GUIDE: Stereo to 5-Channel Surround

[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