GUIDE: Stereo to 5-Channel Surround

[Eye of Horus]

A GUIDE TO MAKE 5 CHANNEL SURROUND FROM STEREO
+++++++++++++++++++++++++++++++++++++++++++++++

This method provides excellent results. In some aspects it even sounds better than AC3 files ripped from DVD. Give it a try and share your thoughts with us !

Here is the software you need.....

- Cooledit
(http://www.syntrillium.com/download/)
- Aurora plugins
(http://www.ramsete.com/Aurora/download/)
- Tools Ambisonics
Included here - see below message

The Tools include 3 wav files you need further in the process and the free programs pack2x2to4.exe and ambidec.exe.

-> a B format signal is a 4 channel signal. W is the sound pressure. X, Y and Z are the harmonics on the pressure field. X is a vector directly in front of the listener, Y is a vector on the side, and Z is the vector of height. Since we'll decode a 2D signal (remember, surround is not 3D audio, it is 2D!), we can forget about the Z signal.

-> The goal here is to convert your stereo CD signal, which is UHJ - encoded, into a B format signal, which carries all the data so that you can decode all the channels you'd want to.

Okay, let's start working !!

-> Rip a stereo track from a CD

-> Using CoolEdit and Aurora, do this.

-> Start CoolEdit.

-> Load the UHJ-W filter file ,a 44.1 Khz file . Copy it to the windows clipboard. Press CTRL+6 for Windows Clipboard and the CTRL+C to copy to that clipboard.

-> Load the stereo signal (the ripped stereo track you want to convert).

-> Launch the "Convolve with Clipboard" Aurora module, and set the following settings:

Channels to convolve: Audio data: Both. Imp. Res: Both. No Crosstalk cancel mode. No remove DC component. No autorange result. No timeReverse Imp. resp. Gain =0.

-> Convolution begins. When it ends, you'll have a stereo file. We need it in mono.

-> Convert Sample type, from Stereo to Mono, with a 50% mix on each channel and 32 bit resolution.

-> Close and restart Cooledit. This is necessary to be sure that the clipboard is empty again ! Make sure you set the Windows clipboard (CTRL+6)

-> Repeat the procedure, but load the UHJ-X and UHJ-Y files, so you end up with three mono files (which will be your w, x, and y signals).

-> Now remember, the "Z" component means "height". So we really don't need to decode it.

-> Close and restart Cooledit

-> Now we have to pack the W signal along with the X signal.
Re-transform W in a stereo wave file, with 16 bits depth, using the Convert Sample Type module in Cooledit. Be sure 100% of the signal goes to the left mix, 0% to the right, and that resolution is 16 bits. Enable dithering.

-> Now switch to the X signal (the X file we've been working on). Resample it to a 16 bit mono file and Copy it to a Cooledit Clipboard. Go back to the W signal, select the entire right channel and paste the X signal file into the right track of the W stereo signal. You'll end up with a stereo file with the W signal on the left channel and the X signal on the right channel. Save it !

-> Close and restart Cooledit

-> Switch to the Y signal. Convert it to stereo, 44.1/16. The Y signal will be 100% on the left channel. Since we don't have a height (Z) signals, the right channel signal should be left empty. Save it !


-> Now you have two stereo signals. Change the extension of the WX file (should be WAV) to "WX". Change the extension of the YZ file (should be WAV, too) to "YZ".

-> Run the Pack2x2to4.exe file from a DOS prompt. It will ask you the names of the WX and YZ files. It will then save a 4 channel WXYZ file suitable for use with AmbiDec.
! So make sure the extension of the saved file is .WXYZ

-> Use ambidec. It's a program that is used to decode a B-Format file (the one you made) to produce a Wave file with speaker feeds. Those files will be suffixed in .wxyz.
You can decode the signal into any rig type. If you are using a standard 5.1 system, you'll want to decode it in "Pentagon".

-> Use it like this:

ambidec -r Pentagon (input file) (output file)

-> The file you'll end up with is a 5 channel wave file. Now we must encode it to DD 5.1
Load the entire file into SoftEncode. The program automatically recognizes the feed as a 5 channel wave file, and presents it as such. Encode it with a setting of 640 K/s to get the higest possible quality in DD5.1 .


-> Burn the result and enjoy ! The results are really superb !!

OR

Use Bspilt to extract 5 wav files from the wxyz signal and use Surcode DTS to make a DTS.

Commandline for Bsplit :
besplit -core( -input song01.wav -prefix g:\song01 -type wav -demux )

song01.wav = song01.wxyz
g:\song01 = the place where the 5 mono wavs are saved

Use the files in this order :

Left front : song0101
right front : song0103
center : song0102
lfe : stays blank
left surround : song0104
right surround : song0105

NEW METHOD FOR THE LAST STEPS
=============================

After making the wxyz file, there is now an alternative for Ambidec and Besplit. My friend Andreas brought this goodie to my attention !
First the link where you can download the program VVMIC, which stands for Visual Virtual Microphone.
http://mcgriffy.com/audio/ambisonic/vvmic/
Download, install and run the program.
Here follow the steps :
1. Click on input file and chose your wxyz file. You can chose between a wxyz extension or a wav extension.
2. Click on output file and give it a name
3. Select multiple mono files
4. Change number of outputs to 5
5. Select link pairs and select all directives
6. Click on G-format
7. Now click on PLAY and you will end up with 5 mono files !

In surcode, use the same order to make a DTS from those 5 files as described above (1-3-2-blank-4-5)

Results are a fraction less then the Ambidec/Besplit method, but that's mainly, because there is no gain control. The volume is lower and it sounds a little bit les crispier, brighter.

But.... it is very easy and saves a lot of time and it is very fast !



Good luck and let us know something about your results !

Eye of Horus with the much appreciated help of Dr. Freudiana

DSPguru : We hope you will be able to include this

[theReal]

I have one naive, amateurish question about this :)
If the source is stereo, why is the result going to sound better as a 5 channel file?

Is the result just like if I'm telling my 5.1 soundcard to play stereo files on all channels (and apply some spatial effects, maybe)?
But, there must be a benefit over this, otherwise you wouldn't go through all the hassles, right?

[DJ Bobo]

@ Eye of Horus
This is all very nice, but even without testing (I don't have a 5.1-system), I can tell you, you won't get real 5.1 sound. I think, as theReal said, you're just applying some kind of surround effect, nothing special. I think it works in a similiar way like Dolby Surround Pro-Logic II.

But I will take the opportunity and ask you about something really impotant for me: how to generate high frequencies from a source that have frequencies up to 11KHz only in a good quality?
I remember someone having asked such a question but there was no good answer I think. You seem to be an expert in sound matters, so may be you can help :D

[Eye of Horus]

Originally posted by theReal
I have one naive, amateurish question about this :)
If the source is stereo, why is the result going to sound better as a 5 channel file?

Is the result just like if I'm telling my 5.1 soundcard to play stereo files on all channels (and apply some spatial effects, maybe)?
But, there must be a benefit over this, otherwise you wouldn't go through all the hassles, right?

Of course there is !
I suggest you read this article :

http://s2n.org/Articles/Ambisonics.html

It is a complete explanation and it also gives inside why this system never made it to the mainstream. It was invented in the 70's !!
And.... all experts (I am definitely not !) say this is the best !

just give it a try once and then you will know why !

best wishes,

Eye of Horus

[Eye of Horus]

Originally posted by DJ Bobo
@ Eye of Horus
This is all very nice, but even without testing (I don't have a 5.1-system), I can tell you, you won't get real 5.1 sound. I think, as theReal said, you're just applying some kind of surround effect, nothing special. I think it works in a similiar way like Dolby Surround Pro-Logic II.

But I will take the opportunity and ask you about something really impotant for me: how to generate high frequencies from a source that have frequencies up to 11KHz only in a good quality?
I remember someone having asked such a question but there was no good answer I think. You seem to be an expert in sound matters, so may be you can help :D

Oh no !! I am definitely NOT an expert !
But... I did some reading on the subject (a lot).
In one of my first posts I asked if people could help me on this subject ! A friend promised me to tell me how to do it, but after 6 weeks I almost gave up hope. And.... today the instruction finally was there :-)
I immediatly tried it with an instrumental from my favorite group (the Alan Parsons Project) and I still have that feeling....this can't be true !! But....it is :-)
So no "expert" opinions here. The only thing I will say about it, is that it is NOT like Dolby at all ! If you wish it can even encode to surround from 10, 11, 12 or whatever number of speakers !

I know we always take for granted that true surround is : a solo guitar from the left-front, voices from the Center, Bass thru the LFE, etc.etc. The article will give you a complete different angle !

http://s2n.org/Articles/Ambisonics.html

It tells all !

Just read it !

best wishes,

Eye of Horus

[MvB]

@eye

If i'm not mistaken you need for correct reproduction of ambiosonic sound a special microphone. the standard stereo signal (please correct me if i'm mistaken) doesn't include enough information to build the ambiosonic matrix. At least you would need two microphones placed orthogonal together while recording to include the sound comming from the back? I've just had a short peek at the paper you indicated but they talk about a special microphone for the recording. Even if you throw out the height ... if i understood correctly a normal stereo signal is not really a useable source. But maybe i've misunderstood the whole thing.

[Eye of Horus]

Originally posted by MvB
@eye

If i'm not mistaken you need for correct reproduction of ambiosonic sound a special microphone. the standard stereo signal (please correct me if i'm mistaken) doesn't include enough information to build the ambiosonic matrix. At least you would need two microphones placed orthogonal together while recording to include the sound comming from the back? I've just had a short peek at the paper you indicated but they talk about a special microphone for the recording. Even if you throw out the height ... if i understood correctly a normal stereo signal is not really a useable source. But maybe i've misunderstood the whole thing.

For live-recordings : yes, a special microphone is needed.
And indeed normal stereo hasn't enough information.
That's why we convert it ! :-) The 3 pulse files add room characteristics to the stereo sound.
Just try it out, the results will amaze you !
I still cannot believe the results ! I did a few other tracks and still every track sounds very good.

best wishes,

Eye of Horus

[theReal]

How do you play it back? Do you have a DVD Burner and play it through your DVD Player?

[Eye of Horus]

Originally posted by theReal
How do you play it back? Do you have a DVD Burner and play it through your DVD Player?

????
It is a standard WAV file ! You can burn it to CD, just like WAV's encoded with DTS.......
So I burn it to CD and play it on my standalone DVD-player...... I don't have a 5.1 card in my PC :-(

Eye of Horus

[theReal]

I don't have a standalone DVD player, so I didn't know you could play DTS wave files from normal CD's. I thought you had to have an Audio-DVD to do that.

[Eye of Horus]

Originally posted by theReal
I don't have a standalone DVD player, so I didn't know you could play DTS wave files from normal CD's. I thought you had to have an Audio-DVD to do that.

They are just normal wavs, but of course you need an decoder to hear them. A normal cd player attached to stereo only gives white noise when you play these. And yes, when you have a 5.1 card in your PC, you should be able to play them.....

Eye of Horus

[Eye of Horus]

Originally posted by Eye of Horus


Tomorrow I will start trying to make a script or batchfile to automate the process in Cooledit..... (never done that, so cannot promise anything !)


Well, I tried and tried and tried :-(

There are a view problems in Cooledit. I will write them here and also edit the guide. It is important to follow these line to be sure you will get good results !

1. A script in Cooledit is useless ! You still have to be at your PC because you have to press "Enter" or "OK" for quite a few times. So that's not a real solution....

2. IMPORTANT !!!!
Cooledit has some major problems with the clipboards.
a. When you are in one of the first steps in the guide, you have to put the UJH-x.wav in the WINDOWS clipboard. Do NOT use one of the Cooledit clipboards !!
b. When you are finished with the UJH-x part, then close Cooledit and restart the program. Make sure to clipboard is set to Windows and is EMPTY !
You have to do this, because I found out that when you copy to the clipboard, the second time you do this, the first content isn't always correctly removed. And I couldn't find an "Empty Clipboard" function. So that's why it is absolutely necessary to restart CE !
c. When you try to do several steps at once, like Convolve with clipboard, then immediatly conversing to MONO 32 bit and then conversing to Stereo 16 bit. When you have done this for the W-wav...no problem ! But then you must Save the file and restart CE.
d. When you have done X-wav....Save too ! And....restart.
e. When you want to combine W and X, make completely sure that when you are copying X-wav to the clipboard, that you have one of CE's clipboards and NOT Windows' clipboard.

A tip : when you want to convert a whole cd, it is fastest when you load in all the wavs and one of the UJH-.wavx, and do the steps involved with all the files. Then save them all, restart CE and to the next step...... This way it saves you a lot of restarts and reloads... !!!


Another thing is the order of the WAV files as given in the DD5.1 encoder : Leave it as it is. ! It looks unnatural in the graphics in the DD5.1 screen, but it is NOT !!

When you use Bsplit to make five separate MONO wavs from the 5_streams_wav, you end up with lets say track01_01.wav, track01_2.wav, track01_3.wav, track01_4.wav and track01_5.wav.
This is the correct order for using them in Surcode to make a DTS :

Left Front : 01-1
Right Front : 01-3
Center : 01-2
LFE : left blank
Left Surround : 01-04
Right Surround : 01-5

Yesterday I did Water Colours and Seen One Earth from Pete Bardens.
And.... the results were SUPERB !
Full natural surround in DTS ! When you really want to hear what this whole method is about : use the DTS encoder instead of the DD5.1 encoder ! The sound is just more...fresh, clear, bright .

In the beginning some people asked if it really is worth the troubles and time. I can really say : YES ! After experiencing that sound, I understand more what I did read on the Internet about it !


Anyone else tried it yet ?

Last point : someone asked about the microphones to record. Well, there are quite some UJH encoded cd's available. As you can read in the article (link in the "guide") the writer quotes a review of "Stereotomy" by Alan Parsons. This is one of the UHJ encoded cd's. This one sounds excellent when you listen to it in stereo.And those who have read the article : all the channel information you need for whatever number of channels is already in that file ! So when you do the complete process on this CD, you get the ultimate result. Full perfect surround which throws a huge shadow on whatever other surround system !!!!!

I am not an expert on this, I only believe my ears ! They don't lie to me......
I found a list with a few titles online and will do a search if they are still available .


Well, we still need a bit more automated process. Is there any way (DSPguru?) to put the two DOS programs from the Ambisonics package and the Bsplit program in one ? So all the steps from thesong.wxyz file to 5 separate MONO wav files ??

I will adjust the guide later........ this was long eough for now :-)

best wishes....... and please share some reviews !!

Eye of Horus

[DSPguru]

Originally posted by Eye of Horus
Is there any way (DSPguru?) to put the two DOS programs from the Ambisonics package and the Bsplit program in one ? So all the steps from thesong.wxyz file to 5 separate MONO wav files ??i can surely implement the 'Pack2x2to4.exe' part inside BeSplit. i belive my code might even be faster.
as for ambidec,
since i don't have time to dig into this, i would have to ask you for direct links for info about its operation, algoirthm, etc'...

then, i'll see what i can do.. :)

[Eye of Horus]

Originally posted by DSPguru
i can surely implement the 'Pack2x2to4.exe' part inside BeSplit. i belive my code might even be faster.
as for ambidec,
since i don't have time to dig into this, i would have to ask you for direct links for info about its operation, algoirthm, etc'...

then, i'll see what i can do.. :)

Here is the link to the site of the maker :

http://www.muse.demon.co.uk/mn_index.html

regards,

Eye of Horus

[ivan_alias]

Is there any other way to do this at all that may be easier?

I'm going to give this a go with some Michael Jackson, but the process is quite a long one!


I want to thanks Eye of Horus for his / her hard work on this though! I don't want to seem ungrateful :)

[Eye of Horus]

Yes , there is a faster way to do this, but the results are worse.
There is a commercial available program, that does basically the same but without use of Aurora or Cooledit. It's all done in one program and IMHO it's main drawback is that it doesn't do any resampling with 32 bits, but it does all with 16 bits files. The endresult is the same but it sounds more dull, flat. Just like when you convert a 44.1/16 to a 22.050/8 wav.
The program is for sale but you can ask the maker by email a key for 30 days.

The FIReverb Suite 2nd edition
Available at : http://www.catt.se

I know my method is very timeconsuming, but it gives good results !

regards,

Eye of Horus

[jansb000]

In the guide you tell us to use SoftEncode to change the 5-channel wave into DD5.1. According to the documentation this task could be done with BeSure. Have you tried this?

(BeSure is free - SoftEncode is not and seems to be no longer available on the web)

PS: In another post from me I mention problems with BeSure, but for the time being I guess that these problems are caused by wrong parameters that I set and not by a real problem with the tool.

[Eye of Horus]

When you use Besure....... at the end it starts.....Softencode !!!!

Contact me private at my email address !

regards,

Eye of Horus

[DSPguru]

Originally posted by Eye of Horus
When you use Besure....... at the end it starts.....Softencode !!!!no, it doesn't.
BeSure uses BeSweet to encode to DolbyDigital & uses SurCode to encode to DTS.

[Eye of Horus]

You're right !
I mixed up Softencode and Surcode :-)

BTW did you shine your light on implementing pack 2x2to4 and ambidec ?

It would be marvellous !!!!!

regards,

Eye of Horus

[iGorland]

Just wanted to say: Thanks! It does take time to go through the whole procedure for one CD, but the DTS sound at the end is great on my CD player with optical output!

[kempfand]

@ DSPguru:

Just also want to say that I look forward to the ambidec implentation.

I also like the sound, and some automation will help alot.

Regards,

Andreas

[DIggedy]

Use the files in this order :

Left front : song0101
right front : song0103
center : song0102
lfe : stays blank
left surround : song0104
right surround : song0105



Why is there no LFE file? I'd imagine adding an LFE channel would just be a matter of using left front & right front converted to mono followed by a low pass filter... I have my own method of creating 5.1 conversions using only Sound Forge and have not tried this method yet... am interested to see how it compares.

[lucindrea]

i'm curoius on how you do your 5.1 ...
i can take a stero file and make a mono for center and lfe .. just cant figure how to remove the center from the left and right .. also the dynamics of the rear are hard to get working also

[drpaulng]

If what the soundforge's impulse (delay effects) has already done with close simulation to various environments. Why bother use this AURORA thing when the CoolEdit Pro plug-in costs 200USD with so slow in multiple steps to simulate Ambisonics?

[Eye of Horus]

Originally posted by drpaulng
If what the soundforge's impulse (delay effects) has already done with close simulation to various environments. Why bother use this AURORA thing when the CoolEdit Pro plug-in costs 200USD with so slow in multiple steps to simulate Ambisonics?

And what does SF cost ?????

You forget there is a difference between reverb only and Ambisonics (read the websites). And only adding reverb doesn't convert your stereo to 5.1 DTS !! And it doesn't simulate ! It IS Ambisonics !
Yes, it simulates the room responses, but that's not the only thing it does !

Take for instance the program from www.catt.se .
You can make 5.1 from stereo with added reverb. Simple direct way. Why do you think they included a separate option to encode to Ambisonics ????

Eye of Horus

[DSPguru]

Originally posted by kempfand
@ DSPguru:

Just also want to say that I look forward to the ambidec implentation.

I also like the sound, and some automation will help alot.

Regards,

Andreas if i only had more free time.. :(

[drpaulng]

Can you show us a sample to convince us to spend 200USD for the plug-in. I lost interest after the non-registering bomb. How can ambisonics be evaluated when we never get in touch with it? We've got no idea or any impression on this technology at all. With a good AV amplifier such as the DENON 3802, one can get instant surround simulation with DTS NEO 6, Dolby Prologic II, and even one more, called DENON 5 or 7 channel stereo. I call all of them simulated surround because it is true that the multi-channel sound is generated from a stereo source. If you give me a chance to use the original multi-track recorder, I'll make my own version of "true" surround by feeding some of the tracks discretely like the new DVD Audio such as the Eagles' Hotel California. The DENON 5 or 7 channel stereo is the closest simulation to the DVD Audio 5.1 sound, for example, Fleetwood Mac's RUMOURS is a very good example when you feed the stereo in and get the DENON 7 channel out. A brief listening moment would fool you into believing that you are hearing from the 5.1 mix. If DENON can give us their simulation surround like this, why doesn't the AMBISONICS? Just go license such DSP firmware to all kinds of brands of AV amplifier. We just can't bear to see such good thing buried dead...but please, at least let us have a slight touch on this. Show us an .ac3 file sample please.

Sound Forge 6 is 349.97USD (I just checked this on the web).
Cooledit Pro 2 is 249USD...and the AURORA plug-in package is 200USD.

[Eye of Horus]

Originally posted by drpaulng
Can you show us a sample to convince us to spend 200USD for the plug-in. I lost interest after the non-registering bomb. How can ambisonics be evaluated when we never get in touch with it? We've got no idea or any impression on this technology at all. With a good AV amplifier such as the DENON 3802, one can get instant surround simulation with DTS NEO 6, Dolby Prologic II, and even one more, called DENON 5 or 7 channel stereo. I call all of them simulated surround because it is true that the multi-channel sound is generated from a stereo source. If you give me a chance to use the original multi-track recorder, I'll make my own version of "true" surround by feeding some of the tracks discretely like the new DVD Audio such as the Eagles' Hotel California. The DENON 5 or 7 channel stereo is the closest simulation to the DVD Audio 5.1 sound, for example, Fleetwood Mac's RUMOURS is a very good example when you feed the stereo in and get the DENON 7 channel out. A brief listening moment would fool you into believing that you are hearing from the 5.1 mix. If DENON can give us their simulation surround like this, why doesn't the AMBISONICS? Just go license such DSP firmware to all kinds of brands of AV amplifier. We just can't bear to see such good thing buried dead...but please, at least let us have a slight touch on this. Show us an .ac3 file sample please.

Sound Forge 6 is 349.97USD (I just checked this on the web).
Cooledit Pro 2 is 249USD...and the AURORA plug-in package is 200USD.

Ok... a reply....
First of all I want to state two things.... English is not my native language. I'm from the Netherlands and only learned English in highschool, which is about 30 years ago. So : sometimes you need to read between the lines. OK ? When something is not quite clear, please ask me to rephrase and don't start to yell immediatly ! :-)
Second : I don't want to be dragged in a role where I have to defend something and where you are the attacker. I don't like Mac<>PC wars, I don't like Quad<>DD5.1 wars, I don't like LP's<>CD's wars !

I wanted to understand more about surround in general and I decided to do a search on the Internet. I have read a lot about Quad, Dolby, DTS etc. I am not an expert in DSP, I am not an expert in whatever soundprocessing and I am certainly not THE expert to explain the advantages of Ambisonics. I don't even know if I can do that with my knowledge of English, but I try......
After reading a lot, I asked a simple question in an Audio Newsgroup : How can I make 5.1 from stereo ?
I got several reactions and some were just like yours : use your hardware ! Well, I got a nice Yamaha receiver that got excellent reviews from users all over the world and especially they liked the excellent DSP built into it. I can emulate 5.1 with it and yes... it sounds like real 5.1. Just like you say about the Denon.
But.......... I don't like 5.1 and I don't like quad. I always had that feeling of something missing in both. As I already stated : I am NOT an expert, I react only by my own feelings, by what I hear and try to explain it as simple as possible. Quad : this has IMHO nothing to do with surround, it is just extended stereo. And in some mixes it is terrible. A drummer from the left front channel, a singer from the right front channel, a guitarplayer from the left rear and a bass from the right rear. But..... where the heck is the band ?????
In quad they sometimes overact the separation of the channels. And this brings you to another thing missing, which will come back in the 5.1 mixes and I will discuss that later on in this reply.
5.1 : I am very disappointed with a lot of music in 5.1. Comparing it with quad, it is at least more surround, but most of these mixes suck. There are only a few producers who use that format to the limit and one of them is Alan Parsons with his DTS version of "On Air" . IMHO still the best available 5.1 mix.
As you know, the DTS catalog is small..... I've heard a lot of the original DTS cd's (and then I mean the music that was officially released on DTS !). Nothing comes close to "On Air"
For movies however, 5.1 is excellent ! I love the way some of the producers give you the feeling you are "in the movie". I DID dive in some of the scenes of "Saving private Ryan", because I got the impression someone was shooting at me from behind :-)
Back to music however, there are not that many 5.1 mixes (DD or DTS) that do something extra to the music. A lot of these can be heard in stereo without losing much of the effect. Why? Because on most tracks there is no effect ! There isn't hardly any music that in 5.1 gives you the same sensation as some movies do ! I hope I have put this part in an understandable way in words !
So obviously there must be something wrong in quad and 5.1 .
And here I come on to what that is : the Sweet Spot.
Quad and 5.1 can only be appreciated fully when you sit on the sweet spot and your room is capable of placing the speakers where they SHOULD be ! Is that surround ? One spot in your room where you can fully enjoy it, but moving one meter to the left or right and away goes the good effect ! Is that surround ?? I don't think so !
In my definition surround means you are surrounded by sound. Sound coming from everywhere... And that's not the case with quad or 5.1 ! Oh, so they say : then we just add some more speakers. 6.1 , 7.1 , 8,1...... where the heck does that stop ????
And here I come back to the DSP's in receivers : yes, some do an excellent job and give you the feeling you are listening to 5.1, but the above mentioned disadvantages still are there. And there is also another thing to keep in mind : not everyone has the ability to spend huge bucks ! Not on hardware and.... not on software. that's why a lot use illegal software , but illegal hardware ??? LOL !! It is not there !! So a lot of us stick to the cheaper range of the "surround" sets. And.......... get disappointed ! You will be surprised to know how many people there are with complete surround sets, cheaper than one piece of Soundforge !

Ok, back to where I was :-)
So because I was disappointed, I looked further and then someone pointed me to Ambisonics. I visited all the websites available and read all about it. I didn't understand more than half about the mathematical explanation, I didn't have a clue about recording, but as these were university professors, I believed them. I had no reason and not the knowledge to judge their opinions ! But what I did understand out of all these websites, was that there was a system that could give you REAL surround. A system where it didn't matter if you moved one meter from the sweet spot. I was impressed and needed to give it a try. And I did.... but only after I got in contact with another Alan Parsons fan who pointed me to a review of AP's album "Stereotomy". Woowww, Alan recorded that in Ambisonics and the reviewer was very enthusiastic about the sound in stereo. (We are talking vinyl here !!! not CD !!!) This reviewer (and he was not the only one) said he had never heard an album which such a clear sound that gave you the impression of some sort of surround. (Mind you : we are still talking about stereo vinyl !) So I thought : if such a big producer ("Dark side of the Moon" ! and all the Alan Parsons Project albums) already used a system in his recording in the beginning of the 80's that gave a critic reviewer a sense of surround and this same producer now made the only reasonable DTS cd...then I just have to try it ! If he does it, it can't be wrong, huh ? Of course I am biased : I am a fan !
Ok, that same guy gave me instructions on how to do it and with his help I developed the guide as published here.
And now comes the difficult part for me : explain the difference between Dolby Digital and DTS and Ambisonics. Well, I tried above a part and only can add here this : the "sweet spot" in ambisonics is a lot bigger than whatever other surround system. Ambisonics is not based on separation of channels. Recordings with only two Ambi compatible microphones contain all the information ! You don't need 5 or more microphones.
And as Stereotomy was recorded with Ambi compatible mics : a conversion to DTS (actually : 5.0) was just the next logical step.
And boy oh boy.... it sounds so good ! There is only one judge here : my pair of ears ! They won't lie to me ! They don't try to convince me what is a better format, they just let me listen to music and decide for their own what I like most !
OK, so I converted a stereo to DTS (5.0) and it sounded superb ! But...that stereo already had all the information in it because it was originally recorded with Ambi mics. So I searched further and found out that you can emulate that kind of recording by adding "roompulses" to the stereosound and encode it to Ambisonics.
And so I came to the final guide !

Ok, that was the history and the reason behind it, now of course you want to hear a sample ?

Tomorrow I will publish somewhere online one sample that you can play on your set. I will give you the address tomorrow. It will be a conversion from a stereo track to a DTS track. Just burn it on a CDRW and play it. If you don't like the music.... don't let that stand in the way of the experience of REAL surround.

Is this now the ultimate soundexperience in surround ? No, it is not ! Ambisonics has some disadvantages too ! I believe (Note : I am not technical, only repeat what I understand from what I've read) it also works with differences in soundpressure. And these differences sometimes lead to an effect where instruments are floating between speakers. I cannot explain this in English :-( Something like someone is singing and you hear his voice in one speaker and one thousand of a second later in another speaker and then back again...
So not every music has the same great effect ! It works better when the stereo signal already had a clear separation and it definitely works better with symphonic music and music where sound is already in stereo floating from left to right or with sound that has a lot of pressure. (Pink Floyd, Yes, ELO, Alan Parsons, Techno, House !)
it works worse on music like traditional rock or music with only a few instruments (For instance a singer with an acoustic guitar. But hey : does that work good in Quad or DD or DTS ??)

Well, it was a long reply.... tomorrow an example !

good night.........


Eye of Horus

[Eye of Horus]

Ok, I promised a track .
It is a 7 minute wav-file. just burn it on a CDRW, so you can always delete it :-)
(DON'T FORGET : IT IS DTS, SO ONLY PLAYABLE WHEN YOU HAVE A DTS SYSTEM !!!)
It is about 77 MB, so I had to place it on 2 different servers.
Use Winrar3 to extract the wav.

Here are the links :

http://thor.prohosting.com/miempie/Track06.part01.rar
http://thor.prohosting.com/miempie/Track06.part02.rar
http://thor.prohosting.com/miempie/Track06.part03.rar
http://thor.prohosting.com/miempie/Track06.part04.rar
http://thor.prohosting.com/miempie/Track06.part05.rar
http://www.xs4all.nl/~appfan/Track06.part06.rar
http://www.xs4all.nl/~appfan/Track06.part07.rar
http://www.xs4all.nl/~appfan/Track06.part08.rar
http://thor.prohosting.com/miempie/Track06.part09.rar

BTW.... they won't be there forever !!!

March 20, 2003 : REMOVED THE TRACKS.

So download them as soon as you can......
And..... give us your comments here !!

best wishes,

Eye of Horus

[lucindrea]

hmm .. slight problem on this end ... and this may or may not be the fourm for it but here goes
i have a audigy gamer connected to creative's inspire 5700 speakers connected via spidif out and i have the card to send eveything digitaly ... anyone know how i can listen to DTS tracks? the 5700 has the dts logo on it .. the thing does great with DVD's , i just cant get anything with DTS to play anything but white noise ... acctully an ac3 plays fine unless it has the wav ext. needed for cda then it's also noise.

[MaTTeR]

Originally posted by lucindrea
i just cant get anything with DTS to play anything but white noise ...
Most likely it's because the card and/or driver is molesting the stream and then the amp isn't recognizing it as a "pure" DTS stream. Awhile back I ended up running a cable from the digital out connector on my CDROM to the amp directly in order to get around this problem. It's a solution but not a very pretty one. Think we have a few threads in the forum also discussing this as well....not sure how long ago it was though.

[Eye of Horus]

Did some reading.....
Problem is that it assumes a 48 Khz stream with DTS and the DTS on CD is of course 44.1 Khz.
You shouldn't be able to play an official DTS CD (www.dtsonline.com) too.

Don't know about how to fix it..... suppose you have to do some searching ! Some in alt.audio.dts had the same problem.......

EoH

[DIggedy]

i'm curoius on how you do your 5.1 ...
i can take a stero file and make a mono for center and lfe .. just cant figure how to remove the center from the left and right .. also the dynamics of the rear are hard to get working also


I've yet to work that out myself but don't mind just using the original stereo channels for Front L&R, I find it gives the front soundstage a 'presence' that still improves the sound. How do you create your LFE after converting to mono? I use the Paragraphic EQ to isolate the low frequencies.
I also know what you mean about the dynamics of the rears... what I do to help this is to use Paragraphic EQ on the original stereo file to isolate the lower end (about 80-60hz down) then Paste mix this signal into the rear channels.
If you guys are interested I might do a guide to my process... I'd be interested to see what others think of the end results.

[vvlad]

Hi
I should say that it is very interesting technology
but whe I've tried to do it myself I don't get superb result
it seems that I do something wrong

problem:
AC3 i've got and tried to play in WinDVD plays faster then source file
Does anybody get same problem?

[drpaulng]

It's true.
This Track06.part09 is right. I just made a mistake by looking at the file size alone. I'll listen to the track first. Thank you very much.

[drpaulng]

http://home.gigigaga.com.tw/paulng/Amisonics.pdf

Please refer to the above pdf document for my quick report on the issue. This report is made with a brief listening session with my first impression. I do appreciate what Eye of Horus has shown us all. But I am a little sketpical on the selection of stereo tracks for encoding because not every stereo tracks recorded into a CD is pre-encoded in ambisonics (just the same for Dolby Prologic). So I dig a little deeper and discussed in my report.

[Eye of Horus]

[QUOTE]Originally posted by drpaulng
[B]http://home.gigigaga.com.tw/paulng/Amisonics.pdf

Tried the URL for several times now, on several days. Cannot get any connection !

EoH

[arman68]

I have read the full thread and articles linked from various posts.

It is very exciting news and I cannot wait to try it out. Like you I have always been disatisfied with current surround technology and felt that something was missing. Maybe it comes from having worked as musician in various band, therefore knowing how music should sound from 'inside'.

The only problem I have is I cannot decode DTS, and many people are like me.

However my hifi can decode Dolby Surround sound. I am well aware there would be a loss converting the tract to dolby surround, but it should still preserve a great deal of the ambisonic surround mix. After all, from what I understand, the ambisonic process is more of a recording and encoding process than decoding.

Let me know what you think.

[Eye of Horus]

As long as it is DD5.1 > no problem !
But encode it with 640 Kb/s for the best quality.
And DON't reduce the rear channels with -3 DB (If I remember correctly, that is standard in softencode).
I prefer DTS only because it has a better sound (1536 Kb/s), but to say the difference is huge.....no !

BTW still wish that like most modern standalone DVD players can play MP3, they make a model that can play AC3 ! (like the SB does)


regards,

EoH

[drpaulng]

:confused: Don't know why people cannot see the pdf, I add another address here:

http://www.geocities.com/drpaulng/Amisonics.pdf
or
http://home.gigigaga.com.tw/paulng/Amisonics.pdf

Please refer to the above pdf document for my quick report on the issue. This report was made with a brief listening session with my first impression. I do appreciate what Eye of Horus has shown us all. But I am a little sketpical on the selection of stereo tracks for encoding because not every stereo tracks recorded into a CD is pre-encoded in ambisonics (just the same for Dolby Prologic). So I dig a little deeper and discussed in my report.

I have tried making some MONO to STEREO, then extended to 5.1.
DTS encoding is better than DD encoding, absolutely!

[Eye of Horus]

Originally posted by drpaulng
:confused: Don't know why people cannot see the pdf, I add another address here:

http://www.geocities.com/drpaulng/Amisonics.pdf
or
http://home.gigigaga.com.tw/paulng/Amisonics.pdf

Please refer to the above pdf document for my quick report on the issue. This report was made with a brief listening session with my first impression. I do appreciate what Eye of Horus has shown us all. But I am a little sketpical on the selection of stereo tracks for encoding because not every stereo tracks recorded into a CD is pre-encoded in ambisonics (just the same for Dolby Prologic). So I dig a little deeper and discussed in my report.

I have tried making some MONO to STEREO, then extended to 5.1.
DTS encoding is better than DD encoding, absolutely!

OK, I am glad that some of the original scepticism is gone !
But yes..... I tried to fool you and you didn't fall for it !
My compliments !!!!
The track was from the indeed with Ambisonics recorded stereo albumn from Alan Parsons "Stereotomy".
Well.... now a fairer test. This is from a normal recorded stereo CD where the Ambisonics is added with my software method. For all of you who are still hesitating if it is worth the effort and time !!
It is a track from Pete Bardens. A bit new age, but Bardens played and wrote the rock album "Keats", which was actually Alan Parsons but without the orchestra and synthesizers.
So try this track ! Because this is really the ultimate test !! A stereo track converted into B-format (the WX and YZ files) and encoded into DTS.

http://thor.prohosting.com/miempie/Track01.part1.rar
http://thor.prohosting.com/miempie/Track01.part2.rar
http://thor.prohosting.com/miempie/Track01.part3.rar
http://thor.prohosting.com/miempie/Track01.part4.rar
http://thor.prohosting.com/miempie/Track01.part5.rar

REMOVED the tracks on March 20, 2003.



Happy listening !

EoH

[drpaulng]

The song.rar's are not ready to be downloaded...again?

ftp://drpaulng.myftp.org/

Try these 2 files, one .mp3 (mono) the other one .ac3 5.1 channel. Compare the difference.

These files won't last forever...(like your files).

[specise_8472]

After fixing the bug in the Aurora Plugin that kept crashing cooledit I am very impressed with the results. I am playing around at the moment with adding the LFE channel into DTS encodes. I have found that if just isolating the low frequencies (85hz - 25hz) from the original track and using them does not work as for some reason the length is different, and surcode complains.

Anyway thank you for giving your time to come up with a way of doing this, will give me hours of pleasure.:)

[Eye of Horus]

Originally posted by specise_8472
After fixing the bug in the Aurora Plugin that kept crashing cooledit I am very impressed with the results. I am playing around at the moment with adding the LFE channel into DTS encodes. I have found that if just isolating the low frequencies (85hz - 25hz) from the original track and using them does not work as for some reason the length is different, and surcode complains.

Anyway thank you for giving your time to come up with a way of doing this, will give me hours of pleasure.:)

Interesting idea !
Let me explain : I had a subwoofer that was not amplified and couldn't use the LFE channel on the receiver. The receiver has the option to leave out the LFE and mix it through the 2 front speakers. Cables from the receiver to the left and right front channels went into the subwoofer and came out again. So the subwoofer got it's bass from the two channels. So in my situation it didn't matter that it was 5 channels instead of 5.1 !
Two days ago however, my new surround speakers arrived with an amplified subwoofer that has two options of connecting to the receiver. One is as I described and the other is to the LFE channel on the receiver. As the cables were lying there already, I made the same connection as I had with the old sub.
Now first my question : What's the best option ? This way of connecting or use the subwoofer with the signal from the LFE on the receiver ?

Ok, on to an answer or advice : You end up with 5 mono channels . Try the bass from those ! There shouldn't be a problem then I suppose with the length !
Don't forget you added something to the original stereo file (the roompulses !) during the conversion.

BTW I haven't tried it yet... it is just a suggestion (or speculation !)

kind regards,

EoH

[Eye of Horus]

Originally posted by vvlad
Hi
I should say that it is very interesting technology
but whe I've tried to do it myself I don't get superb result
it seems that I do something wrong

problem:
AC3 i've got and tried to play in WinDVD plays faster then source file
Does anybody get same problem?

See above the reply to Lucindrea

EoH

[specise_8472]

I have just tried your suggestion.

I mixed the left and right channels, after besplit, into a single mono file, then isolated the low frequencies. Surcode took this without a problem. The lfe is there and can be heard. Sounds alright. But on my sound setup, when no LFE present I think the amp automatically sends all low freauencies to the sub anyway. As a lot of music CD's give the sub (powered btw) a good workout.
I actually tried isolating the lows then convoled them with W (just a random pick) and added this as the Z channel. The result was interesting. There was a noticable increase in the sub response, but the drawback was that there was a decrease in the very high ranges.
I think ambi-decode notices more low and saturates the higher in response.

How did you come up with the convole files in the first place?

[Eye of Horus]

Originally posted by specise_8472
I have just tried your suggestion.

I mixed the left and right channels, after besplit, into a single mono file, then isolated the low frequencies. Surcode took this without a problem. The lfe is there and can be heard. Sounds alright. But on my sound setup, when no LFE present I think the amp automatically sends all low freauencies to the sub anyway. As a lot of music CD's give the sub (powered btw) a good workout.

I think I dig into the manual again !
I was afraid that connecting the sub to the LFE would lose bass when playing a 5.0 or 4.0 CD. Or put the bass in the front speakers !?
I hope some more people can shine a light on this......
BTW can you tell the exact steps you used, so I can add it to the guide ??


I actually tried isolating the lows then convoled them with W (just a random pick) and added this as the Z channel. The result was interesting. There was a noticable increase in the sub response, but the drawback was that there was a decrease in the very high ranges.
I think ambi-decode notices more low and saturates the higher in response.

How did you come up with the convole files in the first place?

Well, In the guide I told that I developed the method together with a friend . He supplied me with the room pulses.....

regards,

EoH

[specise_8472]

BTW can you tell the exact steps you used, so I can add it to the guide ??

1 - Take the left channel after besplit and turn it back into stereo 100% left.

2 - Take the right channel after besplit and paste it into the right channel of the stereo file you just created.

3 - Filters/FFT filter - make a custom filter that only allows the low frequency range you want to be isolated. Filter the file.

4 - Return it to a mono file and then use as LFE channel in surcode.

I played around with the lfe file a bit and found that if you use the Amplify filter on it, you produce a marked difference in the LFE responce. I LLLOOOVVVEEE having the floor shake/vibrate under foot in the lounge (concrete floor) (VIRUS DTS is good for this during the storm at the beginning). Try 200% amplification then go from there depending upon personal tastes and what the track is/should be?.

I am still experimenting with different approaches and will keep you imformed. As I said before - "hours of pleasure".

Cheers

[drpaulng]

With the aid of somebody great here in this forum, I finally get in touch with the ambisonic codec. I'll try to make another listening report comparing my method (not ambisonics) and the newly acquired ambisonics.

The sample I made a few days ago was derived from a MONO sound source. After generating a stereo image by the pan/expand filter in CoolEdit, the surround channels were generated by adding the concert hall effect with a large cut down of direct original compoents. A 0.02sec delay was added to make a sense of back/surround rebound. To complete the 5.1 mix (actually 4.0 should be alright) the center was simply a original mono with some cut in dB level, while the LFE is made from the center mono via low frequency filtering at 120Hz roll-off. The result was close to what is heard in the official production (only a short video clip). But I was still unsatified with my simple method. Hopefully, if the ambisonic codec works, I'll adopt this method. Before I start the ambisonic project, I would like to ask what is the environmental condition that the impulses (WXYZ) for the convolving derived from. I think it is important because different kind of impulses should make different ambisonic sounds.

[Eye of Horus]

Originally posted by drpaulng
With the aid of somebody great here in this forum, I finally get in touch with the ambisonic codec. I'll try to make another listening report comparing my method (not ambisonics) and the newly acquired ambisonics.

The sample I made a few days ago was derived from a MONO sound source. After generating a stereo image by the pan/expand filter in CoolEdit, the surround channels were generated by adding the concert hall effect with a large cut down of direct original compoents. A 0.02sec delay was added to make a sense of back/surround rebound. To complete the 5.1 mix (actually 4.0 should be alright) the center was simply a original mono with some cut in dB level, while the LFE is made from the center mono via low frequency filtering at 120Hz roll-off. The result was close to what is heard in the official production (only a short video clip). But I was still unsatified with my simple method. Hopefully, if the ambisonic codec works, I'll adopt this method. Before I start the ambisonic project, I would like to ask what is the environmental condition that the impulses (WXYZ) for the convolving derived from. I think it is important because different kind of impulses should make different ambisonic sounds.

I will ask my friend which values he used. If I remember correctly he adjusted them to his own roomsize which is a small room.
Again I want to repeat here that you can make your own roompulses with the software from http://www.catt.se . You can input your own room size and create pulses. It is an excellent piece of software, but it's implementation of Ambisonics is not that good. Converting is all done with 16 bit, so the end results sound a bit flat and dull. But the roompulses have nothing to do with it, so ask by email the maker a 30 days key, make thousands of roompulses in these 30 days and keep the pulses ! :-)
The package includes several orchestrahalls, but you can make your own too by inputing the size of your room, the listening position, the reverb and absorbtion etc. etc..

BTW what are your opinions about the second sample ?

regards,

EoH

[Eye of Horus]

Originally posted by specise_8472
[B]BTW can you tell the exact steps you used, so I can add it to the guide ??


3 - Filters/FFT filter - make a custom filter that only allows the low frequency range you want to be isolated. Filter the file.

You need to tell exactly the steps and how you've done it. This description is too vague for beginners !
So can you please make it simpler, so that even I know what to do ?
:-)
The guide should be on a newbie level, so it must be clear and precise.

TIA,

EoH

[DIggedy]

What's the best option ? This way of connecting or use the subwoofer with the signal from the LFE on the receiver ?

For true 5.1 use the LFE output to connect to the sub. Connecting the sub through the fronts you'll be missing information when you listen to a true 5.1 source, which would be a waste, especially when you have the ability to do so. The only disadvantage is as you said when listening to 5.0 the sub will be inactive. Maybe you could connect your old sub to the fronts and the new one to the LFE channel giving you the best of both worlds (and monstrous bass when watching movies!). Mixing the LFE into the fronts is a good option when you don't have a sub at all, but I wouldn't recommend it when you are using a sub (the sub will not only be reproducing the bass from the LFE channel but also the bass from the front channels, which could produce 'boomy' and muddled results). Do your front speakers have a high dynamic range? If they can cope with a full range signal then you won't lose bass when listening to 5.0.

But on my sound setup, when no LFE present I think the amp automatically sends all low freauencies to the sub anyway. As a lot of music CD's give the sub (powered btw) a good workout.
It depends on the source (and your reciever settings). If it is a 5.0 discrete source, you won't get bass in the LFE channel, but CDs are 2.0 stereo so the reciever will apply a low pass to the signal and send it to the LFE.

[drpaulng]

What we have now are a few choices of 5.1, 6.1, and even 7.1 encoded sounds realized by MLP, Dolby Digital, and DTS (hopefully the future ambisonics). After listening repeatedly to the modern commercial packages in DVD Audio, you'll get a general feeling of better sound playing through MLP (DVD Audio), than DTS, and the last acceptable is Dobly Digital. But it is not very natural to hear in many cases the background vocals being placed in your back. Take an example of Black and White Night of Roy Orbison (DVD Video), the background vocals and strings were placed in your back. Another example is Olivia Newton-John's One Woman's life Journey (DVD Audio). You'll find that both Olivia and Elliot Scheiner agree to put their name together for the 5.1 mix as producers...but you'll also get a feeling of UNREAL backup vocals standing in your back too. What happened to these commercial packages? They are exciting enough to fill your room with surround sound discretely distributed among the front and back/side speakers, but it is absolutely unnatural to let the front stage components sounds in your back! So, perhaps I would repack these into ambisonic by first downmixing the 5.1 to stereo for a better surround (more natural but not so exciting). HOWEVER, what I did last night following the steps posted at the very beginning on this topic ended up in a slightly accelerated 5.0 mix. What is the problem behind this slight acceleration? :confused:

My impression for the second sample is: it's natural and you get a feeling of some sound going past you from front to rear.

Right now I have a project trying to bring the MONO to 5.1 surround. Anybody has idea on this?

[drpaulng]

LFE is intended to be played it's role only a brief moments of earthquaking lows in a movie so that you won't sacrifice the details in the front mix. In music playing, we already have 2 large speakers delivering all lows and highs in stereo. Playing 2.1 sound would be unnatural if you have the LFE volume tuning too high to make the sound BOOM. The Ambisonics 5.0 is OK for everyone who has already enjoy his stereo. If you separate the lows to the LFE from the front speakers, the necessary reason is that you have a brief moment of great extra lows to be delivered that would harm (make distortion at normal level) the front speakers had it not been separated. However, since everybody has an impression of incompleteness in 5.0, a slight touch of 0.1 added should be alright as long as the final mix is balanced with no exaggerated lows. A balanced lows and highs sound is natural and sweet.

In the movie, an ordinary dialogue scene would be about 70dB. A sudden boom from a thunder or bomb would bring out a 110dB shaking lows. If the dialogue and booming lows should be mixed together at 2.0, the diaglogue level of 70dB at 2m away from the speaker would need a very small amount of power. However, for a 110dB of shaking low frequency bomb at the same setting, the 2 front speakers would need to deliver 120W individually. For small speaker system such as those satellite speakers plus a subwoofer around a computer, a 2.1 re-distribution of low frequency energy is a good choice, but for the audiophile system with plenty of headroom in every speakers, even 2.0 mix would be OK. For music playing, there is very few occasions which need the real use of LFE like that in a movie, the LFE is thus not an absolute necessity.

[vvlad]

Originally posted by drpaulng
HOWEVER, what I did last night following the steps posted at the very beginning on this topic ended up in a slightly accelerated 5.0 mix. What is the problem behind this slight acceleration? :confused:

I got the same problem
For me reason is that when I encode multichannel wave (44.1 /16) to ac3 (640 /48) - ac3 file can be played through spdiff (I tried it on PowerDVD) but playes accelerated

when I produce ac3 (640 44.1) in SoftEncoder file is played well in PowerDVD through 4 or 6 channel
but not through sp/diff

I have asked this question here - and got some answers - the reason is that ac3 (or DTS i'm not expert in this things) standart should be 48 stream not 44.1

now I'm going find Surcode DTS and try encode there

sorry for my English I hope anybody will understand something...

[drpaulng]

May be I should try to encode ambisonics in 48kHz/16bit (if it allows).

[vvlad]

Originally posted by drpaulng
May be I should try to encode ambisonics in 48kHz/16bit (if it allows).

Ambisonic Decoder - Copyright 2000 Richard W.E. Furse.
Build: MNLib v1.15, 44.1kHz, Oct 1 2000.
Usage: ambidec [<optional parameters>] <input> [<output>]

Start screen of ambidec
So this version does not suppoer 48kHz/16bit

it seems for 48 ambidec should be rebuild by athors...

[DIggedy]

I have asked this question here - and got some answers - the reason is that ac3 (or DTS i'm not expert in this things) standart should be 48 stream not 44.1
DTS can be 44.1 - for DTS CD it must be encoded @ 44.1. That must be why I didn't get any speedup whatsoever when I tried it this way.

[Eye of Horus]

Today I will buy a cable for the LFE and I guess the best thing is to just try it out.
I've read teh manual again and it says that if a subwoofer is connected to the LFE, it will send all signals below 90 Hz to the LFE. You can even have this in 2.0 mode or chose to have these through the front speakers.....

So I guess it is an option for me. (long live Yamaha !)

I'll keep you all posted on the results......


EoH

[specise_8472]

To Eye of Horus
Glad to hear about the LFE cable solved.

I have been plying around with the CATT FIREVERB & CATT ACOUSTIC programs lately. I was interested in hearing the differences in encoding over your method. I was particually intereseted in the fact that the program will handle sample rates upto 96000 at 32 bit. At the moment I am modeling my speaker setup so as to get the best "sound" out of it.

But work has been very hectic at the moment. A ray of sunshine though :( I tried to rip my calf muscle in two today at work, so am on light duties:D So I hope to be able to spend time at home playing around some more.

[drpaulng]

The accelerated song.ac3 in 44.1kHz/16bit format should be a bug in PowerDVD because the same song when encoded in Dolby Digital wav instead of .ac3, plays normally (burned to a CD-R as Dolby Digital CD).

Softencode accepts both 44.1kHz and 48kHz and the Dolby Digital 5.1 Mixing manual also states about the Dolby Digital CD (just like the dts CD). It is however not commercially recognised because most of the DD 5.1 we heard are in form of DVD Video which is encoded in 48kHz instead of 44.1kHz.

While it is quite alright with ambisonic surround production derived from mere 2 channel stereo source, I would like to see if anybody could provide more room acoustic impuleses for the appropriate setting for different kinds of environments such as live concert or in small club/bar. Sonic foundry provides with all kinds of impulses derived from different environments. With my method, I make use of these impulses for the appropriate surrounding effect that suit the sound stage. The sound that is heard in backseat row within a single environment would be closely simulated with greater cut of direct component. The front row audiences would thus have a much more direct components accordingly. A careful design of relative balance in the reverb compoents generated by the impulse characteristics as well as the relative sound level between the front and surround speakers would bring out a lot of different combinations in 5.1 mix sound design dervied from the stereo scratch. With suitable sound delay (in tens of mini-sec), a more dramatic effect could be generated too. Softencode provides with "90 degree phase shift" option for the surround mix where Surcode does not apply (even then, no sound editors seem to have this 90 degree phase shift effect). My MONO to 5.1 project is now nearly complete. A more detailed listening report would be out soon after I clear all my jobs in hands.

[Eye of Horus]

Originally posted by drpaulng
The accelerated song.ac3 in 44.1kHz/16bit format should be a bug in PowerDVD because the same song when encoded in Dolby Digital wav instead of .ac3, plays normally (burned to a CD-R as Dolby Digital CD).

From what I've read in alt.audio.dts the problem is the soundcard (in that peculiar case a SB audigy)!



Softencode accepts both 44.1kHz and 48kHz and the Dolby Digital 5.1 Mixing manual also states about the Dolby Digital CD (just like the dts CD). It is however not commercially recognised because most of the DD 5.1 we heard are in form of DVD Video which is encoded in 48kHz instead of 44.1kHz.
Side note : only the newer version of Surcode !!


While it is quite alright with ambisonic surround production derived from mere 2 channel stereo source, I would like to see if anybody could provide more room acoustic impuleses for the appropriate setting for different kinds of environments such as live concert or in small club/bar.


See my tip about www.catt.se !
Perhaps a good idea to built a bank of pulses ??

[drpaulng]

ftp://drpaulng.myftp.org/Compare.rar

After using the method of Eye of Horus, I got my amisonic 5.0 mix compared side by side with the official 5.1 and my own 5.1 mix. Here is the report...it won't stay there forever...so :D .

[drpaulng]

ftp://drpaulng.myftp.org

http://drpaulng.myftp.org/Compare.rar

I don't know if there is any problem...but it seems to be on the side of the ISP. May be you should try to use ftp. If it won't work, I'll try the hard way to find some other places to hold it.

[AudioVideoMaster]

Well I gave Eye of Horus's procedure a whirl tonight on about 7 minutes of stereo audio I captured from a Looney Tunes cartoon (44.1Khz/16bit). I used Soft Encode for the 5.1 encoding; don't have Surcode.

The results:
Well the 5.1 version does sound abit better/wider (I guess that's how to explain it) than the original 2 channel version. But I developed pops and or cracks in the 5.1 audio that wasn't there in the 2 channel version. (Especially in the rear channels). They may be there in the original....it may just be this procedure amplifies them.

Another problem was that I couldn't make a VOB file with the resulting 5.1 audio file because it was sampled at 44.1Khz and not 48Khz. Which is DVD standard I believe.

My conclusion is that I still like the way my hardware receiver emulates the surround sound automatically from a stereo source.

Just my 2 cents. :)

Comments are welcome.

[Eye of Horus]

Originally posted by AudioVideoMaster
Well I gave Eye of Horus's procedure a whirl tonight on about 7 minutes of stereo audio I captured from a Looney Tunes cartoon (44.1Khz/16bit). I used Soft Encode for the 5.1 encoding; don't have Surcode.

The results:
Well the 5.1 version does sound abit better/wider (I guess that's how to explain it) than the original 2 channel version. But I developed pops and or cracks in the 5.1 audio that wasn't there in the 2 channel version. (Especially in the rear channels). They may be there in the original....it may just be this procedure amplifies them.

Another problem was that I couldn't make a VOB file with the resulting 5.1 audio file because it was sampled at 44.1Khz and not 48Khz. Which is DVD standard I believe.

My conclusion is that I still like the way my hardware receiver emulates the surround sound automatically from a stereo source.

Just my 2 cents. :)

Comments are welcome.

Ok, then here my comment :
a. pops and cracks ? or you did something wrong or it was indeed in the original. If it was in the original.....then what do you expect ??? Getting something good from something bad ???
b. Considered resampling the 5 mono 44.1 files to 48 ????
c. So you cannot put it in VOB and you prefer the receiver ! Please gimme some clues how you put the sound from your receiver in a VOB !
(just kidding !)

But.... my main criticism still stands : using a bad file will just give you a bad file back !
Now try it again with a good sample and then please try to get a hold of surcode. The difference between a DTS encoded file and a DD5.1 is already like the difference between AM and FM radio !

Just my 3 EURO cents :-)

EoH

[AudioVideoMaster]

Originally posted by Eye of Horus

b. Considered resampling the 5 mono 44.1 files to 48 ????
EoH

Well I started out from the very begining with a 2 channel 48khz file. I continued to stay at that sampling until I got to the 'ambidec.exe' part and it told me that it would not accept my 48Khz .wxyz file, they needed to be in 44.1Khz sampling. So I went back a couple steps changed the sampling rate to 44.1Khz. Went thru the steps until I had a 5 channel 44.1Khz wave file.

I knew about the 48Khz standard for DVD's. So I tried to bring the 5 channel wave file back into Cool Edit and convert the sample rate. But all it done was merge all five of the wave files together into one mono wave form. I went ahead and converted the sampling rate and saved it fine, hoping that the channel info in the file might still be there. But when I brought it into Soft Encode it only showed up as a mono left channel.

As far as the hardware receiver thing I refered to, all I meant was that when the audio is sent out via S/PDIF from my computer and sent to the receiver, the receiver emulates the surround sound quite well. I have the Cambridge Soundworks DTT2500 5.1 system. When I have the Dolby Surround option selected it emulates the surround sound from a stereo source from a music file on the computer or a TV show etc. (I watch TV on the computer here in the dorm room. It saves me from having to bring a regular TV with me. :) )

[lucindrea]

use the method for creating 5 wavs for DTS and convert each wav to 48khz .. soft ejncode should handle tose just as easly as the 5 chan wav.

[AudioVideoMaster]

Originally posted by lucindrea
use the method for creating 5 wavs for DTS and convert each wav to 48khz .. soft ejncode should handle tose just as easly as the 5 chan wav.

OK, I'll give it a go. Thanks! :)

[drpaulng]

http://www.kellyindustries.com/

Anyone has experience on this software?

[iGorland]

Without trying to diminish the Eye of Horus's very nice guide, there is another way to get 5.1 ac3 or dts sound out of a CD. For those with big bucks in pockets, get Minnetonka's Mx51 software. Load your CD track, and it will produce 6 mono tracks that you can later encode in AC3 or DTS format (including LFE). In addition, you will have a complete control over panning and editing of the sound. Demo version is also available somewhere on the net.

[oddball]

If you are suitably bored enough maybe you can go through Horus process. To be honest I just got a new amp with Dolby Pro Logic II and I find it simply superb! I don't have a problem with 'sweet spots' it sounds good no matter where I sit within the image. One thing I did have to do however was adjust the 'spread' across the front pair and center speakers. It was all set to come from the center with very little in the stereo pair. After balancing this using some recommendations from another website it sounds stunning!

The website also does a review of the different methods like DTS Neo 6 and he wasn't too impressed. He gives DPL II the thumbs up.

Link here http://www.hometheaterhifi.com/volume_8_1/dolby-prologic2-3-2001.html

[drpaulng]

Somebody suggested Mx51 for 5.1 surround panning. However, after some initial Demo testing, I failed to get started with the demo version. However, I found the 5.1 surround panning in Sonic Foundry ACID Pro 4.0. It is the easiest method to get stereo to 5.1 with software like these (Panhandler, Mx51...) but the sound would be different from ambisonics. After some tests I'll try to write a report on this.

[drpaulng]

iGorland's Suggestion for Mx51 is good. However, I did download the demo version but failed to initiate the program. So I tried to look for something else on the web and found this one: Sonic Foundry ACID 4.0.

This program can handle surround 5.1 panning too. I tried it with satisfactory result by applying the very same principle to modify the surround cannel live ambiance characteristics derived from real acoustic impulse (concert hall in this case of live recording). The original panning scheme from ACID is very simple. With a careful planning of the center, surround and LFE in sound pressure levels, one can get a balanced surround mix according to his need. A real time monitoring is best achieved by output through 3 stereo sound cards.

http://www.geocities.com/drpaulng/ACIDsurround.jpg

[sebus]

Any chance fgor "idiots guide" on the subject

Thanks

sebus

[drpaulng]

http://www.geocities.com/drpaulng/

This is An Introduction to the making of surround sound from stereo exemplified by ACID Pro 4.0.

[sebus]

Really nice of you

Thanks

sebus

[oddball]

Of course the biggest downside is that you are compressing the CD stereo to a lossy format.

[specise_8472]

AMBISONICS is still the way to go:

Shortcomings of conventional 5.1
In what ways is G-format actually superior to conventional 5.1 surround? To answer that question we need to consider briefly the shortcomings of conventional approaches. Conventional surround (and stereo!) recordings localize sound sources simply by means of level. To place something mid-way between front and rear left, for example, you simply apply equal levels to these two loudspeakers. This is not very satisfactory, however, for several reasons. Human hearing relies on different combinations of level, phase and arrival time -- as well as other factors -- to localize sound sources, and not solely on level. Loudspeakers in a conventional stereo pair are generally placed 60 degrees apart with respect to the listener. In this configuration, the ears can hear sound coming from both speakers and differences in level are interpreted as phase differences and provide localization information. This effect begins to fail as the speakers are moved further apart, and by the time you get to a 90 degree front stage -- as was found in quad setups and is still recommended in some quarters today -- a significant "hole in the middle" has developed (it's one reason for the center front channel in 5.1).

Level-only localization is poor at best in the rear, and virtually nonexistent at the sides. As a result, conventional surround tends to suffer from poor inter-speaker imaging. Sounds are sucked into the speakers and it is hard to get them to appear from anywhere else -- as a result, some engineers and producers have taken to deliberately placing sounds only in the loudspeakers: a simple and effective way out, but one that is rather limiting in creative terms. Another problem is the fact that as only level is involved, moving about within the listening environment changes the positions of sounds in the replayed image. Move to the left and you hear more of the left speakers, so the balance appears to move to the left. Move to the rear and the image moves backwards. This is not satisfactory: not only is the image unstable, it is only "right" in a very small spot in the center of the array -- the "sweet spot".

Ambisonics, on the other hand, relies on phase as well as level and other cues to provide localization information. As a result, it is less sensitive to listener position. In fact, a favorite sales technique for Ambisonics is to set up the system driving near-field monitors in a studio control room and ask the client to come into the room during replay of an Ambisonic recording. A remarkably stable image inside the speaker array is generally experienced from outside the array, much to the client's delight. And the image changes little as you enter the replay area and sit down (unless, of course, you go right up to a loudspeaker). Contrary to belief in some circles -- largely on the part of people who think they know the equations on which Ambisonics is based, and therefore know what it would sound like if they ever listened to it properly -- the "sweet spot" for Ambisonics is very large.

Ambisonics also allows sounds to be positioned anywhere inside the listening area. A sound panned around the room at constant speed a constant distance from the listener should stay that way and sound the same all round -- it shouldn't jump from one speaker to the next, changing its timbre as it goes, and moving closer to and further from the listener, which is all too often the case with conventional 5.1. In Ambisonics, it behaves impeccably.

Judge for yourself.

[drpaulng]

Ambisonics is good...only that we lack support from both hardware and software in the commercial world.

Thanks to Eye of Horus who made the guide.

I think the alternative methods of making 5.1 surround from stereo source should be widely discussed. The more suggestions and methods, the more we can choose to suit everybody's particular need.

As a listener, we have limited choices....but as a sound designer, one can choose among lots of different combinations, pack them and share them to the listeners. The only bad thing we have to face is that we are limited to the stereo as our source of surround making. If we can access the original multi-track materials, I believe the surround making could be near-perfect with full manipulation.

[Kilyan]

Hy!

My question is about using a vhs stereo of a movie, with the one on dvd. The ac3 5.1 of the dvd can be decodec with besweet:

"D:\besweet\BeSweet.exe" -core( -input "g:\Bridget_assets\bridget_eng.ac3" -output "g:\Bridget_assets\bridget-" -6ch ) -profile( ~~~~~ Default Profile ~~~~~ )

Do I get the original mono files with this command line? if not please tell me which one to use.
So I got the 6 separate channels. Do the surrounds stay shifted by 90 degrees? or in what state are the mono files?

I also make a stereo mix of it, to synchronise with the english ac3.
The surround and lfe generally don't have dialogue in it. So I intend to use them with the vhs stereo to make a new ac3 5.1.

Splitting the vhs stereo to 2 48kHz mono, identical in length to the original ac3, and making a mix 50-50 of the vhs to a mono for the center it should be ok.

My main problem is that I only have a stereo system. So the sound will be strange or bad or what in surround?

What options should be checked or iinchecked in scenarist ac3 encoder?
Surround 90 shift for example? if the source was ac3 as above? does it contain the 90 shift in the mono files or not?
Surround and center mix level?
Dialogue normalization?


I was wondering if somebody already tried this already?

Thanks

[amorfeusz]

Thanks for a great guide Eye of Horus!

I just read on http://www.ramsete.com/Ramsete/convolution.htm

that CoolEdit Pro also has a convolution module, it uses an imp file format but you can create these from the three wave files - and then convolve.

However with the generic settings it doesn't seem to produce the same result as the Aurora convolution.. but I could be doing it wrong since I'm a complete noob in this respect.

Is there any way of using CoolEdits own convolve for the UHJ->B conversion?

Also, I found a great little program, ClipClear,to clear your windows clipboard by simply clicking on an icon in the system tray. It's freeware

http://www.moonsoftware.com/files/cclr311.zip

No need to close and open cooledit then.

-Amorfeusz

[FLeXO]

wow! works nice here i get some great quality only i can only edit with cooledit 2000 when i try to use cooledit pro 2.0 i get all sorts of error's anyone had some luck with pro?. ow and when i try to convert a large file like 80minutes the plugin tell's me i don't have enough memmory (i have 512MB!). shorcomming of the plugin? or am i doing something wrong?.

[amorfeusz]

I'm using CoolEdit Pro 2.0 build 2095 and have had no problems - apart from the Aurora Russian Roulette crash :)

I also tried converting a whole CD at once but even with my 768MB it wouldn't let me - Aurora doesn't appear to use virtual memory - I'll throw in my extra 256 later and will see what that changes. What I am just finishing now is a CD I ripped in two parts, each about 300mb. I'm planning to combine the 2 DTS waves later.

[FLeXO]

@amorfeusz

i think it has something to do with my memory cous you have some more then me and since pro uses more memory i think that's the problem here.

[drpaulng]

Does anybody know which Direct X plug-in does the job of shifting phase (+90 degree and -90 degree), so that we can make a surround track with left +90 degree and right -90 degree phase shift for the dts encoder (Softencode has the ability to add 90 degree phase shift to the surround track).

[specise_8472]

To drpaulng

I have been looking for this sort of plugin for a while, and no luck.

But goto
http://www.gamasutra.com/features/sound_and_music/19981218/surround_03.htm
and this is a good thesis on phase shifting and the pitfalls to look out for.
There is also source code to make your own software. (I have never got around to compiling and trying:( )

BTW I have been scrounging around the net a lot lately and have found some very good NUENDO plugins for B format processing and also some VST plugins for automaticaly creating ,stripping from surround channels, a LFE channel and also recombining the LFE back into surround channels : in case you don't want LFE channel.

Goto
http://www0.york.ac.uk/inst/mustech/3d_audio/welcome.html (http://)
for the B format processors

As for the LFE plugins I don't know where to find them :confused:

[drpaulng]

Maybe we can first encode using Softencode with phase shift for the surround track and then decode to individual mono wav files for input to the dts encoder. A collateral way - tedious but I think it should work.

[FLeXO]

I found a LFE splitter. but it's hard to get. Nuendo Surround Bundle

http://www.audiomidi.com/common/cfm/product.cfm?Product_ID=1150

[Eye of Horus]

Originally posted by FLeXO
wow! works nice here i get some great quality only i can only edit with cooledit 2000 when i try to use cooledit pro 2.0 i get all sorts of error's anyone had some luck with pro?. ow and when i try to convert a large file like 80minutes the plugin tell's me i don't have enough memmory (i have 512MB!). shorcomming of the plugin? or am i doing something wrong?.

I have 386 MB and it only converts upto 10 minutes. So I even had to split a 12 minute track :-(

EoH

[Eye of Horus]

Originally posted by amorfeusz
Thanks for a great guide Eye of Horus!

I just read on http://www.ramsete.com/Ramsete/convolution.htm

that CoolEdit Pro also has a convolution module, it uses an imp file format but you can create these from the three wave files - and then convolve.

However with the generic settings it doesn't seem to produce the same result as the Aurora convolution.. but I could be doing it wrong since I'm a complete noob in this respect.

Is there any way of using CoolEdits own convolve for the UHJ->B conversion?

Also, I found a great little program, ClipClear,to clear your windows clipboard by simply clicking on an icon in the system tray. It's freeware

http://www.moonsoftware.com/files/cclr311.zip

No need to close and open cooledit then.

-Amorfeusz

Thanks for this excellent tip !!

About the results.... when you use the Gatt package, results too are disappointing. I still prefer to do it "my way" !
It seems they all work the 16 bit soundfile directly and the endresult is degraded in quality.
I did DSOTM from Pink Floyd a few days ago..... love it !!!
I think we shouldn't compare Ambi with others but just accept it as another format !

EoH

[exrty]

sorry for this question But when i'm try yo play ac3-cd on my dvd player, it doesn't decode it and play it as a std PCM.

My dvd player is a scientific labs with an internal 6 channels analog decoder (DD).

Is there any way for the DVD player to know that the stream is AC3 ?

Regards.
exrty

[sebus]

What is ac3 CD? How did you make it?
Do you mean Dolby Digital CD where the wave files got created with Sonic encoder?

sebus

[Eye of Horus]

Originally posted by exrty
sorry for this question But when i'm try yo play ac3-cd on my dvd player, it doesn't decode it and play it as a std PCM.

My dvd player is a scientific labs with an internal 6 channels analog decoder (DD).

Is there any way for the DVD player to know that the stream is AC3 ?

Regards.
exrty

I wish there was !!!
It would save a lot of converting and space on CDR's......
The SB soundcard plays them without any problem, but there is no standalone DVD player that can handle AC3........

EoH

[drpaulng]

AC3 and DTS wav files are both acceptible formats for making audio CD. You can put AC3 wav files mixing with dts wav files within the same CD. The only problem is that you need to have an AV amplifier capable of decoding them into Dolby Digital and dts surround and the only way out is through the digital link, not analog. However, some high-end DVD player can decode them and output to the 6 channels analog output.

Softencode can output in both .ac3 and .wav format, Surcode dts cd can output in .wav adn .dts format. The .wav format for AC3 and DTS can only be listened to after being multiplexed into the vob files (in this case 48kHz/16bit) or burned into the CD (44.1kHz/16bit), while .dts and .ac3 can be directly decoded by softDVD player such as PowerDVD.

[drpaulng]

Anybody can shine us a light? How do we use Nuendo surround and it's Direct-X plug-ins.

[drpaulng]

http://www.muse.demon.co.uk/vspace/vspace.html

Vspace seems to be the solution for generating ambisonics encoded files without going through the tedious prepration procedure with Cooledit Pro. However, the example.vsp link is broken.

[specise_8472]

Here is Example Script:)
Have fun:cool:

No attachment:( Email me for script

[Eye of Horus]

Originally posted by drpaulng
http://www.muse.demon.co.uk/vspace/vspace.html

Vspace seems to be the solution for generating ambisonics encoded files without going through the tedious prepration procedure with Cooledit Pro. However, the example.vsp link is broken.

There are others...... like the CATT package, but the big question still is : is there any degradation in the sound ? I tried several ways to achieve the best results and the CE-method still stands as a rock in that aspect !

regards,

EoH

[Eye of Horus]

Originally posted by iGorland
Without trying to diminish the Eye of Horus's very nice guide, there is another way to get 5.1 ac3 or dts sound out of a CD. For those with big bucks in pockets, get Minnetonka's Mx51 software. Load your CD track, and it will produce 6 mono tracks that you can later encode in AC3 or DTS format (including LFE). In addition, you will have a complete control over panning and editing of the sound. Demo version is also available somewhere on the net.

Tried that one.... but... too difficult for me :-(
Perhaps easier if there was a Dutch version......
But anyway : For me the problem is in the panning and editing.

There are other ways and plugins too, but my problem is that they almost all expect a certain level of knowledge of soundprocessing. I don't have that !

EoH

[Eye of Horus]

Originally posted by drpaulng
Somebody suggested Mx51 for 5.1 surround panning. However, after some initial Demo testing, I failed to get started with the demo version. However, I found the 5.1 surround panning in Sonic Foundry ACID Pro 4.0. It is the easiest method to get stereo to 5.1 with software like these (Panhandler, Mx51...) but the sound would be different from ambisonics. After some tests I'll try to write a report on this.

Indeed it would.....
I don't think you can easily get the same results.
I find all those tryings of finding a way to make 5.1 from stereo a bit amusing, because : YOU CANNOT COMPARE IT WITH AMBISONICS.
It has nothing to do with quality or 5.1 . Ambisonics is just a completely different system !
You can make excellent 5.1 from stereo, you can buy commercial excellent 5.1 music, but.... IMHO you cannot compare it with Ambisonics.

I believe you wrote it yourself in one of your previous letters : listening to a group of musicians where the background voices are coming from the back channels. Completely unnatural ! And not surround at all !
I think that's still the main difference : original commercial 5.1 relays to much on tricks, while Ambisonics relays on surround !
Dolby and DTS are responsible for this by naming their systems surround..... IMHO it has nothing to do with surround !

EoH

[Eye of Horus]

Originally posted by specise_8472
AMBISONICS is still the way to go:

Shortcomings of conventional 5.1
In what ways is G-format actually superior to conventional 5.1 surround? To answer that question we need to consider briefly the shortcomings of conventional approaches. Conventional surround (and stereo!) recordings localize sound sources simply by means of level. To place something mid-way between front and rear left, for example, you simply apply equal levels to these two loudspeakers. This is not very satisfactory, however, for several reasons. Human hearing relies on different combinations of level, phase and arrival time -- as well as other factors -- to localize sound sources, and not solely on level. Loudspeakers in a conventional stereo pair are generally placed 60 degrees apart with respect to the listener. In this configuration, the ears can hear sound coming from both speakers and differences in level are interpreted as phase differences and provide localization information. This effect begins to fail as the speakers are moved further apart, and by the time you get to a 90 degree front stage -- as was found in quad setups and is still recommended in some quarters today -- a significant "hole in the middle" has developed (it's one reason for the center front channel in 5.1).

Level-only localization is poor at best in the rear, and virtually nonexistent at the sides. As a result, conventional surround tends to suffer from poor inter-speaker imaging. Sounds are sucked into the speakers and it is hard to get them to appear from anywhere else -- as a result, some engineers and producers have taken to deliberately placing sounds only in the loudspeakers: a simple and effective way out, but one that is rather limiting in creative terms. Another problem is the fact that as only level is involved, moving about within the listening environment changes the positions of sounds in the replayed image. Move to the left and you hear more of the left speakers, so the balance appears to move to the left. Move to the rear and the image moves backwards. This is not satisfactory: not only is the image unstable, it is only "right" in a very small spot in the center of the array -- the "sweet spot".

Ambisonics, on the other hand, relies on phase as well as level and other cues to provide localization information. As a result, it is less sensitive to listener position. In fact, a favorite sales technique for Ambisonics is to set up the system driving near-field monitors in a studio control room and ask the client to come into the room during replay of an Ambisonic recording. A remarkably stable image inside the speaker array is generally experienced from outside the array, much to the client's delight. And the image changes little as you enter the replay area and sit down (unless, of course, you go right up to a loudspeaker). Contrary to belief in some circles -- largely on the part of people who think they know the equations on which Ambisonics is based, and therefore know what it would sound like if they ever listened to it properly -- the "sweet spot" for Ambisonics is very large.

Ambisonics also allows sounds to be positioned anywhere inside the listening area. A sound panned around the room at constant speed a constant distance from the listener should stay that way and sound the same all round -- it shouldn't jump from one speaker to the next, changing its timbre as it goes, and moving closer to and further from the listener, which is all too often the case with conventional 5.1. In Ambisonics, it behaves impeccably.

Judge for yourself.

Advantage when English is your native language ! I wish I could have said it this way :-)

EoH

[Eye of Horus]

Originally posted by amorfeusz
I'm using CoolEdit Pro 2.0 build 2095 and have had no problems - apart from the Aurora Russian Roulette crash :)

I also tried converting a whole CD at once but even with my 768MB it wouldn't let me - Aurora doesn't appear to use virtual memory - I'll throw in my extra 256 later and will see what that changes. What I am just finishing now is a CD I ripped in two parts, each about 300mb. I'm planning to combine the 2 DTS waves later.

There is a stable version available, but you have to contact me privately !

(appfan@xs4all.nl)

EoH

[Eye of Horus]

Originally posted by drpaulng
Anybody can shine us a light? How do we use Nuendo surround and it's Direct-X plug-ins.

I cannot shine much on it, but I once used it for the following :

I took a lot of midi files and made 4.1 wavs with it. I left the centerchannel blank. These files were for the prupose of Karaoke ! A separate mic-amplifier for the center channel and....... excellent surround Karaoke !!

That's the only thing I ever did with Nuendo... again a piece of software that needs almost an academic level to understand and use :-(

EoH

[drpaulng]

Yes, Dear Eye of Horus, I (I think we) do appreciate your work. I was questioning about the up-sampling and down-sampling procedure (and so I think it's tedious). If this up and down sampling actually helps so much in producing good quality ambisonics, somebody involved in the development of ambisonics should be aware of. All other methods of ambisonics making should be widely discussed in order to reach the best result. The other question on the issue is that the impulse file EoH supplied to use all is limited to one single room acoustic parameter (size, sound source relative position among the 6 walls) which is largely unknown. This is absolutely important for people who want to make ambisonics from a stereo track because some sound source were recorded live in a small club, but some were in a concert hall...VSpace.exe seems to be a very versatile solution if you set the room acoustic parameters right for the sound. Today, people tend to record or derive sound source from bit rate of 48kHz or higher or bit depth of 24bit or higher rather than the 44.1kHz/16bit CD standard. With the old impulse sample rate at 44.1kHz, it seems to be very much not enough and we are all tired from down-sampling and up-sampling works during ambisonics production...so, there is room for future improvement.

For all other methods of surround production, I urge everyone who is interested in this, to pour out his/her opinions and experiences in this forum.

(Some typo errors were corrected)

[Eye of Horus]

Originally posted by drpaulng
Yes, Dear Eye of Horus, I (I think we) do appreciate you work. I was questioning about the up-sampling and down-sampling procedure (and so I think it's tedious). If this up and down sampling actually helps so much in producing good quality ambisonics, somebody involved in the development of ambisonics should be aware of. All other methods of ambisonics making should be widely discussed in order to reach the best result. The other question on the issue is that the impulse file EoH supplied to use all is limited to one single room acoustic parameter (size, sound source relative position among the 6 walls) which is largely unknown. This is absolutely important for people who want to make ambisonics from a stereo track because some sound source were recorded live in a small club, but some were in a concert hall...VSpace.exe seems to be a very versatile solution if you set the room acoustic parameters right for the sound. Today, people tend to record or derive sound source from bit rate of 48kHz or higher or bit depth of 24bit or higher rather than the 44.1kHz/16bit CD standard. With the old impulse sample rate at 44.1kHz, it seems to be very much not enough and we are all tired from down-sampling and up-sampling works during ambisonics production...so, there is room for future improvement.

For all other methods of surround production, I urge everyone who is interested in this, to pour out his/her opinions and experience in this forum.

I cannot say this enough : www.catt.se where you can download software to make your own roompulses ! Ask the maker for a 30 day key by email. And make whatever pulses you want. The program comes with pulses from some famous concerthalls.

I have impulses in 48 KHZ and 32 bit too ! But..... from the same room. I didn't try it yet , but I think you can make them with the CATT software too.
I also wish there was a faster and more automatic way of convolving than Cooledit. About the up- and downsampling.... Just leave the 32 bit steps in Cooledit and do all steps in 16 bit and see (or better : hear) what happens then....... !
You don't even need the whole convolving routine to check the difference..... just take a stereo source, load the left and right channel twice in Cooledit and make a stereo from the 4 files with the same method. (left+right as the new left channel and left+right as the new right channel, together in one new stereo file). Listen to the differences when using 16 bit and when using 32 bit.....
Then you will know why :-) And then you will also see that it isn't Ambisonics that need the resampling, but making a new stereo from an existing one IS !

I think this also is the explanation why the results in the CATT suite sound worse than in Cooledit....... But for making pulses it is an ideal program !

BTW as far as all those experiments I see in 5.1 ....... CATT suite can do that too ! Actually the Ambisonics was added later !

I didn't do that before, but I wonder if the Aurora plugins will be usable in other programs too. Because however tedious all those steps in Cooledit are, the fault is in Cooledit. You should be able to automate such a process, but I couldn't manage it in CE. I wouldn't complain if CE got the same batch conversion routines as SF !! I don't mind a process of 10 hours, as long as I don't have to sit next to it to push the right buttons at the right moment :-)

Anyone willing to check this out ?

OTOH Vvmic allows only one file at a time...... I do Ambidec and Besplit always in one batch, all the files at once ! And I think I have to dig in the old MS DOS manuals again to find that you should be able to add the Pack2x2to4 too !

I hope DSPGuru is still watching this thread and can help us with more automation !



EoH

[kempfand]

Acoustic Mirror Plug-In:

Understanding that the Sonic Foundry tools (Acid Pro 4, Sound Forge) would offer better automation (scripting, batch files) than CoolEdit, I did a bit of searching and found the Acoustic Mirror Plug-In.

The help file says: "The Acoustic Mirror (PlugIn) uses convolution as the principle behind its processing".

Also, if I remember correctly, the Sonic Audio tools can handle the 32 bit steps in EoH method.

Being at work, I couldn't try it out yet, but it sounds very promising. Anyone has tried or has experience with the "Acousic Mirror" ?

I'll report back here once I had a chance to test this.

Regards,
Andreas

[drpaulng]

The acoustic mirrors are real impulses derived from some famous concert halls and some specific environment such as bathroom and some strange shaped acoustic space (ie long corridor) that can be found in everyday life.
However, Sonic Foundry set them at 44.1kHz sample rate. This is the shortcoming because you have to down-sample the 96 or 48kHz recordings to 44.1kHz in order to use them. However, you may bypass this if you use the other "unreal" Direct X reverb plug-in. They are equally good with no limitation of sample rate.
In order to fully realize the 5.1 setting, you have to monitor the production in real time. In my opinon, Mx51 is very handy if you have an ASIO 2.0 soundcard such is the Audigy 2 which can output the 6 channels immediately to the 5.1 speaker setting at real time.

I have tested the FIReverb today. It is very handy for ambisonics production. It can let you design almost infinite number of acoustic environments and the quality is good (however, Mr.Eye of Horus seems does not very much agree with the sound quality it produces).
http://www.catt.se/

[Eye of Horus]

Originally posted by drpaulng
The acoustic mirrors are real impulses derived from some famous concert halls and some specific environment such as bathroom and some strange shaped acoustic space (ie long corridor) that can be found in everyday life.
However, Sonic Foundry set them at 44.1kHz sample rate. This is the shortcoming because you have to down-sample the 96 or 48kHz recordings to 44.1kHz in order to use them. However, you may bypass this if you use the other "unreal" Direct X reverb plug-in. They are equally good with no limitation of sample rate.
In order to fully realize the 5.1 setting, you have to monitor the production in real time. In my opinon, Mx51 is very handy if you have an ASIO 2.0 soundcard such is the Audigy 2 which can output the 6 channels immediately to the 5.1 speaker setting at real time.

I have tested the FIReverb today. It is very handy for ambisonics production. It can let you design almost infinite number of acoustic environments and the quality is good (however, Mr.Eye of Horus seems not very agree with the sound quality it produces).
http://www.catt.se/

About Sonic Foundry : I have a 2 channel SB Live. I tried this method a long time ago, but couldn't manage to get a good surround effect over the 5 speakers. Perhaps it's different when you own a 5.1 card. But in my situation it was useless ! (every minor change had to be burned and tested in my standalone DVD-player)
And this was the case for all the plugins for SF. 5.1 cards are not that long on the market.....

About FiReverb : did you compare 2 equal tracks ? I missed a bit clarity, brightness and dynamics....... Perhaps I should try to convert the original stereo to 48 Khz first, then use FIReverb and then downsample to 44.1 Khz again. Nice next project :-)

(Busy now with ripping and editing one of the best movietracks......)

regards,

EoH

[drpaulng]

The CREATIVE Audigy 2 is out now for only 1-2 month. This new "consumer" level soundcard can decode MLP of the DVD Audio. The price is reasonable when you compare it to the "professional" soundcards such as the CardDeluxe. The almost useful feature for this card is that it has the ASIO 2.0 interface so that it can put you through to the 5.1 or 6.1 system by direct link to softwares that support the Steinberg ASIO 2.0 standard. These softwares include Nuendo, ACID, and Mx51...

It is very convenient to monitor your surround sound production at real time rather than guessing by try-and-error methods. A cheaper way to attain the same monitoring control is try to buy 3 soundcards (or 2 plus the built-in card in the motherboard) of different brand names. Three brand names because it is much easier to differentiate each other. A total of 6 channels is now availble for real time monitoring. However, 3 sound cards may crash and cause a lot of trouble. I have very limited experience with Nuendo but ACID Pro is relatively easier to manage. However, when you have used Mx51, you should agree on my conclusion that you are in total control. The surround channels are very easily set up and modified by different Direct X reverb plug-ins including the Sonic Foundry's acoustic mirrors. There are a lot of such kind derived from real acoustic environments or unreal but very handy similated reverb effects for creating the surround tracks. Just choose the appropriate room acoustic parameters to suit your need. However, ambisonics is a totally different story. You have to pre-produce the surround sound and decode it into 4 or 5 wav files before puting them to Mx51 that may help you to add a LFE and complete the 5.1 surround. Most sound engineers would suggest you do not bother to use the 0.1 channel for music production because it is not needed at all. The LFE is only needed for movies to add headroom for extreme situations that happen only briefly in some action scenes such as explosion and earthquake. Therefore, if the 0.1 channel is added, just let it be a low volume "dummy" track as to "fool" people to believe that they are listening to full 5.1 surround rather than an "incomplete" 5.0 surround.

[Eye of Horus]

Originally posted by drpaulng
The CREATIVE Audigy 2 is out now for only 1-2 month. This new "consumer" level soundcard can decode MLP of the DVD Audio. The price is reasonable when you compare it to the "professional" soundcards such as the CardDeluxe. The almost useful feature for this card is that it has the ASIO 2.0 interface so that it can put you through to the 5.1 or 6.1 system by direct link to softwares that support the Steinberg ASIO 2.0 standard. These softwares include Nuendo, ACID, and Mx51...

It is very convenient to monitor your surround sound production at real time rather than guessing by try-and-error methods. A cheaper way to attain the same monitoring control is try to buy 3 soundcards (or 2 plus the built-in card in the motherboard) of different brand names. Three brand names because it is much easier to differentiate each other. A total of 6 channels is now availble for real time monitoring. However, 3 sound cards may crash and cause a lot of trouble. I have very limited experience with Nuendo but ACID Pro is relatively easier to manage. However, when you have used Mx51, you should agree on my conclusion that you are in total control. The surround channels are very easily set up and modified by different Direct X reverb plug-ins including the Sonic Foundry's acoustic mirrors. There are a lot of such kind derived from real acoustic environments or unreal but very handy similated reverb effects for creating the surround tracks. Just choose the appropriate room acoustic parameters to suit your need. However, ambisonics is a totally different story. You have to pre-produce the surround sound and decode it into 4 or 5 wav files before puting them to Mx51 that may help you to add a LFE and complete the 5.1 surround. Most sound engineers would suggest you do not bother to use the 0.1 channel for music production because it is not needed at all. The LFE is only needed for movies to add headroom for extreme situations that happen only briefly in some action scenes such as explosion and earthquake. Therefore, if the 0.1 channel is added, just let it be a low volume "dummy" track as to "fool" people to believe that they are listening to full 5.1 surround rather than an "incomplete" 5.0 surround.


For my next computer (probably March next year) I was thinking of upgrading the soundcard to a professional one ! (M-Audio 1010 !).
But for now I have to do with what I have :-)

About LFE : Most receivers will have the option to put low frequencies through the LFE channel, even if it is not present. So why bother anyway ?

EoH

[DIggedy]

The CREATIVE Audigy 2 is out now for only 1-2 month. This new "consumer" level soundcard can decode MLP of the DVD Audio. The price is reasonable when you compare it to the "professional" soundcards such as the CardDeluxe. The almost useful feature for this card is that it has the ASIO 2.0 interface so that it can put you through to the 5.1 or 6.1 system by direct link to softwares that support the Steinberg ASIO 2.0 standard. These softwares include Nuendo, ACID, and Mx51...

How does this work? I wasn't aware that the Audigy2 had 6 channel outputs so how does it output the 5.1 signal? If it is via the optical or coaxial digital outputs what format is it encoding to for the 5.1 signal to be decoded?

EDIT: Ignorant me! I just looked up the Audigy 2 and it looks like it DOES have 5.1 outputs on the internal card itself... looks like I'll be picking up one of these babies very soon!

[drpaulng]

Through the ASIO 2.0 standard, CREATIVE Audigy 2 can let you monitor the 3 stereo (analog) buses (5.1 output) at real time. It has the standard stereo output like any other sound cards on the market. It's digital output can be set to (1) by-pass mode for both Dolby Digital out and dts digitial out, and (2) decoded mode for Dolby Digtial out.
The only difference when comparing to the "professional" sound card is that it still has "higher" but ignorable harmonic distortion in analog recording and replay. It claims to have 106dB signal-to-noise ratio for analog recording. I deem it a "semi-professional" card rather than "consumer level" card. Judging for it's versatility, it worths buying because no professional sound card offers other features such as game-playing, midi and IEEE 1394 (for DV capture, for example) at the same time.

[terriblemonkey]

Of course there is also the M Audio Revolution for less money and probably better drivers that is coming out this month.

[drpaulng]

http://www.m-audio.net/products/consumer/index.php#

Little is known about this new comer. Anybody could help us knowing more about the M Audio Revoluation 7.1 sound card?

[terriblemonkey]

http://www.digitalconnection.com/Products/Audio/revolution.asp

also there is plenty more info on avsforum.com just search for revolution ,thats where I found it originally.

[Eye of Horus]

Originally posted by terriblemonkey
http://www.digitalconnection.com/Products/Audio/revolution.asp

also there is plenty more info on avsforum.com just search for revolution ,thats where I found it originally.

Haven't read all on the avsforum yet, but the specification sounds good. That is : if you can record 7.1 in 24/96 and output it !
It says 24/96 and it says 7.1, but....... together ????

I wish that 1010lt was a bit cheaper :-) (Need those analog outputs !)

EoH

[kempfand]

I played around with the Sonic Foundry tools (Acid Pro 4, Sound Forge), but stopped, since they don't offer good scripting / batching.

This is really a pitty, since I think it would be really nice to have a 'push-burron' solution for Ambi-conversions. They sound so well (stating but my my personal taste here). And HW-based decoders are either not available anymore (Onkyo) or way too expensive for my budget (Meridian).

Btw: JJLopez has developed Ambiovolover (http://www.gaips.upv.es/Beginning.htm under DownLoad), but it doesn't work on my machine. Anyone tried this tool and has it working ?

Regards,
Andreas

[Eye of Horus]

Originally posted by kempfand
I played around with the Sonic Foundry tools (Acid Pro 4, Sound Forge), but stopped, since they don't offer good scripting / batching.

This is really a pitty, since I think it would be really nice to have a 'push-burron' solution for Ambi-conversions. They sound so well (stating but my my personal taste here). And HW-based decoders are either not available anymore (Onkyo) or way too expensive for my budget (Meridian).

Btw: JJLopez has developed Ambiovolover (http://www.gaips.upv.es/Beginning.htm under DownLoad), but it doesn't work on my machine. Anyone tried this tool and has it working ?

Regards,
Andreas

Yep.... I did ! And........ it didn't work on my machine too !
But I redownload it right now and perhaps something changed.....

EoH

[Eye of Horus]

Originally posted by kempfand
[B]I played around with the Sonic Foundry tools (Acid Pro 4, Sound Forge), but stopped, since they don't offer good scripting / batching.


Sound Forge has an excellent batch routine !!!
Only..... you need to get it separate from SF 6 ! It isn't included anymore in the main program !

If Cooledit had the batch that is available for SF, we could make the process completely automatic !

(contact me privately for more "info" about SF's batch processor !)

EoH

[theReal]

Sound Forge has an excellent batch routine !!!
Only..... you need to get it separate from SF 6 ! It isn't included anymore in the main program Ah! That's where the batch processing went in SF 5, I guess.

When I was still using SF 4.5, I was often using the batch processing for normalizing and encoding to mp3 (at that time it was one of the best solutions for me to get that job done with good quality and the Fraunhofer mp3 codec).
I was recently looking for the batch processing in SF5 and was wondering where it went to... I then came to the conclusion that Sonic Foundry had just stripped this useful tool for some stupid reason. Now I know the reason is they want to earn more money....*?$#+ *#?´#!!!!! ;)

[Eye of Horus]

Originally posted by theReal
Ah! That's where the batch processing went in SF 5, I guess.

When I was still using SF 4.5, I was often using the batch processing for normalizing and encoding to mp3 (at that time it was one of the best solutions for me to get that job done with good quality and the Fraunhofer mp3 codec).
I was recently looking for the batch processing in SF5 and was wondering where it went to... I then came to the conclusion that Sonic Foundry had just stripped this useful tool for some stupid reason. Now I know the reason is they want to earn more money....*?$#+ *#?´#!!!!! ;)

Indeed, because the Batch processor alone is $ 150.= !!

EoH

[Eye of Horus]

Originally posted by Eye of Horus
Yep.... I did ! And........ it didn't work on my machine too !
But I redownload it right now and perhaps something changed.....

EoH

Downloaded it and still didn't work (Division by Zero).
Contacted the maker, but mail bounced back as undeliverable.

EoH

[Trebort]

I have a very cheap (£15) CMI8738 based sound card and using WDM drivers get 3 stereo outputs avaiable from n-Track Studio and SONAR 2. No 5.1 as such in these apps but if you don't need dynamic panning then 5.1 mixes are no problem.

SONAR 2 is excellent with WDM drivers and from browsing around it seems WDM is better than ASIO where supported.

Does the AUDIGY 2 realy offer up three ASIO stereo ouputs ?

[kempfand]

Contacted the maker, but mail bounced back as undeliverable.

For José_Javier_López: did you use " jjlopez@dcom.upv.es " ? That one should work...

I have the SF Batcher 5.1 up and running (thanks for your offer). This looks very promising, and I will try things out over the weekend (guess the 'Acoustic Mirror' can be used for the convolution part).

Regards,

Andreas

[Eye of Horus]

Originally posted by kempfand
For José_Javier_López: did you use " jjlopez@dcom.upv.es " ? That one should work...

I have the SF Batcher 5.1 up and running (thanks for your offer). This looks very promising, and I will try things out over the weekend (guess the 'Acoustic Mirror' can be used for the convolution part).

Regards,

Andreas

Great minds think alike, huh ?
Just finished the "acoustic mirror" method with the pulses we use in Cooledit. For the rest I used the same method as in Cooledit (stereo>32 bit mono etc.). The results are very, very disappointing.
I think convolving is something else then just adding the pulses. You lose a lot of detail in acoustic mirror. And there are some strange reverbs in it. Alos the channel separation is less. I cannot recommend this method !
(But it was damn faster !!!)

Yes, I used the same emailaddress, but it bounced back with "unknown user".

Ok, on to the next test : which is better ?
1. ripping a 24/96 track and converting to 16/44 or
2. recording the output from WinDVD in 16/44.

Will be busy the next days ! LOL !

Did step 1 and 2 and made Ambi from step 2. Sounds very good.
Now I will make Ambi from step 1 !

Keep you all posted !

EoH

[Eye of Horus]

Hi all,

Someone asked me by email to publish my batch process here.

here you go :

When saving processed files in Cooledit, make sure you name the wx and yz files like this :
01wx.wav 01yz.wav
02wx.wav 02yz.wav

(this is different from what I wrote in the guide. In the guide I said : 01.wx 01.yz)

Make sure Pack2x2to4.exe, Ambidec.exe and Besplit.exe are all in the same directory as the files.

Here is the batchfile.......... (you can change the driveletter to your own needs.) When you have only 8 or 11 or whatever number of tracks.... don't worry, it will skip to the next program.)

Pack2x2to4.exe 01wx.wav 01yz.wav 01wxyz.wav
Pack2x2to4.exe 02wx.wav 02yz.wav 02wxyz.wav
Pack2x2to4.exe 03wx.wav 03yz.wav 03wxyz.wav
Pack2x2to4.exe 04wx.wav 04yz.wav 04wxyz.wav
Pack2x2to4.exe 05wx.wav 05yz.wav 05wxyz.wav
Pack2x2to4.exe 06wx.wav 06yz.wav 06wxyz.wav
Pack2x2to4.exe 07wx.wav 07yz.wav 07wxyz.wav
Pack2x2to4.exe 08wx.wav 08yz.wav 08wxyz.wav
Pack2x2to4.exe 09wx.wav 09yz.wav 09wxyz.wav
Pack2x2to4.exe 10wx.wav 10yz.wav 10wxyz.wav
Pack2x2to4.exe 11wx.wav 11yz.wav 11wxyz.wav
Pack2x2to4.exe 12wx.wav 12yz.wav 12wxyz.wav
Pack2x2to4.exe 13wx.wav 13yz.wav 13wxyz.wav
Pack2x2to4.exe 14wx.wav 14yz.wav 14wxyz.wav
Pack2x2to4.exe 15wx.wav 15yz.wav 15wxyz.wav
Pack2x2to4.exe 16wx.wav 16yz.wav 16wxyz.wav
Pack2x2to4.exe 17wx.wav 17yz.wav 17wxyz.wav
Pack2x2to4.exe 18wx.wav 18yz.wav 18wxyz.wav
Pack2x2to4.exe 19wx.wav 19yz.wav 19wxyz.wav
Pack2x2to4.exe 20wx.wav 20yz.wav 20wxyz.wav
ambidec.exe -r Pentagon 01wxyz.wav 0001.wav
ambidec.exe -r Pentagon 02wxyz.wav 0002.wav
ambidec.exe -r Pentagon 03wxyz.wav 0003.wav
ambidec.exe -r Pentagon 04wxyz.wav 0004.wav
ambidec.exe -r Pentagon 05wxyz.wav 0005.wav
ambidec.exe -r Pentagon 06wxyz.wav 0006.wav
ambidec.exe -r Pentagon 07wxyz.wav 0007.wav
ambidec.exe -r Pentagon 08wxyz.wav 0008.wav
ambidec.exe -r Pentagon 09wxyz.wav 0009.wav
ambidec.exe -r Pentagon 10wxyz.wav 0010.wav
ambidec.exe -r Pentagon 11wxyz.wav 0011.wav
ambidec.exe -r Pentagon 12wxyz.wav 0012.wav
ambidec.exe -r Pentagon 13wxyz.wav 0013.wav
ambidec.exe -r Pentagon 14wxyz.wav 0014.wav
ambidec.exe -r Pentagon 15wxyz.wav 0015.wav
ambidec.exe -r Pentagon 16wxyz.wav 0016.wav
ambidec.exe -r Pentagon 17wxyz.wav 0017.wav
ambidec.exe -r Pentagon 18wxyz.wav 0018.wav
ambidec.exe -r Pentagon 19wxyz.wav 0019.wav
ambidec.exe -r Pentagon 20wxyz.wav 0020.wav
BeSplit.exe -core( -input 0001.wav -prefix d:\track01 -type wav -demux )
BeSplit.exe -core( -input 0002.wav -prefix d:\track02 -type wav -demux )
BeSplit.exe -core( -input 0003.wav -prefix d:\track03 -type wav -demux )
BeSplit.exe -core( -input 0004.wav -prefix d:\track04 -type wav -demux )
BeSplit.exe -core( -input 0005.wav -prefix d:\track05 -type wav -demux )
BeSplit.exe -core( -input 0006.wav -prefix d:\track06 -type wav -demux )
BeSplit.exe -core( -input 0007.wav -prefix d:\track07 -type wav -demux )
BeSplit.exe -core( -input 0008.wav -prefix d:\track08 -type wav -demux )
BeSplit.exe -core( -input 0009.wav -prefix d:\track09 -type wav -demux )
BeSplit.exe -core( -input 0010.wav -prefix d:\track10 -type wav -demux )
BeSplit.exe -core( -input 0011.wav -prefix d:\track11 -type wav -demux )
BeSplit.exe -core( -input 0012.wav -prefix d:\track12 -type wav -demux )
BeSplit.exe -core( -input 0013.wav -prefix d:\track13 -type wav -demux )
BeSplit.exe -core( -input 0014.wav -prefix d:\track14 -type wav -demux )
BeSplit.exe -core( -input 0015.wav -prefix d:\track15 -type wav -demux )
BeSplit.exe -core( -input 0016.wav -prefix d:\track16 -type wav -demux )
BeSplit.exe -core( -input 0017.wav -prefix d:\track17 -type wav -demux )
BeSplit.exe -core( -input 0018.wav -prefix d:\track18 -type wav -demux )
BeSplit.exe -core( -input 0019.wav -prefix d:\track19 -type wav -demux )
BeSplit.exe -core( -input 0020.wav -prefix d:\track20 -type wav -demux )

Now copy this and paste all these lines into Notepad and save it (all files !) as process.bat

Next time you only have to doubleclick on process.bat, drink a complete pot of coffee and you end up with all the files ready for Surcode DTS !!!!!!!

Eye of Horus

[fluss]

Hello
I have not yet risen to the heights discussed in this thread,
but a quick question that somebody might car to reply to.

I have burnt (wit nero) DTS and DD5.1 test tracks and demos from
http://www.sr.se/multikanal and played them on my DVD player.

I have converted .mp3 files to .ac3 with besweet )dspguru.doom9.org
Is there a fast utility or procedure to convert .ac3 to .wav, my DVD or CD player cannot play mp3's.

Yours Fluss

[Eye of Horus]

Originally posted by fluss
Hello
I have not yet risen to the heights discussed in this thread,
but a quick question that somebody might car to reply to.

I have burnt (wit nero) DTS and DD5.1 test tracks and demos from
http://www.sr.se/multikanal and played them on my DVD player.

I have converted .mp3 files to .ac3 with besweet )dspguru.doom9.org
Is there a fast utility or procedure to convert .ac3 to .wav, my DVD or CD player cannot play mp3's.

Yours Fluss

Hi Fluss,

You already have the tool needed !
Use Besure/Besweet and you can convert them.......

Eye of Horus

[fluss]

Thanks for the hint :-)

I have now tried besweet with ac3-wav and it worked.
But ... this was not exactly what I wanted,
The wav files are 10 times larger than the original
192kbit mp3.

I am looking for a way to make a CDR that I can play in my DVD player with a play time of at least 2 hours.

My DVD player can't play mp3 records ( as some new ones can),

Is there a simple way to produce a vcd/svcd with just a picture
and compressed sound ?

-- Fluss

[Eye of Horus]

Originally posted by fluss
Thanks for the hint :-)

I have now tried besweet with ac3-wav and it worked.
But ... this was not exactly what I wanted,
The wav files are 10 times larger than the original
192kbit mp3.

I am looking for a way to make a CDR that I can play in my DVD player with a play time of at least 2 hours.

My DVD player can't play mp3 records ( as some new ones can),

Is there a simple way to produce a vcd/svcd with just a picture
and compressed sound ?

-- Fluss

Yes, WinOnCD 5 and 6 have that option...........

EoH

[Eye of Horus]

Originally posted by Eye of Horus
Great minds think alike, huh ?
Just finished the "acoustic mirror" method with the pulses we use in Cooledit. For the rest I used the same method as in Cooledit (stereo>32 bit mono etc.). The results are very, very disappointing.
I think convolving is something else then just adding the pulses. You lose a lot of detail in acoustic mirror. And there are some strange reverbs in it. Alos the channel separation is less. I cannot recommend this method !
(But it was damn faster !!!)

Yes, I used the same emailaddress, but it bounced back with "unknown user".

Ok, on to the next test : which is better ?
1. ripping a 24/96 track and converting to 16/44 or
2. recording the output from WinDVD in 16/44.

Will be busy the next days ! LOL !

Did step 1 and 2 and made Ambi from step 2. Sounds very good.
Now I will make Ambi from step 1 !

Keep you all posted !

EoH

This weekend I did the experiment.
I have a 24/96 stereo track and tried to find out the best method for making Ambi surround.

1. direct recording from the DVDA to 16/44 with the SB soundcard (full duplex)
2. Ripping and converting to 16/44
3. doing the complete proces in the CATT package and use 24/96 pulses

@1. simple and fast. Made Ambi with the Aurora/Cooledit method.
On a scale of 1-10 this one gets 7 points
@2 . converted the ripped 24/96 track with Soundforge 6 to 16/44 and then used AUR/CE. This one gets a 9 !
@3. Made 24/96 pulses with the CATT package and did all the Ambisonics in 24?96 with CATT. At the end I converted them back to 16/44. Result : a 6 !

Conclusion : Allthough it is very timeconsuming, the combination of Aurora and Cooledit still gives the best results !!

Andreas found a reply to a post from the maker of the Aurora package in which he confirmed what I (we) already heard from our own ears.... the using of 32 bits floats gives far better results than doing the trick in 16 bits, or 24 bits .

Here a quote from his post :
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.

Now we can only hope that in future there will be a Cooledit version that is fast, stable and has a good batch routine.........

Eye of Horus

[SallyDog]

@EoH

could these be used as the wxy files? if so, which degrees would be be suited for w, x and y.

btw, thanks for the guide. i'm hooked on dts now.

http://www.saunalahti.fi/~arox/tracks

[Quilty]

You could do all this, or ... er... play the original CD and set your amp to Pro-Logic Mode.

Whey-hey ... hardware support!

Or maybe I am missing what you are up to ;)

Q.

[Eye of Horus]

Originally posted by Quilty
You could do all this, or ... er... play the original CD and set your amp to Pro-Logic Mode.

Whey-hey ... hardware support!

Or maybe I am missing what you are up to ;)

Q.

Yes you are missing something. Perhaps it's a good idea to visit some of the Ambisonics websites and reading all about it before making statements like this ?

Ambisonics is NOT comparable with other systems like Dolby Prologic, Stereo, Dolby Digital or whatever. It is a system on its own !
You cannot compare a mono sound with a Dolby Digital (or can you ?) and you also cannot do that with Ambisonics and Dolby !



Eye of Horus

[Quilty]

Yes, but correct me if I am wrong but your system is just "guessing" information? On the basis that practically no recordings are done with two mics these days - things are mixed from a whole load of mono tracks - there is no accurate data to decrypt .. a setting of Hall or Concert mode on an amp would do a similar thing, in a slightly different way.

Q.

[Eye of Horus]

Originally posted by Quilty
Yes, but correct me if I am wrong but your system is just "guessing" information? On the basis that practically no recordings are done with two mics these days - things are mixed from a whole load of mono tracks - there is no accurate data to decrypt .. a setting of Hall or Concert mode on an amp would do a similar thing, in a slightly different way.

Q.

Repeat : read the websites........

My English is not good enough to explain the principles, but when you read everything here, you will find enough links to sites that explain it perfect.

The only setting on an amp that would give the same results is an Ambisonics setting..... and there are no amps with that :-)

Really, you shouldn't compare..... just try it out once and then give us your conclusions.

The only one who is guessing at this moment is..... YOU ! LOL !!

Eye of Horus

[drpaulng]

Originally posted by Quilty
Yes, but correct me if I am wrong but your system is just "guessing" information? On the basis that practically no recordings are done with two mics these days - things are mixed from a whole load of mono tracks - there is no accurate data to decrypt .. a setting of Hall or Concert mode on an amp would do a similar thing, in a slightly different way.

Q.

You are right. Almost all "studio" recordings are artificially made with no natural sound stage setting. However, all instruments (including vocals) can be re-recorded by the ambisonic method by specific parameters setting the environment, ie room size, relative positions of the listener and the instruments. The computer program then calculates the new sound generated from the original stereo tracks and places to the appropriated surround speakers. But then, you are not "guessing" if you know precisely what to do...no single setting satisfies all.

[hippoth]

I tried using the guide for a 5.1 audio cd but there is something going wrong...I have a problem using the "Convolve with clipboard" module...I always get an error:

"Convolution effect can not access Clipboard Data. Another application or window has the Clipboard opened."

I have tried the small tool "ClipClear" what Amorfeusz has recommended but it also does not help :(

Can anybody help me?:confused:

[Eye of Horus]

Originally posted by hippoth
I tried using the guide for a 5.1 audio cd but there is something going wrong...I have a problem using the "Convolve with clipboard" module...I always get an error:

"Convolution effect can not access Clipboard Data. Another application or window has the Clipboard opened."

I have tried the small tool "ClipClear" what Amorfeusz has recommended but it also does not help :(

Can anybody help me?:confused:

Perhaps you need a clipboard viewer to see which application is making use of it ?
I think these can be found at www.download.com

EoH

[SallyDog]

re: clipboard problem

Are you sure you are setting the Cool Edit clipboard to use the Windows clipboard instead of the Cool Edit clipboard?

You will get the error message you are seeing if you are copying the impulse file to the Cool Edit clipboard by mistake.

Give it a try and good luck.

Sallydog

[hippoth]

Originally posted by SallyDog
re: clipboard problem

Are you sure you are setting the Cool Edit clipboard to use the Windows clipboard instead of the Cool Edit clipboard?




hi SallyDog...
I didn´t know about the possibility for such settings...can you tell me where I make such setting?
It would be fine :p

[SallyDog]

hippoth

From the main Cool Edit toolbar chose:

Edit
then "Set Current Clipboard"
then "Windows" (at the bottom)

At least this is how it is set up in Cool Edit Pro 1.2

SallyDog

[hippoth]

people who can read are clearly in advantage...it was a stupid question :rolleyes:

it was my fault that I didn´t read the guide from Eye of Horus exactly...I mean I also forgot to copy UHJ-W filter to the !Windows!-clipboard...now it works fine an I will hope to get a crisp 5.1 signal :D
one more question to the 5.1 signal...is a real 5.1 AC3 DolbyDigital what my 5.1 receiver will interpreat as a DolbyDigital (to be seen on a small red LED)???

by the way...thanks for your help SallyDog...I feel a bit shamed :stupid:

[SallyDog]

Don't worry about it...hope the results are good.
I use the Surcode DTS method and the playback is great.

SallyDog

[hippoth]

hi SallyDog,
I think I also will take the DTS method because so groovy :D

[hippoth]

my question goes especially to DSPguru!

when finish with 5 channel wav is it possibel to split it with BeSplit or should I do it before the whole procedure from a 2 channel wav into a 5 channel
I ask it because it very arduously to do the procedure with 20 tracks :(

[DSPguru]

you can use BeSplit '-demux' switch to split 5ch wave into 5 mono waves, or you can feed BeSweet directly with the multichannel wave and it'll execute the Surcode DTS encoder.

[hippoth]

hi DSPguru,
you don´t understand what I mean: when I have a stream form a dvd with 20 different audio tracks and I do the whole procedure from this 2.0 channel stream to a 5.1 channel or DTS stream...do I have to split the stream into the 20 tracks before the whole procedure or can I do it after it...I ask because I don´t know if Besplit can handle a DTS or 5.1 wav stream...everything clear?

[DSPguru]

BeSplit can split dts, 5.1wav, ddwav, dtswav, aac, and more... ;)

[hippoth]

hi master DSPguru...you should write all the formats into the readme of Besplit...less questions! :sly:

ok...I have my first DTS Audio CD (Best of David Bowie...I just made the 10th best tracks)...it was a hard work because the whole procedure is not the shortest one and I had to cut 7-8 seconds quiet on each track (the quiet was on the beginning of the tracks depending of the video stream) and so I had to cut all the tracks by hand :mad:

my question to all the guys who made such a DTS CD: do you feel that the rears are a little to loud?...I don´t like it despite the result sound not bad :D ... does it depend on the non exists of all the correct direction-information (can I say it?..in german it´s called "Richtungsinfomationen") ;)

[pafkosta]

Guys... this link on the begining of the post doens´t work! Can anyone tell me were I can find this somewere else?!
Tks
(http://www.ramsete.com/Aurora/downl...rora32Beta7.zip)

[SallyDog]

this will get you there.

http://www.ramsete.com/

SallyDog

[hippoth]

hi SallyDog,

what do you think about wahr I said...do you feel that the rear speakers are a little to loud? :confused:

[SallyDog]

hello hippoth

Actually, I think that the rear speakers are not loud enough, but that may be because of the way I have them placed (higher than the front and center speakers)

I am not sure that we are using the same source material. All of my recordings have been made from 2 channel wave files (ripped from audio cd's or converted from mp3's). When you talk about the video streams, I wonder if you are not using DVD as source.

Anyway, you should be able to use Cool Edit to reduce the volumn of your 04 and 05 mono wave files after running bsplit and before encoding with Surcode.

Let me know if that gives you a better result.

regards
SallyDog

[hippoth]

hi SallyDog,
I took a DVD as source...the audio stream was a 16-bit LPCM 2.0 channel wave
and...yes you are right I can try it with cooledit to reduce the volume...I will tell you more about the results (but can spend some time...because I don´t have the time :rolleyes: )

can you tell me is it possible to play the dts-waves...I tried it with windvd but I got just noise...after burning I could play them with my cd player...but I want to test them on the pc!
I have some dts *.vob files...there it is no problem to play them windvd...but why not with the dts-waves??

[SallyDog]

can't help you with PC playback. I haven't figured out a way to do it either.

oh well....keep trying.

[pafkosta]

I have two questions for you and i would be in many tks if someone could answer me...
The fisrt is that i have a few DIVX conversion of some of mine DVD. I was wondering if it is possible to tranform the avi sound into Ac3 DD 5.1. using this guide. If so can i then after in the authoring add the file that was created so that i end up with a normal DVD with 5.1 sound?
The second is that i can´t seem to find out where in cool edit - i´m using cool edit pro- is the "convolve with clipboard" aurora module option. I would in many tks if someone could tell me this.

[hippoth]

Originally posted by pafkosta
i have a few DIVX conversion of some of mine DVD. I was wondering if it is possible to tranform the avi sound into Ac3 DD 5.1. using this guide. If so can i then after in the authoring add the file that was created so that i end up with a normal DVD with 5.1 sound?at the beginning I think you have a DVD with DD 5.1 and you can end with a DivX in DD 5.1 ;) ...or what do you want to know :confused:
The second is that i can´t seem to find out where in cool edit - i´m using cool edit pro- is the "convolve with clipboard" aurora module option. I would in many tks if someone could tell me this.
you should copy all the *.xfm files into the cooledit folder and then you can find all under "Effects" -> "Aurora" -> "Convolve with Clipboard" ... thats all whats written in the guide :sly:

[m4551m0]

@Eye of Horus
i'm following your guide
"A Guide from Stereo to 5 channel Surround !"
and I'm just at the end, going to use the ambic.exe,
but for some strange reason the command line you supplied:
ambidec -r Pentagon (WXYZ.WZYZ) (output.wxyz)
doesn't work to me the program keep saying
unable to find wav file WXYZ.WXYZ
the file is in the same directory of ambidec.exe, I tried also to rename to WXYZ.wav but no success.
So i was ready to try the alternative method using Visual Virtual Microphone but the link to the program you supplied its down.

So here's my questions:
Is Visual Virtual Microphone a free program?
Can you upload it and post a link to the forum??

thanx for the attention to this
looking forward to try your method
best
m.

[SallyDog]

After you run pack2x2to4, you should have a file (whatever you named it), with an extension of wxyz. This is the file you refer to (the input file) with ambidec. Don't even use a name for the output file, one will be created automatically.

You might need to carefully re-read the guide, it does seem complicated until you made a few conversions, but, it does work.

Regards,
SallyDog

[m4551m0]

SallyDog

thats what I did:
1. I had 2 files named WX.WX and YZ.YZ
2. run run pack2x2to4 and get WXYZ.WXYZ

run ambidec as soon as point it to WXYZ.WXYZ
the program says
unable to find wav file WXYZ.WXYZ

???

u know where to get
Visual Virtual Microphone???

[lucindrea]

try nameing it wxyz.wav and see if it will take it.

[SallyDog]

@M4551m0

Did you make sure that wxyz.wxyz is in the same folder as ambidec?

re VV Microphone - do a google search. VVM is out there and it's freeware.

SallyDog

[m4551m0]

It worked today!
Don't tell me how 'cause I dunno.
Yesterday was too late.
Going to encode in ac3 an dts.

thanx lads

max

[jslombar]

Eye of Horus

Where is your latest Guide to DD5.1 from Stereo? I can't seem to locate it? If it's the first posting in this post, pardon my ignorance.

Thanks,
jslombar

[Malow]

some one can give a alternate mirror for aurora?
i can´t open www.ramsete for a month!!!!

i´m in brazil, this may be the problem... heheheh

the site dont open for me....

or send-me by mail!

alpha.pro@terra.com.br

Malow

[Bert Schlichter]

Thanks to the guide from EoH we can convert stereo files into 5 channel Ambisonic DTS or DD5.1...Great EoH ;-)
For those of you who want to use Ambisonic in their music-clip DVD's. Here is a small addendum to this guide to keep the audio files 100% in sync with the video. As i was told the Aurora plug-in produces so called 'room pulses' and therefore increase's the filesize with two times 0.3715 along the whole 'EoH guide' route.
So we have to correct this to keep the video in sync.

Here are the steps:

1: Use Smartripper to rip the various chapters.
2: Use TMPG to demux the .vob to a .m2v and and a 48 Khz .wav file.
3: Use Cooledit to down-sample the 48 Khz file to a 44.1 Khz .wav file.
4: Follow the 'EoH guide' exactly until you have got 5 mono files produced by Bsplit.
5: Cooledit: Pulldown menu Timebar - Define custom frames - Custom Time Code Display - Put: 10.000 frames/sec. 6: Find for each file the exact start-pulse and delete 0:3715 towards the middle from that point.
7: Find for each file the exact end-pulse and delete 0:3715 towards the middle from that point.
8: Use Cooledit to up-sample each 44.1 Khz file to a 48 Khz .wav file.

Use SurCode (DTS) or SoftEncode (DD 5.1) and your favourite Authoring program to make a Ambisonic DVD!

Have Phun,

Bert Schlichter :-)

[specise_8472]

I have just found a program that allows you to convolve audio.
Best of all its FREE!

http://shoko.calarts.edu/~bcassidy/hog/

Hog audio. Better than russian roulette with Aurora.

[Eye of Horus]

Originally posted by specise_8472
I have just found a program that allows you to convolve audio.
Best of all its FREE!

http://shoko.calarts.edu/~bcassidy/hog/

Hog audio. Better than russian roulette with Aurora.

Interesting..... but how to make this into surround 5.1 ?

EoH

[specise_8472]

I really just put this link up so as to give people a freeware alternative to Aurora. Looking at it, there are a lot of options to convole in this program.
If I get time I will try the different alternatives and see what the resulting mix sounds like.

[Eye of Horus]

Originally posted by specise_8472
I really just put this link up so as to give people a freeware alternative to Aurora. Looking at it, there are a lot of options to convole in this program.
If I get time I will try the different alternatives and see what the resulting mix sounds like.

That would be nice !
Almost every soundprogram has the possibility to convolve nowadays.
But only that routine on it's own doesn't make 5.1 from stereo.

So it would be nice if someone could do a comparison.....

At this moment (IMHO) I have this order based on the sound :
1. Aurora/Cooledit method
2. VVmic
3. Catt

I left out all those higend, highpriced receivers with their DSP effects :-)

EoH

[chrisguy]

Ok, I'm really excited about this but I having a very hard time. I read that you need to get the Tools Ambisonics (it's the third requirement). I guess this has the 3 wav files you need in it. Problem is, I have no clue as to where to download the tools. There is no link under where they are mentioned and I think it says see below, but I don't see anything below. And also, is Aurora functional, or do you have to buy that to do this.

Any help would be appreciated. Thanks in advance,
Chris

[Eye of Horus]

Originally posted by chrisguy
Ok, I'm really excited about this but I having a very hard time. I read that you need to get the Tools Ambisonics (it's the third requirement). I guess this has the 3 wav files you need in it. Problem is, I have no clue as to where to download the tools. There is no link under where they are mentioned and I think it says see below, but I don't see anything below. And also, is Aurora functional, or do you have to buy that to do this.

Any help would be appreciated. Thanks in advance,
Chris

The link is removed. I don't know why !
Try :
http://home.wanadoo.nl/appyhappy/Tools-Ambisonic.zip
and
http://www.xs4all.nl/~appfan/setupa32.EXE

EoH

[snowcrash101]

cheers for sharing this method EOH and posting/email me the links :devil:

I've encoded 8 songs into this material, and it is a very interesting sound, the closest i can relate to it being as if your listening to the sound live, rather than standard DTS/5.1 audio CD remix's which just utilise the speakers in different variations (i.e. singer in left speaker, guitarist in right, drums in rear)...

And I find it really good as this is more like reality than a the "standard" surround mixing style (i.e. when do you hear a band who run around you in a circle, or has the lead singer popping up all around you...).

I'm playing about with making ambisonic audio on video atm but i think this will be good for mainly live music events (concerts) rather than films. Due to the TV giving you information - as person or object placement in relation to where you are viewing the film/video from (screen) and "standard" surround mixing is logically more correct in your mind (i.e. noise from something off screen is from a speaker behind you - not in most of the speakers...). Then again this is theory and i havent made/authored a DVD (yet) with Video footage and Ambisonic 5.1 ;) - I'll wait and see before I decide weather or not converting some already made videos is worth the hassle when I make a proper test DVD.

Cheers,

Snowcrash. :D

[Eye of Horus]

Originally posted by snowcrash101
cheers for sharing this method EOH and posting/email me the links :devil:

Thanks :-))


I've encoded 8 songs into this material, and it is a very interesting sound, the closest i can relate to it being as if your listening to the sound live, rather than standard DTS/5.1 audio CD remix's which just utilise the speakers in different variations (i.e. singer in left speaker, guitarist in right, drums in rear)...

And I find it really good as this is more like reality than a the "standard" surround mixing style (i.e. when do you hear a band who run around you in a circle, or has the lead singer popping up all around you...).


It does sound more natural than a lot of these fancy mixes !
I sure agree with you :-)))


I'm playing about with making ambisonic audio on video atm but i think this will be good for mainly live music events (concerts) rather than films. Due to the TV giving you information - as person or object placement in relation to where you are viewing the film/video from (screen) and "standard" surround mixing is logically more correct in your mind (i.e. noise from something off screen is from a speaker behind you - not in most of the speakers...). Then again this is theory and i havent made/authored a DVD (yet) with Video footage and Ambisonic 5.1 ;) - I'll wait and see before I decide weather or not converting some already made videos is worth the hassle when I make a proper test DVD.

Cheers,

Snowcrash. :D :D

It's possible to make an Ambi soundtrack, but you must keep in mind that the original files are made a fraction of a second longer, because the pulses are added to the original tracks. This can lead to out-of-sync problems.
Earlier in this thread you can find more on this subject........... and how to fix it.

kind regards,

EoH

[daphy]

Hi Eye of Horus

first I would like to tribut respect for your work!!!

I´ve tried your guide and it works nearly perfect, sometimes the sound gets to much peak so it sounds a little wired (sorry for my bad English :( )

I bet you for a big faviour:

Is it possible to do one 'complete' guide with all tricks and hints and publish them an the doom9 guide side :rolleyes:

My lack of knowledge of the English language leads (me) to the point, that I have to search through all the postings to find my mistake (wired sound because of to much peak I´ll guess). This seams to me nearly unpossible (184 postings - puh - I guess this means hours for me) :confused:

So I´ll bet you to write this guide for me (and I think for all those idealistic guys out there which wanted to get a real optimum of sound out of their stereo recordings). All I can do is offer you my personal help as support for this guide - please send pm!

One curious thing left:
I downloaded from this link the cooledit plugin (http://www.xs4all.nl/~appfan/setupa32.EXE) and as I start the file it explained me that there are no cooledit installtion found on my pc - 2.1 must be enough I guess :cool:

CYA Daphy

[snowcrash101]

daphy - i had to install the older version of cooledit (v1) to get the aurura pack installed..

i installed V2 over the top (but couldnt get it working)...

so just used V1 instead.

[Eye of Horus]

Originally posted by daphy
Hi Eye of Horus

first I would like to tribut respect for your work!!!

I´ve tried your guide and it works nearly perfect, sometimes the sound gets to much peak so it sounds a little wired (sorry for my bad English :( )

I bet you for a big faviour:

Is it possible to do one 'complete' guide with all tricks and hints and publish them an the doom9 guide side :rolleyes:


Yep, I will do that one of these days......


My lack of knowledge of the English language leads (me) to the point, that I have to search through all the postings to find my mistake (wired sound because of to much peak I´ll guess). This seams to me nearly unpossible (184 postings - puh - I guess this means hours for me) :confused:


I'm also non-english !!!!!!!!!! :-)


So I´ll bet you to write this guide for me (and I think for all those idealistic guys out there which wanted to get a real optimum of sound out of their stereo recordings). All I can do is offer you my personal help as support for this guide - please send pm!

One curious thing left:
I downloaded from this link the cooledit plugin (http://www.xs4all.nl/~appfan/setupa32.EXE) and as I start the file it explained me that there are no cooledit installtion found on my pc - 2.1 must be enough I guess :cool:

CYA Daphy

The solution is somewhere in the thread too :-)

De-install CE 2.1
Install 1.0 or 1.2
Install Aurora
Install 2.1 in the same directory as 1.0 or 1.2

There is also another solution somewhere in the thread if I remember right............

The guys who made the install routine for Aurora made it when there was no 2.1 version of Cooledit.


for now.........


best regards,

EoH

[SallyDog]

You really should not have to un-install and then re-install.

When you run the set-up for the plug-ins, it put all of the *.xfm files into a folder (probably called programs\cooledit). All you have to do is move the xfm files to the existing folder for CE 2.1
This works fine for me.


Nevermind:mad:

I screwed up. To make a long story short, just follow EoH suggestion to install Aurora.

Sorry

[Malow]

just create a reg to install the version 2.0 without v1.2

-------------------------------------------------

Windows Registry Editor Version 5.00

[HKEY_LOCAL_MACHINE\SOFTWARE\Syntrillium\cepro]
"MainDir"="C:\\PROGRAM FILES\COOLEDIT"



---------------------------------------------------

CHANGE THE PATH WITH RIGHT WAY TO APP.

Malow

[Malow]

oops! use "\\" instead "\" on path, and this reg header is for Xp regs, i can´t remeber winme, 98 header.

Malow

[matrix]

Now you have two stereo signals. Change the extension of the WX file (should be WAV) to "WX". Change the extension of the YZ file (should be WAV, too) to "YZ".
Please excuse my ignorance, but how do you change the extension? CEdit won't let me do it.

[SallyDog]

Just rename them from within Windows explorer.

[matrix]

Yes I tryied that, but the extension is still wav

[Covak]

Say, I can play the 5ch wav file made by ambidec just fine (in WinDVD, PowerDVD, or Winamp using hardware-accelerated DirectSound). Only problem is that the channels don't seem to be in the right order.

Anyone have any ideas/suggestions as to how I might fix that?

[lucindrea]

then your explorer is set up wrong ...

tools/folder options
click view tab
click "show hidden files and folders"
UNCHECK "hide file extensions for knowen types"

if that doesnt work you can use the old fasitioned way ( command prompt )


start/run/cmd ( or command if you have 98 )
if your files are on a differnt drive then C: .. type (letter):
type cd /(path to the WX,YZ files)
dir
move filename1.wav filename1.WX
move filename2.wav filename2.YZ



here is a quick command prompt ver i just did to show you how it should look

C:\WINNT>f:

F:\>cd \sounds

F:\sounds>dir
Volume in drive F is dma133-1
Volume Serial Number is 9422-3758

Directory of F:\sounds

06/23/2003 07:20p <DIR> .
06/23/2003 07:20p <DIR> ..
06/23/2003 07:19p 0 soundfile.WX.wav
06/23/2003 07:19p 0 soundfile.YZ.wav
2 File(s) 0 bytes
2 Dir(s) 20,060,389,376 bytes free

F:\sounds>

F:\sounds>move soundfile.WX.wav soundfile.WX

F:\sounds>move soundfile.YZ.wav soundfile.YZ

F:\sounds>dir
Volume in drive F is dma133-1
Volume Serial Number is 9422-3758

Directory of F:\sounds

06/23/2003 07:20p <DIR> .
06/23/2003 07:20p <DIR> ..
06/23/2003 07:19p 0 soundfile.WX
06/23/2003 07:19p 0 soundfile.YZ
2 File(s) 0 bytes
2 Dir(s) 20,060,389,376 bytes free

F:\sounds>

[Covak]

A little more searching and I was able to answer my own question.

http://www.cs.bath.ac.uk/~rwd/mctools.html

Great little set of command line tools for playing with multi-channel wavs.

[matrix]

@ lucindrea
You were right. I had the explorer set up wrong.
Thanks. I appreciate it.

[Covak]

One of the first parts of the guide says:

-> The goal here is to convert your stereo CD signal, which is UHJ - encoded, into a B format signal...

But my stereo CD signal sure isn't UHJ-encoded. Them's some rare CDs that were actually UHJ-encoded (for Ambisonics). This whole procedure is designed to decode UHJ-encoded material. So I'm not sure what the heck we're actually doing by applying the process to regular stereo material.

I guess it's like applying Dolby Pro-Logic (I or II) to a normal stereo signal? I just thought this process (based on that snippet from the guide) was actually designed to be used on regular stereo material.

[Eye of Horus]

Originally posted by Covak
One of the first parts of the guide says:

-> The goal here is to convert your stereo CD signal, which is UHJ - encoded, into a B format signal...

But my stereo CD signal sure isn't UHJ-encoded. Them's some rare CDs that were actually UHJ-encoded (for Ambisonics). This whole procedure is designed to decode UHJ-encoded material. So I'm not sure what the heck we're actually doing by applying the process to regular stereo material.

I guess it's like applying Dolby Pro-Logic (I or II) to a normal stereo signal? I just thought this process (based on that snippet from the guide) was actually designed to be used on regular stereo material.

It is ! I guess it's my bad English :-)
Instead of "which is" read "which will be".
In Dutch in this case you can use the word "is" to give it the same meaning.

And I strongly advice you to read the whole thread ! It answers your PL question !

regards,

EoH

[Covak]

Ah so :)

BTW, thanks for the guide. I had never heard of Ambisonics before I saw your guide yesterday (I was recently impressed with DPLII's music mode, and I wondered what methods there might be for making DTS versions of up-mixed stereo music), and I spent a good while reading all about it.

I'm still experimenting with the process, but so far the results are interesting. I'm a bit confused as to channel ordering though...

- The channel order of ambidec? Using the default rigs, I dunno, but I guess we can use custom rigs to get the channels to be in any order we want. I'm toying with this just now.

- The channel order for AC3, DTS, and multi-channel WAVs? I'd hope they're all the same, but I don't know. I guess I can check that myself too, with a little test clip.

One thing that confuses me about your process though is the conversion to 32 bits. It seems to me that, following your guide, there's no processing done at all with the 32 bit waveforms. We convert to 32 bit and then convert back to 16 bit before we've done anything with the 32 bit wave. At first I thought "Maybe the stereo->mono conversion works better in 32 bit." But CE actually seems to do stereo->mono before it does 16->32 bits, so that can't be it. It seems to me that all it accomplishes is some dithering...

[Malow]

anyone got a strange volume level on channels, on the firts time i "reorder" hehe, the chanels, but now again, i have strange levels... even from a "mono like" setero, were levels are almost same.

left front - "volume at 95%"
right front- "volume at 75%"
center - "volume at 70%"
left surr - "volume at 75%"
right surr- "volume at 95%"

and YES, i put on the correct order, i see dozen of times, an its correct. extracting to wav to convert in dts, or direct transcode the wav in softencode, same channel order.... same levels...

any help i apreciated.

[Covak]

Are you using the "Pentagon" rig? Check out the channels of the wav ambidec puts out for that:

ch1 = LF (left front)
ch2 = LS (left surround/rear)
ch3 = CS (yes, center rear!)
ch4 = RS
ch5 = RF

And I'm pretty sure that because of how the Ambisonic decoder works, you can't just rotate the channels around to get the right sound. The "Surround1" rig is what you want (and what the guide should suggest, I believe):

ch1 = LF
ch2 = CF (regular old front center this time :)
ch3 = RF
ch4 = LS
ch5 = RS

Now if you do what the guide suggests, and put the channels into a DTS or DD encoder in the order 1, 3, 2, none, 4, 5, then things will be right.

Really though, a good custom rig specification is the way to go!

[Malow]

tanks for the help, i using pentagon.
im doing exactly as the gide, step by step, an with the order of guide:

Left front : song0101
right front : song0103
center : song0102
lfe : stays blank
left surround : song0104
right surround : song0105

i do the same with a music now, and the same occours,...
the better i can do now is adjust the levels os each channl in vegas to not move the order in vegas, but keep the volume levels "leveled"..
hehe...

tnks kovak

Malow

[Covak]

Is it the exact same problem now (with the Surround1 rig)? Or are you getting different volume levels (but still incorrect ones) than before?

And how are you getting those volume level measurements?

I think messing with volume levels will screw-up the effect.

[matrix]

Can somebody give me a hand please?

I tryed this thing several times, but I get stuck at runnung ambidec.
I can run Pack2x2to4.exe right from the directory where it is(tools-ambisonic), but ambidec for some reason, when I doubleclick on it, just opens and closes quickly. So I copyed it to windows dir. and try to run it from a command prompt, but it allways tells me it cannot find the file.
I don't know what to do. I hope somebody can tell my what I can do.
I would really appreciate any help on this.

Thank you.

[Covak]

If you copy ambidec to the folder you're working in (where the .wxyz file is), and switch to that folder at the command prompt and then run ambidec, you should be alright.

[matrix]

Thanks for the sugestion, but still no go. Just like when I run it from windows, it can't find the file. I tried rename it, but it won't go.
But why can't I run it from its current location, like I can run pack2x2to4 ?
This is so frustrating. I red the whole thread a few times, and I think I did everything the way I'm supposed to, but this damn program keeps me from finishing.

Anyway, thank you for trying to help, and if there's anything else I can try, let me know.

[Covak]

pack2x2to4 isn't entire command line based (it asks you questions), which makes it easy to run from anywhere (including Explorer). But for ambidec you have to specify everything on the command line.

Let's say you have everything in F:\Temp...

F:\Temp>dir
...
01/10/2000 13:51 258,048 ambidec.exe
26/06/2003 12:03 896,463 sample.wxyz
...

Then you'd go:

F:\Temp>ambidec -rSurround1 sample.wxyz

If you wanna chat me: ICQ# 18969614, AIM: GaryCartel, MSN: covak@hotmail.com, it might be easier to help that way if you still need it :)

[matrix]

Has anybody tryed ambidec with a large file?
I have this movie soundtrack, 135min long, and I see some other people are thinking to do this.
I run it through the whole process, and I ended up with a 135 min wxyz file. But then, when trying to run it in ambidec,(I had some problems, but with the great assistance from Covak, I was able to do that too)... well, I got an 81 min long wav file.
Now I don't know if I did something wrong in the process, but I don't think so since the wxyz file was the right length.
Somehow ambidec did only 81 min out of it. And I run it through ambidec a few times. The command was the one provided in the guide.
ambidec -r Pentagon stereo.wxyz and once ambidec -r Surround1 stereo.wxyz
Every time I ended with the short file.

So it mekes me belive ambidec has some problems with large files.?
Can someone confirm this or have I really done something stupid.
Anyway, is there a way to split the wxyz file?
If I still had the wx and yz files, I could do it, but I deleted them.
I'm curious though, I did something wrong or it can't process large files.

Regards

[Eye of Horus]

Originally posted by matrix
Has anybody tryed ambidec with a large file?
I have this movie soundtrack, 135min long, and I see some other people are thinking to do this.
I run it through the whole process, and I ended up with a 135 min wxyz file. But then, when trying to run it in ambidec,(I had some problems, but with the great assistance from Covak, I was able to do that too)... well, I got an 81 min long wav file.
Now I don't know if I did something wrong in the process, but I don't think so since the wxyz file was the right length.
Somehow ambidec did only 81 min out of it. And I run it through ambidec a few times. The command was the one provided in the guide.
ambidec -r Pentagon stereo.wxyz and once ambidec -r Surround1 stereo.wxyz
Every time I ended with the short file.

So it mekes me belive ambidec has some problems with large files.?
Can someone confirm this or have I really done something stupid.
Anyway, is there a way to split the wxyz file?
If I still had the wx and yz files, I could do it, but I deleted them.
I'm curious though, I did something wrong or it can't process large files.

Regards

Never tried such a big file, because Aurora and Cooledit won't accept such lengthy ones !

I didn't try it yet on my new PC, but my Pentium 3 500 with 384 MB couldn't handle files longer than 9-10 minutes.

EoH

[matrix]

Never tried such a big file, because Aurora and Cooledit won't accept such lengthy ones !
Oh, but they do. Both of them. That's what I used to get as far as ambidec.
Well, it takes forever even with a P4 2.66Ghz, let alone the random aurora crashes, but I can do the job.
I'm doing it again right now, but I'm just going to split it before.
Hopefully I'll see soon what I spent all this time for.

Matirx

[matrix]

Originally posted by Bert Schlichter
Thanks to the guide from EoH we can convert stereo files into 5 channel Ambisonic DTS or DD5.1...Great EoH ;-)
For those of you who want to use Ambisonic in their music-clip DVD's. Here is a small addendum to this guide to keep the audio files 100% in sync with the video. As i was told the Aurora plug-in produces so called 'room pulses' and therefore increase's the filesize with two times 0.3715 along the whole 'EoH guide' route.
So we have to correct this to keep the video in sync.

Here are the steps:

1: Use Smartripper to rip the various chapters.
2: Use TMPG to demux the .vob to a .m2v and and a 48 Khz .wav file.
3: Use Cooledit to down-sample the 48 Khz file to a 44.1 Khz .wav file.
4: Follow the 'EoH guide' exactly until you have got 5 mono files produced by Bsplit.
5: Cooledit: Pulldown menu Timebar - Define custom frames - Custom Time Code Display - Put: 10.000 frames/sec. 6: Find for each file the exact start-pulse and delete 0:3715 towards the middle from that point.
7: Find for each file the exact end-pulse and delete 0:3715 towards the middle from that point.
8: Use Cooledit to up-sample each 44.1 Khz file to a 48 Khz .wav file.

Use SurCode (DTS) or SoftEncode (DD 5.1) and your favourite Authoring program to make a Ambisonic DVD!

Have Phun,

Bert Schlichter :-)

In step 6: Find for each file the exact start-pulse
What start-pulse are you\is he refering to, and how do you find it?

[Malow]

is the same prob in the first others encodes,
the volume level is not procise, is just a "medium" too you imagine the "peak level" of the audio channel.

i do another on encode again, and just on convolve the channels with the wav´s files provided by ambitool, the level of audio go down.

and im doing exactly as the guide. after use ambidec the channels packed is the same right? only one channel is added and others remain intact right? im seeing now, is right on convolve the levels is modified... maybe use another settings on convolve with aurora?

Thank you for the help.

Malow

[sazanon]

First attempt to script Eyes of Hours Ambisonic Guide

1.- Soft I use
--------------
1.1 - Cool Edit Pro 2.1
1.2.- Aurora's Convolve with Clipboard Module (Convolve.xfm vers.1.0.0.1)
1.3.- Ambisonic Impulse files UHJ_W_4416, UHJ_X_4416 and UHJ_Y_4416 placed in c:\Cool Edit Pro\Imps. Modify the script according to your needs.

2.- Procedure
-------------
2.1 - Load your original stereo 16 bits 44.100 Hz wav file in CEP
2.2 - Options --> Scripts & Batch Processing --> Open New Collection --> Ambisonic.scp --> WXYZ --> Run Script
2.3 - At the end of the process you'll have
2.3.1 - Ambisonic W file in Clipboard 1
2.3.2 - Ambisonic X file in Clipboard 2
2.3.3 - Ambisonic Y file in Clipboard 3
2.3.4 - Ambisonic WX file in Clipboard 4
2.3.5 - Ambisonic YZ file in Clipboard 5

So you just have to copy WX & YZ to your H.D.
Set Current Clipboard 4/5 --> Paste to New --> File save as WX/YZ.wav

3.- Questions I ask myself
--------------------------
3.1 - Why is it that in the script I have to reverse channels in WX to get the same wav file as when I do it manually ?
3.2 - Why YZ has a lower volume as when I do it manually ?

4.- Remarks
-----------
4.1 - Sorry for my english, it's not my mother tongue
4.2 - What ring should we use for a AC3/DTS system ?, Pentagon ?, Surround1 ?
4.3 - Any enhancement or CORRECTION to this script are welcome

5.- Well, here it goes :

Collection: Ambisonic

Title: WXYZ
Description: Ambisonic
Description: ----------------
Description: Load your original Stereo 16 bits 44,100 Hz file first
Description:
Description: At the end of the script, you'll have the 5 processed files as follows :
Description:
Description: Wav W in Clipboard 1
Description: Wav X in Clipboard 2
Description: Wav Y in Clipboard 3
Description: Wav WX in Clipboard 4, Copy to Hard Disk as WX.wav
Description: Wav YZ in Clipboard 5, Copy to Hard Disk as YZ.wav
Description: *************************************************************************
Description: Debe cargarse previamente el Wav original
Description:
Description: Al final del proceso, los 5 wavs procesados estaran como sigue:
Description:
Description: Wav W en el Clipboard 1
Description: Wav X en el Clipboard 2
Description: Wav Y en el Clipboard 3
Description: Wav WX en el Clipboard 4, Copiar a Disco Duro como WX.wav
Description: Wav YZ en el Clipboard 5, Copiar a Disco Duro como YZ.wav
Mode: 2

Selected: none at 0 scaled 2958028 SR 44100
cmd: Channel Both

Selected: 0 to 2958027 scaled 2958028 SR 44100
cmd: Cut
1: 4

cmd: Paste Special
1: 1
2: 1
3: 3
4: 1
5: 0
6: 1
7: C:\Cool Edit Pro\imps\UHJ_W_4416.wav
8: 2
9: 4

cmd: Cut
1: -1

cmd: Paste Special
1: 1
2: 1
3: 3
4: 1
5: 0
6: 0
7: C:\Cool Edit Pro\imps\UHJ_W_4416.wav
8: 2
9: 4

cmd: Aurora\Convolve with Clipboard
1: 69
2: 69
3: 72
4: 16384
5: 131072
6: 2958028
7: 44100
8: 44100
9: 0
10: 0
11: 0
12:
13:
14: 32767
15: 0
16: 0
17: 0

cmd: Convert
1: 44100
2: 1
3: 32
4: -1
5: 0.5
6: 0.5
7: 1
8: 0
9: 150
10: 1
11: 1
12: 16

cmd: Cut
1: 0

cmd: Convert
1: 44100
2: 2
3: 16
4: -1
5: 1
6: 1
7: 1
8: 0
9: 150
10: 1
11: 1
12: 16

cmd: Paste Special
1: 1
2: 1
3: 3
4: 1
5: 0
6: 1
7: C:\Cool Edit Pro\imps\UHJ_X_4416.wav
8: 2
9: 0

cmd: Cut
1: -1

cmd: Paste Special
1: 1
2: 1
3: 3
4: 1
5: 0
6: 0
7: C:\Cool Edit Pro\imps\UHJ_X_4416.wav
8: 2
9: 4

cmd: Aurora\Convolve with Clipboard
1: 69
2: 69
3: 72
4: 16384
5: 131072
6: 2958028
7: 44100
8: 44100
9: 0
10: 0
11: 0
12:
13:
14: 32767
15: 0
16: 0
17: 0

cmd: Convert
1: 44100
2: 1
3: 32
4: -1
5: 0.5
6: 0.5
7: 1
8: 0
9: 150
10: 1
11: 1
12: 16

cmd: Cut
1: 1

cmd: Convert
1: 44100
2: 2
3: 16
4: -1
5: 1
6: 1
7: 1
8: 0
9: 150
10: 1
11: 1
12: 16

cmd: Paste Special
1: 1
2: 1
3: 3
4: 1
5: 0
6: 1
7: C:\Cool Edit Pro\imps\UHJ_Y_4416.wav
8: 2
9: 1

cmd: Cut
1: -1

cmd: Paste Special
1: 1
2: 1
3: 3
4: 1
5: 0
6: 0
7: C:\Cool Edit Pro\imps\UHJ_Y_4416.wav
8: 2
9: 4

cmd: Aurora\Convolve with Clipboard
1: 69
2: 69
3: 72
4: 16384
5: 131072
6: 2958028
7: 44100
8: 44100
9: 0
10: 0
11: 0
12:
13:
14: 32767
15: 0
16: 0
17: 0

cmd: Convert
1: 44100
2: 1
3: 32
4: -1
5: 0.5
6: 0.5
7: 1
8: 0
9: 150
10: 1
11: 1
12: 16

cmd: Cut
1: 2

*************************************************************

Selected: none at 0 scaled 0 SR 44100
cmd: Channel Both

Selected: none at 0 scaled 0 SR 44100
cmd: Paste
1: 1

cmd: Convert
1: 44100
2: 1
3: 16
4: -1
5: 0.5
6: 0.5
7: 1
8: 0
9: 150
10: 1
11: 1
12: 16

cmd: Cut
1: 3

cmd: Paste
1: 0

cmd: Convert
1: 44100
2: 2
3: 16
4: -1
5: 1
6: 0
7: 1
8: 0
9: 150
10: 1
11: 1
12: 16

cmd: Channel Right

cmd: Paste
1: 3

cmd: Channel Both
cmd: Amplitude\Channel Mixer
1: 0
2: 1
3: 1
4: 0
5: 0
6: 0
7: 0
cmd: Cut
1: 3

cmd: Convert
1: 44100
2: 1
3: 32
4: -1
5: 0.5
6: 0.5
7: 1
8: 0
9: 150
10: 1
11: 1
12: 16

cmd: Paste
1: 2

cmd: Convert
1: 44100
2: 2
3: 16
4: -1
5: 1
6: 0
7: 1
8: 0
9: 150
10: 1
11: 1
12: 16

cmd: Cut
1: 4

End:

[jslombar]

To Matrix,

In step 6: Find for each file the exact start-pulse

I'm not sure if this is correct. After I did the ambisonic method,
My wave files were .3715 seconds longer not .743 seconds The DVD I did was Rush Chronicles. To correct the sync problem (audio following the video), I used AC3 Delay Corrector V1.7 on the ac3 file, and used a -372ms delay. Hope this makes things alot easier.


jslombar

[matrix]

My wave files were .3715 seconds longer not .743 seconds
Yeah, same here. I only cut .3715 from the beginning, and I was OK.

Thanks for your answer

matrix

[hammerstein.dacm]

Thanks, looks like a nice guide!
Only missed the link to the obligatory
ambisonic tools.

Is it thinkable to use also mpeg 1 layer 2
audio as input for the creation of this
artificial dolby surround sound?

It would be nice to burn old vhs material
on dvd with a choice for the mpeg 1 layer 2
stereo-sound and a according to this guide
constructed dts track.

But I understand that the lenght of a whole
concert would give serious desynchronisation
issues?


regards

dh

[Eye of Horus]

Originally posted by hammerstein.dacm
Thanks, looks like a nice guide!
Only missed the link to the obligatory
ambisonic tools.

Yep, they were removed. I don't know why....


Is it thinkable to use also mpeg 1 layer 2
audio as input for the creation of this
artificial dolby surround sound?

Excuse me ? When you call it artificial Dolby surround, then you don't have to read or use it. or you didn't understand it.
Use an AC3 encoder. Bye !!!


It would be nice to burn old vhs material
on dvd with a choice for the mpeg 1 layer 2
stereo-sound and a according to this guide
constructed dts track.


You can use it with mpeg 1 layer 2 , but then you are using already compressed material. Don't expect the end result to be as good as when using CD tracks !


But I understand that the lenght of a whole
concert would give serious desynchronisation
issues?


read the rest !


regards

dh

regards,

EoH

[hammerstein.dacm]

EoH!

Sorry, no offense was intended, had the idea
that only careful mixing of the physical instruments
during a recording session could result in real surround
or dts sound.

But my background knowledge indeed is very modest,
I only seek for a practical way to improve the
soundtracks when burning great old music-videos
on DVD.
So far I only managed to get 6 wav-channels from
the mpa soundtrack I captured with my Hauppage PVR 350
from my old videotapes using Besweet.
This seems to result only in ultra speedy soundtracks.

I can imagine that it is boring to get such an ignorant
newby in this discussion, sorry for that.

dh
PS Would be obliged if you could put the lost link back.

[Eye of Horus]

Originally posted by hammerstein.dacm
EoH!

Sorry, no offense was intended, had the idea
that only careful mixing of the physical instruments
during a recording session could result in real surround
or dts sound.

But my background knowledge indeed is very modest,
I only seek for a practical way to improve the
soundtracks when burning great old music-videos
on DVD.
So far I only managed to get 6 wav-channels from
the mpa soundtrack I captured with my Hauppage PVR 350
from my old videotapes using Besweet.
This seems to result only in ultra speedy soundtracks.

I can imagine that it is boring to get such an ignorant
newby in this discussion, sorry for that.

dh
PS Would be obliged if you could put the lost link back.

I suppose reading the whole thing will bring you where you want :-)
I know it's a lot, but it helps to understand the process and it's dfferences from the "Dolby" way.
The guide is pretty easy to follow. Just give it a try with one of your favorite tracks.

Kind regards,

EoH

[nuked]

I read the whole thing and it didn't describe the specific mix used for stereo compatibility at all. But anyway, if you don't start with something encoded with a real x and y signal then that's that. There is out of phase stuff that can be pulled to the back using the basic dolby surround trick and it sounds really cool and has some actual realistic merit to it, some artificial pro-logic steering and maybe it sounds cooler but you can't really determine where the signal really came from from a regular stereo source at least, unless as said it was encoded with a special mic in the first place. I believe you can add some room characteristics that may sound awesome, but that's a sound effect, not a realistic representation. All there is is sum and diference, nothing else, ***UNLESS***you can fully deconvolve the phase information from the amplitude information, but I think this requires some fancier frequency analysis, not just mixing matrices. That would be impressive. Then maybe I could be convinced some extra information could be pulled out, but I'd still have to be convinced. Is this what's going on? If not basically seems like you're doing a round about dolby surround upmix by using the diference signal (which CAN sound AWESOME! ) , and why bother to encode that in an AC3?

nukeD

[Eye of Horus]

Originally posted by nuked
I read the whole thing and it didn't describe the specific mix used for stereo compatibility at all. But anyway, if you don't start with something encoded with a real x and y signal then that's that. There is out of phase stuff that can be pulled to the back using the basic dolby surround trick and it sounds really cool and has some actual realistic merit to it, some artificial pro-logic steering and maybe it sounds cooler but you can't really determine where the signal really came from from a regular stereo source at least, unless as said it was encoded with a special mic in the first place. I believe you can add some room characteristics that may sound awesome, but that's a sound effect, not a realistic representation. All there is is sum and diference, nothing else, ***UNLESS***you can fully deconvolve the phase information from the amplitude information, but I think this requires some fancier frequency analysis, not just mixing matrices. That would be impressive. Then maybe I could be convinced some extra information could be pulled out, but I'd still have to be convinced. Is this what's going on? If not basically seems like you're doing a round about dolby surround upmix by using the diference signal (which CAN sound AWESOME! ) , and why bother to encode that in an AC3?

nukeD

"Then maybe I could be convinced..........."

The only thing that convinced me when wroting the guide in the first place : my own ears !
I advice you strongly to do the same instead of all this theoretical bla bla !
And yes there are different ways to make sound awesome, if you're a professional or at least know how to use professional software and when you are pretty much into music in the first place.
Ooh, you can get such excellent results with Reason or MX5.1 or.....
Problem is that no one writes an understandable easy to follow guide for that :-) You need to have a technical background to be able to use these "tools" :-) or even read the manuals !

Where is the part of putting that into AC3 ????????????

EoH

[daphy]

@ Eye of Horus

youre right!;)
i dont think the sense of this guide is to convince the whole world ;)

@ nuked
love it or leave it :sly:

CYA Daphy

[Eye of Horus]

Originally posted by daphy
@ Eye of Horus

youre right!;)
i dont think the sense of this guide is to convince the whole world ;)

@ nuked
love it or leave it :sly:

CYA Daphy

THNX !!! :-)

EoH

[nuked]

"Where is the part of putting that into AC3 ?"

I apologize, by ac3 I just meant more than 2 channels, dts, 5 channel wav whatever you like. My point really just was why bother encoding in 5 channels what can be encoded in 2. Don't get me wrong though. If this "effect" sounds cooler than dolby's effects then of course, in the end it's about what makes you smile when you hear it. But don't jump on the defense when people in passing and with no offense call it an effect. Unless you can prove otherwise, I think most of us are just saying it seems like an "effect", which is ok. Personally I believe it's a little more than an effect, as is dolby surround (the pro logic additions are just effects), I just don't believe it's any less of an effect than dolby surround either.

nukeD

[nuked]

Ok, well your link does NOT describe the mixing process in detail, but
I'm starting to find more information that does. Interestingly
enough the calculations are in fact done in frequency space
and true phase information as a function of frequency is used..
ie sine vs cos, 90 degree phase as oposed to just non
linearly-indepent polarity info used in normal time-space mixing(dolby).
I've also found references
saying that ambisonic mics are not used anymore because you can't
really go back from UHJ tyo B-format anyway. This will always be true
of course cause it's impossible to keep some of the original L/R
elements from leaking into the front/back info and that's a problem
with surround sound as well. However, it does seem this may do a be
able to do little better job than surround of doing that, since there is
more information to work with. This is all from a few
google hits of not so official sources of info. Maybe I (or someone) will
find something better and more comprehensive about it. Interesting though.

It's not clear to me personally yet, if real info is gained for
material that wasn't encoded with an ambisonic mic, but maybe it is.
If dolby can get some info from regular stereo, maybe this can do it a
little better.

nukeD

[Eye of Horus]

This is how I came on it.
I read a lot of the websites and most of the times I didn't understand it. My maths didn't go that far :-)

But from what I read, it must be possible to get info from stereo and to add info to stereo.
At that same time I was discussing the possibilities of this with a good friend , who told me about Stereotomy, the Alan Parsons Project album. APP is my favorite group (www.thealanparsonspages.com shameless plug for my own homepage :-)) and that friend told me that Stereotomy was recorded in Ambisonics, which is a system that adds surroundness to a soundstream. I read some old reviews of that album and reviewers stated : this is the best sound ever heard in stereo on vinyl. It gives you a sense of surround sound, without losing the band. ?? = a lot of the quad albums were : voice from left front, guitar from right front, drums from rear left and bass from rear right, but where the heck is the band ??
Well, I thought and so did my friend, if our favorite artist which also is one of the greatest producers of all time (Dark Side of the Moon, Year of the Cat etc.) used Ambisonics, we want to know more about it and try to develop a method so we can use it too on our stereo albums !
My friend developed the method, I wrote a guide for it !
I don't understand everything from the mathemetics, I'm not an audio professor like the guys who developed Ambisoncs. I'm just a user who wants to get the best out of his cd's without having to spend thousands of Euro's on a top system.
And... it works ! On most albums it's definitely an improvement ! Don't ask me why... but it is !

Three months ago I bought a Philips SACD player. This player has an upsampling function for stereo CD's. It upsamples them to 192 KHz and that signal is send to the receiver. They sound a lot better !
So here is another mystery I don't understand : why do stereo CD's made in 44.1 Khz sound better when upsampled to 192 Khz ? (11.025 Hz upsampled to 44.100 Hz does NOT sound better !)

I don't understand it, but I hear the difference !

The same goes for Ambisonics !

And for me that's all that counts !

Another point : I already said it above , but it needs some more information : we all assume that the method of Dolby is surround. But it isn't ! It is separation (at least DD 5.1 is !).
For me surround means : surrounded by sound.
The sweet spot of Ambisonics is so large that you can easily move a meter to the right without losing the effect. Try that with DD 5.1 !

well.... enough for now.....
just some thoughts.... or as they say : my 2 cents

kind regards,

Eye of Horus

[nuked]

Cool, yeah floyd stuff has awesome surround quality even in stereo
playback on good speakers. I'd heard they used some advanced
microphone techniques.

I don't know so don't take my opinions on it. But I'm guessing
frequency upmixing just smooths out the waveforms. I've heard
extreme audiophiles claim that the square edges at 44 khz can produce
audible effects, even if we're not directly sensitive to those
frequencies. It's topic of debate and some people still claim vynil
is better because of it. I don't think my ears are that
good. Or maybe they are maybe I just don't pay quite that much
attention.

[StoneRoses]

Eye of Horus,

Thank you for all your hard work. I really appreciated.

But please correct your guide (on the first page) a bit.

Ambidec.exe rig setting should be "Surround1" instead of "Pentagon". This will save a lot of time for newbies (including myself) to figure it out. Most of us not read this thread to page 12.

[Eye of Horus]

Originally posted by StoneRoses
Eye of Horus,

Thank you for all your hard work. I really appreciated.

But please correct your guide (on the first page) a bit.

Ambidec.exe rig setting should be "Surround1" instead of "Pentagon". This will save a lot of time for newbies (including myself) to figure it out. Most of us not read this thread to page 12.

First : thanks for the compliments.... !!

I don't know if there has been an update for Ambidec, but anyway...
All the other software (the roompulses) etc. have been tested with the Pentagon setting and work fine !
In the time I wrote the guide, "surround" in Ambidec didn't give good results. I can't remember if it crashed or just didn't give what we wanted. Or perhaps it wasn't even there... The pentagon option always gives good results ! And.... 5 speakers is just fine for the ones without a subwoofer !

I don't quite understand what you mean with "this saves a lot of time "..... and with the "should be" remark.....

Can you please explain in what sense it saves time and when you say should be, why should it be ?


TIA....

kind regards,

EoH

[StoneRoses]

I follow your guide and got strange volume problem like Malow got (see Page 10 of this thread) Covak suggest Malow to use "Surround1" rig instead of "Pentagon" and explain a bit more about the rig.

I did some research to confirm Covak's finding and found info about default rig parameters that "ambidec.exe" used for decoding the B-format file. (from the ambidec author's website)

Here is the URL:
http://www.muse.demon.co.uk/ref/speakers.html

The speaker position relative to the listener are present in [X,Y,Z] vector format. (X+ is the direction toward the listener, Y+ is on the left and Z+ is the direction above the listener)

And here is my quick and dirty drawing of "Pentagon" and "Surround1"

http://www.thai.net/stoneroses/ambisonics/Rig.jpg

On that website, there is the nice b-format player called "Ambisonic Player", it have the illustrations of some rigs on its options menu.

[Eye of Horus]

Originally posted by StoneRoses
I follow your guide and got strange volume problem like Malow got (see Page 10 of this thread) Covak suggest Malow to use "Surround1" rig instead of "Pentagon" and explain a bit more about the rig.

I did some research to confirm Covak's finding and found info about default rig parameters that "ambidec.exe" used for decoding the B-format file. (from the ambidec author's website)

Here is the URL:
http://www.muse.demon.co.uk/ref/speakers.html

The speaker position relative to the listener are present in [X,Y,Z] vector format. (X+ is the direction toward the listener, Y+ is on the left and Z+ is the direction above the listener)

And here is my quick and dirty drawing of "Pentagon" and "Surround1"

http://www.thai.net/stoneroses/ambisonics/Rig.jpg

On that website, there is the nice b-format player called "Ambisonic Player", it have the illustrations of some rigs on its options menu.



isn't your drawing incorrect about the pentagon.... ?

shouldn't it be something like this :


Pentagon

0
0 0

0 0

Surround

0 0 0


0 0


And although I'm open to any suggestion : Did you try it out and listen to the differences ?

And.... please keep this in mind : I wrote the guide with the material I got . Over a year ago. I cannot recall exactly why we used pentagon or what was wrong with the surround option, back then.
I have said that before in this long living thread..... I am NOT an expert in Ambisonics. When you point me to a mathematical explanation about why your opinion is right, I can only say : I don't know, because I don't understand those mathematics ! . PERIOD !

When I started this thread it was very simple : a friend let me listen to some music he converted to the format the way described in my guide. I thought it sounded excellent !! (One of the best conversions I did is IMHO still "Jeff Waynes - War of the Worlds").
My goal was to make that whole difficult process as clear as possible to as many people as possible, from beginner to the one that can handle and understand those mathematics. And please don't forget that a year ago, there was nothing else that came close to my guide ! (And I mean the "how to convert stereo to 5 channel surround with Ambisonics") And believe me.... I did search and search and search.... :-) There just wasn't anything that made it clear and that could be used by a beginner. I didn't write Ambidec, I didn't record the pulses, I didn't write Aurora, I didn't write Cooledit. I just wrote the guide that explained the "How to....".

So , to resume...... I don't doubt what you say, I say : convince me with material, not with theory. Let my ears make the decision !

In other words..... post a converted sample to alt.binaries.sound done with the Pentagon option and the same sample done with the Surround option.....

In this case my ears will give better judgement than my brains :-)

with kind regards,

Eye of Horus

[Covak]

If you run "ambidec -lv" it will give you the details on its standard rigs. The coordinate system is as shown in the picture here:
http://www.ramsete.com/aurora/conversion_between_uhj_and_b.htm

StoneRoses drawings are accurate.

I think Visual Virtual Microphone, with its 5.1 preset, is much better than ambidec anyway.

VVMic is here: http://mcgriffy.com/audio/ambisonic/vvmic/

I believe the 5.1 preset is based on G-Format (the "5.1" button used to be labelled "G-Format" in earlier versions), which pretty much makes it THE way to decode B-Format for standard 5.1 setups.

More on G-Format here: http://www.ambisonic.net/gformat.html

[StoneRoses]

EoH,

According to the info on ambidec website. Pentagon rig is like it was in my drawing.

I don't understand all the mathematic behide ambisonic eigther. The reasons why I pointed you to that page are mainly for the standard rig setup information not the equations.

The speaker positions of "Pentagon" rig are:

<X, Y, Z>

1: <0.8090,0.5878,0.0000>
2: <-0.3090,0.9511,0.0000>
3: <-1.0000,0.0000,0.0000>
4: <-0.3090,-0.9511,0.0000>
5: <0.8090,-0.5878,0.0000>

You can compare other rig types with the illustration from "Ambisonic Player"
See this picture for some of them (no Pentagon and Surround1 rig in the player)
http://www.thai.net/stoneroses/ambisonics/SomeRigs.gif

Here is the info on that page
Stereo
1: <0.0000,1.0000,0.0000>
2: <0.0000,-1.0000,0.0000>

Square
1: <0.7071,0.7071,0.0000>
2: <0.7071,-0.7071,0.0000>
3: <-0.7071,-0.7071,0.0000>
4: <-0.7071,0.7071,0.0000>

Cube
1: <0.5774,0.5774,-0.5774>
2: <0.5774,-0.5774,-0.5774>
3: <-0.5774,-0.5774,-0.5774>
4: <-0.5774,0.5774,-0.5774>
5: <0.5774,0.5774,0.5774>
6: <0.5774,-0.5774,0.5774>
7: <-0.5774,-0.5774,0.5774>
8: <-0.5774,0.5774,0.5774>

You can also use the custom rig for decoding (e.g. using real measurements from your room), it's very easy to do.
http://www.muse.demon.co.uk/utils/ambidec.html


BTW, I did not do serious listening test on this subject. (I just read this thread about 2 days ago) :)

I noticed that somethings may went wrong after I have converted a few songs with "Pentagon" rig, many 5 channels songs have significant difference in level between left and right. (The stereo version did not.) If you see the Pentagon's set up you may guess why this happened.

Anyway, I owed you much for the guide. Thank you.

[StoneRoses]

Originally posted by Covak
I think Visual Virtual Microphone, with its 5.1 preset, is much better than ambidec anyway.

VVMic is here: http://mcgriffy.com/audio/ambisonic/vvmic/

I believe the 5.1 preset is based on G-Format (the "5.1" button used to be labelled "G-Format" in earlier versions), which pretty much makes it THE way to decode B-Format for standard 5.1 setups.

More on G-Format here: http://www.ambisonic.net/gformat.html

I've try that and have a question obout the channel order and LFE.

Is this correct?
Ch1 - LF
Ch2 - RF
Ch3 - C
Ch4 - LFE (It's full range, should I have to filtered this channel?)
Ch5 - LS
Ch6 - RS

[Eye of Horus]

Originally posted by StoneRoses
I've try that and have a question obout the channel order and LFE.

Is this correct?
Ch1 - LF
Ch2 - RF
Ch3 - C
Ch4 - LFE (It's full range, should I have to filtered this channel?)
Ch5 - LS
Ch6 - RS

Where do you get that 6th WAV from ????????

I think you better leave out the LFE.
A lot of setups don't have a LFE anyway and a lot have a LFE where the signal fed to that LFE coomes from the two front speakers. (A not amplified LFE !). And beside that every DD and DTS receiver can make it's own LFE from the input.
The 5 speaker options is the best here.

EoH

[StoneRoses]

I used 5.1 profile in VVM for decoding B-format file. There are 6 speakers in that profile. Speaker #4 located at the center [0,0,0] so it may be used as the input for LFE channel.

[Covak]

I'm not sure what's up with that LFE channel. It didn't appear in the beta versions. It does have the advantage of making a wav with all 6 channels in the right order now though. Whether or not that LFE channel should be used, I dunno. I'm inclined to say no, because as EoH said, any system will handle 5.0 material properly, while I don't think all systems handle full-range LFE channels so well.

[StoneRoses]

G-format speaker setting follows ITU-R BS.775-1 recommendation. (Info from Richard G. Elen of ambisonic.net)

here is the speaker setting for use with ambidec.exe

ITU-R BS.775-1
x y z
FL 0.866025404 0.5 0
FR 0.866025404 -0.5 0
C 1 0 0
SL -0.342020143 0.939692621 0
SR -0.342020143 -0.939692621 0


This is the illustration that compare VVM, ambidec's Surround1 and ITU-R BS.775-1 speaker array.
http://www.thai.net/stoneroses/ambisonics/Compare.gif

According to Richard G. Elen, the LEF channel should not be used for G-format decoding.

[Covak]

Ah, now that is some handy info (and a nice diagram), thanks!

A nice bit of B-Format music: http://pcangelo.eng.unipr.it/Public/B-format/Prodi/Bach/B-format/

I was surprised to find that it sounds a good bit better (to my ears with my speaker setup, at least) decoded with ambidec and that G-Format rig you specified versus VVMic's 5.1 preset.

[Eye of Horus]

Originally posted by Covak
Ah, now that is some handy info (and a nice diagram), thanks!

A nice bit of B-Format music: http://pcangelo.eng.unipr.it/Public/B-format/Prodi/Bach/B-format/

I was surprised to find that it sounds a good bit better (to my ears with my speaker setup, at least) decoded with ambidec and that G-Format rig you specified versus VVMic's 5.1 preset.

From the original guide, somewhere at the end........
"Results are a fraction less then the Ambidec/Besplit method, but that's mainly, because there is no gain control. The volume is lower and it sounds a little bit less crispier, brighter."

EoH

[joshbm]

Isn't abfpan a free utility that does this kind of stuff?

This is a quote from the website --->
ABFPAN:
pans a sound Ambisonically in a circle round the listener, between given start and end positions. Multiple rotations are possible.
Output is either a standard (decoded) quad soundfile, or Horizontal B-Format (standard WAVE file). The latter can be converted to the new WAVE-EX format by COPYSFX.
If so-- is there a way to incorperate this method into a freeware utility such as abfpan of MCTools?

Here is a reference:
http://www.bath.ac.uk/~masjpf/CDP/cmcrefmn.htm#ABFPAN

The download site:
http://www.bath.ac.uk/~masjpf/CDP/swfree.htm
(Scroll down to Multi-Channel Toolkit)

This tool set is definately worth the download! I use interlx, channelx, and copysfx constantly.

It would be nice to see a freeware version of this (and totally automated for newbies like me lol).

-joshbm

[Eye of Horus]

Originally posted by joshbm
Isn't abfpan a free utility that does this kind of stuff?

This is a quote from the website --->

If so-- is there a way to incorperate this method into a freeware utility such as abfpan of MCTools?

Here is a reference:
http://www.bath.ac.uk/~masjpf/CDP/cmcrefmn.htm#ABFPAN

The download site:
http://www.bath.ac.uk/~masjpf/CDP/swfree.htm
(Scroll down to Multi-Channel Toolkit)

This tool set is definately worth the download! I use interlx, channelx, and copysfx constantly.

It would be nice to see a freeware version of this (and totally automated for newbies like me lol).

-joshbm


Yeah, not only for newbies !!

Problem is that (as far as I know) there is no freeware utitlity that can handle and process in 32 bits the way CE does !

EoH

[Mug Funky]

@Eye of Horus:

just stumbled upon this (i'm normally rockin' it in the DV or avisynth forums), and i noticed the links for the aurora plugins don't work.

is there anywhere that these can be downloaded from?

another thing (this may have been answered) - i have Dark Side of the Moon in SQ quad format, but only have a regular stereo cartridge. do you think i'd be able to recover the full image using these tools? it would rock to have the album in 5.1ch AC3.

hope people still read this thread...

[SallyDog]

@Mug Funky

It's alive. Try this link.

http://www.ramsete.com/aurora/

SallyDog

[cooper99]

I'm using SoftEncode DD5.1 and I've tested using SurCode DTS, but I'm unable to play the wav files on my pc. All I hear is loud noise, but when I burn the files on cd and play them on stand-alone dvd player, it's fine. Why can't I play the .wav on PC. My board is the A7N8X deluxe and should be able to handle decoding DD5.1 and DTS, if I'm not mistaken.

[wejgomi]

"Here is the software you need.....

- Cooledit
(http://www.syntrillium.com/download/)
- Aurora plugins
(http://www.ramsete.com/Aurora/download/)
- Tools Ambisonics
Included here - see below message"


but the tools aren`t included in the message anymore..
Where can I get that two programs and 3 .wav files included in "Ambisonics Tools" package ?

[matrix]

Try this link

http://home.wanadoo.nl/appyhappy/Tools-Ambisonic.zip

[Eye of Horus]

Originally posted by Mug Funky
@Eye of Horus:

just stumbled upon this (i'm normally rockin' it in the DV or avisynth forums), and i noticed the links for the aurora plugins don't work.

is there anywhere that these can be downloaded from?

another thing (this may have been answered) - i have Dark Side of the Moon in SQ quad format, but only have a regular stereo cartridge. do you think i'd be able to recover the full image using these tools? it would rock to have the album in 5.1ch AC3.

hope people still read this thread...

No, you won't and........

Don't bother ! DSOTM is probably the most posted one in the DTS usenet group. (alt.binaries.sounds.dts)

There are 4 or 5 different versions available !

EoH

[Eye of Horus]

Originally posted by cooper99
I'm using SoftEncode DD5.1 and I've tested using SurCode DTS, but I'm unable to play the wav files on my pc. All I hear is loud noise, but when I burn the files on cd and play them on stand-alone dvd player, it's fine. Why can't I play the .wav on PC. My board is the A7N8X deluxe and should be able to handle decoding DD5.1 and DTS, if I'm not mistaken.

Problem is probably that it only handles 48 Khz surround streams.
Can you play AC3 in 48 Khz on your PC ? If you can, you need special codecs to play 44.1 Khz surround (as on DTS CD !).

EoH

[Eye of Horus]

Hi all,

This method is now more than one year old.
There were quite some new developments and improvements the last year and I decided to make a new guide based on the new methods.
You can find it here

http://forum.doom9.org/showthread.php?s=&threadid=60137

kind regards,

EoH