GUIDE: Stereo to 5-Channel Surround

[Eye of Horus]

Originally posted by ursamtl
Oops, sorry. I posted just the text without using the http:// button on the forum interface because I didn't want to repost a link to a file that might potentially be corrupted. I didn't notice that the forum software automatically converted it to a link.

No need to apologize : perhaps it gives some extra visitors :-)



Well, did you try the dipole without moving your speakers? I found the results were better with my speakers at +-25°. At +-10° there was very little sense of width to the sound at all.


Yep ! I try out every bidule :-)


[b]
I look forward to the new bidules. As for the 32 bit conversion, yes, I have done this a couple of time for comparison sake and it does provide a marginally cleaner sound especially in high-frequency details such as cymbals, etc., but I didn't notice any appreciable difference in terms of the Ambisonic effect. I'm lucky in that, for one thing, I can monitor my mixes in real time on my PC so I don't always burn DTS CDs. In addition, my DVD player reads CDRWs so when I do write burn DTS CDs, they're temporary tests to see how the sound is transferring to my 5.1 system. Once I settle upon an upmix method that gives me the results I seek, I'll do some proper full CDs and start with 32-bit files. I still haven't found a satisfactory answer concerning the whole question of dithering. It's generally agreed that going from 32-bit to 16-bit with no dithering reduces sound quality, but so far the Surcode documentation hasn't mentioned whether it downsamples to 16-bit and if it does so, does it use dithering. All it says it that the program accepts 16- or 32-bit files for input. The CD redbook spec is 16-bit, but it's not clear that a DVD player recognizing a CD but playing a DD or DTS wave file does so only if the file is 16-bit.

By the way, yes, your earlier post did clear things up between us. I'm at work and don't have time for a detailed reply, but yes, there are surely cultural nuances to our communication that can cause misunderstandings. For example, I once worked in an office with some Swiss and German personnel who were working on a contract for the Canadian govt. here. They would get into discussions that seemed to me like they were ready to haul out weapons and kill each other, yet walk out of the office at the end of the day smiling at each other! Here I was the poor Canadian fellow shocked into thinking I was about to witness murder! These folks told me this was simply their European temperament and nothing I should be concerned about. Perhaps we Canucks are just too polite! Plus, having worked with them for awhile I found I was adopting some of their attitude.

Have a great weekend!
Ursa


LOL !!! How recognizable !!!

Even some of my friends in the USA (for over 8 years now) still have problems in a discussion with me :-) They always think I'm angry or too sharp.

Have a good (productive !) weekend too !

kind regards,

EoH

[kempfand]

Ursa: Regarding the Stereo Dipole method I was posting here, I wrote it is a prototype, as the 2 "only" speakers for the Ambisonics (placed in the back) are probably not enough to make it a full AmbioPhonics by the books. Do do things correctly, there are at least 4 speakers (square) or even more, as well as convolution with real space impulse responses. Also, the dipole spreads the image in the front onyl (ideally over 180 deg), but there are methods which use a 2nd dipole in the back.

I have also read from two sources (Farina being one of them) that a frontal dipole works best if it comes from above (i.e. two closely spaced speakers at the ceiling to produce a dipole). I can't really report any personal experience on this, as my other half would probably get a nervous breakdown if I changed the living room too much ... :D ... but maybe one day ... :)

On the 32 bit conversion of the feeding Stereo-in, I quote here a Farina post from April 2002 from SurSound, which you/all might find interesting.
Now CEP2 instead defaults to the most usual mode.
In any case, being a float number, no dither is required (nor any special trick) when rescaling. Most availablòe libraries of loops and sound effects are actually 32-bits waveform. All known software smoothly converts betweeen 32 and 16 bit waveforms, so probably You are already using a lot of 32-bits WAV files, and You do not know this.... CEP, as any other program natively supporting 32-bits files (such as Audio Mulch, Cubase, Sound Forge, etc.) plays without problems a 32-bits file over a 16-bits sound board.

This does not pose any conversion problem, because the sampling rate is unaffected, and the bit depth can always be increased without any particular trick.

This is advantageous also when the original sampling was also at 16 bit. 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.

Apart from the storage space on the HD (which is very cheap now), all the computations made in float-32bits are nowadays very efficiently performed in the FPU of the computer, which can outperform the most powerful DSP chips by a factor 16 or more, if the computer is properly programmed (making extensive use of the SIMD instructions, for example). This means that, in practice, che computational load is lower working in 32-bits float than in 16-bits integer! Regards,
Andreas

[ursamtl]

Hi,

I've just added a zip file containing three SIR delay compensation groups to the FTP server and to my web page at www.geocities.com/ursamtl.

This zip file contains three Plogue Bidule groups I've created as alternatives to loading dummy instances of SIR in your bidules to compensate for its sample delay of 8960 (in SIR v.1005). The savings in CPU usage can be significant, ranging from 5-10%


SIRDelComp2 provides 2 channels in and out with a 8960-sample delay on each. This replaces one dummy instance of SIR
SIRDelComp4 provides 4 channels in and out with a 8960-sample delay on each. This replaces two dummy instances of SIR
SIRDelComp4+2 also provides 4 channels in and out with a 8960-sample delay on each, plus 2 channels directly through (pins 5 & 6) for those who want 5.1 channels on one neat bar. For example, L, R, C, LFE, might have delay compensation with channels 5&6 connected to one instance of SIR elsewhere in the bidule.


When I conducted tests on my PC (Athlon Thunderbird 1.1GHz), the results were as follows:
Average CPU usage
-----------------
1 SIR with stereo impulse + 1 dummy SIR: 28%
1 SIR with stereo impulse + 2 dummy SIRs: 32%
1 SIR with stereo impulse + SIRDelComp2: 20%
1 SIR with stereo impulse + SIRDelComp4: 22%
1 SIR with stereo impulse + SIRDelComp4+2: 22%
2 SIRs with stereo impulse + SIRDelComp2: 27%
2 SIRs with stereo impulse + 1 dummy SIR: 33%


Although I did no testing of memory usage, obviously there should be some saving given that SIR is 1.6MB and the groups are a few KB!

Enjoy,
UrsaMtl

[Shayne]

Originally posted by kempfand


........So you must set 'never' for the WAVE Format Extensible (with Preference)........

Regards,

Andreas

Sorry for the late reply but thank you Sir. All is fine now and my computer no longer needs to live in the past.

I have checked pin outs and this release seems to be consistent with besweet (all is well). I did find myself dropping the gains on lfe down to -6.5 db but we are batching again.

Weird being a forced beta tester. But free is good.

Peace

[kempfand]

Originally posted by ursamtl
I have yet to discover convincing impulses for giving me that "being there" feeling. For example, at work I can listen to music through a nice set of headphones, but the plugins or impulses suggested when I search on the net for "out of head" sounds just don't seem to work. In theory, with the right impulses I should be able to sit there and have the music sound as if I had a PA sitting in front of me, but it hasn't worked so far. But, I digress...Check the samples at the bottom of The FIReverb Suite audio demonstration (http://www.catt.se/suite_music/index.htm) for beautiful "out of head" binaural listening experience.

The FIReverb Suite is a superbe package IMHO (as well as the small version CATT-Acoustic (http://www.catt.se/)

Cheers,
Andreas

[kempfand]

Originally posted by Shayne Weird being a forced beta tester. But free is good.[/B]

Plus it's one of the rare examples where beta is labelled as beta. Far too often, betas is sold for $. Also, I've never seen an example where the company (Plogue) was so responsive to implement enhancements, and help for bug fixing.

I'm not a fan of advertising, but I'll be in what will hopefully be a line when the free-version changes to "fee-version".

Cheers and peace,

Andreas

[Shayne]

Well you would think a batch process routine would need to be implemented to make it truly useful that there would be any sort of line.

But they need not listen.

Peace

[Umma]

Well you would think a batch process routine would need to be implemented to make it truly useful that there would be any sort of line.

I wanted a batch routine, too, when I first started with it, but Plogue crashes on me after 5 or so straight uses without closing it down and restarting it. I just use one big wav file now.

Plogue wasn't even developed with this sort of music manipulation in mind. I may be wrong, but I think it was mostly meant for midi or homegrown electronic music. It would be nice if they took out the midi part and sold a "mini-Plogue" to those of us who like to fool with music conversions such as those we are doing over here. :)

[kempfand]

Originally posted by specise_8472
Have just uploaded to the FTP server some new toys.
Daphy will put on Web server when he can.

It is a very good Mono to stereo conversion utility. And feeding the result into one of the above 2 to 5 methods produces astounding results. Better than I had hoped for. I myself personally use my Allinone process.

/// snip ///

For those interested - it is basically feeding the mono signal through two banks of cascaded filters. Each bank is a 19 pole Allpass filter. Then feeding the resultant signals through my x-talk filter to produce better channel seperation.

Species: I tested it (both the original one, as well as the 44.1 & 48 ones). Setup: Mono Signal -> Mono_To_Stereo -> Audio File Recorder. The levels of the recorded L and R are extremely different. Same results when EoH runs this.

Guess it's just a small wrong routing somewhere, and would appreciate if you could re-check.

Kind regards,
Andreas

[Tantulus]

Well,finally I got around to experimenting with Bidules. I decided to add the "ambient hall" impulse from NoiseVault to SIRs on the SAD5.1 bidule. At that time I also decided to spring for a better center speaker and a powered subwoofer. This gave me a chance to try out my "SIR" SAD5.1 adaptation in a decent listening room. Honestly, I was blown away! My cramped living room never did justice to surround sound. Anyway, I felt the soudscape fill the room and the Impulse Response gave enough ambience that when I shut my eyes I could actually imagine being in a concert hall. My impression is that at least with orchestral music one can get a sense of "broader-than-room-size" with a decent speaker setup. Although, at times I felt that I was just bathed in the music rather than sensing the music orginated from the front as in a concert hall, I was very satisfied. Just as mentioned earlier in the thread, the salesman wanted to know how this was done and called others in the store to hear the disc.

Also, as Kempfand noted, there are discs that capture the ambience of the enviroment where they are recorded in stereo and I found with these recordings I get an added benefit of nice localization of instruments. Ambience was augmented without the need for using impulse responses. I guess the results of encoding into 5.1 ultimately depends on the orginal recording process, the speaker setup, the room acoustics and the bidule used. Once again, hats off Kempfand; my receiver has DPLII, DTS, and Neo6 settings. None of them come close to the bidules produced in this forum. At best I get some broading of music and tighter vocals in the center channel with TV or CD's.

Anyway, that's my two cents!

Finally, please, please, could somebody explain what the AmpIn and AmpOut settings in the HNM filter does? I looked through the whole thread I can only find references to settings but not their meaning.

Also, some recordings tend to be too bassy and no amount of changing the low pass control on the subwoofer corrects this. I know I can add a gain control but can this be adjusted with the HNM filters connected to LFE?

A less timid but greatful,
Scott

[ursamtl]

Learning to like Ambisonics
This forum has several guides or suggestions for converting 2-channel stereo to some variation of surround sound, usually 5.1. Each method is fun to experiment with, and each presents something different. I’ve been playing around with them and trying different combinations for a few weeks now. What follows is the result of my experiments so far. I’m quite pleased with the sound it provides and I hope you’ll enjoy it too.

Basically, I took the equations on Angelo Farina’s web site and created a bidule group that converts 2-channel stereo (UHJ) to B format (WXY, the Z is null because no height info is available). This group is a basic Ambisonic encoder with no controls for azimuth, elevation, distance, or anything. It just creates the basic signals.

Then I used different combinations of the equations on Richard Furse’s web site to put together a similar basic bidule group to serve as an Ambisonic decoder. I tried the calculations for the “Surround” speaker layout, but the sound wasn’t that impressive, and Furse’s site points out that this results in symmetry distortion. I then considered the Pentagon suggested by EoH and kempfand, but when I looked at Furse’s suggested equations, I discovered that they do not form a truly equilateral pentagon. Thus, rotating this by 36° to provide three front channels and two back results in a sort of lopsided listening field. To lay out one's speakers to match the calculated points of the rotated Pentagon, one would have to place the center speaker roughly three degrees to the left of center, the left speaker at 74° from center and right speaker at 71° from center. Even then, the center of the rotation would not be centered between the left and right speakers.

http://stevethomson.ca/audio/guides/Pentagon36.gif

I got better results from the Octagon 2 layout. Since it calls for 8 speakers, and there are only five available in my target 5.1 system, I created phantom speakers by splitting the signals for each of the side left, side right, and center surround speakers and feeding them through the two speakers on either side of each phantom.

http://stevethomson.ca/audio/guides/Oct2layout.gif

I tried the equations for both the strict spherical harmonics layouts and the controller opposites. As Furse points out, the latter results in a more diffuse image. I found the spherical harmonics layout sounded better.

When I put this all together and tested it, the sound was quite good, much more expansive than any other Ambisonics I’d heard to date, yet still providing that sometimes uncanny sense of realism that simple Dolby Surround upmixes just can't convey.

I then played around with various enhancements to the input stereo signal, such stereo widening, brightening the stereo edges with EQ, etc. These provided some interesting results, but it got complicated and CPU intensive.

Then I thought about the nature of the signals, if the Y signal consists of the spherical harmonics representing the width of the soundfield, then boosting it with respect to the others should spread the sound out. I tried it, and it worked! More importantly, nothing else in the sound suffered. Stereo wide effects generally kill tight, accurate bass response, but this widened Ambisonic image did not. I also tried the same trick on the X channel to see if a similar improvement would occur in the depth of the soundfield, but it didn’t provide such a spectacular result. However, applying reverb to the X channel--even simple mono reverb--does provide some amazing results.

So, I put it all together in a bidule consisting of nested groups, and learned how to put together some controls in a bidule. This resulted in the following controls, which are available from a control panel when you double-click on the UpMix Studio group in the center of the bidule:

http://stevethomson.ca/audio/guides/Controls.gif

Input Level A standard bidule Gain. The standard setting is at 0dB.
LFE? Checking this box causes all really deep bass centered around 32Hz or so to be redirected to the LFE channel. If the box is unchecked, the channel will be null. If you have monitoring capabilities with a subwoofer on your PC, try disconnecting all other outputs but the fourth from the left, and you'll hear exactly what's going through the LFE channel. By the way, the LFE channel is not affected by the Bass Boost controls below.

Enable Soundfield and Bass Boost? I put this control there for the purists. If you don’t want any width control or bass boost, and wish to just upmix to 5.1 using the basic Ambisonic implementation in this bidule, uncheck this box. If you do want the enhancements, check it!

Soundfield Width Gives you the chance to boost the Y or width channel, thus spreading out the sound. This also tends to make the highs a little crisper, but there is no EQ involved. This one control provides an incredibly spacious yet accurate soundfield. Of course, it all depends on the source file. Try, some Floyd or Roxy Music's Avalon.

Bass Boost Level Adjusts how much boost there is to the bass frequencies below the crossover frequency.

Bass Boost Crossover Frequency Moves the upper limit of the bass boost. If you’d like a more solid bottom but without much more bass, Move this all the way to zero and use the Bass Boost Level to boost the deepest bottom end a bit. This puts a nice bottom on drums without overpowering anything.

And if you really like the sound of the grand canyon…..
If you have a fairly dry piece of music wish to add some reverb, try adding it only to the X channel. I was astounded by the huge, professional sound image I got when I did this. I also tried running both the X and the Y channels through the two channels on a SIR VST. It was pretty massive, but almost too much so. Have fun, play a bit.

http://stevethomson.ca/audio/guides/AddSIR.gif

Monitoring
To get the most out of this system, it’s best to monitor the sound while you make the adjustments. If you don’t have a surround sound system on your PC, just hook up the standard 2-channel output. I’ve found that this gives you a really clear idea of what the sound of the final mix will be. Do not hook up the surrounds, the center or the LFE to this, however, as the sound will be inaccurate. It amazes me just how nice even a 2-channel mix sounds through this bidule. I think I'll use it to do some new mexies for listening in the car.

Note: You’ll have to remove this monitoring sound output device if you want to process in Plogue Bidule’s offline mode.

http://stevethomson.ca/audio/guides/Monitor2chan.gif

Of course, if you have a surround system and a multichannel ASIO setup, you can monitor the whole mix.

http://stevethomson.ca/audio/guides/MonitorASIO.gif

So, have fun. As I said, this provided me with some very nice surround mixes. Unless you add some reverb, the natural ambience of the original recordings is all you have, and most of the time, that’s all you need.

<b>You can download the resulting bidule from

My website on Geocities (http://www.geocities.com/ursamtl/audio.html), another link

here (http://stevethomson.ca/audio/guides/UUMStudio.zip)

or an offline processing mode version from the Projects\Ursa's UpMix Studio folder on the NeedfulThings server (http://www.needfulthings.webhop.org).
Thanks to daphy for doing the mod for offline processing and linking the player and recorder!

As for what to do with the multichannel wave file, that’s been covered extensively in this thread and others. If you’re encoding to AC3 using SoftEncode, just load the file in and launch the encoder. Otherwise, use Besweet to split the 6-channel file and then encode using your preferred AC3 or DTS encoder. By the way, as mentioned earlier in this thread, converting your input wave files to 32 bits does indeed make a difference. The sound difference may be subtle, but you will notice more detailed high frequencies, etc.

For more reading, check out the following:

This whole fascinating thread! (http://forum.doom9.org/showthread.php?s=&threadid=60137)
Thanks to EoH, kempfand, kpex, species, daphy, andy and all the others, you’ve started something very, very cool.

Conversion between UHJ and B-format (http://www.ramsete.com/aurora/conversion_between_uhj_and_b.htm) by Angelo Farina.

First and Second Order Ambisonic Decoding Equations (http://www.muse.demon.co.uk/ref/speakers.html) by Richard Furse.


UrsaMtl, June 6, 2004

Moderator Edit: Fixed Image links per request of ursamtl

[kempfand]

ursamtl: Thanks for sharing your work. It's always good to see what others do, and playing with the various possible schemes in Bidule is so much fun, as you also mention. A few remarks:

1) In the current uploaded bidule (June 6), I believe that the decoding maatrix for C is broken (in "B-Octo2SH-to-5.1"). It should be C = 0.1768xW + 0.2500xY (but currently is C = 0.1768 x W + 0.1768 x X). The "0.2500"-box is there but not linked, so it probably is just broken.

2) It is true that "Spherical Harmonics" decodes result in a (slightly) loss of directional information, but it should be added that "Controlled Opposites" produce a larger listening area, and guarantee that speakers will never generate an `anti-phase' signal.

I did many tests some time back and personally prefer the "controlled opposites", but it's a question of taste (not right or wrong).

3) I analyzed your modified Octagon2 decode, and it clearly produces a slightly more narrow soundfield as compared to the Pentagon decode. This is also expected, as the different speakers receive different W's, i.e. you would expect some 'distortions' for the soundfield (in the case of the modified Octagon2 some kind of squeezing along the x-axis, especially in the FL and FR areas of the soundfield).

I would suggest you still give the Pentagon decode a try, as it produces a perfectly equilateral pentagon (72 deg between each speaker). You can use 2 options:

i) Using BFprocEdit (rotate the soundfield by 36 deg) and then Emigrator (decode to the Pentagon), as on the guide on the 1st page of this threat or

ii) Modify the decoding matrices and use a bidule scheme similar to the one you did:

For Spherical Harmonics:
speaker________________Weights
__________x_______y_________W_______X_______Y
C ___1.0000__0.0000____0.2828__0.4000__0.0000
FL___0.3090__0.9511____0.2828__0.1236__0.3804
SL__-0.8090__0.5878____0.2828_-0.3236__0.2351
SR__-0.8090_-0.5878____0.2828_-0.3236_-0.2351
FR___0.3090_-0.9511____0.2828__0.1236_-0.3804

For Controlled Opposites:
speaker________________Weights
__________x_______y_________W_______X_______Y
C____1.0000__0.0000____0.2828__0.2000__0.0000
FL___0.3090__0.9511____0.2828__0.0618__0.1902
SL__-0.8090__0.5878____0.2828_-0.1618__0.1176
FR___0.3090_-0.9511____0.2828__0.0618_-0.1902
SR__-0.8090_-0.5878____0.2828_-0.1618_-0.1176

In summary, I think you did a great job, and I'm sure it was great fun an a good deal of deep understanding of how Ambisonics works, and what it does and does not.

Kind regards,

Andreas

[kempfand]

Originally posted by Tantulus I guess the results of encoding into 5.1 ultimately depends on the orginal recording process, the speaker setup, the room acoustics and the bidule used.
Tantulus: Think you made a key statement here, so I am re-quoting it. A big deal depends in fact on the original recording and how it was mixed . As a matter of fact, I know some repurposing experts, who use LP's as source, just because they contain the 'right' mix for what they want to do. Now I'm not suggesting to use LP's as source, but I think it makes the point.

As to the AmpIn and AmpOut settings with the HNM filter, I am sorry I cannot really help, as I never use an LFE, and if it is part of a bidules, I just rely on the settings published. I simply think there is too much danger of "things going wrong", unless of course one knows exactly what one does.

To quote from recent posting from the QuadrophonicQuad forum, from a guy I highly respect for his knowledge:

The biggest problem with nearly all of the DTS-CD's is that they are home cooked, and essentially qualify as bootlegs.
If the person who did the original transfer was working on, say, PC speakers or HiFi speakers then they used an inaccurate monitoring environment, and as the studio saying goes "If you can't hear it you can't mix it".
Another big area for error is the extensive use of the LFE channel as a sub channel, when it is by definition a completely different thing.
The Sub channel exists to extend the bass response in systems that do not have 5 full range monitors/speakers, and NOT as a subwoofer.
When you start encoding music with an LFE channel, you run the following risks.

A/. The end user has not set things up correctly, and that pumpimg bass just isn't there.

B/. You can never know all the crossover frequencies that end user sub/sattelite systems use, so phase cancellation is a big problem. There is also the reverse of this, and what is piped to the LFE may well add to what is already in the bass end of the main speakers, giving the "bass heavy" problem you describe

C/. (The most common) the system for playback is simply not properly calibrated, the sub is too loud, the balances between all the other 5 channels are phase delayed & comb filtered and the result is an unpredictable bloody mess.

D/. There is a bass management setting on the AV amp, and it is conflicting with what is piped to the LFE Channel.

E/. Incorrect crossover frequency set in the DTS conversion. In an ideal world, you do not use the LFE but instead use an LFE splitter to emulate the effects of Bass Management. Set the X-Over to 80Hz and the slope to 24dB/Octave, or even steeper if possible. Never, ever encode with the LFE active unless you know that your monitoring environment is 100% accurate & properly calibrated. And only then with extreme caution.

In short, do not use the LFE channel unless you are doing film scores or the 1812 overture, and trust to the Bass Management in the AV amps to sort everything out. It is what it is there for.


Kind regards,

Andreas

[ursamtl]

@kempfand,

Thanks for the feedback. Just a couple of brief comments for now as I have some balcony gardening to take care of.

Originally posted by kempfand
1) In the current uploaded bidule (June 6), I believe that the decoding maatrix for C is broken (in "B-Octo2SH-to-5.1"). It should be C = 0.1768xW + 0.2500xY (but currently is C = 0.1768 x W + 0.1768 x X). The "0.2500"-box is there but not linked, so it probably is just broken.

Actually the numbers I chose are intentional. I did forget to delete the unused 0.2500 constant. You see the Octagon is designed for a soundfield where the center speaker is at the same distance from the listening point as the others. Since most surround systems situate the center speaker on the same plane as the fronts, I felt it would be more accurate to attenuate the center somewhat. After doing a scale drawing in Illustrator (which I will share once I resolve some problems linking to graphics), I determined that since the center speaker on the same plane is 0.7071 the distance (interesting how that number keeps coming up :), I attenuated the center by that amount. As for there being different values going to different speakers, of course there are. That's how the phantom speakers are created. For example, the front left speaker receives 0.1768 for it's own signal, plus 884 (half of 1768) for the phantom left side speaker. This totals 0.2652. For the rear left speaker, I took this amount representing the rear left plus half the rear side, plus an additional 884 for half of the rear center phantom speaker. In fact, all the virtual speakers in my implementation of Furse's Octagon 2 receive the same amount of sound pressure with the exception of the center, the reason for which I just explained.


2) It is true that "Spherical Harmonics" decodes result in a (slightly) loss of directional information, but it should be added that "Controlled Opposites" produce a larger listening area, and guarantee that speakers will never generate an `anti-phase' signal..

Yes I read that too, but surprisingly, to me controlled opposites don't sound bigger. They actually remind me of the sound of audio that is out of phase, in that they produce an overly diffuse soundfield with nothing in focus. I personally found the spherical harmonics sounded much more appealing and more like the kind of professional 5.1 sound I so like.

3) I analyzed your modified Octagon2 decode, and it clearly produces a slightly more narrow soundfield as compared to the Pentagon decode. This is also expected, as the different speakers receive different W's, i.e. you would expect some 'distortions' for the soundfield (in the case of the modified Octagon2 some kind of squeezing along the x-axis, especially in the FL and FR areas of the soundfield)..

Your statement really surprised me. In all the tests I've done on all the bidules over the past few weeks, the Pentagon decoded bidules sounded much more narrow to me. Until I actually tried the Octagon 2 layout (which I believe species 8472 came up with first in this thread. I simply added the phantom speakers). I dismissed Ambisonics and completely unacceptable for producing surround sound.

I would suggest you still give the Pentagon decode a try, as it produces a perfectly equilateral pentagon (72 deg between each speaker)..

I did try a Pentagon layout and it sounded really compressed and lifeless compared to Octagon. Even a 4-channel cube layout is much better. Once took the calculations on Furse's web site and plotted them in Illustrator, I realized tht it does not seem to produce an equilateral pentagon (assuming the point of rotation is the geometrical center at -0.0489, ambisonic X). Check the diagram I just posted in my original message. Worse, once rotated 36° to the left, the soundfield is completely distorted. I already edited my message this morning once I had time to measure the angles in my drawing (which I'd left at work).

Even if the rotation is centered on the 0,0 listening position, there is still a problem with playing back a file with harmonics calculated for a wide soundfield, but on sound systems with real speakers placed differently. The ideal position for the front left speaker in the rotated Pentagon is at 74° from center, a full 29-44° from the suggested placement in a typical 5.1 system. the Right speaker is at 71°, or 26-41° from the typical position. Therefore, if you play back a file decoded through the Pentagon, the width of your soundfield is being compressed between 45-85°!! No wonder I found the Pentagon bidules lacking in width! I'm really surprised that you actually ended up making such a conclusion about the Octagon 2 layout. Sit down and draw it out, you'll see. I prepared some graphics for my post last night but there were some problems with linking to my geocities site. I'm going to place them on another site and edit my post. I think if you look at them and consider the geometry of it all, there's no way you can conclude that the Octagon 2 soundfield is compressed , certainly nowhere nearly as much as the rotated Pentagon.

Anyway, time to tackle the garden. By the way, did you try the effect of widening the soundfield with the slider I implemented? It sounds amazing!

Regards,
Steve.

[kempfand]

@ursamtl/Steve:

Thanks for the explanations. Few comments from my side.
did you try the effect of widening the soundfield with the slider I implemented? It sounds amazing! I tried, and the effect IS truely amazing. As indicated before, I also like the way you implemented this finding using bidule, as it makes testing so easy.
I felt it would be more accurate to attenuate the center somewhat. After doing a scale drawing in Illustrator (which I will share once I resolve some problems linking to graphics), I determined that since the center speaker on the same plane is 0.7071 the distance (interesting how that number keeps coming up , I attenuated the center by that amount. That makes it clear to me now.
As for there being different values going to different speakers, of course there are. That's how the phantom speakers are created. For example, the front left speaker receives 0.1768 for it's own signal, plus 884 (half of 1768) for the phantom left side speaker. This totals 0.2652. For the rear left speaker, I took this amount representing the rear left plus half the rear side, plus an additional 884 for half of the rear center phantom speaker.Understood. But:
In fact, all the virtual speakers in my implementation of Furse's Octagon 2 receive the same amount of sound pressure with the exception of the center, the reason for which I just explained. I disagree :) : Pressure (W) going into C: 0.1768, each of FL & FR: 0.2652, each of SL & SR: 0.3536.

Now: This is just a factual statement (i.e. observation) I made, i.e. not saying 'right' or 'wrong'. In fact, I have seen unpublished decodes from well respecetd experts in this area who also apply different pressures to the decodings to the speakers.

3) I analyzed your modified Octagon2 decode, and it clearly produces a slightly more narrow soundfield as compared to the Pentagon decode. This is also expected, as the different speakers receive different W's, i.e. you would expect some 'distortions' for the soundfield (in the case of the modified Octagon2 some kind of squeezing along the x-axis, especially in the FL and FR areas of the soundfield).. Your statement really surprised me. In all the tests I've done on all the bidules over the past few weeks, the Pentagon decoded bidules sounded much more narrow to me. Until I actually tried the Octagon 2 layout (which I believe species 8472 came up with first in this thread. I simply added the phantom speakers). I dismissed Ambisonics and completely unacceptable for producing surround sound. You clearly see the mentioned deviation when you plot your layout using the CATT-FIReVerb Suite, whereas there are no deviations for the rV and rE deviations when using a Pentagon decode.

But same remark as above: Just an observation :) As a side-remark, it's interesting to see how people have have & are putting patent protection to 'their' decoding specs (IMhO one of the reasons why Ambisonics never really made it to the masses). My personal opinion is summarized well in Anegelo Farina's presentation from May 2004 (at the Berlin AES convention): "We are still learning what is the best way to render ... over a standard 5.0 (or 5.1) setup".

I did try a Pentagon layout and it sounded really compressed and lifeless compared to Octagon. Even a 4-channel cube layout is much better. Once took the calculations on Furse's web site and plotted them in Illustrator, I realized tht it does not seem to produce an equilateral pentagon (assuming the point of rotation is the geometrical center at -0.0489, ambisonic X). Check the diagram I just posted in my original message. Worse, once rotated 36° to the left, the soundfield is completely distorted. I already edited my message this morning once I had time to measure the angles in my drawing (which I'd left at work). This is actually the point I am most keen to understand better. Maybe I just have 'bricks' in front of my eyes, but plotting the Pentagon (rotated by 36 deg) gives a perfect equilateral Pentagon in my plots (see the x,y coordinates I gave above). In fact, I was and am using this to construct signed filters to decode B-format to 5 speakers (using 15 filters in total, 3 for each of the Pentagon-rig speakers, using 2 instances of Prinstine Space to do the 16 concurrent convolutions).

I think if you look at them and consider the geometry of it all, there's no way you can conclude that the Octagon 2 soundfield is compressed , certainly nowhere nearly as much as the rotated Pentagon. In addition to my remark above (i.e. Octagon2_MOD is compressed a bit on FL & FR, as per CATT-FIReVerb analysis), I think a 'complication' comes by the fact that there are 2 aspects:
a) decoding for a specific speaker configurarion. If this was the only criteria, we should all go for the Surround-decoce (= ITU-5.1), which is what we actually do not.
b) actual speaker-set-up in our listening room, which delivers the created output.
I other words: A decode for a specific speaker configuration (be it Pentagon or Octagon2_MOD or xyz) CAN sound give good results, as a function of listener's preference and actual speaker set-up (which most often is not the standard ITU-5.1).

Hope the gardeing was a succes. Time to get some sleep here.

Regards,

Andreas

[Tantulus]

Thanks Kemfand for your reply:

Part of my problem with the LFE channel is that I have too damn many controls on my AV receiver (Onkyo TX-DS797) not to mention my subwoofer. Since I use my system for DVD's and music I decided to take the bidules as is and do the adjustments on the receiver so that I can switch from one to the other.

B/. You can never know all the crossover frequencies that end user sub/sattelite systems use, so phase cancellation is a big problem. There is also the reverse of this, and what is piped to the LFE may well add to what is already in the bass end of the main speakers, giving the "bass heavy" problem you describe

I have a switch on the subwoofer that allows my to adjust the phase which seems to be set correctly for my ears.


I have a dial that allows my to adjust the low pass from 200htz to 60. The manual for the subwoofer recommends setting the low pass at 200hts because I'm using the decoder's SUB OUT jack. However, this takes on to much of the bass and results in the bass heavy problem. Although its a pain to keep changing the Lowpass it does allow me the flexibity of how much bass I want to capture from the other speakers depending on the format.

C/. (The most common) the system for playback is simply not properly calibrated, the sub is too loud, the balances between all the other 5 channels are phase delayed & comb filtered and the result is an unpredictable bloody mess.

I went to "Radio Shaft" and bought a SPL meter to set the levels of all the speakers to 75 decibel sound pressure. However, again by rereading the manual I found that I can vary the levels while monitoring the music to get the reponse I'm looking for.

Set the X-Over to 80Hz and the slope to 24dB/Octave, or even steeper if possible

I'll check out this setting on the subwoofer.

I feel I do need the LFE. It brings out the lower register instuments such as cellos, bass's and bassons. I'm curious why you do not use the LFE. Do you have speakers with good bass response or do you use the receiver?

Anyway, your respones has spurred my on to find the ideal settings. Thanks again.

Now to Ursmtl:

I noticed in your diagramns that you have the listener placed in a central location relative to the speakers. However the manuals suggest that the surround speakers should be placed in line with the listener about 3 feet above the ears. Are you implying there is a sweet spot for ambisonic 5.1? Also, I recently purchased surround speakers that can switch from dipole to monople to bipole. Would any of these settings be better for the surround effect or do I need back surround speakers?

Thanks everyone for your assistance!

Scott

[Tantulus]

Sorry I forgot one more thing. For those of you with Windows 2000 Pro or XP. I'm trying to write a script using Windows Script Host to automate the encodeing process. However, it's going to take some time. Is anybody trying to do this?

[Eye of Horus]

Hi all,

After 2 months of hard work, discussions all over, listening, rewriting, discussions again, adjustments, etc. etc.,
Kempfand and EoH proudly present our new bidules.
These bidules are completely different from everything published till now and in our opinion so are the results.
Our goal was to come as close as possible to the HardWare Repurpose guys and..... in our opinion on some tracks we even did better.
If you agree with us remains the question
We would love to hear your feedback !

NEEDED
--------

1. Besweet b28
2. MDA VST
3. Voxengo stereo VST (freeware !)
4. SSRC.exe
5. bidules and groups

DOWNLOAD
---------

The whole package can be downloaded here :
www.app.demon.nl/Kempfand-EoH-06-09-2004.rar

or on Needfull Things......

SETUP
------

1. Install Besweet, start.bat and SSRC.exe in the same folder
2. Install MDA and Voxengo Stereo into the VST folder of Plogue
3. Install the 3 bidules in the layout folder of Plogue
4. Start Plogue Bidule and rescan for plugins and groups
5. Make sure the filerecorder is saying : 16 bits !!!!

USAGE
------

There are 3 bidules which handle different kind of music.

1. Voice-Center.bidule
This bidule handles music with a singer in the center of the stereo. This is checked quite easily by putting your receiver into DPL mode and listen very carefully to the voice. If it's coming from the Center speaker only, then you should use this one.
After conversion, the voice comes out of the Center only.

2. Voice-Non Center_or_instrumental.bidule
This one is useful for songs where the voice in not (or not only) in the center and for instrumentals. After the conversion you will hear the voice from all speakers, but the accent is on the Center

3. Instrumental.bidule
Use this one or the previous for tracks without voice.

4. For a first run, don't change a thing in the bidules, but do make sure the filerecorder is using 16 bits. Of course your source need to be converted to 32 bits floating first for the best results !!

5. Save your output to the Besweet folder

6. Run start.bat, but first edit (right-click and chose "edit") it to suit your settings. Here is the batch routine :

BeSweet.exe -core( -input 01.wav -output f:\01- -6ch )

ssrc.exe --twopass --normalize f:\01-FL.wav f:\01-New-FL.wav
del f:\01-FL.wav

ssrc.exe --twopass --normalize f:\01-FR.wav f:\01-New-FR.wav
del f:\01-FR.wav

ssrc.exe --twopass --normalize f:\01-C.wav f:\01-New-C.wav
del f:\01-C.wav

ssrc.exe --twopass --normalize f:\01-SL.wav f:\01-New-SL.wav
del f:\01-SL.wav

ssrc.exe --twopass --normalize f:\01-SR.wav f:\01-New-SR.wav
del f:\01-SR.wav

copy f:\01-LFE.wav f:\01-New-LFE.wav
del f:\01-LFE.wav

As you see I use 01.wav as name for the 6channel WAV file in the Besweet folder (in my case on e and I do the processing to drive f:\ .
Change these values to the ones you wish to use. (Don't use spaces in the filename !) If you want the input WAV in the same directory, leave out the harddisc letter , but make sure it has another name after "-output" in the Besweet line !

In my case it will output 6 mono 16 bits 44.1 Khz WAV's to harddisc f: .

As usual you can use these to make a DTS cd......

EXTRA INFORMATION
--------------------

Voxengo stereo VST uses presets and we got a very good result with a preset of "pretty wide" . If you wish to experiment with these settings, it's very easy. Just doubleclick on the VST in Plogue and adjust to another preset or make your own. However : our results are all based on the "pretty wide" setting and we don't take responsibility for a worse conversion when using other settings

The same goes for the normalize routine. With these bidules we found that making all 5 mono files at the same peak level, gives the best result. It can be that you wish to have less sound in the rears. Just edit the batch and replace the setting for SSRC to your own preferences.

Regarding the messages about the LFE : all three routines don't have a separate LFE output ! The LFE file you see in the routines is empty !


Well, this is about it..... happy testing and please share your thoughts here !

kind regards,

Kempfand and Eye of Horus

[MaroonMike]

EOH,

Are you suggesting that this new method is an improvement over the Ambisonics method? Can you explain how this method is different (from a sound perspective?)

Thanks for your work on this. Mike

[AllTimeSToneD]

once i start the "start.bat" i get following error:

Error 58: Error : Unknown Input-File Format : "01.wav".
Quiting...

i named my wave file also 01.wav i also tryed to fix the wave file using besliced but still getting the same error :(

[kempfand]

The 3 new bidules make usage of Gerzon technology (Gerzon 1997 Groups & Bidules (http://forum.doom9.org/showthread.php?s=&postid=508560#post508560)), but are much more elaborated than the basic groups.

We are actually keen on your feedback & opinion as to how they do, but I wouldn't see them as or call them "competing" to Ambisonics, as they try to achieve a different result.

Cheers,
Andreas

[kempfand]

http://forum.doom9.org/showthread.php?s=&postid=499028#post499028

Course assuming you are using adjusted paths to the 01.wav too (f:\)

[AllTimeSToneD]

Hehe its weird but yes i use "f:\" for all my wave processings too :D
Thanks for the link! Haven't seen that option at all :)

[ursamtl]

EoH & kempfand,

Just some initial comments since I don't have time tonight to do more thorough testing.

It's obvious you put some thought and experimentation into your new bidules and I congratulate you for this. My initial impression is that the Voice-Non Center or instrumental bidule has quite a bit of potential. This first version produces a nice large soundfield, but it's a bit too "reverby" for my taste. What I mean by this is that it seems to "color" the sound as if it were running through a multichannel reverb unit with the wet level a bit too high. Perhaps it's because of the delays in Stereo Touch. For example, I just took the song "Avalon" by Roxy Music and ran it through the bidule and then did a DTS CD. The percussion in this song really seemed doubled, at points where the highs were flammed by the doubling and the bass suffered from lack of focus. For comparison, I ran the same song through the UpMix Studio bidule with the wdith control on full and put it as the second track on the DTS CD. I still got the big soundfield, but without the doubling on the percussion.

For another test, I took Pink Floyd's High Hopes. Processed through the Voice-Non Center or instrumental bidule, the bells at the beginning sounded like they were in an enclosed space, say a huge arena. The birds chirping and the fly buzzing through seemed overly loud and reverberated. The UpMix Studio bidule version sounded as if the bells were outside and the fly and birds sounded as if they they were right there.

Then I tried some classical guitar music, and I found that the delay and reverb seemed to overwhelm the details in the guitar sound.

Having said all that, I think with some adjustment, this bidule has some potential. As for the other two, I didn't test them very much because after listening to them, I found that their sound is basically not much different from a good, old-fashioned stereo difference signal (L-R) with a bit of bass thrown in, plus a single mono center on the voice-center. I've been playing around with difference signals since the mid-70s when I found a circuit in an electronics magazine for hooking a speaker up across the two positive terminals of an amp. This can produce a very ambient sound from properly recorded stereo material, but it can also sound really weird if instruments are hard panned. An out of phase version is basically the surround channel in Dolby Surround encoding.

Anyway, I'm sorry if this sounds overly negative, but to my ears, the Voice-Non Center or instrumental bidule is the only one with potential. The other two actually sound far worse than my first MatrixMixerEmulator bidule. Could it be that some connections are missing from the two that sound like just a difference channel?

Please don't take this the wrong way. I appreciate that you both put a lot of effort into this, but you did ask for feedback. Perhaps you could explain some of the inner design of the bidule and we can all work together to improve them to the point where they meet our needs better.

Regards,
Ursa.

[ursamtl]

Originally posted by Tantulus
Now to Ursmtl:

I noticed in your diagramns that you have the listener placed in a central location relative to the speakers. However the manuals suggest that the surround speakers should be placed in line with the listener about 3 feet above the ears. Are you implying there is a sweet spot for ambisonic 5.1? Also, I recently purchased surround speakers that can switch from dipole to monople to bipole. Would any of these settings be better for the surround effect or do I need back surround speakers?

Thanks everyone for your assistance!

Scott

Hi Scott,

Actually, if you do a search around the net for "ITU-5.1", you'll find some diagrams showing the surround speakers at rear angles of about 45° from the center with the fronts at about 30° from center. I've seen some manuals suggest putting the surround speakers point towards each other, others suggesting pointing them towards the listener, and still others suggesting that one point them directly towards the front. The problem with some of this is that it comes from the days of the original Dolby Surround, when both surround speakers were fed with the same signal, an out-of-phase difference signal (front L-front R) with the highs rolled off at 7kHz. The intention was not to create any accurate directionality in the rears, but simply to give the impression of being surrounded by ambience. By shifting the signal out of phase, the ambience is not really directional. The actual ITU-5.1 diagrams I've seen point the speakers towards the listener. As for the fronts, the ideal listening spot is an equilateral triangle with the listener at one point and the L and R at the other points. Of course the center is right in the middle.

As for there being an Ambisonic 5.1 sweet spot, well, yes by definition, the listening point is in the center. I'm starting to come to the conclusion that what we're doing is not "real" ambisonics. If you check Angelo Farina's web site, where I got the formulas for converting stereo to B-format, he states that they are for converting "UHJ" to WXYZ. He states that "UHJ is a 'standard' stereo waveform, which includes information capable of driving a complete horizontal surround system (for example, equipped with 5, 6 or even 8 loudspeakers)." I've written him for confirmation on this, but what he's basically saying is that UHJ is a "stereo" waveform in that it is two channels, but it is unlike regular stereo waveforms in that it also contains encoded extra surround information that, when decoded, allows one to recontruct a complete horizontal surround soundfield. I did some more searching around the net for UHJ info, and the info I found seems to confirm this.

Therefore, taking a regular commercial stereo recording and running it through any of the Ambisonic bidules and/or VSTs we've been discussing is not going to produce true ambisonics. Obviously it does do something. The observation I made last weekend that increasing the Y channel with respect to the W and X increases the perceived width of the soundfield, seems to confirm that the encoding formulas create combinations of sum and difference signals plus phase information in each signal. If one listens to the Y channel isolated from the others, it sounds a lot like a standard difference signal (L-R) but with better frequency response. This was confirmed in the UpMix Studio bidule when I increased the width, the bass remained strong and focused. Doing that with a normal stereo wide effect reduces bass response dramatically. Therefore, encoding a regular stereo signal and processing it as real Ambisonic material certainly seems to have a lot of potential. Of course if part or all of the mix is recorded properly, it [i]is[/] really Ambisonic, but most off-the shelf recordings nowadays feature a whole hodge-podge of recording techniques.

Anyway, I have to get some sleep. I didn't get a chance yet to address the whole issue of LFE use and Kevin Shirley's (mixed the Led Zeppelin DVD) response to my email concerning LFE use. Briefly, he does use it, but makes sure the bass on a system without LFE sounds good and strong enough for those who do not have subwoofers. One thing I found interesting in his response and also in his musings as revealed in the diary on his web site (http://www.cavemanproductions.com) is that he does completely separate mixes for stereo and 5.1 on a DVD. Some engineers would probably just do the stereo and then recycle it as the fronts for a 5.1 mix, but Kevin approaches the two formats as separate entities and mixes them as such.

My feeling as well is that a properly adjusted system with a bass management system renders the whole point moot. When I first set up my system, I had to tweak the sub settings and phase switching to get the sound right, but once I did, the system sounds great. I recommend a test DVD or one of those THX test tracks that are found on some commercial DVDs one rents. The whole notion of avoiding the LFE channel seems ridiculous and a waste of a potential resource.

Ok, the last of a huge multipart download is complete and my better half is going to leave me if I don't come to bed, so goodnight!

If you have any further questions, don't hesitate to ask!

Regards,
Ursa

[kempfand]

ursamtl/Ursa,

Speaking for myself, I do not take your comments as overly negative, and I value the initial feedback which we were looking for. I and a few others like the 3 new bidules very much, and I think they beat the existing bidule methods, but they still can be improved (hopefully), and yes, you are right: maybe a joint effort gets us there. Maybe the bidules do not beat the outcome of a professional audio engineer who sits for 2 weeks to do an upmix, but this wouldn't be a fair comparison, as the bidules do a number of CD's in just an hour.

I would suggest (to all readers also) that you try the new bidules on a large number of various pieces of music. You might be disappointed if you just shoot for the most difficult pieces of music (did that myself in the first place). You might also be disappointed if you shoot for music you know well & love very much, as you might have too harse expectations to the outcome. You might be very happy if you compare what they do to standard DPL2 upmixes.

The overall idea of the new bidules ("inner design" as you say) is simply as follows: Reproduce using Software (SW) what some of the Hardware (HW) Repurposing wizzards did & do.
As crazy as this might sound, I believe that SW Repurposing these days can do it, maybe even better. The same applies for Ambisonics, where the increased precision of internal processing of the SW tools just beats the established old HW units.

The idea of HW repurposing "per se" is IMHO not really a huge secret: "All" it does (i.e. what the HW people do) is to to apply (as an example) a little bit of SRS II and use that for the C, apply DPL 2 and use it for L & R, and finally apply a Sansui decode for the Surrounds. Some of the best HW Repurpose works I know of are done that way or similar (maybe even applying small variations to different parts of the album, or even to different parts within the same track).

As far as I can see, the real difficulty for Repurposing is to select the 'right' music, and often the right mix of the same album. As mentioned earlier, some even take "old" LP's, just because they contain the right mix. After all, isn't it surprising that there are not too many works around which were HW repurposed ? I think it is because of the reasons I outlined (not all music suitable; often various mixes around, some working well, some not).

"Reverby" thing: This is something I noted too. I peronally like it, but you can play with the settings of Voxengo's Stereo VST to change this. Plus there is quite a number of other mono_2_stereo schemes or VST's (i.e. using a simple 'pseudo stereo' scheme: S->L', S->Inverse->R', or specise_8472's more elaborated mono_to_stereo groups mentioned and published earlier in this threat)

"Color": Again try changing the settings of Voxengo's Stereo VST to suit your taste. Speaking about "coloration" (not sure if you mean that one), this also can happen with Ambisonic decodes (esp. on the ITU-5.1).

Finally, a brief comment to your reply to Tantulus/Steve: The methods described here do proper Ambisonics, but the feeding material is not true UHJ. Most importantly, it lacks the 3D information (Z-axis), but who cares, as most of us don't have a 3D speaker setup.

The key thing is that feeding Stereo through an Ambisonics upmix still can produce surprisingly results on "Stereo". In fact, it often does ! For very dry recordings, Ambiophonics is probably a better way to do things (i.e. a Stereo Dipole plus Ambisonics as 'supportive' means 'only'). See also: Farina's comments regarding the concern about the usage of UHJ-to-Surround conversion for not-UHJ stereo recordings (http://forum.doom9.org/showthread.php?s=&postid=419102#post419102)

Regards,

Andreas

[brock101]

Hi EOH And kempfand

I have been following your Adventures with regards to stereo manipulation :D
Sadly every guide which i have tried, i have had the same problem occur many many times, when importing the mono wav files into surcode, some wavs cause surcode to warn me that the wavs are shorter than the others :angry:

Has anyone else had the same problems
as i have seen no one else mention this problem

I had given up on converting some of my cd collection into dts... due to the frustration caused... but i decided to give it one last shot due to the new way ... sadly no luck :(

may i add out of the 20+ attempts i had 1 success
i did war of the worlds sadly disk 1 worked but not disk 2 -_-

Regards
Brock101

[ursamtl]

Originally posted by brock101
Sadly every guide which i have tried, i have had the same problem occur many many times, when importing the mono wav files into surcode, some wavs cause surcode to warn me that the wavs are shorter than the others :angry:

Has anyone else had the same problems
as i have seen no one else mention this problem
[/B]

Hi Brock,

I haven't had this problem per se, but I have noticed that you have to clear Surcode from the first set of files you were working with before starting a new project. It retains the files loaded into the previous session when you first open it. If you then try to load individual files without first starting a new project (Ctrl+N) or clicking the C beside the channels you want to empty, you'll run into problems. Also, are you following the advice from the original guide and exporting to one 6-channel wave file? I recall reading some talk of just connecting six individual 1-channel file recorders in a bidule as a way of skipping the splitting with Besweet step. I haven't tried this but if I remember correctly, it causes problems as well.

Don't give up! all this experimentation can be fun.

Regards,
Steve.

[kempfand]

Hi brock101,

Welcome to the forum ! :)

I am having this "error" from time to time as well, and so do others I am aware of. I am also assuming you use the 6-channel Audio File Recorder (i.e. not 3 x 2-channel recorders).

As we found out, this "error" (i.e. warning upon import into SurCode about the mono-wav's having different lengths) does not matter, i.e. has no negative impact on the DTS-packaging. I have tested this on a very long file (nearly one hour), and didn't hear anything out of sync even near the end of the playing time.

It is really difficult to say where this happens^. Could be with Bidule (which I doubt, because Plogue looked into this I think), could be BeSweet, could even be SurCode, could be slow drives or too many other processes running during the demux.

In any case, I would encourage you to try some and see if you really hear something 'out of sync' at the very end.

Good luck,

Andreas

[Eye of Horus]

Originally posted by brock101
Hi EOH And kempfand

I have been following your Adventures with regards to stereo manipulation :D
Sadly every guide which i have tried, i have had the same problem occur many many times, when importing the mono wav files into surcode, some wavs cause surcode to warn me that the wavs are shorter than the others :angry:

Has anyone else had the same problems
as i have seen no one else mention this problem

I had given up on converting some of my cd collection into dts... due to the frustration caused... but i decided to give it one last shot due to the new way ... sadly no luck :(

may i add out of the 20+ attempts i had 1 success
i did war of the worlds sadly disk 1 worked but not disk 2 -_-

Regards
Brock101

Hi Brock,

I have had these problems too, but as kempfand said : nothing to worry about. Did you try the different I/O and DSP settings for the buffers in Bidule ? You could try increasing these to see if that solves the problem.....

Anyway, glad you enjoy working with our stuff, but it would be better to get the results you want !
I have done WotW myself several times. Actually it was the very first one I ever did, with the Cooledit/Aurora method. I also used it on several other bidules to compare. And..... the voice-center bidule gives an excellent result !
BTW I never had problems with the second disc of WotW !

I hope you will have better results in future.

Another tip : try to run on a clean machine (no other programs or tools or virussoftware running....

kind regards,

EoH

[ursamtl]

Originally posted by kempfand
ursamtl/Ursa,

Speaking for myself, I do not take your comments as overly negative, and I value the initial feedback which we were looking for. I and a few others like the 3 new bidules very much, and I think they beat the existing bidule methods, but they still can be improved (hopefully), and yes, you are right: maybe a joint effort gets us there. Maybe the bidules do not beat the outcome of a professional audio engineer who sits for 2 weeks to do an upmix, but this wouldn't be a fair comparison, as the bidules do a number of CD's in just an hour.

I would suggest (to all readers also) that you try the new bidules on a large number of various pieces of music. You might be disappointed if you just shoot for the most difficult pieces of music (did that myself in the first place). You might also be disappointed if you shoot for music you know well & love very much, as you might have too harse expectations to the outcome. You might be very happy if you compare what they do to standard DPL2 upmixes.


I think your comments might underline where we differ on this whole thing. If I'm going to invest the time to put into this experimentation, I don't want to do some rush job and neither do I want to spend my time on music that isn't my favorite or that isn't too complicated. As I said in an earlier post, I can do tests on CDRW so I'm not ending up with a bunch of written CDs wasted because I'll never listen to them again. It's just that I believe that quality results are possible, otherwise, I could use my time much more wisely by slapping on the Pro Logic II, cranking up some good music and getting on with the other tasks that fill my life outside of work.

I understand that's it's great to be able to automate this process as much as possible, and maybe that's what most people are looking for, but since each piece of music is unique and presents different demands, I think it's worth taking the time to tailor the upmix for each one. Hence, the ability to adjust and monitor the bidule's effect is important to me. I was happy to discover that my last bidule processed the music in such a way that the adjustments I provided are still easily heard if someone doesn't have a surround setup on his or her PC. It's so rewarding to be able to do this kind of adjustment to make the music the best it can be. I know it's not everyone's goal, but I don't think I should compromise my goal just because a bidule might not do justice to my favorite music or to a particular piece of music that's a bit more complex. Let's strive for excellence here and if whatever level we do attain will be higher than if we simply aimed for something that's a bit more convenient. :)


The overall idea of the new bidules ("inner design" as you say) is simply as follows: Reproduce using Software (SW) what some of the Hardware (HW) Repurposing wizzards did & do.
As crazy as this might sound, I believe that SW Repurposing these days can do it, maybe even better. The same applies for Ambisonics, where the increased precision of internal processing of the SW tools just beats the established old HW units.

The idea of HW repurposing "per se" is IMHO not really a huge secret: "All" it does (i.e. what the HW people do) is to to apply (as an example) a little bit of SRS II and use that for the C, apply DPL 2 and use it for L & R, and finally apply a Sansui decode for the Surrounds. Some of the best HW Repurpose works I know of are done that way or similar (maybe even applying small variations to different parts of the album, or even to different parts within the same track).

This is interesting info. Where did you find this?

Finally, a brief comment to your reply to Tantulus/Steve: The methods described here do proper Ambisonics, but the feeding material is not true UHJ. Most importantly, it lacks the 3D information (Z-axis), but who cares, as most of us don't have a 3D speaker setup.

The key thing is that feeding Stereo through an Ambisonics upmix still can produce surprisingly results on "Stereo". In fact, it often does ! For very dry recordings, Ambiophonics is probably a better way to do things (i.e. a Stereo Dipole plus Ambisonics as 'supportive' means 'only'). See also: Farina's comments regarding the concern about the usage of UHJ-to-Surround conversion for not-UHJ stereo recordings (http://forum.doom9.org/showthread.php?s=&postid=419102#post419102)

Actually, I wrote Angelo Farina yesterday on this subject and he was kind enough to respond today. I didn't think to ask him for permission to quote his answer, but basically he confirmed my suspicion that we've been doing is not capable of producing the proper Ambisonic result. I don't think he'd mind my quoting the following from his email to me:
Written by Angelo Farina to UrsaMtl on June 10, 2004
In a hiearchy scale, I would rate the surround recovered from a not UHJ recording, converted to B.format and finally decoded to 5.1 as 1 over 5. The same score for the original Dolby Surround decoder.
He did suggest that taking B-format impulse responses and convoluting the original stereo track with them, combining the results to one file and treating it as a B-format file by decoding it to derive the 5.1 feeds is quite a bit better. He rated it 4/5 as apposed to a true 5.1 recording, which he rated as 5/5.

Anyway, you are right in saying that processing a non-UHJ stereo file as Ambisonics does in fact produce an interesting result. I've got a couple of ideas I want to test out on a version 2 of UpMix Studio and then I'll post it.

Have a good weekend!
Steve (Ursa)

[Shayne]

Originally posted by Eye of Horus
Hi Brock,

I have had these problems too, ....

kind regards,

EoH
I have never seen this problem in 1934 encodings and therefore would wonder if it has something to do with 44.1 hz (i always use 48).

Peace

[alastor]

HI kempf and eoh.

I tested your new bidules, voice center, and my conclusion....

! Great Works, sounds wonderful ¡

Only a comments :

In my computer the modication for besweet d'ont work, may be because my processor ia AMD, i d'ont Know.

Usualy i use the six mono recorders without a problem, i use the old bidule for more of 25 films without incidents.

Best Regards

[ursamtl]

Originally posted by alastor
In my computer the modication for besweet d'ont work, may be because my processor ia AMD, i d'ont Know.


Hi Alastor,

Your computer being an AMD shouldn't make any difference. I've been using a 3-year-old Athlon Thunderbird 1.1 GHz for all this surround processing without any problems at all.

Regards,
Steve

[alastor]

Hi all

if any is interested :

I use these procedure for six mono recorders, six beacuse i added two filters hnm for lfe, power off in bidule, open input. indicates names of all outputs, play in input. Power On.

In my case all the files ever all the same lenght. Remember i only works with film soundtracks, i d'ont know if this procedure is also valid for small files.

Bye

[trooper11]

ok i have a question, i was trying out the SAD5.1 Bidule using the guide that was provided here:

http://www.dtsac3forum.digitalzones.com/SAD51Bidule.htm

i thought i did everything correctly, i finished using bidule and i saw the out put of the one large wav file. i went to try and use besweet and i created the start.bat file, but everytime i try to run it, the screen just pops up quicklt then closes. i belive its saying error, unknown input file format. here is the line i pasted into the bat file:

BeSweet.exe -core( -input 01.wav -output 01- -type wav -6chfloat ) -ssrc( --rate 48000)

what am i doing wrong here? after i get the mono files, im going to be using them to convert to ac3 dd files, if i can just get this step to work. thanks.

[ursamtl]

Originally posted by trooper11
ok i have a question, i was trying out the SAD5.1 Bidule using the guide that was provided here:

http://www.dtsac3forum.digitalzones.com/SAD51Bidule.htm

i thought i did everything correctly, i finished using bidule and i saw the out put of the one large wav file. i went to try and use besweet and i created the start.bat file, but everytime i try to run it, the screen just pops up quicklt then closes. i belive its saying error, unknown input file format. here is the line i pasted into the bat file:

BeSweet.exe -core( -input 01.wav -output 01- -type wav -6chfloat ) -ssrc( --rate 48000)

what am i doing wrong here? after i get the mono files, im going to be using them to convert to ac3 dd files, if i can just get this step to work. thanks.

I ran into the same problem when I tried it. I got very wrapped up in trying different Ambisonic stuff so put the SAD51 on the back burner. I think it might have to do with long filenames in Windows and having to enclose them in quotation marks. As I said I didn't have time to test this theory, but I seem to recall reading something to that effect. I just found another source for ambisonic formulas so I want to try that out a bit more first, but I do intend to get back ot SAD51 for comparison purposes.

Regards,
Steve

[Eye of Horus]

Originally posted by trooper11
ok i have a question, i was trying out the SAD5.1 Bidule using the guide that was provided here:

http://www.dtsac3forum.digitalzones.com/SAD51Bidule.htm

i thought i did everything correctly, i finished using bidule and i saw the out put of the one large wav file. i went to try and use besweet and i created the start.bat file, but everytime i try to run it, the screen just pops up quicklt then closes. i belive its saying error, unknown input file format. here is the line i pasted into the bat file:

BeSweet.exe -core( -input 01.wav -output 01- -type wav -6chfloat ) -ssrc( --rate 48000)

what am i doing wrong here? after i get the mono files, im going to be using them to convert to ac3 dd files, if i can just get this step to work. thanks.


Everything in the line is correct, but....... Do you have the latest Plogue Bidule version ? (0.6601)
If so : goto the edit menu
Chose preferences
Goto the disk I/O tab
Make sure use WAVEFORMATEXTENSIBLE is set to "never"

I bet you have set it on "always" (default setting) or "adaptive".

Good luck and please let me know if this was the solution.

BTW when you like the output of SAD51inBidule, I recommend you to try our latest 3 bidules too !

Kind regards,

EoH

[trooper11]

hmm ok, well i ownder why long filenames would be giving me this proble, my file name is 01.wav. could it be the location of besweet? it is ina folder that has spaces, which i know has caused problems in the past with other things, but this time the error was for the file type. i wish someone would take a look at this.

[ursamtl]

Originally posted by trooper11
hmm ok, well i ownder why long filenames would be giving me this proble, my file name is 01.wav. could it be the location of besweet? it is ina folder that has spaces, which i know has caused problems in the past with other things, but this time the error was for the file type. i wish someone would take a look at this.

Yes, I think the filename problem has to do with the Besweet location, but EoH is probably on track with the setting in Bidule. I know that has caused quite a few problems from what I've read.

Steve.

[kempfand]

Ladies & Gentlemen,

I think the discussion over the previous few posts is going into the wrong direction, as these technical problems have null and nothing to do with the discussion about the Audio Methods discussed in this threat (which reads "Stereo to 5-Channel Surround").

If you have problems with BeSweet & GUI, SSRC, recording & splitting multichannel WAV's, short/long filenames etc. etc. etc.:

1) Use the technical threat(s) on this as agreed earlier: 6ch Wave to dts with BeSweet (thread split) (http://forum.doom9.org/showthread.php?s=&threadid=60907)
2) Always post your logfile :logfile:
3) Use the Search function. 99% of the problems have been described before, and a solution was given.



@ ursamtl:

This is interesting info. Where did you find this? From discussions which took place on the Usenet Newsgroup in middle 2002.



Actually, I wrote Angelo Farina yesterday on this subject and he was kind enough to respond today. I didn't think to ask him for permission to quote his answer, but basically he confirmed my suspicion that we've been doing is not capable of producing the proper Ambisonic result. Wrong interpretation from your side on Angelo Farina's email (based on what you quoted).
The methods described here do propoper Ambisonics.
If we feed the methods with 'Stereo only', that's a point which has nothing to do with the method per se.



Written by Angelo Farina to UrsaMtl on June 10, 2004
In a hiearchy scale, I would rate the surround recovered from a not UHJ recording, converted to B.format and finally decoded to 5.1 as 1 over 5. The same score for the original Dolby Surround decoder.He did suggest that taking B-format impulse responses and convoluting the original stereo track with them, combining the results to one file and treating it as a B-format file by decoding it to derive the 5.1 feeds is quite a bit better. He rated it 4/5 as apposed to a true 5.1 recording, which he rated as 5/5.

:devil: :devil: :devil:

That's exactly what many people here do as of mid 2002, with excellent & proven results: GUIDE: Stereo to 5-Channel Surround (as of 07/2002) (http://forum.doom9.org/showthread.php?s=&threadid=29277).
Note that this exactly does what Farina describes on his public site.

The new guides on Ambisonics GUIDE: Stereo to 5-Channel Surround (as of 08/2003) (http://forum.doom9.org/showthread.php?s=&threadid=60137) just make usage of newer tools (VST's, Bidule or Audiomulch) and expand the topic into Ambiophonics.

Cheers,
Andreas

[trooper11]

ok id love to post a logfile, if it gave one! using besweet, no log file was outputted, how do you get a log file to output?

also, i searched for long file name , to see if there was anyhting in the forum, and no there were no solutions that i found for my problem. i also tried input file type unknown, also no results to help. so if oyu could point me to the proper thread to ask questions, ill move there.

[ursamtl]

Originally posted by kempfand
Ladies & Gentlemen,

I think the discussion over the previous few posts is going into the wrong direction, as these technical problems have null and nothing to do with the discussion about the Audio Methods discussed in this threat (which reads "Stereo to 5-Channel Surround").

Andreas, if people have problems with instructions you give in this thread, then it's ridiculous to tell them they should not post their problems here! After all, the people reading this thread who may have a helpful response for someone with a problem may not read some other thread, and thus the person with the problem misses a chance to get help. That's rather unfair, don't you think? You can't control a discussion. You can perhaps remind people to stay on topic, but if you bring information into a discussion, you cannot control how others will react to that information.

Wrong interpretation from your side on Angelo Farina's email (based on what you quoted). The methods described here do propoper Ambisonics. If we feed the methods with 'Stereo only', that's a point which has nothing to do with the method per se.

Please don't try to tell me how to think or how to interpret someone's response to an email I sent him. I asked him if I was correct in assuming that feeding a non-UHJ signal through the Ambisonic encoding and decoding process is not true Ambisonics. I believe his answer as I quoted it was quite clear. He said it was no better than the original Dolby Surround!

Look, if Ambisonics by definition is the science of manipulating sound information in three steps: 1. recording sounds a certain way, or convolving sounds to give them missing information, 2. encoding the recorded information a certain way, and finally 3. decoding the information to present it a certain way. If one of these steps is missing, it is not truly Ambisonics. What Mr. Farina suggested was to convolve the stereo signals with true B-format impulses. This would in fact create a true B-format file by sort of grafting the soundfield information from the IRs onto the original stereo file. The resulting file would be a substitute for the recorded file, and thus fulfill the requirements for the first step of the process, thereby making it truly Ambisonic. Seems pretty clear to me!

Originally posted by ursamtl
He did suggest that taking B-format impulse responses and convoluting the original stereo track with them, combining the results to one file and treating it as a B-format file by decoding it to derive the 5.1 feeds is quite a bit better. He rated it 4/5 as apposed to a true 5.1 recording, which he rated as 5/5.

Originally posted by kempfand
:devil: :devil: :devil:

[QUOTE]That's exactly what many people here do as of mid 2002, with excellent & proven results: GUIDE: Stereo to 5-Channel Surround (as of 07/2002) (http://forum.doom9.org/showthread.php?s=&threadid=29277).
Note that this [B]exactly does what Farina describes on his public site.

The new guides on Ambisonics GUIDE: Stereo to 5-Channel Surround (as of 08/2003) (http://forum.doom9.org/showthread.php?s=&threadid=60137) just make usage of newer tools (VST's, Bidule or Audiomulch) and expand the topic into Ambiophonics.

Hmmm, who are you calling a devil, me or Angelo Farina? He was the one who gave me the ratings! Now as for the rest of your quote, yes, the original guide uses convolution, but if you read the beginning of this thread, which appears at the top of the Audio Encoding forum, you present it as a newer way of doing Ambisonics through VSTs (in fact finding the old guide takes a bit of reading for someone who comes across this forum for the first time). The Ambiophonics and other methods do not come up until later on.

In fact now that I think about it, you yourself said in response to me in this thread in a message dated May 27, 2004:

3) I agree with your point about "coloration" with Ambisonics, but this can be avoided by doing it right. Note that the most of the Ambisonics methods here (such as the guide at the beginning of this threat) are suited for the masses. Real Ambisonics uses signed filters (real-world impulse repsonses) to creat the B-format, as well as creating the decode to xyz speakers. See Conversion between UHJ and B-format for an outline on howto use signed filters for the B-format.

Sorry to throw your words back at you, but you did say this.

Now why don't you stop telling me what I understand from an answer I received from one of the foremost experts in the field, and stop telling people that they shouldn't complain here about problems they're having with information presented here. Perhaps I'm mistaken, but it almost seems as if you're frustrated because you want people to talk about your new bidules, but they keep talking about other things such as problems with information presented earlier in the thread, or debating the pros and cons of LFE.

I'm sure others are like me and haven't had the time to thoroughly test the new ones. I gave you my preliminary impression the other night, but I'm going to do some more testing on the weekend and I'll have more to say about the new ones as well. Since the title of the thread is GUIDE: Stereo to 5-Channel Surround, any discussion of any method to achieve this result should be fair game, as well as problems encountered by those following the instructions given here. If you don't want this, then perhaps you should ask the Doom9 management to make this a regular thread because it appears as if it's an overall thread for discussing all methods for converting 2-channel to surround.

Have a nice weekend!
Steve

[kempfand]

Originally posted by trooper11
ok id love to post a logfile, if it gave one! using besweet, no log file was outputted, how do you get a log file to output?
trooper11: In Bidule, did you set 'never' for the WAVE Format Extensible (with Preference), and use 16 or 32 bit for the Audio File Writer ?

Andreas

[trooper11]

ok well i have a question for a problem im having using besweet. i have a wav file i made using bidule. its a 32bit wav file and im wanting to split it into 6 mono files. here is the command line i tried using and placing it in a text file called start.bat:


BeSweet.exe -core( -input 01.wav -output 01- -type wav -6chfloat -logfilea "G:\program_setup\Sound_tools\Stereo_to_Surround\BeSweet_1.5b28\BeSweet.log" -ssrc( --rate 48000)

but i get the error 58 saying unknown input file format, pointing to "01.wav". here is the log:


BeSweet v1.5b28 by DSPguru.
--------------------------
Error 58: Error : Unknown Input-File Format : "01.wav".
Quiting...
[00:00:00:000] Conversion Completed !

Logging ends : 06/11/04 , 16:42:49.



would this mean the file i produced in bidule is bad or do i need to change soemthing in the command line?

[trooper11]

ok see there you go, that helped alot, no i had the wave extensible format on adpative, since there was no mention of it in the guide for the SAD5.1 Bidule. ill try it on never and see if that helps. i did use 32bit floating on output though. it may be soemthing that should be added to the SAD guide

also, in the guide it shows using 44100 as the sample rate, but the wav file im inputting is 48000, so shouldnt i set the sample rate at 48000?

[kempfand]

Steve: I think your comments are not appropriate. Asking that people use a search function or scroll a few pages up to see if a problem was reported before is a senesful request.

You might call this unfair, but that's how things are in life. Again read the title of this threat: "GUIDE: Stereo to 5-Channel Surround". Information I bring is mostly is on that topic. I repsect your helping hand to people's technical problems, but again: This should not the threat for it.


Please don't try to tell me how to think or how to interpret someone's response to an email I sent him. If you mention it here, partially quote it, and make your conclusions which you post here, I can comment on it, and I will do if I think it helps to clarify. If you don't like it, don't post it in a forum.


What Mr. Farina suggested was to convolve the stereo signals with true B-format impulses. This would in fact create a true B-format file by sort of grafting the soundfield information from the IRs onto the original stereo file. The resulting file would be a substitute for the recorded file, and thus fulfill the requirements for the first step of the process, thereby making it truly Ambisonic. Seems pretty clear to me! Seems pretty clear to me too. Again: That's exactly what Angelo Farina describes on his public server, and what is described in some of the guides here. So I don't see us (you and me) having a difference here.



Hmmm, who are you calling a devil, me or Angelo Farina? My native language is not English, but I think you turn around words here in a misleading way. Please stop this.

It should be clear that the 'devil' is not calling you or anyone devil (I would write that out if that was my intention, which it isn't). The :devil: is meant to express my non-understanding when you praise a method as "rating 4/5", and people here actually use this method for 2 years now.


Now why don't you stop telling me what I understand from an answer I received from one of the foremost experts in the field, and stop telling people that they shouldn't complain here about problems they're having with information presented here. See my comment above. As this is not your personal web-page but a public forum, I feel free to comment, and I believe I have always done in an appropriate and helpful way. Asking that people take reasonable effort to use the Search function or read if problems were solved before doesn't seem wrong to me (maybe a Moderator can comment ?).


Perhaps I'm mistaken, but it almost seems as if you're frustrated because you want people to talk about your new bidules, but they keep talking about other things such as problems with information presented earlier in the thread, or debating the pros and cons of LFE. Could also be the other way round. You didn't seem to like the comments that many people are happy with a Pentagon decode, and that your modified Octagon2 decode has some compressions in the soundfield.


I gave you my preliminary impression the other night, but I'm going to do some more testing on the weekend and I'll have more to say about the new ones as well. As I said, I am grateful for any feedback, and I really credit you gave quick and honest feedback. It is well understood that it is a fair game, and the only implication with the 3 new bidules was that we felt they do better than the previous on (basically SAD51). I always made it clear (I hope) that I don't see it in competition with the Ambio methods, as the intention is somewhat different (i.e. soundfield overall for Ambi, illusion of discrete speakers for the repurposing methods).

Regards,
Andreas

[trooper11]

ok well your help did fix my problem. the only reason i posted ym quesiton here is becuase this is the thread about the guides that have been posted, so my question was about a problem whne using your guide, so that seems reasonable. and i had used the search function before i posted.

i wanted to ask about other bidules you mentioned, you said oyu had 3 new ones, and that they are updates to SADbidule5.1? or are they just other options? are these other bidules avialable on the ftp or on a site? thanks

[kempfand]

The 3 new bidules I was mentioning are here: 3 new exciting bidules !! (http://forum.doom9.org/showthread.php?s=&postid=508505#post508505)

They are on the FTP, but you can also download directly from www.app.demon.nl/Kempfand-EoH-06-09-2004.rar (http://www.app.demon.nl/Kempfand-EoH-06-09-2004.rar)

I personally see them in comparison to the SAD5.1 method, because they try to create an illusion of discrete speakers, where as the Ambimethods are more about the overall soundfield. Just my reading of course.

Regards,
Andreas

[kempfand]

Add-on:

Originally posted by ursamtl Since the title of the thread is GUIDE: Stereo to 5-Channel Surround, any discussion of any method to achieve this result should be fair game, as well as problems encountered by those following the instructions given here. If you don't want this, then perhaps you should ask the Doom9 management to make this a regular thread because it appears as if it's an overall thread for discussing all methods for converting 2-channel to surround. Again please stop turning around words.

If you want Doom9's management to change things, you do it, not me, because this threat did run very well until a couple of weeks.

Have a nice day.
Andreas