GUIDE: Stereo to 5-Channel Surround

[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