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Old 4th September 2003, 21:16   #61  |  Link
joshbm
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First off, I would like to say EoH you are awesome! I have been working around with KpeX's SAD5.1 and it is very nice it does a good job, yet it was lacking something. Your new Ambisonics method works great! Hats off to you :-)!!!

If you guys are wondering how to get 6 mono wavs, well it is quite simple to do using CDP Multichannel Toolkit.

You can download it from there website here:
http://www.cs.bath.ac.uk/~rwd/mctools.html

or a direct link to the zip file here:
http://www.cs.bath.ac.uk/~rwd/mctools.zip

Now once you download it, you will be working with the command-line tool called "channelx". Since it is a command-line tool you will need to create a new batch file with the following in it:

@ECHO OFF
channelx -och.wav YOUR_6CH_WAV.wav 1 2 3 4 5 6

This will separate all the channels within a 6 channel wav file. You can also specify which channels to take out with this tool. This is definately a handy set of tools.

Last edited by joshbm; 4th September 2003 at 21:34.
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Old 4th September 2003, 22:19   #62  |  Link
Eye of Horus
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Quote:
Originally posted by joshbm
First off, I would like to say EoH you are awesome! I have been working around with KpeX's SAD5.1 and it is very nice it does a good job, yet it was lacking something. Your new Ambisonics method works great! Hats off to you :-)!!!

If you guys are wondering how to get 6 mono wavs, well it is quite simple to do using CDP Multichannel Toolkit.

You can download it from there website here:
http://www.cs.bath.ac.uk/~rwd/mctools.html

or a direct link to the zip file here:
http://www.cs.bath.ac.uk/~rwd/mctools.zip

Now once you download it, you will be working with the command-line tool called "channelx". Since it is a command-line tool you will need to create a new batch file with the following in it:

@ECHO OFF
channelx -och.wav YOUR_6CH_WAV.wav 1 2 3 4 5 6

This will separate all the channels within a 6 channel wav file. You can also specify which channels to take out with this tool. This is definately a handy set of tools.
Thanks for the kind words......

I had these tools already since more than a year, but now they're a bit outdated.....

You know, there's also a tool called wav2to6.exe..... does exactly the same :-)

And the Besweet tools, Soft Encode also....there are more...

As long as there's no alternative (except doing it in a soundeditor) to the OTA -g max command..... I'm more than happy with Besweet !!!

But it's always good to look at other tools too !

So....thanks for the links !

kind regards,

EoH
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Old 4th September 2003, 22:20   #63  |  Link
bitsnbytes
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Speaker Arrangement

Now looking at the pentagon layout in emigrator
speakers should proly be adjusted to match... Has anyone tried this different speaker layout?

dunno if the wife would let me turn the room into a pentagon hahahahahah



@plogue david

Is the 6 seperate recorders a bug or a processor / resources problem?
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Old 4th September 2003, 22:33   #64  |  Link
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You create 6 separate mono recorders and sync their start parameters toghether?

sadly this trick is not very sample accurate.
the only trick i know is to press rec on the first one from series of linked recs and then start the audio processing (circle icon on top of the app) then do the inverse after.

Even then all files might not have the same end point, idd need to check.
What we tought about doing was a single module that saved multiple mono files OR a multichannel one, so all mono files would be synched.

We might add this in a future version.

Cheers
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Old 4th September 2003, 22:45   #65  |  Link
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@ davidv; Good to see you here. Bidule (despite beta) is indeed a great tool, and I would like to say 'Big Thanks' for it. If you / your team can implemnet an offline (render) mode, that would really rock :-)


@ EoH:
Quote:
I'm more than happy with Besweet !!!
The 'old' method with Ambidec (which does normalisation by default) and BeSplit achieves exactly the same. In other words: After B-Pan & B-Proc, the B-file could be saved with B-Rec, with subsequent Ambidec & BeSplit, and this would achieve the same as Emigrator & BeSweet -ota( -G max ).


@ bitsnbytes: Good catch. Using only B-Pan & Emigrator (w/o B-Proc), you would have to re-arrange the speaker set-up, as there would be a SC (surround center, in your back so-to-speak).

In order to avoid this, we used a 'trick': Rotating the sound-field by -36 degrees counter-clock-wise with B-Proc. Result is that you don't need a SC (surround center) speaker.

I know it might sound a bit confusing. It might help if you just draw the speaker-setup from Emigrator on a piece of paper. Channel-1 is slightly turned counter-clock-wise (by -37 degrees). Channel-3 is the surround-center (at the end of the minux-X axis).

Then, by literally turning the sound-field by -37 degress (counter-clock-wise), channel-1 becomes C, channel-2 becomes FL, cannel-3 becomes SL, etc.

Regards,

Andreas

Last edited by kempfand; 4th September 2003 at 22:56.
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Old 4th September 2003, 22:54   #66  |  Link
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Re: a long the 24 bit 32 bit story

Quote:
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
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Old 5th September 2003, 01:30   #67  |  Link
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Hello David, Welcome,

Quote:
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

Quote:
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.


Quote:
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.

Quote:
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


Cheers,
Dg.

Last edited by DSPguru; 5th September 2003 at 09:31.
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Old 5th September 2003, 02:13   #68  |  Link
bitsnbytes
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Thanks For The Gain!

just what I was missing... I added 3 gains
utilizing 5 slots ... NOT using the 6th one for LFE as the HNM filter adds OutAmp ...Gain
I set it to +10db much better
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Old 5th September 2003, 05:09   #69  |  Link
specise_8472
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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.
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Old 5th September 2003, 05:37   #70  |  Link
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Quote:
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.
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.

Last edited by DSPguru; 5th September 2003 at 05:41.
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Old 5th September 2003, 07:21   #71  |  Link
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@ 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
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Old 5th September 2003, 09:26   #72  |  Link
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the spike at the beginning cannot be avoided, since it's actually data - not samples, and in fact, this is useful data : it's the peak value of all channels.

Quote:
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):
Quote:
BeSweet.exe -core( -input L:\s2s\result.wav -output L:\s2s\temp- -6chfloat ) -ota( -g peak )
but if you don't feel like sitting infront of the computer until the BeSweet process ends, use the commandline from the BeSweet FAQ to transcode directly to DTS. (end-to-end floating-point process)


note : don't forget to create logfiles - for future discussions!

Last edited by DSPguru; 5th September 2003 at 13:27.
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Old 5th September 2003, 13:37   #73  |  Link
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/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 ?
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Old 5th September 2003, 15:50   #74  |  Link
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Quote:
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 !!!!

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

EoH

Last edited by Eye of Horus; 5th September 2003 at 15:52.
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Old 5th September 2003, 16:08   #75  |  Link
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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 ) ?
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Old 5th September 2003, 16:41   #76  |  Link
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Quote:
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
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Old 5th September 2003, 16:43   #77  |  Link
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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..

Last edited by DSPguru; 5th September 2003 at 16:46.
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Old 5th September 2003, 22:10   #78  |  Link
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Quote:
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
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Old 5th September 2003, 22:25   #79  |  Link
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Quote:
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
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Old 5th September 2003, 22:53   #80  |  Link
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Last edited by bitsnbytes; 6th September 2003 at 00:17.
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