't post it in a forum.
As I just said, I believe in free speech, but I don't believe in your telling me how I should think. That is what you were doing.

Seems pretty clear to me too. Again: That's exactly what Angelo Farina describes on his public server, and what is described in some of the guides here. So I don't see us (you and me) having a difference here.

The difference is that I have claimed that the first page of this guide presents something that is, in my analysis of the facts, in IMHO, not truly "Ambisonic." It applies the principles and methods of Ambisonics, but it is not complete. I feel the UpMix Studio (http://forum.doom9.org/showthread.php?s=&postid=506996#post506996) bidule I uploaded provides a better result than the original guide, but I don't claim it to be true Ambisonics. Anyway, I don't care what we call it, if it sounds good, that's all that really matters.

The reason I've made such a point out of this is that the tone you used in responding to my posts seemed as if you consider any deviation from the actual layout angles or equations presented as some sort of heresy! All I'm saying is that, since it's not truly Ambisonic, the only thing that should matter is the sound..

My native language is not English, but [B]I think you turn around words here in a misleading way. Please stop this.That's a pretty strong accusation. Please provide me with examples if why you feel this way. I have a 12 years experience as a second language teacher, so perhaps I can explain these misunderstandings in a way you might understand. I have not written anything with the intention of misleading anyone. I've been trying to enjoy this whole process of experimentation. However, it seems that if I express an opinion that you don't happen to like, you immediately jump on me. Perhaps I'm wrong, but that's the feeling I get.

It should be clear that the 'devil' is not calling you or anyone devil (I would write that out if that was my intention, which it isn't). The :devil: is meant to express my non-understanding when you praise a method as "rating 4/5", and people here actually use this method for 2 years now.

Reread the message. I did not provide this rating, Angelo Farina did. I simply quoted it. He was rating his method of convolving using true B-format IRs. As I said last message, in the beginning, I didn't know people used this method for two years because the new "guide" that starts this program implies that it supercedes or replaces the old one. By the time I discovered this thread, it was huge. It took quite awhile to find the original guide. I'm not going to go back and requote all the text in the opening message of this thread, but it does say that since a "lot of new software and methods" have been developed, it was time to "to take advantage of these new methods and write a complete new guide." To a native speaker of English, this implies that you consider the old method to be outdated. If one reads deeply enough into the thread, he or she can understand this isn't the case, but to a newcomer to the thread, it seems like the guide at the beginning is the best way to go.

See my comment above. As this is not your personal web-page but a public forum, I feel free to comment, and I believe I have always done in an appropriate and helpful way. Asking that people take reasonable effort to use the Search function or read if problems were solved before doesn't seem wrong to me (maybe a Moderator can comment ?).

Of course you are free to comment, just as I am free to comment that I think it was inappropriate for you to tell people that technical details presented in this forum have nothing to do with this forum! You exact words were:

Originally posted by kempfand
as these technical problems have null and nothing to do with the discussion about the Audio Methods discussed in this threat (which reads "Stereo to 5-Channel Surround").


Could also be the other way round. You didn't seem to like the comments that many people are happy with a Pentagon decode, and that your modified Octagon2 decode has some compressions in the soundfield.

Where have I said that I didn't like comments that "many people are happy". All I ever said was that I wasn't happy with the results of the Pentagon when I tried it and then after doing some research, I explained why, that it compresses the soundfield by as much as 40-80% in the fronts. I found it surprising that as soon as I posted a bidule that produces a much larger soundfield, you immediately said that it was my bidule that compressed the soundfield. It made no sense. It looked more like you were just unhappy with a negative comment about a method you advocated. As an interesting aside to this whole discussion of decoding layouts, I find it interesting, by the way, that Angelo Farina, an expert, uses the Surrounrd layout on his web site.

http://www.ramsete.com/aurora/UHJ-Decoder-Bidule.gif


Anyway, perhaps this is a matter of English not being your first language, but this is the impression I get from your comments. When you misunderstand something I explain, such as why all eight virtual speakers in my Octagon layout do in fact get equal sound pressure, you claim it's not true and I have to re-explain again, but you still come back and tell me I'm wrong. Finally I give up because I start to conclude that perhaps you don't want to understand!

I'm sorry if you feel my saying so is inappropriate, but I felt your request that we stop talking about technical details was also inappropriate.

Regards,
Steve.

[Eye of Horus]

Originally posted by trooper11
ok well your help did fix my problem. the only reason i posted ym quesiton here is becuase this is the thread about the guides that have been posted, so my question was about a problem whne using your guide, so that seems reasonable. and i had used the search function before i posted.

i wanted to ask about other bidules you mentioned, you said oyu had 3 new ones, and that they are updates to SADbidule5.1? or are they just other options? are these other bidules avialable on the ftp or on a site? thanks

I don't want to be a pain in the ..s, but this is also in this thread a few pages back ;)

Anyway : I answered your question about the Besweet problem immediately, but it seems like you didn't read that reply too.......

Not a big deal, but I wrote an almost complete page about these three new bidules together with a mini-guide and the locations where you can download the necessary files.
I don't mind answering questions at all when you have a problem with using them, or when unexpected problems occur. But.... about the new bidules..... you can find it if you try !

I guess you will understand yourself, that it's a bit annoying to have to provide a link everytime to something only a few pages back !

BUT......... don't stop asking when you have a valid question !
I will try to help where possible, but in future we assume you did search first. OK ?

kind regards,

EoH

[ursamtl]

Originally posted by Eye of Horus
You are still a newbie here and you are not the one to tell us how to handle things or how to behave. A bit more modesty would surely suit you.
So this is how you treat newbies?

I would also like to add, that it's pretty rude to answer postings here that are not addressed to you. We can speak for ourselves. We don't need you to answer things addressed to us. And also when you answer, please give a sensible answer and not rubbish, that's useless to the one with the question. (long filenames : 01.wav ???)
It's very rude to tell people who ask for help to go elsewhere. It's also rude to tell people on an open forum that they shouldn't try to help fellow forum members. I explained the long filename aspect as being related to Besweet's location.

Every expert says to leave out the LFE in a conversion, according to you it's ridiculous.Funny, I quoted an award-winning expert who uses the LFE channel. Obviously, it wouldn't exist if it weren't useful.

FWIW : it's MY thread
Oh perhaps you should tell you friend kempfand this, since he just told me it was not my personal web page but a public forum. That would mean it's not yours either!

Now if you have something constructive to say, by all means do so. Otherwise, please stop this inappropriate harrassment of a forum user just because he disagrees with you.

[Eye of Horus]

Originally posted by ursamtl

So this is how you treat newbies?

[/B]
It's very rude to tell people who ask for help to go elsewhere. It's also rude to tell people on an open forum that they shouldn't try to help fellow forum members. I explained the long filename aspect as being related to Besweet's location.

[/B]Funny, I quoted an award-winning expert who uses the LFE channel. Obviously, it wouldn't exist if it weren't useful.


Oh perhaps you should tell you friend kempfand this, since he just told me it was not my personal web page but a public forum. That would mean it's not yours either!

Now if you have something constructive to say, by all means do so. Otherwise, please stop this inappropriate harrassment of a forum user just because he disagrees with you. [/B]

This is my last reply to you (everything else will be ignored from now on):

Perhaps you should read a guide about quoting !
Quoting only a part of a sentence and leaving out the rest, brings it out of context. And that's basically what you do with every response ! Every remark or sentence is manipulated by you in such a way that we will always be the bad guys and you are sooooo good.
Don't make me laugh.

Just leave..... !!

It's a pity we cannot add an ignorelist here. On Usenet I would say only one word to you : PLONK !!!


byyyyeeeee !

EoH

[kempfand]

My native language is not English, but [B]I think you turn around words here in a misleading way. Please stop this.That's a pretty strong accusation. Please provide me with examples if why you feel this way
For example your statement:
When you misunderstand something I explain, such as why all eight virtual speakers in my Octagon layout do in fact get equal sound pressure, you claim it's not true and I have to re-explain again, but you still come back and tell me I'm wrong. Finally I give up because I start to conclude that perhaps you don't want to understand! Fact is that the 5 speakers in your modified Octagon2 receive different sound pressures (not only the C), as I was outlining above. I was also saying that this doesn't imply that it would sounds better or worse than regular decodes, to make it a factual statement on not implay your modified Octoagon2 is bad.

You imply that I am wrong (which I am not), and that you had to "give up" bacause I was too stupid to understand you.

That is just tunrning words the wrong way. I am working for work in an international environment with >40'000 employees world-wide, and I see that kind of behaviour on a daily basis. We call it 'political' (in relation to company politics, not to governmental politics). Most people simply don't like it.

The same 'word-turning' also applies to the compressed sound-field for your decode. I clearly outlined my reasons (did you see how it plots using CATT ? What you get there is pretty much WYSIWYG) in terms of audio). The point that you, in addition, expand the Y to correct for that is not relevant, because I was comparing apples with apples^(i.e. decode 1 vs. decode 2, and not decode 1 vs. decode 2 + modifications).


You surely have excellent language skills, and I claim some scientific skills for my side. So even if you don't like to hear it, I say it again: The methods described here produce 100% Ambisonics. If you (and me and all others) don't feed it with the proper material (UHJ) but just "Stereo", that is a different pair of shoes and has nothing to do with the method used 'per se'.

I hope you see my points, and that we can bring this discussion to a closure, focus on improving the methods, and have some fun.

Regards,
Andreas

[ursamtl]

Originally posted by Eye of Horus
Quoting only a part of a sentence and leaving out the rest, brings it out of context. And that's basically what you do with every response ! Every remark or sentence is manipulated by you in such a way that we will always be the bad guys and you are sooooo good.

Quoting entire long messages in forum threads can make them ridiculously long and needlessly eat up bandwidth. the beauty of the forum format is that if anyone wants or needs to the context of a comment, they can always go back and read the original messages. Both you and kempfand have accused me of somehow manipulating every quote I make to somehow make myself look good and you guys look bad. I'll simply tell him what I told you, I invite you to provide proof of this and then I'd be happy to explain why it is a misunderstanding. I know English is not your first language in either case and I understand that comments can be misinterpreted. However, I repeat what I wrote the last time you and I had a disagreement, you seem to take everything much too personally, as if someone's criticism of a method you advocate is somehow a direct personal attack on you. I don't know you. I have nothing against you personally. I'm merely providing feedback on your bidules. If you can't take criticism, then perhaps you shouldn't post them in a forum but instead on a web page without any interactive messaging. Once more I apologize if you have felt offended by remarks I've made as it was not my intention to offend you. I don't feel I have anything to apologize for, but if it will make you feel better, fine! I continue to reserve the same right as any other participant in this forum to comment positively or negatively on anything you present publically in a forum. That's what it's all about, conversation!

[kempfand]

As an interesting aside to this whole discussion of decoding layouts, I find it interesting, by the way, that Angelo Farina, an expert, uses the Surrounrd layout on his web site.

Note that Anngelo Farina's layout is to illustrate the UHJ (or Stereo for us) to B transcode using convolution.

Speaking about experts, I can only tell you that the layout on the 1st version of that page had mistakes (as well as the bidules which go with it), and that the owner made the appropriate corrections based on my input.

In order to prevent another 'world-turnin', let me just add that the errors on the 1st version of that page were just because of things happened in a hurry.

The 5.1 decode (ITU) on that page is more for illustrative purposes, and this becomes immediately clear if you read some of Farina's papers or look at his decode using SoundWeb processor.

Regards,

Andreas

[ursamtl]

Originally posted by kempfand
Fact is that the 5 speakers in your modified Octagon2 receive different sound pressures (not only the C), as I was outlining above. I was also saying that this doesn't imply that it would sounds better or worse than regular decodes, to make it a factual statement on not implay your modified Octoagon2 is bad.

Ok, let me try this one more time. Three are eight "virtual speaker positions" each of which receives the correct amount of sound pressure. Only five of these are real speakers, with the other three being "phantom speakers." To create those phantom speakers, I sent half of their allotment of sound pressure to each adjacent speaker.

So in a way, we're both right. I send the same amount to my virtual speakers, but not the same amount to the real speakers. There, case closed. :)

You imply that I am wrong (which I am not), and that you had to "give up" bacause I was too stupid to understand you.
No, I'm sorry but this is wrong. I did not use the word "stupid" nor did I imply it. I said that I concluded that you didn't "want" to understand. Meaning, (and excuse my horrible German :-) "Du müchst nicht versteihin." Maybe I am wrong, but that's the feeling I get.

The same 'word-turning' also applies to the compressed sound-field for your decode. I clearly outlined my reasons (did you see how it plots using CATT ? What you get there is pretty much WYSIWYG) in terms of audio). The point that you, in addition, expand the Y to correct for that is not relevant, because I was comparing apples with apples^(i.e. decode 1 vs. decode 2, and not decode 1 vs. decode 2 + modifications).
Ok, I understand, but I think the problem is as I've stated, our goals are "apples and oranges." Since I don't consider this to be "real" ambisonics, I don't care if comparing one layout to another results in a certain plotting on CATT. I plotted the layouts on an Adobe Illustrator file during my lunch break at work. Unfortunately, I didn't think to send the file home or I'd post more diagrams. The whole point is, it doesn't really matter what the plots look like, what matters is the sound! Obviously, I respect the work you guys have done because I took the time to dig a bit more on the Ambisonics side of things without rejecting it outright. I will explore the new bidules more, but as I said in my initial comments, they seemed disappointing to me. If others find them to be good, then great, but I don't feel I should be jumped on and told to shut up because I'm a newbie in this forum just because I might disagree. I may be a newbie here and I may not have a scientific background, but I've been a music fanatic for years and have been playing with various concoctions of homemade surround sound for 30 years. I know what sounds good to my ears, and that's all that matters to me. If you produce something that does sound good to me, I'll be the first one to congratulate you, but if there's a problem with it, I'll also say so. As I said in my postings after testing the new bidules, let's look at them to see if improvements can be made. I'm already working on some improvements and extensions to UpMix Studio and I hope you'll give it a critical listen once I post it. Now yes, let close this discussion and have fun!

Steve.

[ursamtl]

Originally posted by kempfand
Note that Anngelo Farina's layout is to illustrate the UHJ (or Stereo for us) to B transcode using convolution.

Speaking about experts, I can only tell you that the layout on the 1st version of that page had mistakes (as well as the bidules which go with it), and that the owner made the appropriate corrections based on my input.

In order to prevent another 'world-turnin', let me just add that the errors on the 1st version of that page were just because of things happened in a hurry.

The 5.1 decode (ITU) on that page is more for illustrative purposes, and this becomes immediately clear if you read some of Farina's papers or look at his decode using SoundWeb processor.

Regards,

Andreas

That may be so, I don`t know. As a related aside, have you ever plotted Furse's positions for the "Surround1" layout? The surround speakers are at 60° angles behind the listening point instead of the 45° I've seen in diagrams for the actual IUT standard.

[jorel]

@ kempfand(Andreas),Eye of Horus(EOH cos i don't know your name) and all:
first: i can read but can't understand the last posts(and lots more).
second: in my simple(and newby)opinion,this thread in the one of the most importants threads in the whole world about audio.(if don't the best of all threads about audio).
--->Please, don't "fight"(i can't find better wordds)to turn this thread(i can call magnific thread)in a dangerous place to be "closed" by any mod!
ps: i'm not against anyone,i'm just "afraid" about this thread!
ah...and sorry for bore! :o
;) thanks but..please,this thread don't have "price",money can't buy!

[kempfand]

I don't care if comparing one layout to another results in a certain plotting on CATT. I plotted the layouts on an Adobe Illustrator file during my lunch break at work. Unfortunately, I didn't think to send the file home or I'd post more diagrams. The whole point is, it doesn't really matter what the plots look like, what matters is the sound! I really don't care too much about scientific plots myself, but after having done listening tested for dozens of decoding rigs and see how they plot in CATT, I came to the conclusion that CATT is a phantastic tool to work on Ambisonics (and other things), and that the plots you get there are most often WYSIWYG. In the end, I fully agree, the important thing is how things sound.

We tested the pre-set decoding rigs and custom made decodes from Emigrator, CATT, ambidec.exe, VVMix (exe & VST), signed filters (using impulse responses similar to the ones published by Angelo Farina), and the conclusion was always: Pentagon (even if people have a ITU 5.1 in their living room).

For fairness reasons, I clearly mentioned (a) that I do not exclude that there are better decoding set-ups, (b) that even amongst the experts, there is agreement that this is still 'subject to discussion' (cf. Farina's presentation at recent AES in Berlin), and (c) that people have 'stong feelings' about this, in that they put patent protection to specific Amisonic decoding schemes. For (c), the most prominent example is Gerzon's Vienna decoding scheme, which is available in the high-end "$$$" amps by Meridian (and they do very very well, at least according to the "Verdi" listening tests).


I will explore the new bidules more, but as I said in my initial comments, they seemed disappointing to me. If others find them to be good, then great, but I don't feel I should be jumped on and told to shut up because I'm a newbie in this forum just because I might disagree. No problem if you found them disappointing. As mentioned, I value your and anyone's feedback, and I think there is room for improvement. Given today means of Forums and commication across geographies and time zones, we can achieve things that were not possible 10 years back. Might sound 'generic', but I truely believe in this.


I may be a newbie here and I may not have a scientific background, but I've been a music fanatic for years and have been playing with various concoctions of homemade surround sound for 30 years. I think (hope so) I made it clear that your and everyone's feedback is welcome. I have also seen that you have a background with surround methods (otherwise I wouldn't have entered this discussion at all).


I know what sounds good to my ears, and that's all that matters to me. If you produce something that does sound good to me, I'll be the first one to congratulate you, but if there's a problem with it, I'll also say so. As I said in my postings after testing the new bidules, let's look at them to see if improvements can be made. I'm already working on some improvements and extensions to UpMix Studio and I hope you'll give it a critical listen once I post it. I believe in 'free markets' in this aspects, so let the readers & users of thses methods judge. As indicated, I am contributing here for competitive reasons (if I would, I should make my licing out of this). Off-topic remark: I'm not a believer in 'free markets' when it concerns certain social aspects, but this is a whole topic for itself and the wrong place to discuss (still couldn't resist making that statement).

Now yes, let close this discussion and have fun! Yes.

To quote member "Shayne": Peace.

Andreas

[kempfand]

@ jorel: Thanks for your remarks. I really value, especailly from someone who has >750 posts and therefore a proven tackrecord :)

We'll try so.

Kind regards,

Andreas

[trooper11]

@Eye of Horus

hey, yeah youve helped me out alot in the past and i really appreciate it, as far as looking back for info on the thre new bidules, i just kind of got overwhelmed that i would have to read 29 pages to search out info on it, but i guess ill just have to to get links to things, i just wish there was an easier way of finding things is all lol.

you and kemp have done a great job wiht this and i appreciate the work you guys have done, its really been a help for me, sorry if my questions annoyed you guys. keep up the good work.

hey maybe there should be a new sticky devoted just to updates to the method, like guides and bidules, since its so hard to sift them out from this huge thread. you think that would be a good idea?

oh and i just wanted to know what you guys ,EOH and Kemp, use to go to ac3 tracks? or maybe you dont do taht at all. i was thinking of just using besweet to go to ac3, i just wanted to know if im sacrificing quality by using besweet versus another app.

and alos, i read back through the thread and in your guide for hte new ones oyu state that you should set 16bit in the recorder when for the SAd method oyu said 32bit, is there a reason you changed this? also, should i be using 48000 sample when my wav source is 48000?

[ursamtl]

Originally posted by trooper11
hey maybe there should be a new sticky devoted just to updates to the method, like guides and bidules, since its so hard to sift them out from this huge thread. you think that would be a good idea?

Yes, I think this is needed. Daphy and Andy have done a great job putting everything together on the NeedfulThings server, but without some of the context, all the stuff on there can be overwhelming. There is a PDF with the thread discussions, but it still takes some detective work to locate the right information sometimes.

[ursamtl]

Originally posted by kempfand
For fairness reasons, I clearly mentioned (a) that I do not exclude that there are better decoding set-ups, (b) that even amongst the experts, there is agreement that this is still 'subject to discussion' (cf. Farina's presentation at recent AES in Berlin), and (c) that people have 'stong feelings' about this, in that they put patent protection to specific Amisonic decoding schemes. For (c), the most prominent example is Gerzon's Vienna decoding scheme, which is available in the high-end "$$$" amps by Meridian (and they do very very well, at least according to the "Verdi" listening tests).

I think we'll have to agree to disagree on this point, but you see as I mentioned earlier, because I don't see what we're doing as strict, true Ambisonics, it's hard to say that any one rid is the best. Your tests have led you to believe the Pentagon is best. I have tried rotated Pentagons, unrotated (with no center, just a huge, wide surround layout), modified them to widen the Y signals, pumped a stereo wide signal into the input--everything I could think of but both on my PC and my full surround systems, the result sounded small and boxy with some bad phasing happening. The ambiophonics front bidule you posted improved this somewhat (even with my speakers at +-30° instead of 10). then when I took Panorama and hooked it to Emigrator set to Octagon2 with phantom speakers for the three missing speakers, the sound really started to open up. Once I did the bidule groups with the actual formulas instead of the VSTs, it got better. Boosting the Y signal last weekend added even more. With UpMix Studio I am now close to the big, wide commercial sound I heard on true 5.1 mixes but without it sounding gimicky.

I'm listening to some instrumental music right now through your new instrumental bidule, some acoustic guitar. It sounds quite pleasant for this type of music although it's missing definition and accuracy on the individual notes, sort of like smearing them together. I tried it on music with bass and drums and the bottom was really missing. Perhaps crossing over and applying only to the mids might work because the highs get smeared and the lows just aren't there.

A little earlier I was testing the vocal center bidule. I find that I have to crank the Gain Center up to +10 to get a vocal that isn't overwhelmed by it's own reverb. I will say that once I did get the center level high enough, this bidule sounds better than the other two. I know I said the vocal non center one was my favorite initially, but once I got rid of the doubled percusion sounds by backing off the delays, the soundfield spread seemed quite unnatural with cymbals constantly playing behind my right shoulder. It was kind of weird.

My only concern about the vocal center is the same concern I had about my first MatrixMixerEmulator and the Gerzon bidules, that is putting so much dependancy on the fullrange sound of a center speaker. A lot of center speakers are not designed for solid bass for example, they're optimized for dialog. I'm just wondering what your thoughts are on that.

Anyway, que l'on continue...

[KpeX]

@ all

Please keep this discussion as professional and civil as possible. This is an audio processing thread unlike anything else on the web IMO, and I would hate to see it degrade. We have some excellent minds working on these bidules and 5.1 methods, and I ask that you respect each other's intelligence and to respect others' work when posting. As always, thinking before speaking will be greatly appreciated. Cheers,

[Eye of Horus]

Originally posted by trooper11
@Eye of Horus

hey, yeah youve helped me out alot in the past and i really appreciate it, as far as looking back for info on the thre new bidules, i just kind of got overwhelmed that i would have to read 29 pages to search out info on it, but i guess ill just have to to get links to things, i just wish there was an easier way of finding things is all lol.

I agree with you on the search aspect. The search routine used here is not the ideal one :-)


you and kemp have done a great job wiht this and i appreciate the work you guys have done, its really been a help for me, sorry if my questions annoyed you guys. keep up the good work.


Thanks !! We try :-) (that is : if we are not too much distracted by useless discussions :-))))


hey maybe there should be a new sticky devoted just to updates to the method, like guides and bidules, since its so hard to sift them out from this huge thread. you think that would be a good idea?


That's upto the moderators......... who BTW don't seem very active lately ! in the past I remeber that certain psots would have been deleted immedietely :-)


oh and i just wanted to know what you guys ,EOH and Kemp, use to go to ac3 tracks? or maybe you dont do taht at all. i was thinking of just using besweet to go to ac3, i just wanted to know if im sacrificing quality by using besweet versus another app.


Yes, you lose quality ! the freeware AC3 encode doesn't compete with a commercial one like softencode or the one in Scenarist. If you need advice about the settings of commercial products, please contact me privately. I don't think this is the place, but on the other hand : search :-) I know there are some topics about running Softencode in commandline mode and you can find settings there....


and alos, i read back through the thread and in your guide for hte new ones oyu state that you should set 16bit in the recorder when for the SAd method oyu said 32bit, is there a reason you changed this?


Yes...... the reason is that Surcode does internally the same (32 bits to 16 bits). 32 bits WAV's don't play on regular players (like MS Mediaplayer) and they take a lot of space. People with FAT32 file systems won't be able to convert long records in one go in 32 bits, but with 16 bits they can convert longer ones. (Actually 2048/3 = about 682 MB long stereo CD's). We thought this way the whole process would be easier for a lot !


also, should i be using 48000 sample when my wav source is 48000?

Yes ! (And don't forget to adjust Plogue Bidule to 48000 hz too !)

kind regards,

EoH

[Eye of Horus]

Originally posted by KpeX
@ all

Please keep this discussion as professional and civil as possible. This is an audio processing thread unlike anything else on the web IMO, and I would hate to see it degrade. We have some excellent minds working on these bidules and 5.1 methods, and I ask that you respect each other's intelligence and to respect others' work when posting. As always, thinking before speaking will be greatly appreciated. Cheers,

Thanks for the input KpeX !
I fully agree and only wished you'd come up earlier :-)

Anyway : are you still in the conversion "business" ?
Because we never heard any comment from you about our SAD51inBidule, which was based on your conversion method !


kind regards,

EoH

[kempfand]

Originally posted by trooper11
also, i read back through the thread and in your guide for hte new ones oyu state that you should set 16bit in the recorder when for the SAd method oyu said 32bit, is there a reason you changed this? also, should i be using 48000 sample when my wav source is 48000? and Originally posted by Eye of Horus
Yes...... the reason is that Surcode does internally the same (32 bits to 16 bits). 32 bits WAV's don't play on regular players (like MS Mediaplayer) and they take a lot of space. People with FAT32 file systems won't be able to convert long records in one go in 32 bits, but with 16 bits they can convert longer ones. (Actually 2048/3 = about 682 MB long stereo CD's). We thought this way the whole process would be easier for a lot !

Another important reason is, that the process for 3 new bidules includes a normalization on the L,R,C,SL,SR speakers. Note that it is not a normalizaion over the 6-channel WAV (output from Bidule), but a normalization on each of the speakers individually.

This is currently not possibly in a "one go" commandline in BeSweet, so you have to first demux, and then normalize each of the channels.


More on the same topic:
Originally posted by ursamtl
I'm listening to some instrumental music right now through your new instrumental bidule, some acoustic guitar. It sounds quite pleasant for this type of music although it's missing definition and accuracy on the individual notes, sort of like smearing them together. I tried it on music with bass and drums and the bottom was really missing. Perhaps crossing over and applying only to the mids might work because the highs get smeared and the lows just aren't there.

A little earlier I was testing the vocal center bidule. I find that I have to crank the Gain Center up to +10 to get a vocal that isn't overwhelmed by it's own reverb. I will say that once I did get the center level high enough, this bidule sounds better than the other two. I know I said the vocal non center one was my favorite initially, but once I got rid of the doubled percusion sounds by backing off the delays, the soundfield spread seemed quite unnatural with cymbals constantly playing behind my right shoulder. It was kind of weird. I guess you just took the bidules and listened via the ASIO device on the speakers of the PC. That will give bad results. The bidules, as published, are intended to work reasonably well if you follow up with the normalization step, and then listen to the output.

This might sound over-complicated, but we found that you can tweak the gains for the speakers (L+R, C, SL+SR) to work well for one piece of music, whereas it would not work well for the next piece of music.

Normalization seemed a good compromise also for publishing here.

Originally posted by ursamtl
I think we'll have to agree to disagree on this point(of how to best decode), but you see as I mentioned earlier, because I don't see what we're doing as strict, true Ambisonics, it's hard to say that any one rid is the best. Your tests have led you to believe the Pentagon is best. I have tried rotated Pentagons, unrotated (with no center, just a huge, wide surround layout), modified them to widen the Y signals, pumped a stereo wide signal into the input--everything I could think of but both on my PC and my full surround systems, the result sounded small and boxy with some bad phasing happening. The ambiophonics front bidule you posted improved this somewhat (even with my speakers at +-30° instead of 10). then when I took Panorama and hooked it to Emigrator set to Octagon2 with phantom speakers for the three missing speakers, the sound really started to open up. Once I did the bidule groups with the actual formulas instead of the VSTs, it got better.

Emigrator has some shortcomings, which I understand are significantly improved with the commercial version (DecoPro (http://www.gerzonic.net/?prod=24&rub=2)). The standard decodes by Richard Furse, as done by AmbiDec (http://www.muse.demon.co.uk/utils/ambidec.html) have other shortcomings (as one example, they use no shelf filters). I personally had my best decodes with using signed filters. In practical terms, one can either use CATT, or 2 parallel instances of Voxengo's pristine space since you need 15 concurrent convolutions (3 for each of W,X,Y times 5 speakers).

The newest research (driven by Farina, Malham, and others) even explores using one of the 5 speakers for vertical information (Z-axis). Even if the feeding source is 'just' stereo, information for the 3rd dimension (Z-axis) might be re-created by convolution with the a B-Format pulse. This is still in the research status, but I think we will see a whole range of new decoding options over the next couple of years. Lot's of things to play with and have fun :)

Regards,
Andreas

[ursamtl]

Originally posted by kempfand
I guess you just took the bidules and listened via the ASIO device on the speakers of the PC. That will give bad results. The bidules, as published, are intended to work reasonably well if you follow up with the normalization step, and then listen to the output.That's an excellent point and yes I have overlooked it. Before I take the time to burn the CD, I like to tweak the sound as much as possible first. I just found it strange that the center required such a boost in the bidule to begin with.

As I mentioned last night in a message, one concern I have with these new bidules as with the LCR upmix bidules is that the front and rear left and right signals are getting all midrange and high frequencies and one speaker, the center, gets almost all the bass info. Most surround systems don't have a center speaker with as good a bass response as the front left and rights. The center speaker is traditionally designed for dialog. Yes, a high-end system should have a center speaker with good frequency response, but not everyone doing these decodes has a high-end system! If one has a subwoofer and bass management on the amp, this certainly improves things, but it's a waste of the range capabilities of the left and right speakers. Normalization won't change the frequency content of the left and right channels in a bidule, it will only increase or decrease levels as required. Anyway, just a thought.

By the way, try the following for an experiment: take a stereo difference signal and run it through an instance of Stereo Touch. Connect that to the fronts. Do the same with a second instance of Stereo touch and connect it to the rears. Take a stereo sum signal, reduce it by 3dB and connect it to the center. The result should sound familiar. Let's call it "Mr. Hafler's Repurposing Bidule" ;)

The newest research (driven by Farina, Malham, and others) even explores using one of the 5 speakers for vertical information (Z-axis). Even if the feeding source is 'just' stereo, information for the 3rd dimension (Z-axis) might be re-created by convolution with the a B-Format pulse. This is still in the research status, but I think we will see a whole range of new decoding options over the next couple of years. Lot's of things to play with and have fun :) Yes that does sound interesting. The only problem I have with this is like with the Ambiophonics +-10° bidule, having to move speakers from their usual positions. I regularly rent or purchase movies on DVD and enjoy 5.1 surround sound immensely. My TV room has a somewhat challenging shape and it took me awhile to get close to a proper ITU-5.1 setup fit in there. I don't want to start moving stuff around! However, if the Z info is implemented in the surrounds, well that could be interesting, since mine are about a meter above ear level.

Tchuss
Steve.

[trooper11]

@EOH

thanks, i sent you a pm btw

[ursamtl]

Just a note for those using SIR, Christian Knufinke released a new version yesterday.

The web site says v 1.007, but the download is actually v1.008.

http://www.knufinke.de/sir/index_en.html


Regards,
Steve.

[ursamtl]

For those experimenting with designing new surround bidules, I'm posting a couple of links to useful articles I've found while surfing. Although these both relate to mixing or programming for surround sound, they contain information that can help and considerations to keep in mind when designing surround sound bidules or plugins.

This first one has some interesting points regarding use of the LFE and center channels, two things that have come up in this thread:


5.1-channel Sound: From the studio to your home theater (http://www.crutchfieldadvisor.com/ISEO-rgbtcspd/reviews/20030923/5_1_sound.html) by Leslie Shapiro


While this second article is more programming related and deals with DPL, it's still interesting and provides some discussion of sound cancellation.


Dolby Pro-Logic Encoding (http://www.gamasutra.com/features/sound_and_music/19981218/surround_03.htm) by Robert Basler


Anyone else with useful links to practical surround-related information?

Regards,
Steve.

[LAYLA1970]

Hello everybody:

Through a friend of mine, I can heard the post over "Rebuild The Wall".
'Till now, I still surprise with the overall sound. I think it is a forward step to right direction. Here are dinamics with no distortion, well focused any instrumentals, tonnaly very well balanced and all of this produces a typical sense of happiness. God job, guys.

Only I want make a question. IMHO, the conversion lacks of voice definition. It seems to me that the voices are ONE STEP BACKWARD over the other instrumentals. This causes an effect of opacity over all the voices. But, if we are hearding only voices (there are a piece so), these voices sounding really great. If we go to ear the instrumentals (not the voices), that's the same: they are great. But if two of the worlds are joined, this a dissapointment. I can't ear clearly the voices. This effect can make resolve with the new values of the gains? I will say, if the values of center are aumented, the voices are more presents?

Thanks and Good Luck.

[ursamtl]

Originally posted by LAYLA1970
Only I want make a question. IMHO, the conversion lacks of voice definition. It seems to me that the voices are ONE STEP BACKWARD over the other instrumentals. This causes an effect of opacity over all the voices. But, if we are hearding only voices (there are a piece so), these voices sounding really great. If we go to ear the instrumentals (not the voices), that's the same: they are great. But if two of the worlds are joined, this a dissapointment. I can't ear clearly the voices. This effect can make resolve with the new values of the gains? I will say, if the values of center are aumented, the voices are more presents?Hi Layla,

I had the same problem. Kempfand pointed out that I wasn't doing the normalize step. Did you try that? This did help my results a bit. I also modified the center feed to the standard -3dB L+R signal. To accomplish this, Ctrl+click on the "Preservation_3_x_2" group in the center and change the multiplier from 0.4511 to 0.7071. This number produces a -3dB gain, whereas 0.4511 is close to -7dB.

While you are in the group, you can also remove the the unused left and right multiplier, constant, and connection objects, since they are not hooked up to anything but still seem to use CPU resources. After I got rid of them, my CPU usage dropped by an average of 2-3%. I know that is not a big drop, but every little bit helps!:)

Regards,
Steve.

[trooper11]

@EOH i sent you a couple pm's



right now im experimenting with the 3 new bidules and the SAD method. right now though, the SAD mehtod is giving better results, but i think thats my fualt. using hte instrumental bidule, the result was lower in sound then in the SAD method. I think it may be becuase before i loaded the file into the bidule, i had already normalized it, converting it to 32bit as well. As i followed the guide, i know it was normalized again by besweet, so could this cuase the sound to seem lower, dropping out?

ok and my other question for you guys, which of these bidules do you think would work well with audio tracks? By audio i mean movie audio that i want to take to 5.1. Or maybe none of these are suited for that.

[Shayne]

Originally posted by kempfand
and

Another important reason is, that the process for 3 new bidules includes a normalization on the L,R,C,SL,SR speakers. Note that it is not a normalizaion over the 6-channel WAV (output from Bidule), but a normalization on each of the speakers individually.

Regards,
Andreas

normalizing left and right individually would loose the initial stereo would it not? I feel the l and r whether it be front or rear should be grouped. Its the F R C LFE groups that would be nice to be able to normalize independant on the fly but then i would be Happy to just isolate the lfe.

Peace

[ursamtl]

Originally posted by Shayne
normalizing left and right individually would loose the initial stereo would it not? I feel the l and r whether it be front or rear should be grouped. Its the F R C LFE groups that would be nice to be able to normalize independant on the fly but then i would be Happy to just isolate the lfe.

Peace To my understanding, all normalizing does is scan a file for the highest peak level and then either boosts or attenuates the entire file in a linear fashion so that this highest peak reaches a target level.

In the case of the 3 bidules, the "instrumental" bidule has the same signal going to both fronts, and the same signal going to both backs. To check this, set up a binary operator in the bidule as a minus and then hook both signals up to the operator's input and connect the output to one channel of your sound card. If nothing comes out, the signals are the same. Therefore, normalizing these channels independantly will give balanced results for the front between the left and right and the same for the rear.

For the "vocal-center" bidule, the same signal (L-R) is sent to Voxengo to generate a pseudo stereo signal. (Yes, one is a R-L, but if you invert it, you get the same as L-R. You can check this the same way as above). In this bidule, the front left and rear right channels get the same signal. The front right and rear left get the same signal. Since these signals are all generated from duplicate Stereo Touch instances, the levels are basically the same and thus normalizing all these channels independantly should keep things balanced.

The "vocal-non center and instrumental" bidule is a bit more sophisticated in that the improvement groups contain filters as well as sum and difference combinations to create four different channels, but these are all at about the same level. If you try the channel subtracting test I mentioned above on all combinations of these channels, you'll always still hear something (indicating that there is some difference), but the difference levels are low and consistent, so normalizing these individually shouldn't be an issue. I haven't dug into this bidule any further to analyse what's going on because I don't like what it does to music with percussive sounds and I'm busy developing something else I find more interesting.

In the end, follow the bidule author's advice. Andreas said that he got the best results by normalizing the channels individually, so try it like that. Then if you don't like the results, try something else. Half the fun is in the trying!

Have a good weekend!
Steve.

[Shayne]

Originally posted by ursamtl
To my understanding, all normalizing does is scan a file for the highest peak level and then either boosts or attenuates the entire file in a linear fashion so that this highest peak reaches a target level.

In the case of the 3 bidules, the "instrumental" bidule has the same signal going to both fronts, and the same signal going to both backs.

Steve.
Your understanding of normalization would be correct. It finds the highs peak and increases all data points by the same amount(dbs).

If the 3 new bidules merge the original left and right and loose all stereo i would have to stand by my post above and wonder if this is right?

Peace

[ursamtl]

Originally posted by Shayne
Your understanding of normalization would be correct. It finds the highs peak and increases all data points by the same amount(dbs).Just a quick point...it can also work to attenuate as well. For example, if the highest peak is at -1dB and the target normalization level is -3dB, then the process will reduce the entire track level by 2dB.

If the 3 new bidules merge the original left and right and loose all stereo i would have to stand by my post above and wonder if this is right?It's not quite that simple. I'll give a technical explanation because I don't want my comments to cause a misunderstanding and start another flame war!;)

We all tend to think of stereo as two completely different channels of sound. However, if you change the two channels to M (for "Mid" or L+R) and S (for "Side" or L-R) then things can get quite interesting. Adding the two signals together without changing levels gives one composite stereo channel back, while inverting the phase of the S signal before combining the two will give the opposite composite stereo channel.

If you start changing something about either of these before recombining, you get different results. "Stereo wide" effects basically increase the ratio of the S signal to the M before recombining (the better ones do frequency filtering because without it, this effect can kill bass response).

To relate this to the bidules, when I wrote that the same signal is going to both fronts and the same to both rears, I meant that the same "difference" or S signal is going to each front and to each rear. If you then add an M signal to one of these, you can recover some sort of stereo channel. What gives the sense of difference is that the S signal is manipulated by the Stereo Touch plugin to produce two signals that are somewhat different from each other. If you then add the same sum signal to each, you get composite channels that are also somewhat different from each other. However, in the case of the vocal-center bidule for example, no composite channel is created. Instead the M signal is fed to the center channel. Thus, any recombining takes places acoustically through soundwave phase cancellation and addition.

This definitely gives interesting surround effects, but it can vary depending on where the listener sits, and especially on the type of music. The research Andreas translated into his original LCR bidules is quite right in saying that these can provide a wider sweet spot for a stereo image, but if this image is transformed or modified, so is the sweet spot and the results.

Since Stereo Touch does much of its work using delays, any music with percussive, rhythmic sounds could get messy. In fact, if you read the description of Stereo Touch on the Voxengo free VSTs page, (http://www.voxengo.com/freevst/), it says that the plug-in is "most effective on mono pad sounds and alike." I don't know if you're familiar with this term, but a "pad" is usually a synthesizer patch or a real recorded sound that is smooth and continuous in nature, without any sharp percussion or rhythm. So running a complete recording through Stereo Touch can sound bad (especially one with a rhythm section).

Shayne, to get back to your normalizing question, given that the same difference signal is fed through the same effects plugins with the same settings, there should not be any problems at all. If you're in doubt, the best thing for you to do is to find a way to measure the "average RMS level" of each channel. Programs such as Sound Forge or Wavelab allow you to measure this. If the numbers for each channel are close, normalizing should not upset the balance. If, however, one channel has a sound somewhere that's louder than any others then it might end up sounding louder than the others. But, as I wrote, that's unlikely given the construction of these bidules.

Regards,
Steve.

[MaroonMike]

I finally tested out the new bidules (the voice-non center_or_instrumental one) and unless I am doing something wrong, it has way to much reverb for my taste. I felt like the music was "bouncing" all over the place. Sorry...

I still think my Ambisonics bidule (basicly the one from the guide with a little extra reverb added to the rear channels) sounds much better. I tried this on two seperate sound samples tonight.

On a separate note, in a standard 6 speaker setup, are there any benefits going to the 3rd level in ambisonics?

[Eye of Horus]

Originally posted by MaroonMike
I finally tested out the new bidules (the voice-non center_or_instrumental one) and unless I am doing something wrong, it has way to much reverb for my taste. I felt like the music was "bouncing" all over the place. Sorry...

Try setting the Voxengo Stereo Touch to "stage" , and give the Center some extra gain. This will get rid of the reverb.
(The default setting we used in Voxengo was "pretty wide" , because that one gave good results on our testmaterial.)

kind regards,

Eye of Horus

[Eye of Horus]

Originally posted by LAYLA1970
Hello everybody:

Through a friend of mine, I can heard the post over "Rebuild The Wall".
'Till now, I still surprise with the overall sound. I think it is a forward step to right direction. Here are dinamics with no distortion, well focused any instrumentals, tonnaly very well balanced and all of this produces a typical sense of happiness. God job, guys.

Only I want make a question. IMHO, the conversion lacks of voice definition. It seems to me that the voices are ONE STEP BACKWARD over the other instrumentals. This causes an effect of opacity over all the voices. But, if we are hearding only voices (there are a piece so), these voices sounding really great. If we go to ear the instrumentals (not the voices), that's the same: they are great. But if two of the worlds are joined, this a dissapointment. I can't ear clearly the voices. This effect can make resolve with the new values of the gains? I will say, if the values of center are aumented, the voices are more presents?

Thanks and Good Luck.

The CD you mentioned was done with the pretty wide setting in Voxengo . Replace that setting with "Stage", add extra gain to the Center and the results will be far better.
There is also another goodie we found and are using, but you have to contact me or kempfand by PM !
We cannot publish all our secrets here ;)

And a bit of a warning : I wouldn't suggest you to mess in the groups (as someone "advices"). For more gain use the gain settings. They are in the bidule with a purpose, not just as a gimmick !

kind reagrds,

EoH

[ursamtl]

Originally posted by Eye of Horus
The CD you mentioned was done with the pretty wide setting in Voxengo . Replace that setting with "Stage", add extra gain to the Center and the results will be far better.
There is also another goodie we found and are using, but you have to contact me or kempfand by PM !
We cannot publish all our secrets here ;)

And a bit of a warning : I wouldn't suggest you to mess in the groups (as someone "advices"). For more gain use the gain settings. They are in the bidule with a purpose, not just as a gimmick !

kind reagrds,

EoH I'm sorry EoH but I disagree that the results will be "far better." I just tested this and I was still disappointed with the results.

Changing to the "Stage" setting didn't do anything for the percussive doubling problem I've mentioned. Both the voice-center and intrumental bidules still smear rhythmic or percussive sounds. For anyone who wants to hear this, do the following:

[list=1]
Take a song with some well-definied repetitive rhythmic sounds. For example, I just used Pink Floyd's "Comfortably Numb." If you don't have it, any song with well-defined drums (especially cymbals), strummed acoustic guitar, sharp rhythm guitar strums, slow harp plucks, etc.
Load in the Voice-Center bidule and load your song in the Audio File Player.
Replace the Audio File Recorder in bidule with Microsoft Sound Mapper output device. To do this Right-click the Audio File Player, and choose Replace > Audio Devices > Output > MME > Microsoft Sound Mapper.
Load your file into Audio File Player and listen carefully.
[/list=1]
What you hear is the sound that will be in the front speakers of your surrounds after you burn your surround CD with this bidule. Yes I know the two channels still aren't normalized as suggested in the method, but all normalizing will do is vary the volume level of the channel. It won't change the nature of the sounds in the channel. If there's a problem with the sound information contained in a channel, normalization will not erase it, but only change the volume level of the channel.

Back to Comfortably Numb. Listen to the door knocks and the guy saying "Time to go...." at the beginning. In the original stereo recording, both are panned to the left as the artists originally intended. Through the bidule, however, both come from the center. Ok so that's not the end of the world, this is surround sound, right? Things may get moved.

However, listen carefully to those door knocks. In the original recording they were so sharp and well defined, if you closed your eyes, you would almost jump up to go and answer your own door! Yet through the bidule, the realism is diminished. That's because the original audio waveform's transients have been mangled and delays echo the sound in both speakers. It still doesn't sound bad, but the sharp focus of the original recording is lost. Surround sound shouldn't do that.

Listen to the cymbals in the verses and the strummed acoustic guitar in the chorus. They no longer sound real and well-definied. Surround sound shouldn't do that.

The tight locked-in bass and kick drum notes are also smeared. It's also harder to hear these because most of their sound energy is contained in the sum of the original stereo signals that is sent to the center channel.

This brings up another point I've mentioned in recent posts. If you've got a monster center speaker with great bass and treble response, you'll be ok. Most 5.1 systems have a center speaker that's optimized for dialog. Bass response in these is rolled off. Since there's sound going to your LFE in this bidule, the only deep, solid bass you'll get is if your center speaker can produce it.

As for increasing the front center gain to make the vocals more balanced and less reverberated, to do that, you have to bring the center gain up to +10dB. This is if one has a monitoring system hooked up to listen to the adjustments. On most songs, this causes digital clipping and distortion. Yes, you're going to normalize the files afterwards, but once the distortion is introduced, it can't be removed.

Look both EoH and Andreas, I've got nothing against either of you personally, but the more I explore these "exciting new bidules", the more disappointed I get. The logic of spreading the sounds out among the four speakers is good, but the execution mangles the original soundstage too much. If you can find a way to restore the soundstage and position of the instruments as the original artists intended, then these might be a better tool for upmixing. You'd also have to ensure the bass frequencies at least go to the front left and right speakers and/or the LFE. I know, I know, many experts don't believe in using the LFE channel, but others do. Any 5.1 receiver with bass management will direct the bass energy where it belongs anyway, so that point is moot.

Finally, as for "warning" people not to mess with the groups, why not? If the result is better, it makes sense to change the groups. After all the "improvement groups" in these bidules take Gerzon's method for spreading a stereo L and R signal among LCR and apply it to only the stereo difference signal, and route the signals to front and back speakers. That's "messing" with an expert's original intent. If you can do it, why can't we do it with yours?

Now if you choose to take this as a personal attack, I'm sorry but it is not intended that way. If, instead, you take it as it's intended--a critique of the results of your method as it has been presented so far--then let's discuss it. I've already said to Andreas that we should all be working together and contributing here as is the general intent of a forum. Instead of doing this mysterious "we can't reveal all our secrets" routine and telling people to private message you for more, why not share these secrets with the world in a spirit of cooperation?

Regards,
Steve.

[kempfand]

Steve,

No problem for me if the 3 new bidules don't match your expectation. In fact, I see this positive, as it will allow for improvement. Some people are quite happy with them, some are disappointed.

I imply that we all care for music and howto get the best out of it, and improving current methods is not easy or else ther wouldn't be a sometimes heated discussion on this :) or there would be a straight-forward solution.

The "original artist's intend" is of a big theme by itself, and highly disputed even amongst the best know names in academia and industry. We probably could open a separate threat on this, which would become hugegly complex due to all the variables involved (performance, recording, distribution media, reproduction at end users room and equipemt used, and so one). Adding to that is the change over time (20 years back: most performances aimed for live delivers, today: many performances aimed at CD-distribution). IMHO a very interesting topic, and one I've not come to a clear opinion yet.

Back to the bidules: As you correctly noted, we used Gerzon's method for spreading the signal not only to the front, but also to the rears. I would be very keen on your opinion for the 2->5 groups which I published (Gerzon 1997 Groups & Bidules (http://forum.doom9.org/showthread.php?s=&postid=508560#post508560) ). More concrete on 2 setups:
a) Using the "Improvement_5_x_2" Group, and feeding output pins L7/R7 to L/R, feeding L6/R6 to SL/SR, and C5 to C. Or
b) Using groups "Preservation_3_x_2" -> "Preservation_4_x_3" -> "Preservation_5_x_4", and feeding as with (a) the five output pins to the ASIO audio device.
Reason I'm asking is to get feedback on on this modified original intent (of Gerzon), and keep out other complications such as voice removal and mono_2_stereo alternatives.

Regards,
Andreas

[ursamtl]

Originally posted by kempfand
Steve,

No problem for me if the 3 new bidules don't match your expectation. In fact, I see this positive, as it will allow for improvement. Some people are quite happy with them, some are disappointed.

I imply that we all care for music and howto get the best out of it, and improving current methods is not easy or else ther wouldn't be a sometimes heated discussion on this :) or there would be a straight-forward solution.

The "original artist's intend" is of a big theme by itself, and highly disputed even amongst the best know names in academia and industry. We probably could open a separate threat on this, which would become hugegly complex due to all the variables involved (performance, recording, distribution media, reproduction at end users room and equipemt used, and so one). Adding to that is the change over time (20 years back: most performances aimed for live delivers, today: many performances aimed at CD-distribution). IMHO a very interesting topic, and one I've not come to a clear opinion yet.

Back to the bidules: As you correctly noted, we used Gerzon's method for spreading the signal not only to the front, but also to the rears. I would be very keen on your opinion for the 2->5 groups which I published (Gerzon 1997 Groups & Bidules (http://forum.doom9.org/showthread.php?s=&postid=508560#post508560) ). More concrete on 2 setups:
a) Using the "Improvement_5_x_2" Group, and feeding output pins L7/R7 to L/R, feeding L6/R6 to SL/SR, and C5 to C. Or
b) Using groups "Preservation_3_x_2" -> "Preservation_4_x_3" -> "Preservation_5_x_4", and feeding as with (a) the five output pins to the ASIO audio device.
Reason I'm asking is to get feedback on on this modified original intent (of Gerzon), and keep out other complications such as voice removal and mono_2_stereo alternatives.

Regards,
Andreas Good Andreas, that's exactly the spirit I'm talking about. By sharing info and techniques, bouncing them back and forth, getting feedback, etc., we all benefit.

I agree that the "original artist's intent" is a very complex subject. All I was getting at by mentioning that was that Pink Floyd's music by nature was quite detailed and meticulous, so if they mixed a door knock over to the left, it was probably intentional and not due to track limitations. By 1979, multitrack studio technology had advanced enough that it wasn't like old mid-60s recordings where the tracks were limited to 4 or 8 tracks.

I'd be glad to give the groups you mentioned a closer look. Unfortunately, the ftp server is down right now and I can't find them on the needfulthings server. I thought they were included in the original download with the 3 bidules but I can't find the actual group files anywhere. I've checked the bidules and I don't see pins L7/R7, etc. so, I'll give them a look this week.

I'm also working on another bidule that spreads out the front soundfield quite well but still preserves the imaging and seems complemented and reinforced by Ambisonic surrounds. I stumbled upon it by accident when playing around with some hookups. Once I finish testing and implementing some ideas, I'll write a guide and post it.

Have a good week!
Steve.

[Eye of Horus]

Originally posted by ursamtl
I'm sorry EoH but I disagree that the results will be "far better." I just tested this and I was still disappointed with the results.

Changing to the "Stage" setting didn't do anything for the percussive doubling problem I've mentioned. Both the voice-center and intrumental bidules still smear rhythmic or percussive sounds. For anyone who wants to hear this, do the following:

[list=1]
Take a song with some well-definied repetitive rhythmic sounds. For example, I just used Pink Floyd's "Comfortably Numb." If you don't have it, any song with well-defined drums (especially cymbals), strummed acoustic guitar, sharp rhythm guitar strums, slow harp plucks, etc.
Load in the Voice-Center bidule and load your song in the Audio File Player.
Replace the Audio File Recorder in bidule with Microsoft Sound Mapper output device. To do this Right-click the Audio File Player, and choose Replace > Audio Devices > Output > MME > Microsoft Sound Mapper.
Load your file into Audio File Player and listen carefully.
[/list=1]
What you hear is the sound that will be in the front speakers of your surrounds after you burn your surround CD with this bidule. Yes I know the two channels still aren't normalized as suggested in the method, but all normalizing will do is vary the volume level of the channel. It won't change the nature of the sounds in the channel. If there's a problem with the sound information contained in a channel, normalization will not erase it, but only change the volume level of the channel.

Back to Comfortably Numb. Listen to the door knocks and the guy saying "Time to go...." at the beginning. In the original stereo recording, both are panned to the left as the artists originally intended. Through the bidule, however, both come from the center. Ok so that's not the end of the world, this is surround sound, right? Things may get moved.

However, listen carefully to those door knocks. In the original recording they were so sharp and well defined, if you closed your eyes, you would almost jump up to go and answer your own door! Yet through the bidule, the realism is diminished. That's because the original audio waveform's transients have been mangled and delays echo the sound in both speakers. It still doesn't sound bad, but the sharp focus of the original recording is lost. Surround sound shouldn't do that.

Although I said not to reply anymore. I will, because there are again a few lines in your post I am not pleased with. And no, this has nothing to do with a personal attack......

How many times did we say you have to chose a song that suits the bidule ? If one song doesn't give you the result you want, you are immediately disappointed with the method ! You should criticize your choice of music ! What gives good results on one song, doesn't necessarily give good results on another !


Listen to the cymbals in the verses and the strummed acoustic guitar in the chorus. They no longer sound real and well-definied. Surround sound shouldn't do that.

What shouldn't surround sound do ?
Did you ever listen to DSOTM in DTS (the Alan Parsons mix from quad) ? or the latest on SACD ?
Did they intend this in their original stereo mix ???
Changing the nature of the sound is indeed not the intention but again : it also has to do with the choice of music. Some will do better on this bidule while other do better with SAD5.1inBidule or Ambisonics.


The tight locked-in bass and kick drum notes are also smeared. It's also harder to hear these because most of their sound energy is contained in the sum of the original stereo signals that is sent to the center channel.


We do have a solution for this ! see more below !


This brings up another point I've mentioned in recent posts. If you've got a monster center speaker with great bass and treble response, you'll be ok. Most 5.1 systems have a center speaker that's optimized for dialog. Bass response in these is rolled off. Since there's sound going to your LFE in this bidule, the only deep, solid bass you'll get is if your center speaker can produce it.


?? The LFE is empty, so this statement is simply not true ! or worded incorrectly !
yes, there is too much in the center, but NOTHING is steered to the LFE by the bidule ! The LFE is steered by your receiver only !


As for increasing the front center gain to make the vocals more balanced and less reverberated, to do that, you have to bring the center gain up to +10dB. This is if one has a monitoring system hooked up to listen to the adjustments. On most songs, this causes digital clipping and distortion. Yes, you're going to normalize the files afterwards, but once the distortion is introduced, it can't be removed.

Why do you think we added the gains AND the normalisation ??
In the bidule you need to find a good balance between L+R and C with the gains. And then normalize ! If you don't do that, yes, you can get some clipping !
So : increase the gain on the C, decrese the gain on the L+R and then normalize ! THIS WILL GET RID OF THE REVERB !


Look both EoH and Andreas, I've got nothing against either of you personally, but the more I explore these "exciting new bidules", the more disappointed I get. The logic of spreading the sounds out among the four speakers is good, but the execution mangles the original soundstage too much. If you can find a way to restore the soundstage and position of the instruments as the original artists intended, then these might be a better tool for upmixing. You'd also have to ensure the bass frequencies at least go to the front left and right speakers and/or the LFE. I know, I know, many experts don't believe in using the LFE channel, but others do. Any 5.1 receiver with bass management will direct the bass energy where it belongs anyway, so that point is moot.


I have no problem with your disappointment. Just go ahead and say it everytime here with everything we come up with.... I am more interested in the people that are satisfied ;)


Finally, as for "warning" people not to mess with the groups, why not? If the result is better, it makes sense to change the groups. After all the "improvement groups" in these bidules take Gerzon's method for spreading a stereo L and R signal among LCR and apply it to only the stereo difference signal, and route the signals to front and back speakers. That's "messing" with an expert's original intent. If you can do it, why can't we do it with yours?


Did I say you can't do it ? (another example of giving a bad explanation to my words > what we called twisting !) I just gave a warning ! There are also a lot of newbies here who are only interested in input a stereo and output surround. When they start playing with adjusting groups, no problem. But will you answer the emails when they get lost somewhere ?
Not all people have those skills ! And answering between 10 and 20 emails every day about the (in your and our eyes) most simple things, is enough for me. Don't need more !


Now if you choose to take this as a personal attack, I'm sorry but it is not intended that way. If, instead, you take it as it's intended--a critique of the results of your method as it has been presented so far--then let's discuss it. I've already said to Andreas that we should all be working together and contributing here as is the general intent of a forum. Instead of doing this mysterious "we can't reveal all our secrets" routine and telling people to private message you for more, why not share these secrets with the world in a spirit of cooperation?


Two points : I (I don't speak for Kempfand) don't feel any intend to share some things when the outcome is everytime a burning down of what we invented. Positive criticism on how to improve things : EXCELLENT. But this constant : I am so disappointed, doesn't invite me to share anything, let alone work together. That will need some changes first ! A good start for you would be : take a song and find the right bidule to convert it , instead of taking a song and take a random bidule and expect perfect sound. It's the combination that does the trick. Perhaps it will remove a lot of your disappointment. There is no "one method fits all" !! And I said that a lot of times.
I also don't want to sound negative only : try the bidule with for instance "Breakdown" from the Alan Parsons Project. A song with a singer and a huge choir with men and women. With the voice-center bidule the singer comes from the Center only and the choir falls in from all speakers (except the center) with even some separation between the male and female singers (front, rears). An overwhelming effect ! There is also almost no percussion in this song, so that problem is solved too ;)

Second : about the private message : we try to keep up with the spirit of this forum and don't publish links to or cracks for commercial software on the forum. But sometimes you find a program that does exactly what you need (for the voice-center bidule). If any people will give that a shot too, I won't publish it here in public ! But give the link in a PM ! (Is that so hard to understand ? Why that immediate accusation about "not sharing" ?)

Oh, also a third point : "why not share these secrets with the world in a spirit of cooperation" > personally I am insulted by this !
Our work is always shared ! This whole thread and previous ones is sharing of our work. A ridiculous statement from you IMHO !
As long as you make this kind of remarks, I will go back into silent mode again and find other ways to publish our work . (as we already did, BTW)
We shared everything and then someone comes up with such a remark ?
What we don't share is in the spirit of this forum !

You see, it's still not about what you say, but how you say it !
Every invention or improvement came from positive criticism and suggestions to improve things. Negative ones lead to nowhere !
And to me you're still sounding too negative for my taste.

So I wonder : if you cannot find positive things in what we produce : why the heck do you want to work with us ??


Regards,
Steve.

EoH

[jorel]

from EOH (great) explanations:
"What gives good results on one song, doesn't necessarily give good results on another !" this is the resume.
and i'm a witness of his words: ..."I am more interested in the people that are satisfied" cos he is posting "the road" to this results! thanks EOH and Andreas for this magical thread! ;) (EOH my friend, take it easy! :) )

[kempfand]

Originally posted by ursamtl I'd be glad to give the groups you mentioned a closer look. Unfortunately, the ftp server is down right now and I can't find them on the needfulthings server. I thought they were included in the original download with the 3 bidules but I can't find the actual group files anywhere. I've checked the bidules and I don't see pins L7/R7, etc. so, I'll give them a look this week. Daphy now put them to the Needful Things host (http://needfulthings.webhop.org/) (\latest-layouts-temp\gerzon_1997_bidule.rar).

No need to hurry with feedback, [off-topic mode on] as things are pretty busy here with the European championship :) Adding only 2 more teams would make it a real world-championship: Argentina and Brazil :cool: [off-topic mode off]

Andreas

[Eye of Horus]

Originally posted by jorel
from EOH (great) explanations:
"What gives good results on one song, doesn't necessarily give good results on another !" this is the resume.
and i'm a witness of his words: ..."I am more interested in the people that are satisfied" cos he is posting "the road" to this results! thanks EOH and Andreas for this magical thread! ;) (EOH my friend, take it easy! :) )

Hi Jorel,

Thanks for the kind words !!

Yes, I take it easy ;)

BTW I send you a PM !

EoH

[ursamtl]

Originally posted by Eye of Horus
Although I said not to reply anymore. I will, because there are again a few lines in your post I am not pleased with. And no, this has nothing to do with a personal attack......

I don’t see the point in quoting your messages back in my response because you’ll only accuse me of twisting your words.

Ironically, I could accuse you of something similar when you stated that I chose one song and then immediately expressed disappointment. I never said or implied that I used just one song to test your bidules. I gave one example with Pink Floyd’s Comfortably Numb because it provides both percussive and non-percussive sound effects at the beginning plus both percussive and non-percussive music throughout. For testing surround bidules, I actually use about 20 musical recordings in various styles.

I will make one correction to my previous message to you. I apologize because I forgot to type the word “no” in the following: “Since there's no sound going to your LFE in this bidule, the only deep, solid bass you'll get is if your center speaker can produce it.” However, you completely ignored the essential point that all deep bass energy is going to the center speaker instead of the left and rights. Since most center speakers roll off the lows, either someone is going to need a good subwoofer and bass management on the amp to redirect the sound, or else he or she is out of luck! Even with these, the separation of the bass frequencies in the front speakers will be compromised because the center speaker is by nature mono.

Anyway, I could go on, but I don’t see the point--my other posts have adequately expressed my technical answers on these issues for anyone who reads them with attention.

Before I joined this forum, I spent some time reading the threads and I’m sorry to say I developed a negative impression of you and especially how you handle criticism. This is one reason I avoided joining for awhile. I thought perhaps that would change once I got into discussing with you the surround technology covered here. However, every comment I made, you seemed to take as a personal attack of you. This has degenerated to the point where I have zero respect for anything you have to say, and I get the impression the feeling is likewise.

Therefore, if I have comments or questions concerning your bidules, etc., I will address them to Andreas. If I see something you write about the bidules and I disagree with it, I’ll ask him about it. I will continue to contribute to this thread because it is still presented by Doom9 at a public Sticky thread on ways to upmix from stereo to surround and not just your personal thread. I hope Doom9 eventually reorganizes the threads so that everyone who creates an upmix guide in a separate thread is given equal treatment (surroundboy, kpex, bleo, et al) because it seems unfair to newbies coming along that these other methods—all every bit as valid in their own right as yours—scroll off the front pages of the forum. These people are also posting 'roads' to results and contributing to the community.

For now, however, it seems that this thread has evolved into a general forum for discussion of upmixing to surround. Although you may have started it--and I do congratulate you for doing so--it is now a forum for discussion of various methods by various people. As such, it is a valuable resource and everyone who contributes to it deserves praise, just as the people I mentioned above and others deserve praise for taking the time to contribute their ideas.

[ursamtl]

Originally posted by kempfand
Daphy now put them to the Needful Things host (http://needfulthings.webhop.org/) (\latest-layouts-temp\gerzon_1997_bidule.rar).

No need to hurry with feedback, [off-topic mode on] as things are pretty busy here with the European championship :) Adding only 2 more teams would make it a real world-championship: Argentina and Brazil :cool: [off-topic mode off]

Andreas

Ok, well I can give you my initial impressions of one bidule, the Preservation_5_Speaker_v2_REC_.bidule. Personally, I think this is the best, most realistic sounding upmixing bidule I've heard since I discovered this forum! It is light years ahead of both the 3 bidules you posted recently, the ambisonics one that starts this thread, etc. I still have to test bleo's DPLII upmix but for sheer simplicity and results, this definitely rocks! I'm also surprised in that its fronts sound almost exactly like the new bidule I was working on. They both tend to spread the stereo soundfield out beyond the front speakers without leaving a hole. My only negative comment would be that the surrounds are a little strong for my taste. Personally I don't like surrounds that present part of the music information directionally. Neither do I want them to be simple reverb of the front channels, but I like them to create a realistic sound space. I added a gain object to the rears of Prev5v2 and at about -4dB it definitely produces nice sounding surround without the cheap, gimicky sound. To be honest, those last three bidules reminded me of a cheap Radio Shack analog delay box I had for my guitar 20 years ago! This new one sounds like something out of a recording studio! In addition, for those with slower PCs, this might be a great solution since it runs at about only 5% CPU. There's some work to do on the surrounds but it's a minor detail.

So, although I have some testing to do with the others and exploration of the guts, I'd say this is a winner. I ran about a dozen songs through it quickly on the PC ASIO out. I've been doing this so much now that I'm able to anticipate how things will sound through the big 5.1 system and I have a good feeling about this.

Regards,
Steve.

[Eye of Horus]

[QUOTE]Originally posted by ursamtl
I don’t see the point in quoting your messages back in my response because you’ll only accuse me of twisting your words.
{/quote]


Of course, why should you react to what I say ??
No, you don't, you only post another tirade about how bad I am.
Here in Holland we call someone like you a guy with an incredible plate before his head !

This was a nice forum......

EoH

[DSP8000]

@ ursamtl,

I can't find Preservation_5_Speaker_v2_REC_.bidule on the needfull things server, can you give me link for it.
Personally, I follow this thread every day and to be honest there is a lot of improvement since the original CoolEdit-Aurora method.
However there is some "fuss" in the air that I don't like and turns this thread in "unneeded" discussion.

Keep up the good work ursamtl,andreas....

DSP8000

[kempfand]

@ Steve: Thanks for your feedback (BTW having the same experience here that running things through the ASIO gives a good anticipation on how it will come out on the big 5.1 system). I also was impressed when I first listened to the Groups and bidules using the Gerzon 1997 approach, and my mental note was a simple statement saying: "Prooven Gerzon Technology". Changing/adding a gain element for the Rears (as you did), or even a gentle reverb (using one instance of SIR) gives so much space for creativity and personal preference.


Originally posted by DSP8000 I can't find Preservation_5_Speaker_v2_REC_.bidule on the needfull things server, can you give me link for it.It's on the host Needful Things host (http://needfulthings.webhop.org/) in \latest-layouts-temp\gerzon_1997_bidule.rar.

When you use them, just make sure you feed the output-pins the right way (the one in the center goes to the C-speaker, the ones at the outmost left & right will go to the Surround, and the ones in the 'middle' go to the front L & R). I've added bidules in the RAR for illustrative purpose.


Originally posted by DSP8000 Personally, I follow this thread every day and to be honest there is a lot of improvement since the original CoolEdit-Aurora method.I think I know what you mean by improvement (i.e. easy of use, speed). I personally prefer to call it variation, as we have so much more choices today than 2 years back. The CoolEdit (Ambisonics) method has it's merits, and so do all the other methods described here and in the Audio section of this forum.

But I also would like to point out that EoH started it all with the original CoolEdit threat here (the Ambisonics method), and that his initial threat was my personal starting point for what I call a wonderful hobby that I wouldn't want to miss.

Regards,

Andreas

[DSP8000]

Thanks Andreas, found it.
Exactly, easy of use, speed...
The original idea it is probably the same(you guys should know), but there is always room for improvement.
With instr.music maybe it's better to experiment, as voice based tracks are a bit different to handle.
All I'm aiming to achieve is complete separation of diff. channels,frequencies,to preserve the original recording sound across 5.1 encoded file without using too much reverberation.
My Creative ZS Pro does nice job with the DTS NEO update driver but, for my taste the rears sounded "plastic"

Thanks again & enjoy the sounds

DSP8000

[ursamtl]

Originally posted by DSP8000
@ ursamtl,

I can't find Preservation_5_Speaker_v2_REC_.bidule on the needfull things server, can you give me link for it.
Personally, I follow this thread every day and to be honest there is a lot of improvement since the original CoolEdit-Aurora method.
However there is some "fuss" in the air that I don't like and turns this thread in "unneeded" discussion.

Keep up the good work ursamtl,andreas....

DSP8000
Andreas has given you the link, so enjoy the bidules! Thanks for the encouragement DSP. Yes, the thread was getting bogged down with unnecessary discussions, but hopefully things will improve now. You're right that there has been a lot of improvement in the bidules. I think you'll find that the the Gerzon 1997 bidules Andreas posted are quite good, at least the one I tried was. If you find the rears a little too loud, be sure to add a gain object and turn it down slightly to balance them. Then if you listen to the fronts you'll hear lots of detail.

Another important point that's been made here several times is to convert the input wave files to 32-bit floating point format before the input. The difference may be subtle, but details on high frequencies, the overall sense of transparency in the music, etc., is improved. If you don't have a commercial program to do this, I've found an excellent freeware DOS utility that converts a file quickly. It's available from the Telecommunications & Signal Processing Laboratory at McGill University here in Montreal and is called CopyAudio. You can download it at ftp://ftp.tsp.ece.mcgill.ca/TSP/AFsp/ and read more about the whole package at http://www.tsp.ece.mcgill.ca/MMSP/Documents/Software/Packages/AFsp/AFsp.html.
The actualy CopyAudio.exe is in the MSVC\bin folder of the extracted files. Simply add -D "float32" plus the input and output filenames to the command line and it converts your file very rapidly. It doesn't provide any onscreen feedback while working. Just wait until the command prompt returns and you'll find your finished file is converted.

There are some other interesting utilities in the AFsp package for comparing audio files, resampling, processing multichannel files, etc. Too bad these aren't VST plugins!

By the way, I noticed you're using a ZS Pro. How do you like it? I'm planning to upgrade my old Live 5.1 later this year and the ZS Pro looks like a good target. Have you tried the kX Project drivers on it? Apparently, they work fine on the ZS Pro and offer improved performance when compared to the Creative drivers. You can check the forums at www.kxproject.com for more. One other question, are you able in Plogue to add a multichannel ASIO out using the Creative drivers? Since Creative never supported ASIO in their official Live drivers, I've completely ditched anything from Creative and use kX exclusively. (Sorry to everyone else for bringing this questions into this thread, but since DSP doesn't accept PMs I have no choice :)

Regards,
Steve.

[ursamtl]

Originally posted by kempfand
@ Steve: Thanks for your feedback (BTW having the same experience here that running things through the ASIO gives a good anticipation on how it will come out on the big 5.1 system). I also was impressed when I first listened to the Groups and bidules using the Gerzon 1997 approach, and my mental note was a simple statement saying: "Prooven Gerzon Technology". Changing/adding a gain element for the Rears (as you did), or even a gentle reverb (using one instance of SIR) gives so much space for creativity and personal preference.Yes, actually I tried a quick run with the free Classic Reverb VST on the rears (find out more about it at http://www.kvr-vst.com/get/717.html). With its "Drum Rooms" preset at about 50%wet-dry, the whole surround mix came alive without actually sounding like added reverb. With the right impulse files SIR can be great too. I just like Classic for quick simple stuff.

One thing I did like about the 3 previous bidules you posted was that you added dithering to the outputs. I never felt we resolved this issue when we discussed it before. I know Surcode (for example) accepts 32-bit input files, but I was never able to find out whether it 1. dithered to 16-bit output, 2. truncated to 16-bit output, or 3. left the output at 32 bits. I seem to recall reading something that led me to believe that the digital DTS wave file is seen by a DVD player as a 16-bit file. I know from experience that the input file has to be at 44.1K for the DVD player to recognize the DTS wave file as a music track, so I assume it recognizes it as 16 bit as well. Therefore, I think dithering the outputs makes sense as one is sure that proper dithering will be done instead of hoping it's done by Surcode or whatever AC3 encoder one uses. What are your thoughts on this?

Regards,
Steve.

[DSP8000]

@ursamtl,

From the kxProject FAQ
"Audigy2 ZS series aren't fully supported at the moment, due to the fact that we do not have an Audigy2 ZS card for testing. However, as soon as we have enough money to buy one, support will be implemented."
I'm happy with the Creative drivers and my PC setup so there's no need to change anything, maybe if they have enough good features in their drivers i'll give it a go.

About CopyAudio, I'll definetely have a look(on the weekend) at the moment I'm using Audition.

Yes, Creative ZS PRO is my choice, I used to have M-Audio Audiophile 24/96 and that is another very good sound card, but not suitable for my needs.
Actualy this is one of the very first semi-pro multimedia/multichannel based cards that performs a bit better at 44.1 kHz. Native 48 kHz is excellent!!! All in one it is very good package.

I do accept PM's just my bigpond email is bugged so I have to use this one ivancosax@hotmail.com ,geez you can tell that I play sax from the email.

Regards,
Ivan

[davidv@plogue]

Hello all

Just a quick note, all audio processing inside bidule is done in 32 bit floats. So your 16/24 PCM files get converted into 32 bits before leaving the "AudioFile Player" bidule.
(there might be better algorithms to go from 16->32, but we use - like many others - the fill in the significant bits, leaving low bits at zero) For tech savy people, you could have a look at the libsndfile sources (LGPL).

Oh course, if you feed files allready in 32 bit float, we pass its data directly to the AudioFile player's outputs without processing.

About that Audigy/live multiple I/O.
Ive got some of those cards and never managed to ouput multiple channels
with them, whatever the settings or kx/official drivers. If anyone knows
how, please tell me :)

Im presonally using a Terratec EWS88MT (8in/8out) card with great ASIO drivers that do buffer size mathching bidule's 512,256,128 and even 64.
So no complaints here.

Continue on the great group making (ive tried one or two)!!
;)

Cheers

[ursamtl]

Originally posted by DSP8000
@ursamtl,

From the kxProject FAQ
"Audigy2 ZS series aren't fully supported at the moment, due to the fact that we do not have an Audigy2 ZS card for testing. However, as soon as we have enough money to buy one, support will be implemented."
I'm happy with the Creative drivers and my PC setup so there's no need to change anything, maybe if they have enough good features in their drivers i'll give it a go.

Yes, I had read that too, but then I asked other users on their forum. Apparently, the drivers work fine. The only thing that's not yet implemented is 24/96. There is talk that this may be implemented in the next version of kX due out any day. The biggest thing I like about the kX drivers (other than the fact it gives me 5ms-latency ASIO on an SB Live) is that their DSP applet is very similar to Plogue Bidule in that one can create custom routings, hook-ups, etc., graphically and use a variety of plug-ins for surround, effects, etc., all accessing the card's DSP. With the right ASIO routings, you can also record these into a file using bidule. Their surrounder plugin is actually another interesting option for upmixing in that it provides a nice surround effect using the card's DSP (which is an EMU chip). Unfortunately, I haven't been able to record to bidule because of a bug in my particular model of Live (it's only a cheap OEM version) that restricts the number of ASIO recording channels . this does not occur on the Audigy1/2 or other Lives. I have been able to record from its output in Cubase on all six channels and it sounded great. It just wasn't as practical as in bidule, where one can link the playback and recording of file player and recorder objects.

Anyway, as I've said on the kX forum, the combination of Plogue bidule and the kX drivers is absolute heaven if someone likes to play around with this stuff.


Ciao
Steve.

[DSP8000]

"Their surrounder plugin is actually another interesting option for upmixing in that it provides a nice surround effect using the card's DSP (which is an EMU chip)"

Hmmm, interesting...

[ursamtl]

Originally posted by davidv@plogue
About that Audigy/live multiple I/O.
Ive got some of those cards and never managed to ouput multiple channels
with them, whatever the settings or kx/official drivers. If anyone knows
how, please tell me :)

Hi David,

First thanks for a wonderful program! Bidule is truly amazing in its flexibility.

As for the Audigy/live multiple I/O thing, I do it all the time without problems. I did notice that you have to have the sample rate set the same in both bidule and the kX ASIO Control Panel. Plus, at 48k, there is one kX duplex I/O Audio Device. Since the kX driver does not support recording at 44.1k, there is only one kX ASIO output device. If you look back through the forum, I posted a pic of this. Of course, you need to have the proper routings set up in the kX DSP applet as well. I'm at work now, but once I'm home, I'll PM you with routing details.

To sum it up, multichannel ASIO output using an Audigy/Live card is possible. You can even output via SPDIF to the digital inputs of a surround receiver.

By the way, [semi-offtopic mode on] have you ever thought of implementing DirectX plugin support in bidule? I do use a free rack from SpinAudio that seems to do this well, I'm just curious about this. There are some really good DX effects out there.[semi-offtopic mode off]

Regards,
Steve.

[kempfand]

Originally posted by ursamtl Another important point that's been made here several times is to convert the input wave files to 32-bit floating point format before the input
...
If you don't have a commercial program to do this, I've found an excellent freeware DOS utility that converts a file quickly. It's available from the Telecommunications & Signal Processing Laboratory at McGill University here in Montreal and is called CopyAudio. Just adding here another option: Voxengo's r8brain (http://www.voxengo.com/r8brain/)

According to Aleksey (the author of r8brain), the tools uses sinc interpolation algorithms. Quality setting selects the number of control points (there can be up to 1000 control points, on Very High quality setting).

http://www.voxengo.com/r8brain/screenshot.jpg

with excellent features:- Freeware
- Reads PCM WAV files of bit depths 8, 16, 24, 32 and 64
- 32 and 64 bit files can be in IEEE floating sample format
- Mono and stereo file support
- Virtually any input and output sample rate
- Batch processing
- Writes 8, 16 and 24 bit PCM WAV files and 32 bit IEEE WAV files
- Reusable royalty-free converter DLL with explained API Think EoH mentioned this nice toy in the first place somewhere in the Forum here, and I've also very good statements on it (in terms of quality) from audio pros.


Originally posted by ursamtl ...
I think dithering the outputs makes sense as one is sure that proper dithering will be done instead of hoping it's done by Surcode or whatever AC3 encoder one uses. What are your thoughts on this?

Nearly all statements from audio pros I have been reading so for strongly recommend dithering, and therefore we indtroduced it with the bidules. I also rememebr having read something about what you mentione for SurCode (dithering, truncation, or leaving things at 32 bit), but I was not able to find the link again.
Some tests done by EoH and myself have lead us to believe that feeding 32 bit into SurCode gives slightly better results as compared to feeding (mda-) dithered 16 bit into SurCode. This still leaves two options: a) SurCode doing a decent dithering on 32 bit input, or b) mda_dither doing a bad job. Unfortunately, there are only a couple of other freeware dithering VST's, so limited choice. Finally, there also is ssrc.exe, which does an excellent job but only on 24 bit (no 32 bit input). In summary: work in progress.

Regards,

Andreas

[ursamtl]

Originally posted by kempfand
Just adding here another option: Voxengo's r8brain (http://www.voxengo.com/r8brain/)

According to Aleksey (the author of r8brain), the tools uses sinc interpolation algorithms. Quality setting selects the number of control points (there can be up to 1000 control points, on Very High quality setting).

Yes, I've tried it. The only problem I have is that if I just want to change a file from 16- to 32-bit without changing the sample rate, it won't work. It seems to require a change to 48 if the input is in 44.1k. If I'm doing a surround CD, I don't want to change to 48 then back to 44.1.

Re the dither, I haven't had time to investigate the mda dither settings. Are they explained somewhere? Perhaps this will change the results.

Steve.

[davidv@plogue]

Originally posted by ursamtl
Hi David,
First thanks for a wonderful program! Bidule is truly amazing in its flexibility.


Hey thanx!

Originally posted by ursamtl
Hi David,
If you look back through the forum, I posted a pic of this. Of course, you need to have the proper routings set up in the kX DSP applet as well. I'm at work now, but once I'm home, I'll PM you with routing details.


Yes the routing details of the kx mixer is what is most intersting.
Idd post them permanently on plogue.com

Originally posted by ursamtl
Hi David,
By the way, [semi-offtopic mode on] have you ever thought of implementing DirectX plugin support in bidule? I do use a free rack from SpinAudio that seems to do this well, I'm just curious about this. There are some really good DX effects out there.[semi-offtopic mode off]

The thought has crossed our minds more than once.
But really its a matter of priorities. The amount of work required (especially for only one platform)
vs the amount of plugins that are DX only. - We had to implement AU plugins on OSX because many plugins went out AU only - where as on PC, a company shoots itself in the feet if it only does DX/DXis.

And as you say theres the SpinAudio trick, even though its not the most stable solution, from my experience.

[ursamtl]

Originally posted by kempfand
Finally, there also is ssrc.exe, which does an excellent job but only on 24 bit (no 32 bit input).Actually I recall reading somewhere that reducing from 32 to 24 bits doesn't really need dithering because the difference would be inaudible, whereas going from 24 to 16 bits definitely benefits from dithering. Therefore converting to 24 bits first might be an option.

Anyway, I'm glad my bringing up dithering in the forum prompted you to add the mda plug-in as it brings the dithering issue back up. We might not have found out from David that we may not really need to convert to 32-bit files before loading a file into a bidule. That's definitely a time-saver. The McGill audio utilities package I mentioned above has a file analyser tool with it so perhaps I'll see if this somehow might compare different 32-bit files in a testing environment. I know there is definitely a quality difference when I listen to 32-bit files outside of bidule, but that could be related to issues with my Soundblaster Live.

Steve.

[Eye of Horus]

Originally posted by davidv@plogue
<snip>
Im presonally using a Terratec EWS88MT (8in/8out) card with great ASIO drivers that do buffer size mathching bidule's 512,256,128 and even 64.
So no complaints here.

Continue on the great group making (ive tried one or two)!!
;)

Cheers

Hi David !

I see you use the same soundcard as I do, so perhaps you can help me with a problem I have with it in combination with Bidule.
In the control panel I select ASIO.
I connect the output in Bidule, but whatever I try, I cannot get any output.
Using the SPDIF or the MS soundmapper gives output on 2 channels.

Can you advice me or show your settings ?

TIA !!

kind regards,

EoH

[davidv@plogue]

Originally posted by Eye of Horus

Using the SPDIF or the MS soundmapper gives output on 2 channels.
Can you advice me or show your settings ?

EoH

Thats odd, ive used the Terratec multichannel ASIO driver for 3 years
whitout a hitch. But ive never tried the SPDIF I/O ever.
On the 8 analog chans its flawless.

I use 256 buffer sizes in the terratec control pannel. (same thing in bidule) so im assured of perfect timings.
Ive also tried all combinations of samplerates in bidule (from 8 to 96Khz) without probs too.
Does your 4x2 MME outputs work?

[kempfand]

Originally posted by ursamtl
Anyway, I'm glad my bringing up dithering in the forum prompted you to add the mda plug-in as it brings the dithering issue back up. As a matter of fact, the discussion if and how to dither already started before it was re-vived here:

- BeSweet/BeSplit : 24 bit Question (http://forum.doom9.org/showthread.php?s=&threadid=76679&)

- 24 bit (http://forum.doom9.org/showthread.php?s=&threadid=74315)

and was actually triggered for me by reading advice a couple of years back:

- Best way to downsample ? (http://forum.doom9.org/showthread.php?s=&threadid=40157)


Originally posted by ursamtl
We might not have found out from David that we may not really need to convert to 32-bit files before loading a file into a bidule. That's definitely a time-saver. EoH and myslef did a few listening test (not published): (a) feeding 16 bit, and (b) feeding upconverted 32 bit (CoolEdit method) in a bidule, then saving both as 32 bit 6-ch wav, demuxing, and SurCoding, and we felt there was a slight audible difference in quality. This seems to confirm David's remark that there are different algorithms.

I also remember some technical elaboration from Farina on why to carefully convert 16 bit source to 32 bit, but I frankly have to dig to find the reference (or re-type from my notes).


Originally posted by ursamtl
The only problem I have is that if I just want to change a file from 16- to 32-bit without changing the sample rate, it won't work. It seems to require a change to 48 if the input is in 44.1k. Good catch ! I just tried the tool again and can confirm this.

Andreas

[ursamtl]

Originally posted by davidv@plogue
Yes the routing details of the kx mixer is what is most intersting.
Idd post them permanently on plogue.comHi David,

Rather PM you as I offered this afternoon, I decided to post this in the forum. I know from messages I've received that there are others who follow the forum and are interested in kX, so rather than repeat, I'll just post the basics here. For those with no interest in this, go to the next message. :)

[Ursa puts on his historian hat for the benefit of the curious]First, let me say that there are some really good guides on the www.kxproject.com web site and in their forums. You can also read there about how these drivers came about as free alternatives to the official Creative ones. The latter have tended to be optimized and focused on gamers and lately DVD playback, whereas the kX drivers were designed for serious music recording and playback.

Ok, so enough history. To put it simply, if you have a Live or Audigy soundcard, kX rocks! [Hat removed]:)

David, I don't know how much you played around with the kX DSP applet (small button at the bottom left of the Mixer or right click on the kX icon in the system tray). It allows hookups something like bidule, although not quite as developed (you can't click on "wires" to delete or move them, only right click on connection ends and choose from the menu).
[list=1]
The default kX routing as installed tries to do too much so the best thing to do is right click on the DSP background and choose "Clear DSP." (If you want to go back to the default layout, you can always choose Reinitialize DSP).

You're now going to add some objects and connect them. To identify the pins, just move your mouse over them to see the pin's label.

Right click and choose Add Effect / Plugin > Default > FXBus. It's best to place this in the left left area of the DSP applet window.

An FXBus represents input signals from inside the PC. Most importantly for use with bidule, FXBus represents the 16 ASIO channels you get as output from bidule (these channels are numbered 0-15). For example, in the following pic, the second output pin of Preservation 5x3 is connected to the fifth kX ASIO (out) input pin, which corresponds to ASIO channel 4. In the kX DSP, the same signal will appear on the fifth output pin from the top of the object, which is FXBus4.
http://stevethomson.ca/audio/guides/kxASIOOut.gif

Right click and choose Add Effect / Plugin > Default > prolog. Place this under the FXBus on the left part of the window. A prolog object represents input signals from outside the PC, such as AC97 for signals coming from the soundcard's line in or microphone jacks.

Right click and choose Add Effect / Plugin > Default > epilog. Place this to the right of the other two objects. The top 14 pins on an epilog object represent the output signals from the card, both analog and digital. The fifteenth and sixteenth pins represent the left and right recording inputs. The remaining pins represent the 16 ASIO input channels (for recording at 48k sample rates).

Finally, between FXBus and epilog, right click and choose Add Effect / Plugin > Basic > Stereo Mix. This will provide two stereo input paths for both the default 2-channel signals from Windows apps and the front left and right of the 5.1 mix.

Important! Be sure to double-lick the Stereo Mix object and move both sliders all the way to the right!

Connect the two output pins on the right of the Stereo Mix object to the top two pins on epilog.

Connect FXBus0 & 1 to the top two left pins on Stereo Mix. This is your standard Windows stereo feed.

The default FXBus outputs for 5.1 are FXBus4-9. Although you can reconfigure to whatever order you want, we'll go with these. Connect them as in the diagram below to:

epilog out0 - Analog Front Left
epilog out1 - Analog Front Left
epilog out2 -
epilog out3 -
epilog out4 -
epilog out5 -
epilog out6 -
epilog out7 -
epilog out8 - Analog Rear LEft
epilog out9 - Analog Rear Right
epilog out10 - Analog Center
epilog out11 - Analog Subwoofer
(these last two may display different numbers when you mouse over them but the connection will be clearly identified).

Note that the following pic is just a portion of the screen to save download bandwidth time.
http://stevethomson.ca/audio/guides/kXDSP.gif

If you plan to use line-in or a microphone, connect the top two pins on prlog to the 15th and 16th pins on epilog (RecL and RecR).

Now click the diskette icon on the right of the window and save your configuration.[/list=1]

As for the kX Mixer, you can use play around with the levels there to see what they do.

Basically, this should work. It's not exactly the way I have my system hooked up because I route my subwoofer and center to both fronts with -3dB on each to balance their volume. I've also redone my FXBus internal routings (right click and choose Tweak) so that my ASIO outs are the first six channels. This way, in bidule I can easily right-click and replace the kX ASIO out device with a 6-channel Audio Recorder (or vice versa) and not have to reconnect anything.

Note that this applies to analog connections only. I haven't even begun to explore the digital outs since my 5.1 receiver is in a room not close to my PC.

I hope this helps. Sorry for the length of the message, but this is such a useful tool for someone with a Live or Audigy who wants to get the most out of any of these upmixing bidules, that I thought it useful to share in this thread.

Regards,
Steve.

Moderator Edit: Fixed image links as per ursamtl's request

[ursamtl]

Originally posted by kempfand
As a matter of fact, the discussion if and how to dither already started before it was re-vived here:

- BeSweet/BeSplit : 24 bit Question (http://forum.doom9.org/showthread.php?s=&threadid=76679&)

- 24 bit (http://forum.doom9.org/showthread.php?s=&threadid=74315)

and was actually triggered for me by reading advice a couple of years back:

- Best way to downsample ? (http://forum.doom9.org/showthread.php?s=&threadid=40157)
Andreas, before I wrote my above statement, I searched the forum for "dither" and confirmed that I was the first in this thread to bring it up, in terms of discussing the need for it when converting back to 16 bits. I wasn't at all looking for "brownie points" or for attention, but in fact I was trying to subtly make a point. You wrote:

Nearly all statements from audio pros I have been reading so for strongly recommend dithering, and therefore we indtroduced it with the bidules.
This implies that you got the idea from reading and then directly decided to implement it. There's no mention that we already discussed this in the forum after I brought it up. Again, I don't want a pat on the back, I'd be writing this right now just the same way if someone else had brought it up, but I do see this as a pattern. Whenever there's a new bidule, instead of just saying here it is, there's a preambule about all the "hard work" that's gone into it. What about all the others who develop guides and bidules? Daphy, for example, makes everyone feel welcome and equally important by immediately putting their work on his forum. His enthusiasm for the joy of the technology is evident. None of them seem to need to talk about all their hard work. What really shocked me was when I clicked on the link in the the following post to this very thread:

http://forum.doom9.org/showthread.php?s=&postid=509574#post509574

and then saw a guide for SAD5.1 bidule that states: "The hard work and hours of testing and trying is a co-operation of Kempfand and Eye of Horus." I searched the guide for mention of the guy who came up with the original concept behind SAD5.1, Kpex, but I didn't see any mention of him whatsoever! Even in the version of the guide posted on EoH's web site, Kpex is mentioned simply as one of several contributers. I'm sorry but this really doesn't seem fair. Yes, I know you might have "improved" on his method as stated in the thread by doing it in a bidule and converting to 32-bits, but at least give the guy credit for having come up with the initial concept!

Anyway, I'm not trying to make a major issue out of this, I just think it's about time someone pointed it out. I'm sure you don't intend to make others feel slighted. To be fair, Andreas, you have gone out of your way a couple of times to praise everyone who contributes, but these little statements counteract that. Yes EoH started this thread. Yes, as I wrote last night, he is to be congratuled for having done so. However, the thread has evolved into a mini-forum on upmixing and as such everyone who contributes should be recognized for their "hard work" and when someone's idea is used, it's nice to at least make a gesture acknowledging that fact. It belongs to all of us now, so I think if we can get rid of all this unpleasantness and implied hierachy of importance, things would go better. So let's share ideas with a little more respect.

Regards,
Steve.

[Umma]

It belongs to all of us now, so I think if we can get rid of all this unpleasantness and implied hierachy of importance, things would go better. So let's share ideas with a little more respect.

Dude, YOU are the one that I've seen the most rudeness and disrespect from. Despite your disclaimers that you mean no offense YOU come across as the bullying Type-A personality. EoH and Andreas's style of presentation and support have never struck me negatively as you are so quick to try and paint. To me, YOU come across, and have been coming across vaingloriously as a petulant bully in a thread of exploration and discovery that has basically been going on for well over a year, now.

If you could just present your findings/experiments/experiences without going off on a soliloquy of pontification this thread will retain its focus and stay on path. Your complaints about them sound to me as if you're describing yourself.

Like it or not, too many chiefs make for many confused indians and EoH and Kempfand have done a superb job of keeping this thread that they began on task.

This thread does NOT belong "to all of us now." Readers and experimenters are priviledged to have had these two share their mindgardens with the world here at Doom9, and they are beyond reproach in their attitude, graciousness, inventiveness, and outright friendliness.

Your discussion of techniques are interesting, but to paraphrase Bill Cosby, a steak dinner served on the lid of a garbage pail makes a wonderful meal very unappetizing.

[Eye of Horus]

When Kempfand and me were busy with the last 3 bidules, we discussed the need to keep the outcoming 6 channel file below the 2 GB border, to avoid too many problems with people who use FAT32 as filesystem.
We were also discussing 16 bit vs 24 bit and the need to normalize afterwards every channel .
We came to the conclusion that the outcoming file in 16 bits would be the most practical and introduced MDA's dither into the biduile.
We never discussed it with an eye on the past here at the forum.
The only reason was what I wrote above. It was absolutely not a matter of reading here and then use the advice to dither. We never discussed that !! [PERIOD !]


For those not technical I will also try to explain the usage of our latest voice-center bidule and what you can expect and it's shortcomings.

The bidule was made with the intention to use a voice only channel in the Center speaker. When you use music with effects that float from left to right (Pink Floyd, Tangerine Dream, Vangelis, Jean Michel Jarre, Kitaro, Alan Parsons Project to name a few), don't use this bidule. Normally the sound that floats, has a higher frequency than the voice of the singer. With the bidule, it tries to float from left to right, but it will not go through the Center. In the Center it's filtered out and thus you get a strange effect. Sonething like a hole in the middle !
The ideal music for this bidule is simpler music with no floating effects form left to right, but nevertheless with a good stereo effect and a singer in the middle. When you play the stereo and would disconnect the left speaker, you shouldn't hear anything from the left speaker except the voice, in the right speaker and vice versa.
Music that suits this is soft rock or a singer with only a few instruments and not too much percussion. On this kind of music the bidule can give excellent results. (Beatles, Country, Eagles...that kind of music).

When you chose a PF song to try it, you will get disappointed.
When you use the right song for the right bidule, you will get satisfying results. Perhaps with some finetuning of the settings of the Voxengo Stereo Touch and the gains, you can even improve.
When you don't get the result you want, drop us a line and perhaps we can advice on getting better results or............ use another bidule for the song you were trying ! The combination of the right song and the right bidule is what does the trick !!

Despite all negativism dropped here, I recall my original statement : Don't go by what others think or by theory, use your own ears. They are the best judges !
Try it and if you don't like the result, try it with another song !
We didn't work for more than 2 months on these bidules to publish something that doesn't work or give bad results only !
Kempfand and I work together for more than a year now and almost daily , we have new ideas and discuss them. Yes, sometimes we come up with an idea anyone else did too before. That's inevitable !
But ... we always make sure the endproduct that we publish here, is good. I think we build up a positive name the last years here, and we try to keep up with that by publishing only the good "inventions" and not the bad ones.
(as a matter of fact : only 3 out of 20 or so, finally made it !)

That someone is constantly trying to discredit our efforts, says more about that person than about us.

kind regards,

EoH

[Eye of Horus]

Originally posted by ursamtl
so I think if we can get rid of all this unpleasantness and implied hierachy of importance, things would go better. So let's share ideas with a little more respect.

Regards,
Steve.

When do you start ??

EoH

[Shayne]

Originally posted by davidv@plogue
Hello all

About that Audigy/live multiple I/O.

Ive got some of those cards and never managed to ouput multiple channels with them, whatever the settings or kx/official drivers. If anyone knowshow, please tell me :)

Continue on the great group making (ive tried one or two)!!
;)

Cheers
I too have never found sound blaster worth their money. Expensive for absolutely no reason but to give you headaches. I currently use the cheap Hercules fortissimo 7.1 card and pass thru is smooth and easy here.

Thanks for the proggy.

Peace

[ursamtl]

Originally posted by Umma
Like it or not, too many chiefs make for many confused indians and EoH and Kempfand have done a superb job of keeping this thread that they began on task.

This thread does NOT belong "to all of us now." Readers and experimenters are priviledged to have had these two share their mindgardens with the world here at Doom9, and they are beyond reproach in their attitude, graciousness, inventiveness, and outright friendliness.


Yes it's indeed a "superb job" to take someone else's ideas, modify them and then claim their your own without giving credit to the person who came up with the idea. That's very "gracious", very "inventive" and extremely "friendly."

If you read varous threads in this forum, you will see what someone else pointed out to me recently, the pattern is very predictable, any criticism--no matter how legitimate--is handled badly and the person criticizing is contantly accused of twisting words.

Does this thread exist for the purpose of someone to feed his own ego by constantly boasting and congratulating himself on his hard work and then publishing something that sounds like a cheap 20-year-old effect box? Or does it exist to share knowledge?

Last time I checked, it was a forum and as such, everyone who contributes to it has a share in it. Call me a bully if you want, I just don't believe in sitting back and letting hypocrisy go unmentioned.

[ursamtl]

Originally posted by Eye of Horus
When do you start ??

EoH
As soon as I see you start to respect others and give others credit for their work instead of claiming it is your own.

As soon as you stop reacting negatively to the smallest criticisms and accusing anyone who does criticize you of twisting words. Just because I dared to criticize your "Ambisonics" method, you've been reacting negatively to every single statement I've made. Even when in the beginning you were at least pretending to be pleasant with me, I detected a sarcastic tone throughout your postings to me.

As soon as you stop rambling constantly about how hard you work and just simply present the work and let it speak for itself, just as everyone else who contributed bidules and files to this forum has done. I didn't see species 8472, desertrat, me, or any of the other contributors doing this. We simply posted our stuff. Again I repeat what I said yesterday, why not adopt the same approach as Daphy and Andy? Treat this as an interesting experience and put everyone on an equal footing. Just because you started a thread on upmixing to surround sound doesn't mean you invented it or that we should pay you some sort of monthly hommage or worship. Does this thread exist for you or for all of us?

[ursamtl]

Originally posted by Shayne
I too have never found sound blaster worth their money. Expensive for absolutely no reason but to give you headaches. I currently use the cheap Hercules fortissimo 7.1 card and pass thru is smooth and easy here.

Thanks for the proggy.

Peace
How have you found the fortissimo's analog performance? I ask this because later this year I'm upgrading my PC and that's one of the cards I'm considering. However, their web site doesn't seem very informative. Does it have a multichannel ASIO driver? This is the most important to me as I interface MIDI keyboards and guitar with my PC and I need low latency for recording and playback. Also, does it show up in a bidule as a multichannel ASIO out? (see my post above about routing the kX driver for a pic. Other than the kX drivers another thing that's making me lean towards the Audigy 2 Zs Pro is the breakout box.

I'll eventually be doing passthrough as well, but for working with these bidules, I like having good analog performance (although digital speakers might be an option too :).

Regards,
Steve.

[Eye of Horus]

Originally posted by ursamtl

As soon as I see you start to respect others and give others credit for their work instead of claiming it is your own.

As soon as you stop reacting negatively to the smallest criticisms and accusing anyone who does criticize you of twisting words. Just because I dared to criticize your "Ambisonics" method, you've been reacting negatively to every single statement I've made. Even when in the beginning you were at least pretending to be pleasant with me, I detected a sarcastic tone throughout your postings to me.

As soon as you stop rambling constantly about how hard you work and just simply present the work and let it speak for itself, just as everyone else who contributed bidules and files to this forum has done. I didn't see species 8472, desertrat, me, or any of the other contributors doing this. We simply posted our stuff. Again I repeat what I said yesterday, why not adopt the same approach as Daphy and Andy? Treat this as an interesting experience and put everyone on an equal footing. Just because you started a thread on upmixing to surround sound doesn't mean you invented it or that we should pay you some sort of monthly hommage or worship. Does this thread exist for you or for all of us?


Whatever ! PLONK !!

EoH

[MC24 Geeek]

Hi I have followed you instruction, but when it comes to using Besweet it does not recognise the wav file. Can you help in any way or am i doing something wronge.
Cheers
Gareth

[Eye of Horus]

Originally posted by MC24 Geeek
Hi I have followed you instruction, but when it comes to using Besweet it does not recognise the wav file. Can you help in any way or am i doing something wronge.
Cheers
Gareth

Hi Gareth,

Do you have the correct settings in Plogue's Bidule ?
Go to the "edit" menu, chose "preferences",
goto Disk I/O and make sure that "use WAVEFORMATEXTENSIBLE" is set to "never" !
Set it to another value and Besweet doesn't recognize the WAV !

I hope this works for you !

kind regards,

EoH

[@ndy]

Hello guys,
since yesterday i have a new ISP with a faster connection.
Solved also today a problem with our dynamic IP.

Have fun and keep in mind that we have three sources:

needful things #1 (http://www.needfulthings.webhop.org)

needfulthings #2 (http://www.freewebtown.com/andiz)

and with upload possibility:

ftp- server (ftp://daphy.mine.nu/incoming/)


Keep surrounding

Yours
@ndy

[kempfand]

Just saw that the announced and long-awaited 64-bit version of Emigrator was released: Emigrator pro64 (http://www.gerzonic.net/?prod=25&rub=2)

Emigrator pro64 is basically the same as Emigrator, but there is one big difference: it uses 64 bit resolution for the decoding and equations. 64 bit resolution improves the audiophile quality many times, because it does not suffer from floating point round off errors (introducing audible / perceptable artifacts to the digital audio signal). The resolution of the decoding equations in Emigrator pro64 is also many times finer.
Please note that this is not freeware (the author asks for $19, very reasonable in my personal opinion). If you really like Ambisonics, I recommend and encourage you to consider this. For the avoidance of any doubts: I have no special relationship to Gerzonic, but the 64 bit decoding precision will make a noticeable difference (a bit as if you feed 32 bit vs. 16 bit to VST processing tools such as bidule).

Anyhow ... just meant as info.

Andreas

[ursamtl]

Originally posted by kempfand
@ Steve: Thanks for your feedback (BTW having the same experience here that running things through the ASIO gives a good anticipation on how it will come out on the big 5.1 system). Andreas,

I've now tested all the groups and bidules from the Gerzon_1997 files you uploaded and so far everything confirms my earlier statement, the Preservation_5_Speaker_v2_REC_.bidule definitely stands out as providing an accurate professional sounding mix, provided the rears are backed off a bit. The others sound similar although not as good to my ears. Actually the Preservation_5_Speaker_v1_REC_.bidule is a bit more balanced in terms of the rear levels, but for some reason I haven't yet nailed down, v2 just seems to sound better.

So, now the next step comes up. I'm sure you're familiar with plugins and progs such as Ambisone, the Spinaudio panners, etc. Perhaps you're also familiar with 3D positional audio. Have you seen anything around on the net concerning this? What would be very cool is to be able to encode some sort of vertical information into a 5.1 system. I know that without speakers mounted above, the reproduction is not going to be possible. I also know that although Ambisonics can carry vertical information, if it's recorded in the first place. However, sound card DSPs have no reached the point where they can produce nearly believable vertical sound, so the question is how do we implement it? If a program such as Ambisone can make sound appear to come from the back even from stereo speakers, surely it should be possible to make sound appear to come from above in a 5.1 system. The same goes for all those two-channel stereo TVs out there with "Virtual surround." I know mine sounds uncanny sometimes with sounds seeming to come from behind and several meters away from the TV (and no, no drugs were involved ;). Since a 5.1 mix gives us the horizontal information only, the missing ambience from over the head would be very cool if somehow synthesized.

Anybody?

[MC24 Geeek]

Hi

Thanks for the reply, I have tried what you said but still have no joy. Here is the error log i get.

Error 58: Error : Unknown Input-File Format : "c:\Agility DVD Files Encore\test.wav".
Quiting...
[00:00:00:000] Conversion Completed !
Logging ends : 06/24/04 , 00:07:55.

Sorry to be a pain but i am very new to all this.
Cheers
Gareth

[kempfand]

Steve,

I can only give some general information to your question.

Ambisone: It's a bit unclear as to what they really do, and they don't disclose anything. I think you better look at at WaveArt's WaveSurroundA VST doing similar things but much better documented and coming from someone with a very prooved track-record (Bill Gardiner).

SpinAudio: Uses XTC (cross-talk concellation). Basically the same topic as the Stereo Dipole I published erlier in this thred (using filters made available by Farina) as well as the XTC VST (including custom-made filter) by Species. SpinAudio's 3D Panner Studio has excellent help files.

For general introduction, see: ISVR Stereo Dipole (http://www.isvr.soton.ac.uk/FDAG/vap/html/sd.html) as well as the related demo WAV files (but you need the closely spaced speakers for the demo to work coorectly).

On howto implement vertical decoding, see Miller's papers (http://www.filmaker.com/papers.htm, with the i, j, k decodings) as well as the VST's published by "Species" here (IMHO, they are excellent, but that's my personal taste).

Regarding "missing ambience from over the head would be very cool if somehow synthesized": I had a very similar thought and once called it "virtual Z restorer/panner" (meaning that Z/horizontal could be created from WXY, even when 'just' using artificall algorithms). That would be really cool indeed.

On howto use one speaker of current ITU-5.1 sytems for vertical informatation (i.e. one of the 5-6 speakers used for decoding vertical data), I just know that Farina, Malham, and others are aware of the the need for more research on this, and that there are efforts on this front, but this is very early work-in-progress and we have to take it as 'teaser' of what to expect over the next 12-24 months.

Sorry for the limited answer (have to leave for a ahort business travel) but hope this helps.

Andreas

[Shayne]

Originally posted by kempfand
Just saw that the announced and long-awaited 64-bit version of Emigrator

Please note that this is not freeware ....

Anyhow ... just meant as info.

Andreas
Funny not even a trial? Would need a trial to get 20 out of me im such a cheap #!?%

Anyway EOH has me tickled pink getting my tunes rocken like they are now .... Thankz.

Maybe you can up load the ftp a sample 32 vs. 64 (same clip) so we can hear the diff.

Peace

[daphy]

@MC24 Geeek
the error 58 is already solved a couple weeks ago ( look here (http://forum.doom9.org/showthread.php?s=&threadid=77271)) I laid a text file (important.txt) on needfulthings ->'plogue' which contains kempfand´s solution! ;)

[ursamtl]

Originally posted by kempfand
Steve,

I can only give some general information to your question.

Ambisone: It's a bit unclear as to what they really do, and they don't disclose anything. I think you better look at at WaveArt's WaveSurroundA VST doing similar things but much better documented and coming from someone with a very prooved track-record (Bill Gardiner).

SpinAudio: Uses XTC (cross-talk concellation). Basically the same topic as the Stereo Dipole I published erlier in this thred (using filters made available by Farina) as well as the XTC VST (including custom-made filter) by Species. SpinAudio's 3D Panner Studio has excellent help files.

For general introduction, see: ISVR Stereo Dipole (http://www.isvr.soton.ac.uk/FDAG/vap/html/sd.html) as well as the related demo WAV files (but you need the closely spaced speakers for the demo to work coorectly).

On howto implement vertical decoding, see Miller's papers (http://www.filmaker.com/papers.htm, with the i, j, k decodings) as well as the VST's published by "Species" here (IMHO, they are excellent, but that's my personal taste).

Regarding "missing ambience from over the head would be very cool if somehow synthesized": I had a very similar thought and once called it "virtual Z restorer/panner" (meaning that Z/horizontal could be created from WXY, even when 'just' using artificall algorithms). That would be really cool indeed.

On howto use one speaker of current ITU-5.1 sytems for vertical informatation (i.e. one of the 5-6 speakers used for decoding vertical data), I just know that Farina, Malham, and others are aware of the the need for more research on this, and that there are efforts on this front, but this is very early work-in-progress and we have to take it as 'teaser' of what to expect over the next 12-24 months.

Sorry for the limited answer (have to leave for a ahort business travel) but hope this helps.

Andreas Thanks for your answers, Andreas. Actually, I've tried Ambisone, Wavesurround, and the Spin Audio stuff. Personally I've found that the Ambisone gives the most realistic sounding horizontal directionality. It does have an elevation setting, but this doesn't seem to accomplish very much. If one closes his eyes and concentrates, there is some sense of difference among the various elevation levels, but how much of it is real and how much is sort of created by the power of "suggestion" or expectation? What I mean by that is, if one expects the sound to give a sense of vertical space, does he or she "fill in" missing psychoacoustic information, or is it actually there?

I still would love to get my hands on whatever the H2O team used for a 2-channel sound file that played during one of their install routines I once tried. It gave an incredibly realistic impression of water bubbling from the front in stereo and then out from the center over the listener's head. It was quite astounding in its realism.

I'm also wondering about Maven 3D Pro. This is a program that looks quite impressive, handles multichannel files, etc., but no one seems to talk about it on here or on any of the other sound forums. I tried the demo once but I didn't have the time to master it to the point where I could do much with it before it expired.

Thanks for the tips concerning Species 8472's VSTs. I'd downloaded them and tried to do something, but the center pages of this thread can tend to be a complete blur. Everyone was uploading bidules and VSTs around the same time and the details on each method didn't seem to get thoroughly discussed.

As for the Ambisonic decoding, I've tried playing around with some stuff but, while I get some interesting horizontal effects--especially when boosting the Y channel width info as in UpMix Studio-- I still haven't had much success with anything vertical. The same old problem arises, trying to apply Ambisonic encoding and decoding to sonic material that was not recorded for Ambisonics. The information just isn't there, so one must--as you said--apply artifical algorithms.

With the right HRTF processing some sense of vertical space should be possible. Or, perhaps some impulse responses with height info could be used to create b-format files with the Z info.

Have a good weekend,
Steve.

[ursamtl]

As I mentioned on the plogue forum, in their default installation, the kX drivers for the Soundblaster Live/Audigy swap the front and rear signals sent to the analog output jacks on the cards. This was done because the AC97 codec chip used for the fronts--especially on the SB Lives--was of inferior quality to the chip used for the rears. In the kX Mixer, there's a switch that will reverse this if a user doesn't want the swap.

Consequently, someone may be unaware of this and wondering why there's no sound when installing the drivers with only front speakers hooked up. In such a case, try either plugging the speakers into the rear channel output or using the switch in the kX control panel.

Regards,
Steve

[Eye of Horus]

I have a strange problem with Bidule/Besweet.
I have in Bidule : a 2 channel file player, one VST and a 6 channel filerecorder.
I have these all in 16 bits, the recorder records a 6 channel file in 16 bits and Besweet makes it 6 mono wav's with the normal order : FL,FR,C,LFE,SL,SR.
In the same layout I change the playing file to a 32 bit floating one and set the filerecorder to 32 bits float.
It writes a 6 channel wav.
This 6 channel wav goes through Besweet with the -6ch replaced by -6chfloat.
Now the order of the 6 mono files is completely mixed up !
LFE=SL, C=SR, FR=LFE, FL=C, SL=FL and SR=FR

Anyone any idea ??

kind regards,

EoH

[Shayne]

Bidule and besweet versions would be? This was a problem with an older version of bidule and besweet but have not had this problem for a while.

Peace

[Eye of Horus]

Originally posted by Shayne
Bidule and besweet versions would be? This was a problem with an older version of bidule and besweet but have not had this problem for a while.

Peace

Bidule 0.6602 and Besweet b28

EoH

[Shayne]

I have tested 6601 and b28 and the 32 bit float output tested by gain decreases is correct. I have not tried 6602 maybe you should try the 6601 version and see if it is the same garbage out. If it is correct than it would be something changed in the .2 version. This has happened before.

Peace

[Eye of Horus]

Originally posted by Shayne
I have tested 6601 and b28 and the 32 bit float output tested by gain decreases is correct. I have not tried 6602 maybe you should try the 6601 version and see if it is the same garbage out. If it is correct than it would be something changed in the .2 version. This has happened before.

Peace

I'm sorry : it's 0.6601 !

Today I tested again with the SRS VST : same problem, but... there is a pattern.

should be - is
FL - SL
FR - SR
C - LFE
LFE - FR
SL - C
SR - LFE

And again only with 32 bits files in a layout I used a minute before in and with 16 bits without any problem......

EoH

[Shayne]

Hmmm?

Perhaps you could post the layout file and we could test here. Would think it is only how you pin out to the recorder and things should not change. As i mentioned before your guide and 32 bit works just fine with pin outs and these two versions.

Peace

[KpeX]

Hi all,

Although things have been more on-topic over the last few posts, I'd ask you all again to keep things civil and on-topic. Anyone not respecting this is begging for an R4 or R16 strike.

As far as credit goes, I think the credit of contributors such as EoH and kempfand goes without saying. A comprehensive guide such as the one presented in this thread helps users more than any simple batch file could do. Think of the doom9 site. Without D9's guides, it would be nothing but a software archive. But his tenacious guides make it one of the most respected AV sites on the web. The bidules and methods used in this thread are well beyond my expertise and I don't feel that any credit needs to be given that hasn't already.

Now hopefully we can all move on and continue this groundbreaking audio conversion discussion.

[Eye of Horus]

Originally posted by Shayne
Hmmm?

Perhaps you could post the layout file and we could test here. Would think it is only how you pin out to the recorder and things should not change. As i mentioned before your guide and 32 bit works just fine with pin outs and these two versions.

Peace

There is not much of a layout file :D
Besides, when I change the order of the pins, it would come back with a 16 bit file !
Anyway : 2 ch fileplayer, a VST that makes 6 channels (it's only 1 VST, not a complex group : A DPL decoder) and a 6 channel recorder. That's really all there is !

EoH

[Eye of Horus]

Hi all,

One of the best commercial stereo to surround VST plugins, is available for a 7 days trial. The retail price is $299.
I can recommend you to try this !
http://www.surroundadvertising.com/store-vst/mall/default.asp

(but please take notice of the above messages about the pin layout on my setup, that is different when feeding 32 bits files instead of 16 bits..........)

An alternative in freeware can be found on :
http://multiphonie.free.fr/acousmodules_en.htm

These guys have a large collection of free VST's. I tried the Spat DPL decoder and was very pleased with the results.
If you put the free Voxengo Stereo Touch to the rears, you can get some incredible results.

They also have more surround plugins, even for 7.1 !
A highly recommended site !

Enjoy !

kind regards,

EoH

[MaroonMike]

EoH,

I have a humble request: would it be possible to put up a page with the various surround "types" and maybe give a brief descritpion of each with the pros & cons? I realize this is highly subjective, but for folks like me who don't get as much time to try out all the bidules & plugins, it might help us zoom in on what we are looking for.

Thanks,
Mike

[daphy]

@MaroonMike

for folks like me who don't get as much time to try out

I think this is one thing we all got in common, so don´t be disappointed when no one wants to start again at very begining (about 2 years ago) to comment every direction which was created in this long time. I once had the idea to this but I gave up because I also one of those who don't get as much time to try out , as married husband sometimes clocks ticking a little different ;)
So one thing is left from this idea: a PDF file with a 'little bit' of structure means a table of contents which leads you direct to the different methodes. I did this in May 2004, and from time to time I ´ll update this 'guide' which is actually nothing more or less than this thread. You´ll find it on the usual places.

[Eye of Horus]

Originally posted by MaroonMike
EoH,

I have a humble request: would it be possible to put up a page with the various surround "types" and maybe give a brief descritpion of each with the pros & cons? I realize this is highly subjective, but for folks like me who don't get as much time to try out all the bidules & plugins, it might help us zoom in on what we are looking for.

Thanks,
Mike

Very good idea !
But..... my time is also limited ;)
Although I'm still on the "sicknote", 24 hours a day is simply not enough.
But... I like this idea so much that I will discuss it with Kempfand and see if we can get something useful.

Perhaps some kind of FAQ......

For sure I will get back to this subject !

kind regards,

EoH

[Eye of Horus]

Originally posted by daphy
@MaroonMike



I think this is one thing we all got in common, so don´t be disappointed when no one wants to start again at very begining (about 2 years ago) to comment every direction which was created in this long time. I once had the idea to this but I gave up because I also one of those , as married husband sometimes clocks ticking a little different ;)
So one thing is left from this idea: a PDF file with a 'little bit' of structure means a table of contents which leads you direct to the different methodes. I did this in May 2004, and from time to time I ´ll update this 'guide' which is actually nothing more or less than this thread. You´ll find it on the usual places.

Hey Daphy,

You had more time : Germany went after the first round :D :D
I still needed two evenings for Holland, but.... that's over too now :mad: :mad:

just kidding !!

kind regards,

EoH

[daphy]

Hey Daphy,

You had more time : Germany went after the first round
I still needed two evenings for Holland, but.... that's over too now
:D :D :D

let´s say 'patt' this time ;)

[ursamtl]

Originally posted by MaroonMike
EoH,

I have a humble request: would it be possible to put up a page with the various surround "types" and maybe give a brief descritpion of each with the pros & cons? I realize this is highly subjective, but for folks like me who don't get as much time to try out all the bidules & plugins, it might help us zoom in on what we are looking for.

Thanks,
Mike

Actually I proposed something similar to this recently to the moderators. The existing surround "sticky" would be replaced with a thread that starts by listing the different guides and methods and providing links to their individual threads or places within a thread. Discussions of surround sound methods in general would continue in the general sticky forum. Each new guide or method that comes along would be a non-sticky thread. The beginning of an individual thread would present the method or guide and then discussions about its technicalities, pros, cons, etc., would follow.

The only remaining question would be what to do with the current sticky thread. Obviously, no one has the time or desire to take this existing thread and split it up, rewrite it, or whatever, so it would still exist, but as a "non-sticky" thread that would be referenced from the sticky list, just like all the surround guides. That would preserve all the valuable information in this thread. Perhaps each contributor of a method could submit the link to his or her posting and then whoever manages the new sticky thread list could add the link.

The problem with the existing setup is that some surround methods never get mentioned in this thread but simply scroll off the first page list of recent threads. I found a couple of the methods I tried entirely by accident. Yes, people can search for them, but if the list were at the beginning of a thread is presented as a "sticky" it should give equal time to all the methods. As I mentioned recently in a reply to kempfand, there were some interesting bidules, vsts, etc., coming up in the middle of this thread, but reading to find them and get info can get really tedious. As well, the thread starts with one guide and for the first few pages, that's all that's discussed, so a newcomer may assume, as I did when I first found it a few weeks ago, that the whole thread focused on the original Ambisonic-based approach. Once I discovered some of the riches buried in the middle, I was surprised and happy!

Daphy and Andy do an amazing job with their server, but downloading the files and then actually implementing them as intended is often not easy. The PDF is good, but it needs constant updating. Anyway, I hope the mods do something because newcomers are missing some potentially great stuff. Yes, this thread is fascinating to read, but it's become somewhat leviathan for people who like you said just want to find something that might be what they're looking for, zoom in on it and have fun!

Regards,
Steve.

[Tantulus]

Well, I compared DPL from my receiver, SAD5.1 with an ambient hall impulse response, the Voice-Non Center.Bidule, the Preservation_5_Speaker_v2_REC_.bidule, using a segment of the last movement of Beethoven's Ninth. I have to agree with Ursamtl that the Preservation_5v2_REc_.bidule is impressive. It even brought out to an extent the natural ambience of the concert hall in addition to filling the room as Ursamtl pointed out. For grins I added the ambient hall Impulse Response to the Preservation bidule (nod to Kempfand)and I was blown away (perhaps too much!) But again, it still all comes down to the engineering of the recording and other factors. For instance, I found that by switching the surround speakers from dipole, to direct or to bipole, I could refine the results of the above methods to my taste. Scattering the surround output produces a better feeling of space with all the above methods but particularly with the Gerzon method. So far I would vote for Preservation_5v2_REC.bidule number one followed by, at least, the Voice-Non Center/Instrumental for choral/orchestral works.

However, SAD5.1 was the winner for a solo instrument backed up by a consort. So at the risk of being redundant, I think the best approach is to really familarize oneself with what each bidle has to offer then listen carefully to the original CD to get a feeling for how it is produced and only then apply the most appropriate surround method.

Thanks everyone for enriching my experience with music!
Scott

[ursamtl]

Originally posted by Tantulus
Well, I compared DPL from my receiver, SAD5.1 with an ambient hall impulse response, the Voice-Non Center.Bidule, the Preservation_5_Speaker_v2_REC_.bidule, using a segment of the last movement of Beethoven's Ninth. I have to agree with Ursamtl that the Preservation_5v2_REc_.bidule is impressive. It even brought out to an extent the natural ambience of the concert hall in addition to filling the room as Ursamtl pointed out. For grins I added the ambient hall Impulse Response to the Preservation bidule (nod to Kempfand)and I was blown away (perhaps too much!) But again, it still all comes down to the engineering of the recording and other factors. For instance, I found that by switching the surround speakers from dipole, to direct or to bipole, I could refine the results of the above methods to my taste. Scattering the surround output produces a better feeling of space with all the above methods but particularly with the Gerzon method. So far I would vote for Preservation_5v2_REC.bidule number one followed by, at least, the Voice-Non Center/Instrumental for choral/orchestral works.

However, SAD5.1 was the winner for a solo instrument backed up by a consort. So at the risk of being redundant, I think the best approach is to really familarize oneself with what each bidle has to offer then listen carefully to the original CD to get a feeling for how it is produced and only then apply the most appropriate surround method.

Thanks everyone for enriching my experience with music!
Scott

Hi Scott,

Nice to hear your experiments are going well. Your discussion of the bipole surrounds reminded me of some reading I did recently regarding THX certification. Part of that standard calls for such an approach for the surrounds.

Stay tuned because if you liked the bidules so far in this thread, you're going to love what I'm working on now! I should finish testing and writing a guide in another week or two. Ironically, I was testing it on the final movement of Beethoven's 9th yesterday as well as on a John Williams recording of Rodrigo's Concierto de Aranjuez. Talk about filling a room with sound! :)

Steve.

[Eye of Horus]

Originally posted by Tantulus
<snip>

So at the risk of being redundant, I think the best approach is to really familarize oneself with what each bidle has to offer then listen carefully to the original CD to get a feeling for how it is produced and only then apply the most appropriate surround method.

Thanks everyone for enriching my experience with music!
Scott

Thanks for the compliment !
And your remark is completely correct : it's impossible to build one method that fits all kind of music !
The combination of the right song with the right method, makes the winner !!

Please give SRS a try too ! ( a few messages above)
I had some excellent results with it and even one of the "quad" boys complimented the conversion !

kind regards,

EoH

[Tantulus]

Thanks for your response, EOH.

I've tried to use Audition's surround panner and found it daunting. Are the vst's your referring to have presets and tutorials?

Scott

[Eye of Horus]

Originally posted by ursamtl
<snip>
Anyway, I hope the mods do something because newcomers are missing some potentially great stuff. Yes, this thread is fascinating to read, but it's become somewhat leviathan for people who like you said just want to find something that might be what they're looking for, zoom in on it and have fun!

Regards,
Steve.

What should be done is a better search routine !

besides that :
A thread becomes sticky when it is proven important enough for a lot of people . The number of visitors to this thread, makes it important enough to get a sticky status.
That people post a lot of non-topic stuff in it, shouldn't be the reason to remove it's status. When all non-topic posts are removed, we come back to the basics ! And I wouldn't be surprised if it becomes less than 15 pages then !!
And as long as there is so much non-topic in it, yes, it will be difficult to find the things you want or need. And I'm sorry to say so (I made myself guilty of that too !) but for instance discussions about hardware drivers for a specific soundcard have nothing to do with this thread !
This thread is already split once, because the former got too much off-topic and became a Besweet commandline discussion.

And actually this discussion is again non-topic and only a waste of space !

So if we all stick to the subject we will get a cleaner and more on-topic thread.

It's not the thread that's wrong, it's the off-topic postings that made it this long.

Yes, I do miss a more active role of the moderators, but I guess this becomes too long for them too !

In the past every now and then a topic was moved to another place because it didn't belong here. I miss that lately !!

My 2 (last) cents, because I will try to stay on-topic from now on !

EoH

[Eye of Horus]

Originally posted by Tantulus
Thanks for your response, EOH.

I've tried to use Audition's surround panner and found it daunting. Are the vst's your referring to have presets and tutorials?

Scott

LOL !! This SRS VST is so simple that I am still surprised that it even has a manual !!

There are almost no settings !

No presets, because it doesn't work that way.

You chose if you want a center speaker in the front only, or also one in the back or none at all. And you chose if you want extra volume to the voice and the LFE channel. If you chose for extra volume it has a standard setting of +5 Db. That's the only thing I advice to change to +3 Db. Sometimes it can clip with +5. (but it even has 7 VU meters, so you can see what's happening and adjust the settings if necessary.
That's all there is ! really !!

Just give it a try with the 7 days trial. It's a pity it's so expensive ! But..... quality obviously has it's price !

Good luck and let us know what you think of it,

kind regards,

EoH

[ursamtl]

Originally posted by Eye of Horus
What should be done is a better search routine !

besides that :
A thread becomes sticky when it is proven important enough for a lot of people . The number of visitors to this thread, makes it important enough to get a sticky status.
That people post a lot of non-topic stuff in it, shouldn't be the reason to remove it's status. When all non-topic posts are removed, we come back to the basics ! And I wouldn't be surprised if it becomes less than 15 pages then !!
And as long as there is so much non-topic in it, yes, it will be difficult to find the things you want or need. And I'm sorry to say so (I made myself guilty of that too !) but for instance discussions about hardware drivers for a specific soundcard have nothing to do with this thread !
This thread is already split once, because the former got too much off-topic and became a Besweet commandline discussion.

And actually this discussion is again non-topic and only a waste of space !

So if we all stick to the subject we will get a cleaner and more on-topic thread.

It's not the thread that's wrong, it's the off-topic postings that made it this long.

Yes, I do miss a more active role of the moderators, but I guess this becomes too long for them too !

In the past every now and then a topic was moved to another place because it didn't belong here. I miss that lately !!

My 2 (last) cents, because I will try to stay on-topic from now on !

EoH Discussing the contents of a thread in the thread is hardly off-topic! Where else are we supposed to discuss it? :)

I think you missed my point. The problem is not off-topic posts, although the thread is full of those (Euro 2004, health problems, surround settings for soundcard drivers, yes we're all guilty :) ). The problem is how newcomers access the information. You wrote a very good guide in the beginning. I'm a technical writer by profession so I'm used to reading this kind of stuff. Yours is well done. As you know very well, I found the results disappointing, but it was not the fault of your guide. However, as I wrote before, someone reading this thread for the first time has to keep reading for a long time before other, equally valid approaches come up. In addition, others have written guides that are also equally valid, but these scroll off the front page before many people have time to read them. Ideally, all these guides, including yours, would appear on the Doom9 Guides page and not just in the forum. They don't, so instead of having to spend a long time searching through a 15 page thread (your estimate after we remove what you consider off-topic), it would be very helpful if people could just access a list of all the methods and go directly to them.

In addition, on most forums I've seen over the years, "Sticky" threads are used by the forum managers to provide important forum-related information, just as the BeSweet FAQ thread. A newcomer such as I was awhile back would naturally come to the conclusion that your guide is the official Doom9-sanctioned approach to creating surround sound. It's not. It's only one of several approaches. As you've said yourself, the best method to use depends on the source music. If there was a simple sticky thread with a list of all the methods, then newcomers would find all the approaches and be in a much better position to evaluate them just as Tantalus has done. For example, in his experience, kempfand's Preservation bidule outranked all the others, yet it only received a passing mention in this thread until kempfand asked me what I thought of it and I posted a message saying how good it sounded! Since then I've had people private messaging me asking how they can get it. This info is there, but it's not apparent, so people just private message instead.

So you see, it's not about off-topic information. It's about information access. I don't know about anyone else, but I don't have the time or desire to read through 14 or so pages of a thread just to find a surround approach that sounds as good as the Preservation bidule. I'm presently putting the finishing touches on something that sounds even better IMHO, and although I'll certainly post details about it in this thread, I'll also provide a complete guide somewhere else so that people don't have to spend forever reading just to stumble upon it.

Maybe we could all share the work in putting a menu together. I'd certainly be willing to help. Would you?

[davidv@plogue]

Hi EoH,

I really dont know what's going on.
Both Bidule 6602 and Besweet b28 seem fine by me
I made two 6 channel files in bidule (not extensible as Besweet doesnt decode them)
and in each channel, a voice saying the channel number:
(available in http://plogue.com/davidv/temp)

and Both

BeSweet.exe -core( -input 06_16bit.wav -output temp- -type wav -6ch ) -ota( -G max )
BeSweet.exe -core( -input 06_32bit.wav -output temp- -type wav -6chfloat ) -ota( -G max )

Give the same mapping

temp-c.wav = 3
temp-FL.wav = 1
temp-FR.wav = 2
temp-LFE.wav = 4
temp-SL.wav = 5
temp-SR.wav = 6

Can you please confirm.

[ursamtl]

Originally posted by davidv@plogue
Hi EoH,

I really dont know what's going on.
Both Bidule 6602 and Besweet b28 seem fine by me
I made two 6 channel files in bidule (not extensible as Besweet doesnt decode them)
and in each channel, a voice saying the channel number:
(available in http://plogue.com/davidv/temp)

and Both

BeSweet.exe -core( -input 06_16bit.wav -output temp- -type wav -6ch ) -ota( -G max )
BeSweet.exe -core( -input 06_32bit.wav -output temp- -type wav -6chfloat ) -ota( -G max )

Give the same mapping

temp-c.wav = 3
temp-FL.wav = 1
temp-FR.wav = 2
temp-LFE.wav = 4
temp-SL.wav = 5
temp-SR.wav = 6

Can you please confirm.

Hi David,

I can confirm that I get exactly the same results as you. Works fine for me.

[Eye of Horus]

Originally posted by davidv@plogue
Hi EoH,

I really dont know what's going on.
Both Bidule 6602 and Besweet b28 seem fine by me
I made two 6 channel files in bidule (not extensible as Besweet doesnt decode them)
and in each channel, a voice saying the channel number:
(available in http://plogue.com/davidv/temp)

and Both

BeSweet.exe -core( -input 06_16bit.wav -output temp- -type wav -6ch ) -ota( -G max )
BeSweet.exe -core( -input 06_32bit.wav -output temp- -type wav -6chfloat ) -ota( -G max )

Give the same mapping

temp-c.wav = 3
temp-FL.wav = 1
temp-FR.wav = 2
temp-LFE.wav = 4
temp-SL.wav = 5
temp-SR.wav = 6

Can you please confirm.

Yep, same result here !
But, that was to be expected !

There is something wrong in Bidule with certain VST's but only on certain computers !
Layla1970 (also member here) gets random results from the filerecorder, I don't get random, but wrong order.
Kempfand reported the same kind of errors.

The two VST's where this happens are :
SRS VST
Spat DPL decoder

And I had it with another one, that we made, too, but ATM I don't remember which one.

So : this doesn't happen always ! That's also what makes it difficult, especially when the filerecorder gives it's output at random !

Resume : everything with the 2 VST works great in 16 bits, but with 32 bits there are problems........

I don't know if it's a fault in the VST's or in Bidule, but as I prefer the outcome in 32 bits, I hope you can find a solution !

with kindest regards,

EoH

[kempfand]

My 2 cents on this: Having s short sticky with links to various methods is a good idea and can be of help for look-up and getting up to speed purposes, but it does in my opinion not replace the need to read through most of the pages of this threat. Where I live, there is a saying going along the lines: "reading educates", and this is what I mean when I advise all newcomers to take the time and read through the details. There is simply no replacement for doing this.

Concerning the pros and cons of each method, this is a function of so many variables (personal taste, source material used, reproduction equipment, even brand of CD-R used, just to name a few), and my preference would be to not use such ratings and just keep it to the technical description & details. Giving 'pros & cons' to the outcome of a method is just a source for endless and mostly unproductive discussions, and I believe that every one here can make up his mind and go the route he or she likes best.

Maybe 2 sticky would make everyone happy. Just my 2 cents.

Andreas

[kempfand]

Originally posted by Eye of Horus
There is something wrong in Bidule with certain VST's but only on certain computers !
Layla1970 (also member here) gets random results from the filerecorder, I don't get random, but wrong order.
Kempfand reported the same kind of errors. I don't have the mapping problem, but still have the problem with >4 GB 32 bit multichannel WAV's and BeSweet demusing. Currently using WaveLab to demux things.

Kind regards,

Andreas

[davidv@plogue]

Resume : everything with the 2 VST works great in 16 bits, but with 32 bits there are problems........

I don't know if it's a fault in the VST's or in Bidule, but as I prefer the outcome in 32 bits, I hope you can find a solution !

with kindest regards,

EoH [/B]

Im sorry but i know how the signal path works in bidule.
as i said ALL signal paths are in 32 bits, and there is no supplemental information that passes through the graph.
The VST plugin has no clue that we record the final output at 32 bits or 16 bits. (or that we change those settings in the file player or recorder)

Please send me your .bidules, and a clear step by step way to reproduce.

[Eye of Horus]

Originally posted by kempfand
My 2 cents on this: Having s short sticky with links to various methods is a good idea and can be of help for look-up and getting up to speed purposes, but it does in my opinion not replace the need to read through most of the pages of this threat. Where I live, there is a saying going along the lines: "reading educates", and this is what I mean when I advise all newcomers to take the time and read through the details. There is simply no replacement for doing this.

Concerning the pros and cons of each method, this is a function of so many variables (personal taste, source material used, reproduction equipment, even brand of CD-R used, just to name a few), and my preference would be to not use such ratings and just keep it to the technical description & details. Giving 'pros & cons' to the outcome of a method is just a source for endless and mostly unproductive discussions, and I believe that every one here can make up his mind and go the route he or she likes best.

Maybe 2 sticky would make everyone happy. Just my 2 cents.

Andreas

Hey Andreas,

Excellent worded !

I agree that some kind of links-sticky-thread can be useful !

Or a FAQ !

About the pros and cons : I think we should make a FAQ to give newbies some kind of direction and describe what the bidules try to achieve. But they still need to try them theirselves. We cannot tell them : this bidule is better or worse , because of all the reasons you said !!

TTYL ;)

EoH

[Eye of Horus]

Originally posted by davidv@plogue
Im sorry but i know how the signal path works in bidule.
as i said ALL signal paths are in 32 bits, and there is no supplemental information that passes through the graph.
The VST plugin has no clue that we record the final output at 32 bits or 16 bits. (or that we change those settings in the file player or recorder)

Please send me your .bidules, and a clear step by step way to reproduce.

I can send you the bidules, but the two that gives the problems are so simple ;)

fileplayer 2 channels > SRS VST (or Spat DPL Decode) > filerecorder 6 channels.


That's really all there is !

But if you wish to have the bidules, please tell me where to upload them to !

BTW there is also some reaction from SRS : they tested in Nuendo and they worked. They didn't test in Bidule and the VST was not designed for Bidule.

Doesn't help much :-) and I thought VST's were universal ?!

What I will do is this : I will test the same in another program (I believe it's called Audiomulch ?) and see what happens there.

BTW could it have anything to do with settings in XP ??
Just wondering......


kind regards,

EoH

[Eye of Horus]

Originally posted by kempfand
I don't have the mapping problem, but still have the problem with >4 GB 32 bit multichannel WAV's and BeSweet demusing. Currently using WaveLab to demux things.

Kind regards,

Andreas

LOL !! That is NO problem here ! Yesterday I did a 7.4 GB mch-WAV without any problem !
And we both have the same setup !!!

STRANGE !!!!

EoH

[kempfand]

@ EoH / David: I think the mapping problem for a few is caused by specific combinations of drivers, dll's, .ax, DirectX, whatever, but not by Bidule a/o BeSweet. If it was, it would reproduce consistently on all machines.

In analogy, there were mapping problems reported for demuxing dts-wav's, and in the end, the solution was/is to use a working combination of azidts, ivaudio.ax, WinDVD, BeSweet (i.e. using specific versions of drivers etc).


@ EoH: But they still need to try them theirselves. That's my point. Even experiments leading to disappointing results are invaluable, as they help to 'recognize' "good" results. In my previous live, I used to be a student for classic piano, and my teacher always advised me to visit and listen to unknown (and therefore sometimes not-so-good) live performances. I was 25 years younger then, and didn't really get the point back then, but I now realize the good advice, as it helped me personally to recognize the "good" performances and not just call them "good" because of what the adds say.

Regards,

Andreas

[Eye of Horus]

Originally posted by kempfand
@ EoH / David: I think the mapping problem for a few is caused by specific combinations of drivers, dll's, .ax, DirectX, whatever, but not by Bidule a/o BeSweet. If it was, it would reproduce consistently on all machines.

In analogy, there were mapping problems reported for demuxing dts-wav's, and in the end, the solution was/is to use a working combination of azidts, ivaudio.ax, WinDVD, BeSweet (i.e. using specific versions of drivers etc).


@ EoH: That's my point. Even experiments leading to disappointing results are invaluable, as they help to 'recognize' "good" results. In my previous live, I used to be a student for classic piano, and my teacher always advised me to visit and listen to unknown (and therefore sometimes not-so-good) live performances. I was 25 years younger then, and didn't really get the point back then, but I now realize the good advice, as it helped me personally to recognize the "good" performances and not just call them "good" because of what the adds say.

Regards,

Andreas

Perhaps I should try a clean install again ?
It's less than a month ago, I did so !

BTW I didn't know you could play the piano at the age of 3 !! LOL !!

grtz,

EoH

[ursamtl]

Originally posted by kempfand
My 2 cents on this: Having s short sticky with links to various methods is a good idea and can be of help for look-up and getting up to speed purposes, but it does in my opinion not replace the need to read through most of the pages of this threat. Where I live, there is a saying going along the lines: "reading educates", and this is what I mean when I advise all newcomers to take the time and read through the details. There is simply no replacement for doing this.

Concerning the pros and cons of each method, this is a function of so many variables (personal taste, source material used, reproduction equipment, even brand of CD-R used, just to name a few), and my preference would be to not use such ratings and just keep it to the technical description & details. Giving 'pros & cons' to the outcome of a method is just a source for endless and mostly unproductive discussions, and I believe that every one here can make up his mind and go the route he or she likes best.

Maybe 2 sticky would make everyone happy. Just my 2 cents.

Andreas Welcome back, Andreas.

I agree that a "pros and cons" discussion would be counterproductive. Yes, reading does educate, but to read the whole thread just to find something specific is not very efficient. If each new method or vidule is presented in its own thread, then added to a master list of methods without any "editorial" information, then people can zero in one specifics and evlauate them for themselves. This current thread is valuable as a discussion are for the theory of surround, etc., but the specifics for each method are better dealt with in a thread for that specific method. At present, if someone decides to post a method in a separate thread, it quickly scrolls off the first page and only those with the curiosity (and time!:)) to read through the thread list get to know about these. Someone new might read the beginning and then go to the last page to see what's new, but what about all the middle stuff? If it were somehow easy to zoom in on these methods, bidules, VSTs, etc., without having to read for a long time, people could spend more time experimenting and less time reading.

[davidv@plogue]

Hi

Unlike Directshow (graphedit and co)'s nighmarish configuration,
Bidule is not influenced in _any_ way by any directshow filters you may have installed, so .ax, directshow, and audio driver do _nothing_ to the computations in the bidule graph OR in the mapping of channels in our file writer, as we dont use any directshow filters to read or write files or to process audio.

all audio/file processing and mangling _appart_ from VSTs is entirely made in PlogueBidule.exe and libdnsfile.dll using cross platform code. reinstalling bidule wont change anything.
Please trust me on that will ya ? :)

As far as the SRS vst plugin is concerned, im afraid that theres NO way im going to install a PACE - protected plugin on my system. EVER.

PACE is EVIL.

i will try out woth the acousmodule one, and get you back on that.
(i fail to see how a VST can grow arms and penetrate bidule's graph code and mess things up)

Now the reason why i ask for a step by step way to reproduce and a specific bidule file is just because i KNOW (im a bidule programmer), that the problem is either in the VST or in the way the cables are wired.

Im trying to isolate the "bug" here.

Ive proven before that Besweet OR the bidule's file writer is NOT to blame. Now there MAY be a bug in the bidule intermodule routing, but if there is one, it is STATIC, it doesnt change if you load a 16 bit wav file or a 32 bit wav file., but maybe only if you add/remove connections between the various VSTs and bidules. (this is why i need the .bidules) or if you load an entirely new bidule (which has different routings)

Cheers

[Eye of Horus]

Originally posted by davidv@plogue
Hi

Unlike Directshow (graphedit and co)'s nighmarish configuration,
Bidule is not influenced in _any_ way by any directshow filters you may have installed, so .ax, directshow, and audio driver do _nothing_ to the computations in the bidule graph OR in the mapping of channels in our file writer, as we dont use any directshow filters to read or write files or to process audio.

all audio/file processing and mangling _appart_ from VSTs is entirely made in PlogueBidule.exe and libdnsfile.dll using cross platform code. reinstalling bidule wont change anything.
Please trust me on that will ya ? :)

As far as the SRS vst plugin is concerned, im afraid that theres NO way im going to install a PACE - protected plugin on my system. EVER.

PACE is EVIL.

i will try out woth the acousmodule one, and get you back on that.
(i fail to see how a VST can grow arms and penetrate bidule's graph code and mess things up)

Now the reason why i ask for a step by step way to reproduce and a specific bidule file is just because i KNOW (im a bidule programmer), that the problem is either in the VST or in the way the cables are wired.

Im trying to isolate the "bug" here.

Ive proven before that Besweet OR the bidule's file writer is NOT to blame. Now there MAY be a bug in the bidule intermodule routing, but if there is one, it is STATIC, it doesnt change if you load a 16 bit wav file or a 32 bit wav file., but maybe only if you add/remove connections between the various VSTs and bidules. (this is why i need the .bidules) or if you load an entirely new bidule (which has different routings)

Cheers

OK ! I will post it tomorrow as an attachment to your emailaddress !
BTW I also think the problem is in the VST, but.... it does work in other programs like Nuendo, as the spokeswoman of SRS said.
Can that be possible, that a VST works good in certain programs only ? I thought it was an universal format.
And if it is indeed the VST that causes the problem, I for sure wouldn't pay $ 299.= for it !! (Despite the excellent results it gives !)

BTW can you explain more about "PACE is EVIL" ?
Does it do anything unwanted to your system ? (I ask this because I just used AD-Aware for the first time on my PC and was surprised by the amount of thrash it found !)

kind regards,

EoH

[davidv@plogue]

Originally posted by Eye of Horus
[B]Can that be possible, that a VST works good in certain programs only ? I thought it was an universal format.
And if it is indeed the VST that causes the problem, I for sure wouldn't pay $ 299.= for it !! (Despite the excellent results it gives !)


i think you may be close to something, theres a whole section of the VST format thats not used at all as my experience is concerned (600 plugins in my folder), and upon looking, it does contain details on expected host "speaker arrangements"
and _maybe_ the plugin expects us to feed it some valid values (prolly cubase/nuendo does that). And its a part of the standard that only some surround plugins require, but thats the first news i have of it.

Seems like i have no choice but to download the PACE bugger,
and check it it require us to feed it values "as if it were" inside nuendo.

Speaking of which, PACE is the most stickest copy protection you can find, it crawls everywhere and its been reported to cause some instabilities in some users PC (allthough im trying to find some proof of that, ill post the links when i do).

Its used as a desperate mesure to block piracy but often timnes bugs the legitimate users more.

[Eye of Horus]

Hi all,

When someone hurt my feelings I tend to ignore all he wrote and also the good points. (hey, I'm human too ;) )
But it's never too late to do something about that......

My proposal : Let's make a top list with direct links to all methods used to make surround from stereo on Doom9.
I will make a start but need some help.

The way I think we should do this :
- Name of the method
- Author of the method
- Link to subject on Doom9
- Direct Link to the place to download

After that list is ready, we can add external (outside of Doom9) links.

So :

I will do this whole thread and find all the bidules and VST's mentioned in here.

Ursamtl : do you want to do the methods outside this thread on Doom9 ?

Daphy : will you do the direct links to your server ? (somethinng like : SAD51inBidule : http://www.needfulthing.org/audio/bidules/SAD51inBidule/SADinBidule.rar).

This way someone can chose not to dive into the subject, but direct download it.....

All others : when you find a link outside Doom9, please add !

You can send it all to me by PM, and I will make a list in the above mentioned format.

Let's ask DSPGuru then to add it as a sticky one above my sticky one....

What do you think ?

kind regards,

EoH

[ursamtl]

EoH,

I'd be happy to do the methods outside of this thread. I'll try and have that to you by the end of the day (the Canadian day. Yours ends 4 or 5 hrs earlier than mine :)).

I think the list this will be a really valuable addition to Doom9.

Regards,
Steve.

[Eye of Horus]

Originally posted by ursamtl
EoH,

I'd be happy to do the methods outside of this thread. I'll try and have that to you by the end of the day (the Canadian day. Yours ends 4 or 5 hrs earlier than mine :)).

I think the list this will be a really valuable addition to Doom9.

Regards,
Steve.

Excellent !!!
I need a little bit longer > Finals of the European Championship Soccer today !

kind regards,

EoH

[ursamtl]

Ok, I've read through the entire list of threads on this forum and sent EoH a list of all the surround methods I could find, 14 in all. This does count the earlier versions of EoH's method. Combine that with all the stuff in the big thread, and we should have a really nice list of methods!

I've also sent him some links to related threads. These contain info that may be useful.

Regards,
Steve.

[heula]

I saw this thread and I tried to make a 6channel wav following this guide. Now everything went well but when making the 6channel wav to 6 mono wavs with besweet, besweet can't open the wav. Nothing happens. Also besweet gui doesn't accept the wav. The wav I created with Ploque Bidule is playable in windows mediaplayer and windvd.
When making mono wavs with a standard 2channel wav it will work in besweet.
Am I doing something wrong or.....? :confused:

Please help me out here.

[Eye of Horus]

Originally posted by heula
I saw this thread and I tried to make a 6channel wav following this guide. Now everything went well but when making the 6channel wav to 6 mono wavs with besweet, besweet can't open the wav. Nothing happens. Also besweet gui doesn't accept the wav. The wav I created with Ploque Bidule is playable in windows mediaplayer and windvd.
When making mono wavs with a standard 2channel wav it will work in besweet.
Am I doing something wrong or.....? :confused:

Please help me out here.

I suppose you have the latest Bidule ? (0.6601)

Goto "Edit " , "Preferences", "disk I/O" and set use WAVEFORMATEXTENSIBLE to "NEVER".

This will fix the problem, but..... you will have to redo your files !

kind regards,

EoH

[heula]

Thanks, this solved the problem.

[MaroonMike]

Stay tuned because if you liked the bidules so far in this thread, you're going to love what I'm working on now! I should finish testing and writing a guide in another week or two. Ironically, I was testing it on the final movement of Beethoven's 9th yesterday as well as on a John Williams recording of Rodrigo's Concierto de Aranjuez. Talk about filling a room with sound

Steve,

Any update on this?

Thanks,
Mike

[ursamtl]

Originally posted by MaroonMike
Steve,

Any update on this?

Thanks,
Mike

Hi Mike,

I deleted my previous answer as I've now posted the bidule and a quickie mini-guide in the thread http://forum.doom9.org/showthread.php?s=&threadid=79862 . Give it a try. I've been quite pleased with the results so far using this bidule.

Enjoy!
Steve.

[Sakuya]

I really appreciate that you have created this guide. I don't understand how a given stereo track can be divided into a surround track since the original had nothing like that. You do mean a 5.1 track for DVDs right? I think 5.1 tracks is a division where some is designed to come out of the back speakers and whatnot. Since my stereo track has voices and background music mushed together, how will Plogue divide that? :confused:

Edit:
Okay, it finished recording and I now have a testresult.wav. Is there a program for me to convert it to AC3 keeping all the channels? I used BeSweet to enter this input line to get the 6 wav files:

BeSweet.exe -core( -input testresult.wav -output D:\TEST\ -type wav -6ch) -ota( -G max )

It gives me this error message:

Error 58: Error : Unknown Input-File Format : "testresult.wav".

I put the WAV file in the same folder as BeSweet.exe so why won't it work? :scared: In Plogue before I followed your guide, I set the Preferences to 48000 Hz since I'm working with that. I tried my best at the BeSweet GUI v0.7 b2 to convert the WAV to AC3 but it shows the same error message above. Is something wrong with the WAV file? I am sure I followed your guide correctly. I also tried using BeSplit to fix the WAV and output as an AC3 but the black command line box popped up and nothing happened. It just stays there with no moving white numbers. :scared:

[daphy]

It gives me this error message:
Error 58: Error : Unknown Input-File Format : "testresult.wav".
did you realy tried this:
http://stevethomson.ca/audio/guides/waveformatextensiblenever.gif
:(

Moderator Edit: Fixed Image link

[ursamtl]

Originally posted by Sakuya
I really appreciate that you have created this guide. I don't understand how a given stereo track can be divided into a surround track since the original had nothing like that. You do mean a 5.1 track for DVDs right? I think 5.1 tracks is a division where some is designed to come out of the back speakers and whatnot. Since my stereo track has voices and background music mushed together, how will Plogue divide that? :confused:Hi Sakuya,

No, almost all the methods described in the guides here actually produce five different audio channels and many also add a sixth LFE channel. They either take information that is already in the 2-channel stereo mix and manipulate it in some way to produce the differences, or else they add artificial ambience to generate a difference. Once you get them to work, I think you'll be quite surprised with the results.

As for your problem with Besweet, it is most likely due to what daphy just posted. Try the following:[list=1] Go to the Edit menu and choose Preferences.

Click the Disk I/O tab.

Set the use WAVEFORMATEXTENSIBLE dropdown list to never.

[/list=1] This should fix your problem.

Happy encoding
Steve.

[Eye of Horus]

Originally posted by ursamtl
Hi Mike,

I deleted my previous answer as I've now posted the bidule and a quickie mini-guide in the thread http://forum.doom9.org/showthread.php?s=&threadid=79862 . Give it a try. I've been quite pleased with the results so far using this bidule.

Enjoy!
Steve.

Link added in the list......

EoH

[Eye of Horus]

Originally posted by ursamtl
Hi Sakuya,

<snip>

Happy encoding
Steve.

Now we have on one page three times the same answer to the same question.......

3 times the question
3 the times the answer
In the last 10 messages !!

People, please search before asking !!
Thanks !

EoH

[ursamtl]

Originally posted by Eye of Horus
Now we have on one page three times the same answer to the same question.......

3 times the question
3 the times the answer
In the last 10 messages !!

People, please search before asking !!
Thanks !

EoH

Hi EoH,

Perhaps a good solution to this recurring question would be to edit your original guide and add a mention of this step. Feel free to use my screen grab, the one that daphy used.

Steve.

[Eye of Horus]

Originally posted by ursamtl
Hi EoH,

Perhaps a good solution to this recurring question would be to edit your original guide and add a mention of this step. Feel free to use my screen grab, the one that daphy used.

Steve.

Yep, I will....
But only after I finally find out how !!
When I try to edit the message, I get the error : your message has too many images. And I should remove some......
(the images were never there in the first place when I started the guide (I had just links to the pictures), but included later by a moderator. And now I read it back it contains some pictures on the wrong place, the layout is a bit of a mess..... but I cannot edit because of the message......)

thanks for the pic !

EoH

[Sakuya]

I did a test DVD with two different audio tracks. One is this:
http://www.planetquake.com/filterfactory/3daudio.html

The other one is the guide by EoH. The first one which has the 5 channels along with the subwoofer is not good for a movie. It divides up the voices and it sounds more like a concert hall than anything (plus, I can do that same effect just by using my receiver).

For EoH's output, it's much better than the above where the voices have less of the echo (but it was still there). Unless there is a stereo2surround which emphasizes more on the background music/other effects and leaves the voices in the center, I think I will keep to stereo.

Another setback was that my source had muffled audio. Normally, it sounds fine. When it gets just a bit louder, it will start to get muffled (like when someone is talking through a pillow). Both of the conversions above makes this worse. I cannot find an audio filter that solves this muffled audio and I have set up a thread asking this question but nobody replied. :scared:

Anyway, this was a pretty good experience. This would work well for CDs and concerts. :D So thank you to those who came up with this.

[daphy]

Hi folks,

I´ve found a nice VST plugin (http://www.needfulthings.webhop.org/files/audio/tools/vst-plugins/unzipped/AnechoicRoomSimulator.rar) to play around with virtual room size parameters.

Please please check it out, wheater it´s worth to add in the vst collection :rolleyes:

[kempfand]

LOL

As far as I remember, this VST does absolutely nothing :p, because that is what an Anechoic Chamber will do (no coloroation, no reflections etc).

Cheers,

Andreas

[daphy]

okay :(

let´s forget about it ;)

[desertrat]

Hi folks,

been a few months since I was last here and it looks like I've got quite a bit of catching up to do. Anyhow the reason for the post is I've got an update to my bidule that I have u/l to the ftp (desertrat_ambi_combo.bidule) and a need to give a bit of explanation to go along with it.

First I would like to ask that these 2 files be removed from the server
desertrat_stereo2ambisonic_ambi_1.5_combo (ver 2).bidule
desertrat_stereo2ambisonic_ambi_1.5_combo (ver 3).bidule

they were put up way to early, before a sufficient amount of testing on my part, sorry if this may have caused anyone and trouble. These 2 bidules will cause distortion on newer source material ie. anything that has a peak close to 0.0 dB.

So now on to the revised method...

The bidule still has a left and right chain, the left side is taken from EoH's ambisonic method and uses the same settings. The right side is based on the ambi 1.5 method with the WXYZ to 5.1 set to J and the three gain modules after it should be set to the same setting.

I normally use one the following settings for the gain modules -10dB, -15dB -18dB -21dB depending on what sounds best for the particular type of music I'm using it on. The gain must be set for a minimun -5dB or you will most likely get distortion in your end results.

The gain modules allow you to control the amount of mix between the left and right sides. The gain modules are on the right side because the right side has a much higher level than the right side, somewhere around -15db to - 18dB seems to be where there's a pretty even mix between the two.

I have found that -15dB setting seems to work pretty well with reggae and dub. -18dB and -21dB work better on ambient music such as Jarre, Eno, etc.

Cheers,
desertrat

[AI_Warrior]

I have to problems
I know this prob had'n very movement lately but i have some questions
1st:BeSweet tell's me that the wav file is invalid
2nd:When im able to split the 6 chan to individual wavs and put it in the Surcode it also says that the wave header is invalid

Wt should i do???

Greetz 2 ALL

[Eye of Horus]

Originally posted by AI_Warrior
I have to problems
I know this prob had'n very movement lately but i have some questions
1st:BeSweet tell's me that the wav file is invalid
2nd:When im able to split the 6 chan to individual wavs and put it in the Surcode it also says that the wave header is invalid

Wt should i do???

Greetz 2 ALL

This subject passed by quite some times and alone in this thread I think the answer is given at least 4 times !! :D

Anyway : start Bidule and go to the settings. Make sure that WAVEFORMATEXTENSIBLE is set to "never". That should fix the problems.
In the version of Bidule used for the guide, this function didn't exist yet. I wanted to update the guide, but I get all kind of remarks about a ""too long message" and cannot save it.....

good luck and kind regards,

EoH

[ursamtl]

Hi AI,

For a graphic that shows the quick solution to this problem, see Step 1 of http://forum.doom9.org/showthread.php?s=&threadid=79862. If you follow the directions there, you'll come up with a file that will work fine in SurCode.:)

Hope this helps.

Steve.

[AI_Warrior]

1st of all thanks for the so quick help
Im sory that there was already an answer for my question, and i repeated.(im new in this thread)
You were right about my problem but now i have another but just in Surcode. And i think its nasty:
Now i load the wavs(ok) but when i set encode he says "Parallel port lock required to encode"
Now this time i checked the web for this issue and nothing.
By the way when i tried de-activate the parallel port the blue screen showed up and system crashed

Can u help m?

[ursamtl]

Originally posted by AI_Warrior
1st of all thanks for the so quick help
Im sory that there was already an answer for my question, and i repeated.(im new in this thread)
You were right about my problem but now i have another but just in Surcode. And i think its nasty:
Now i load the wavs(ok) but when i set encode he says "Parallel port lock required to encode"
Now this time i checked the web for this issue and nothing.
By the way when i tried de-activate the parallel port the blue screen showed up and system crashed

Can u help m?
Do you have a legal copy of SurCode? If so, be sure the dongle is properly installed. If it is and you still get the problem, contact SurCode. If you're using an illegal version well they're often tend unstable.

[AI_Warrior]

Well i reinstaled the program and all went fine but.....i only hear noise. I tried all possible donfigurations. I even tried BeSweet to make the ac3 but nothing..
Everything is alright but the dts/ac3 encoding dont work.

I only ambisonics wonking in pc(WAV)...and i can say its quite impressive, using the realtime DD5.1 provided by nforce(wich has not the briliance of DTS).

Please help m. Its only this that dont work
By the way is there any other DTS encoder than Surcode?

Greetz, and thanks in advance

[ursamtl]

Originally posted by AI_Warrior
Well i reinstaled the program and all went fine but.....i only hear noise. I tried all possible donfigurations. I even tried BeSweet to make the ac3 but nothing..
Everything is alright but the dts/ac3 encoding dont work.

I only ambisonics wonking in pc(WAV)...and i can say its quite impressive, using the realtime DD5.1 provided by nforce(wich has not the briliance of DTS).

Please help m. Its only this that dont work
By the way is there any other DTS encoder than Surcode?

Greetz, and thanks in advance


AI, if you didn't get either the DTS or AC3 encoding to work, then what you were hearing was not "realtime DD5.1" but some simulation provided by nforce.

I haven't heard of any other DTS encoder. Keep trying Besweet AC3Machine. It should work. Don't give up!

[Schwa226]

Hi all!

I build a installation routine what will install all of needed programs and files for the stereo to 5.1 AC3 conversation.

The routine will include:
Sonic Foundry Soft Encode
Plogue Bidule
BeSweet
SSRC
Layouts, Groups and Plugins from the server http://www.needfulthings.webhop.org/

Also a WIN32 GUI to handle Bidule,BeSweet,SSRC and Sonic will be included (still in development).

I already have contact with Daphy about uploading/testing the routine.

For more information: carmenelectra_gc@yahoo.de

schwa226

[ursamtl]

Originally posted by Schwa226
Hi all!

I build a installation routine what will install all of needed programs and files for the stereo to 5.1 AC3 conversation.

The routine will include:
Sonic Foundry Soft Encode
Plogue Bidule
BeSweet
SSRC
Layouts, Groups and Plugins from the server http://www.needfulthings.webhop.org/

Also a WIN32 GUI to handle Bidule,BeSweet,SSRC and Sonic will be included (still in development).

I already have contact with Daphy about uploading/testing the routine.

For more information: carmenelectra_gc@yahoo.de

schwa226

This is a most welcome initiative that will help a lot of newcomers. Well done!

You might want to include AC3Enc since SoftEncode is no longer available and some people are not comfortable using pirated software.

Another suggestion would be to read the registry key for the current VSTPlugins directory and install any plugins there. Then the user could be reminded in a prompt or guide to change the Plogue VSTPlugins folder to point to it. In this way, those of us who have other VSTs installed will have access to all our VST plugins from all hosts. It could be argued that one can always have supplementary VST plugin directories in hosts such as Cubase and Nuendo, but this approach doesn't give Plogue access to plugins outside of its own directory.


I look forward to checking out your routine!

Regards,
Steve.

[daphy]

@ Schwa226

I´ve sent you some further idea for your installing routine -> check your mailbox ;)

[Schwa226]

Hi,

thx Dapyh for the file. I will exchange it in the setup.

Also automatic search for the standart VST plugin folder is now included.

I attached the first BETA of the AC3GUI. Hopefully I didn't forget needed DLLs ;)

Short description about the GUI:
Please use folder/filenames without spaces!!

--
At the first startup you can choose if Sonic or AC3ENC should be used for the AC3 encoding.
--
Needed programs:
BeSweet 1.5b29 (Install folder like C:\Sonic\BeSweet)
Plogue Bidule (Install folder like C:\Sonic\Bidule)

If you are using Sonic:
Sonic Foundry Soft Encode 1.0 (Install folder like C:\Sonic\Bidule)

If AC3ENC:
AC3ENC.DLL, should be located in the same folder than BeSweet
BeSweet 1.5b23 or lower version. Version above have no support for AC3ENC.

Optional:
SSRC if you want to normalize your single mono WAV files. (Install folder like C:\Sonic\BeSweet)


Please test the Beta version of the GUI, I am sure some bugs are included ;)

cu,
schwa226

[daphy]

Please test the Beta version of the GUI, I am sure some bugs are included

you will find on needfulthing at audio/latest

@Schwa226

seamed to be a nice one - it´s too late to test it here :goodpost:



BTW also set new Hypercube Transcoder (3.05) online (see audio/tools/dts-ac3) - feedback wanted ;)

[daphy]

here the link to the first beta version (http://forum.gleitz.info/attachment.php?attachmentid=68724)

[Surlyrich]

I just tried following the guide, but when I press play in biddle i start getting my output file but it just keeps going. My 5 min song ended up being 25 gig before I stoped it. The timer on the plyer doesnt move either when it is playing. Not sure what I have done. Any help would be appreciated as I am trying to make 3 DVD's of all Led Zeppelin's albums in DTS.

Cheers

[daphy]

- 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.


please checkout if you linked the devices correctly :rolleyes:

[Surlyrich]

Yep... They are linked

[ursamtl]

Originally posted by Surlyrich
I just tried following the guide, but when I press play in biddle i start getting my output file but it just keeps going. My 5 min song ended up being 25 gig before I stoped it. The timer on the plyer doesnt move either when it is playing. Not sure what I have done. Any help would be appreciated as I am trying to make 3 DVD's of all Led Zeppelin's albums in DTS.

Cheers Hi Surlyrich and welcome to the forum! The Zeppelin project sounds cool. I've been a major fan since the 70's and in fact I used many of their songs when developing my V.I bidule (http://forum.doom9.org/showthread.php?s=&threadid=79862). It really helps one to better appreciate Page's genius when you get some of those layered guitars into a 3-dimensional space! Plus, Bonham's drums sound like they're sitting in front of you at times! Several members of the Zep mailing list I'm on have tried it and reported great results.

As for your problem, it sounds like you forgot to link the start for the file player and file recorder. When they're linked, the recorder stops at the same time as the player. I wrote a step-by-step procedure foe setting this up at: http://forum.doom9.org/showthread.php?s=&postid=548323#post548323

If you have any other questions, feel free to ask or PM me.

Regards,
Steve.

[freenow]

im newbie.
how i can a wav ar another audio format to ac3 dolby surround 5.1
i have a avi with pcm audio or another format audio
but i want to make audio dolby surround 5.1

sorry for my english..
thanks

[ursamtl]

Hi freenow and welcome to the forum!

Just convert your audio to a wave file and then follow one of the guides. the key is to get the audio into wave file format first. Check the forum by searching and reading some of the thread. You can can also find good information in the Doom9 guides. Then if you have specific questions, there are lots of us here in the forum to help.

Good luck,
Steve.

[freenow]

ok.
thanks
i have output the audio to wav.
now how i can convert to dolby surround 5.1?
who is the easy and best way to do this?

[ursamtl]

Hi again Freenow,

Do a bit of reading and then try out some of the methods. Daphy and Andy have put together an excellent server where you can download all the required software and instructions for each method. You can find it at Needfulthings (http://www.needfulthings.webhop.org/). You can find an extensive list of methods described in this forum at 99% Complete List of stereo to surround methods (http://forum.doom9.org/showthread.php?s=&threadid=79774).

When doing any of the methods using Plogue Bidule, the following two points are important:[list=1] Set Bidule to never output waveformatextensible files as explained and pictured in http://forum.doom9.org/showthread.php?s=&postid=525238#post525238 .
Link your Audio File Player and Audio File Recorder following the instructions in http://forum.doom9.org/showthread.php?s=&postid=548323#post548323.
[/list=1]One thing I might point out is that "Dolby Surround" is not 5.1. It is the original matrixed surround encoding method that eventually became Dolby Pro Logic and then Pro Logic II. I think what you want is "Dolby Digital 5.1", which is five separate full-range channels plus one for low-frequencies.

As I said above, take your time and try out a couple of the methods listed. It can be a lot of fun.

Regards,
Steve.

[@ndy]

Hi everybody,
as I heard from daphy, my FTP-Server (ftp://daphy.mine.nu/) was not reachable. There was an error message like this: "you don`t have the right to access this folder" or so on.
I think that i fixed the problem, so that you`ll have full access including to upload (ftp://daphy.mine.nu/incoming/) your files again.

If you have problems using my service feel free to send me a pm (http://forum.doom9.org/private.php?s=&action=newmessage&userid=46752).

For all of you using my upload (ftp://daphy.mine.nu/incoming/) service:

I will not post more news concerning my FTP-Server in any other thread than this (http://forum.doom9.org/showthread.php?s=&postid=554886#post554886) one.

If you want to be up to date you have to subscribe this (http://forum.doom9.org/showthread.php?s=&postid=554886#post554886) thread to be informed.

Keep surrounding
@ndy

[Eye of Horus]

The guide on the first page is removed by me.

As someone is trying so hard to take things over here (the wannabe-moderator :) ) with his "I know everything better attitude" , I should say : be happy with him and his method.

As long as this jerk goes on, I'm off............

EoH

[DSP8000]

It is very good to have people that understand so much about audio, but people with attitude like yours just make me sick.
On the end of the day this forum is NOT yours, so nobody is taking over something yours.

DSP8000

[KpeX]

Please stay on-topic to the original subject of this thread, or this thread will be closed. As previously mentioned, a copy of the guide is available on ftp://daphy.mine.nu/

[SallyDog]

@EoH

You started it; you should be able to finish it.

Sorry to see you go, but thanks for everything.

Regards

[Eye of Horus]

Originally posted by DSP8000
It is very good to have people that understand so much about audio, but people with attitude like yours just make me sick.
On the end of the day this forum is NOT yours, so nobody is taking over something yours.

DSP8000

I fully agree and nowhere I said it's my forum !
But the guide is my intellectual property and I posted it here for everyone to enjoy. If my efforts on converting stereo to surround are reason for some people to get offensive, I have the right to remove it too ! I also have the right to feel disappointed about the behaviour of someone here, the constant attacks and praising anything other than my guide in the thread I started with my guide. No moderator who did anything about it ! No moderator with a : stay on topic message. (only after a request !)
Messages about the bad results my method should give and immediately sending people to his method...... No operator in sight !

Even questions addressed to me are answered immediately by this person. Why ? He is not EoH, I am !
No moderator who says anything about that behaviour....

All with the reason : free forum, freedom of speech etc. etc.
If being offensive is called freedom of speech here, I don't want to be part of it any longer !

And before someone comes with the question : yes, I tried to solve things ! I tried to reach out ! But it cannot come from me alone. Both sides should be willing to solve the problems. I was !

I regret the days that DSPGuru had an active role here, are over.
I regret seeing all good people (people who could learn me something without constantly nagging about my method) stopped posting...
I regret the fact that Doom9 is no longer the first place to visit when having problems with audio or video, for a lot of people.

I regret that it has to end this way.

And mostly I regret having to leave (or better being forced to leave) because there is also a large group of persons here, that I like and who were friendly and appreciated what I made ! I want to thank them for the nice responses and perhaps I'll see you all somewhere else in future !

kind regards,

EoH

[daphy]

Hi folks,

as Freddy Mercury once said:

The Show must go on ( - even if one have to leave!)

After being forced to remove the old guide, here my first attemp of an easy click guide (version 1.0b) (http://www.freewebtown.com/andiz/audio/guides/first-6WAV-guide.exe)

feedback wanted!

[KpeX]

@EoH

If you have a problem with a forum member, then either use the report post link or post the link to a specific post. You cannot make these blind accusations and say that the moderator is not active in this forum.

If there are posts that you feel violate the rules, please point them out to me. But I fail to see what you are talking about.

This forum is a venue for free speech. Members are free to ask you something. Other members are free to post their answer or opinion to the question, and you are free to post your answer. The original poster has the freedom to use or learn from whatever answers he chooses.

I don't see insults in this thread, or in the other 5.1 discussion threads. I see a lot of discussion and testing of conversion methods. Again, if you feel there are insulting or rule breaking posts, please show me.

[Eye of Horus]

Originally posted by KpeX
@EoH

If you have a problem with a forum member, then either use the report post link or post the link to a specific post. You cannot make these blind accusations and say that the moderator is not active in this forum.

If there are posts that you feel violate the rules, please point them out to me. But I fail to see what you are talking about.

This forum is a venue for free speech. Members are free to ask you something. Other members are free to post their answer or opinion to the question, and you are free to post your answer. The original poster has the freedom to use or learn from whatever answers he chooses.

I don't see insults in this thread, or in the other 5.1 discussion threads. I see a lot of discussion and testing of conversion methods. Again, if you feel there are insulting or rule breaking posts, please show me.

As I said : in the past the moderators played an active role. But now the members must inform the moderators. You say you don't find anything ? And now I have to search to show you examples ?

Sentences like this one about me , were not enough for you to do anything about it :
"Yes it's indeed a "superb job" to take someone else's ideas, modify them and then claim their your own without giving credit to the person who came up with the idea. That's very "gracious", very "inventive" and extremely "friendly."

Just one of many many more. Strange that you don't find them !
If I want to, I can post at least 25 of these quotes, that I consider insulting. Hey, it's MY feeling I have with remarks like this. That you don't care about this, doesn't mean the feeling will be gone for ME ! Calling that "blind accusations" while there are so many examples, is IMHO not fair !

I'm sorry Kpex, but as moderator you did no good job (compared to the old times). I repeat : if freedom of speech is an excuse to let those insults go on, then I don't feel this is the place where I feel comfortable. Your first reaction was a threat to close down this thread. Where were you when those insults started ?

Everyone who reads this whole thread can point you exactly to where it started. Strange that you can't do that yourself !

just my 2 cents......

EoH

[Eye of Horus]

And without any problem my other threads are removed here too ?
Is this the way it works, Kpex ?

The one with the biggest mouth gets the credits ?

EoH

[KpeX]

None of your threads have been removed. The guide list thread has been updated and had to be changed since only you had edit privileges. Would you expect me to keep sticky a thread from which you've removed a guide?

[Eye of Horus]

Originally posted by KpeX
None of your threads have been removed. The guide list thread has been updated and had to be changed since only you had edit privileges. Would you expect me to keep sticky a thread from which you've removed a guide?

If it needed an update, you just decided to remove it and put it back with someone else's name ?
Is this the way it goes here since DSPGuru is no longer active ?
It was my thread...... and you don't justify at all, "giving" it to your friend Ursamtl......

I hope you're not the coward you seem, and let this message stand here, so everyone can see how you use your moderater-ship.....

EoH

[KpeX]

It was ursamtl's idea to begin with, and how was he supposed to update it without edit privileges?

Have a nice break, EoH. Struck, thread closed.