GUIDE: Stereo to 5-Channel Surround

[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?