foo_dsp_fsurround, a new surround processor for foobar2000 0.9.x Options

[Rozzo]

Hi Pro-opt

This post is only to ask your opinion about a couple of things I have been trying these days regarding Dolby surround.

First one is AC3, I'm using Eright Software Super Encoder/decoder to transform my music files into AC3, getting that way Dolby Digital directly from spdif soundcard out. Super uses a ffmpeg library to process the music, there are some mentions of this Dolby encoding system in hydrogenaudio forums, they say it's not such a great thing, but to my ears this Dolby digital gives better sound than Dolby prologic. This Dolby Digital Spdif music bypasses completely all the controls in my X-Fi extreme music, I can change the volume only with the receiver.

Second one is that redocnecxK program that supposedly encodes al your sounds into AC3 and sends them directly to the receiver through spdif. I says supposedly because it doesn't work with X-Fi sound cards.

So my questions at the end are about the difference between hearing AC3 encoded music and music filtered through your Dolby plugin.

Thanks anticipated,
Rozzo

[tebasuna51]

ATSuround Encode2 works, in volume, like coeficients 1/0.7071/0.8165/0.5774, but the rear channels seems phase shifted then can't be recovered exactly by Free Surround (good channel separation) or PowerDVD (and worse channel separation).

With ATSurround Encode2 I don't know how disable LFE (don't be present in dpl II), and seems be amplified because there are big overflows.

If ATSuround Encode2 put one option to disable the rear phase shift and other to disable the LFE in downmix become usable for me. I have a old audio equipment attached to the PC with only dpl decoder then to listen ac3 5.1 need a downmix to dpl and need ffdshow/ac3filter, can't with Foobar. I don't see any plugin with matrix style, like ffdshow/ac3filter, to make the downmix.

Thanks.

This post has been edited by tebasuna51: Feb 16 2007, 03:31

[pro_optimizer]

Hi Pro-opt

This post is only to ask your opinion about a couple of things I have been trying these days regarding Dolby surround.

First one is AC3, I'm using Eright Software Super Encoder/decoder to transform my music files into AC3, getting that way Dolby Digital directly from spdif soundcard out. Super uses a ffmpeg library to process the music, there are some mentions of this Dolby encoding system in hydrogenaudio forums, they say it's not such a great thing, but to my ears this Dolby digital gives better sound than Dolby prologic. This Dolby Digital Spdif music bypasses completely all the controls in my X-Fi extreme music, I can change the volume only with the receiver.

Second one is that redocnecxK program that supposedly encodes al your sounds into AC3 and sends them directly to the receiver through spdif. I says supposedly because it doesn't work with X-Fi sound cards.

So my questions at the end are about the difference between hearing AC3 encoded music and music filtered through your Dolby plugin.

Thanks anticipated,
Rozzo

Hi Rozzo,

since I have never used ffmpeg/redocnecxK I don't really know how good it is and what it does.
But if it upmixes music to ac3 then it must obviously generate the surround channels from somewhere.
And there are basically 4 options:
1: from the front channels by mirroring, delay, echo (e.g. concert hall effect)
you can get this from you sound card, too (by enabling CMSS 3D or some EAX effects)

2: by passive matrix decoding (this is what dolby surround did)
then you can get a similar or better effect with foo_channelmixer.

3: by active decoding like pl1 does
if this is the case it should not sound better than FS

4: by active decoding like pl2 does
in this case it should sound similar to FS (depending on how good/bad the implementation is)

The fact that it is transferred via AC3 should not make a noticeable difference (except if you have really really bad cables/connectors).

Maybe you can post a few links to the programs?

ATSuround Encode2 works, in volume, like coeficients 1/0.7071/0.8165/0.5774, but the rear channels seems phase shifted then can't be recovered exactly by Free Surround (good channel separation) or PowerDVD (and worse channel separation).

With ATSurround Encode2 I don't know how disable LFE (don't be present in dpl II), and seems be amplified because there are big overflows.

If ATSuround Encode2 put one option to disable the rear phase shift and other to disable the LFE in downmix become usable for me. I have a old audio equipment attached to the PC with only dpl decoder then to listen ac3 5.1 need a downmix to dpl and need ffdshow/ac3filter, can't with Foobar. I don't see any plugin with matrix style, like ffdshow/ac3filter, to make the downmix.

Thanks.

Yeah, I got the same results (just a few hours ago, should have checked the board before I started
And your're right with the phase shifting.
Unfortunately it's hilbert filter cuts a lot of the lower and higher frequencies (the passband is approx. 110 Hz to 3.5 KHz).

Do you really need only an AC3 downmixer plugin (without phase shift)?
In this case I could write one... should be a matter of minutes.
Actually I need a replacement (with optional [and perfect] shift) for Encode2, too.

[Acropolis]

I like this component, and been using it for a week, as I think the 5.1 it produces gives me more feeling of "real" (do you call it the dynamic range?) comparing with ATSurround Processor (was using this before your component came out).

I have a little request, can you post a version number in your first post when you update it? because it's quite annoying that I have to find the word "upload" in your post to check if it's updated.

thank you

[tebasuna51]

Do you really need only an AC3 downmixer plugin (without phase shift)?
In this case I could write one... should be a matter of minutes.

Thanks.
Can be used, not only for my strange case (play 5.1 with only dpl surround), but also to convert audio movie tracks from 5.1 to mp3 (or other stereo formats) with dpl II info.

This post has been edited by tebasuna51: Feb 17 2007, 17:24

[pro_optimizer]

I like this component, and been using it for a week, as I think the 5.1 it produces gives me more feeling of "real" (do you call it the dynamic range?) comparing with ATSurround Processor (was using this before your component came out).

I have a little request, can you post a version number in your first post when you update it? because it's quite annoying that I have to find the word "upload" in your post to check if it's updated.

thank you

Ok, from now on.

Do you really need only an AC3 downmixer plugin (without phase shift)?
In this case I could write one... should be a matter of minutes.

Thanks.
Can be used, not only for my strange case (play 5.1 with only dpl surround), but also to convert audio movie tracks from 5.1 to mp3 (or other stereo formats) with dpl II info.

Ok, I uploaded a plugin for this purpose. It uses you mixing matrix.
I also experimented with hilbert filtering and got much better results than with Encode2, but its terribly slow (barely real time) so I have to switch over to fft convolution until I can release that.
Unfortunately this is necessary if you want to downmix DTS tracks (they seem not to have shifted rear channels as opposed to AC3).

[Rozzo]

Maybe you can post a few links to the programs?

Hi Pro-opt,

You can download Super at:

http://www.erightsoft.net/SUPER.htm


"SUPER © Simplified Universal Player Encoder & Renderer.
A GUI to ffmpeg, MEncoder, mplayer, x264, mppenc,
ffmpeg2theora & the theora/vorbis RealProducer plugIn.

If you need a simple, yet very efficient tool to convert (encode) or play any Multimedia file,
without reading manuals or spending long hours training, then SUPER © is all you need.
It is a Multimedia Encoder and a Multimedia Player, easy-to-use with 1 simple click."

Encoding my files into AC3 with this program (6 channels at 394 bps) and sending them to my DENON receiver as DOlby bitstream bypassing all controls in my PC gives impressive acustic results.

YS,
Rozzo

[GeSomeone]

- changed the gain to ~85% to avoid clipping in practically all cases.

I have a feature request could you make that optional?
I don't like it when the sound goes down 1.5dB when I activate fSurround. I use advanced limiter to catch the eventual clipping.

[tebasuna51]

Ok, I uploaded a plugin for this purpose. It uses you mixing matrix.
I also experimented with hilbert filtering and got much better results than with Encode2, but its terribly slow (barely real time) so I have to switch over to fft convolution until I can release that.
Unfortunately this is necessary if you want to downmix DTS tracks (they seem not to have shifted rear channels as opposed to AC3).

Thanks for the news.

Now with a wav6 -> foo_dsp_downmix (Downmix AC3/DTS) -> Free Surround (*) -> New_wav6

(*) With this adjust:
Center Image (1.0)
Dimension (0.0)
Invert rear phase (+0, +180)
Mixing coef A: 0.866
Mixing coef B: 0.5

we obtain 100% identical channels with this considerations:

1) The LFE channel can't be recovered by dpl II, if is included in downmix it appears at Center output channel. The downmix plugin need a selector to disable the LFE channel.

2) The volume is 100 % identical with errors (and crosstalk) less than 0.02 % over the channel test sample. Real signals can produce overflows at downmix and maybe is convenient a slider between the full coeficients values 1/0.7071/0.866/0.5 an the normalized 0.3254/0.2301/0.2818/0.1627.

3) In downmix there are a delay of 12.542 ms for L, R, and C channels. With SL and SR there are 25.083 ms delay. This delay between front and rear channels (25.083 - 12.542 = 12.541) can be a problem with real signals.

4) In upmix (free surround) there are also a delay of 128.0 ms uniform for all channels. The new wav is 53 ms more than original, then the end is cut in 75 ms.

This is only to show my interest in your job, but if you aren't interested in further develop I can understand. For me is enough the actual downmix to play 5.1; to convert 5.1 to dpl2 there are others methods (AviSynth, BeSweet) with 2 pass mode, really I don't know how implement this with Foobar.

The most important is the upmix (Free Surround) because is the better free tool I know to do this. And with the Advanced Controls is perfect to experiment different settings. Only a last petition please, is possible four options for rear phase?:
Keep rear phase (0, 0)
Invert rear Right (0, 180) like PowerDVD movie mode
Invert rear Left (180, 0) Azid-BeSweet style
Invert two rear (180,180) I don't now ... but to complete.

Thanks.

[CZ812CE]

Hi,

I'm very intersted in this component.

BTW, Can I decode my SQ quadraphonic records by using this?
I'd like to convert it into ac3 or dts.

Thanks.

[pro_optimizer]

Hi Pro-opt,

You can download Super at:

http://www.erightsoft.net/SUPER.htm

Thanks, downloaded it. I'll take a look at it when I am done with the other pending stuff.

I have a feature request could you make that optional?
I don't like it when the sound goes down 1.5dB when I activate fSurround. I use advanced limiter to catch the eventual clipping.

Yes, I already changed the gain back to 100% (in 0.3.3) because the channels which produced the clipping were too loud, anyway. There is still some clipping sometimes (the highest which I saw was 105%) but I'll look into this and fix it (hopefully).

Thanks for the news.

Now with a wav6 -> foo_dsp_downmix (Downmix AC3/DTS) -> Free Surround (*) -> New_wav6

(*) With this adjust:
Center Image (1.0)
Dimension (0.0)
Invert rear phase (+0, +180)
Mixing coef A: 0.866
Mixing coef B: 0.5

we obtain 100% identical channels with this considerations:

1) The LFE channel can't be recovered by dpl II, if is included in downmix it appears at Center output channel. The downmix plugin need a selector to disable the LFE channel.

2) The volume is 100 % identical with errors (and crosstalk) less than 0.02 % over the channel test sample. Real signals can produce overflows at downmix and maybe is convenient a slider between the full coeficients values 1/0.7071/0.866/0.5 an the normalized 0.3254/0.2301/0.2818/0.1627.

3) In downmix there are a delay of 12.542 ms for L, R, and C channels. With SL and SR there are 25.083 ms delay. This delay between front and rear channels (25.083 - 12.542 = 12.541) can be a problem with real signals.

4) In upmix (free surround) there are also a delay of 128.0 ms uniform for all channels. The new wav is 53 ms more than original, then the end is cut in 75 ms.

This is only to show my interest in your job, but if you aren't interested in further develop I can understand. For me is enough the actual downmix to play 5.1; to convert 5.1 to dpl2 there are others methods (AviSynth, BeSweet) with 2 pass mode, really I don't know how implement this with Foobar.

The most important is the upmix (Free Surround) because is the better free tool I know to do this. And with the Advanced Controls is perfect to experiment different settings. Only a last petition please, is possible four options for rear phase?:
Keep rear phase (0, 0)
Invert rear Right (0, 180) like PowerDVD movie mode
Invert rear Left (180, 0) Azid-BeSweet style
Invert two rear (180,180) I don't now ... but to complete.

Thanks.

1) Hmmm, I think the downmix will sound pretty dull with LFE disabled, don't you?
If I got you right, you decode the downmix with an external dpl1 decoder which can't do bass management.
Have you tried to disable the center speaker (I think this is called phantom surround in dpl1 decoders)?
OTOH, if you decoded the downmix on the PC, you could use the bass management of your soundcard or I could implement similar functionality in FS itself (I'd be interested if there is anyone who needs bass mangement right in FS).

2) Yeah, maybe this is necessary. Another option for now would be to put an Equalizer at -6db before the downmixer.

3) Sorry, that delay was a mistake (it's a holdover of my earlier phase shifting experiments).
I'll fix that in the next version.

4) Hmmmmm... This not easy to solve. Unfortunately there is a tradeoff between delay and fidelity.
The smaller you make the delay, the fewer separate frequency bands FS can use for steering. And this results in more crosstalk, steering glitches etc.
Currently I use 2048 bands. So one needs 4096 samples and because it's overlapped, it sums up to 6144 samples. At 48Khz this is 128ms delay. For music it's no big problem of course. But if anyone wants to use FS as decoder for a movie player he needs to delay the video stream, too.
Maybe one should add a slider for the window length in the expert controls (with a clear warning).
Btw: The first three phase modes you proposed are ok, I'll support all of them.

This is only to show my interest in your job, but if you aren't interested in further develop I can understand. For me is enough the actual downmix to play 5.1; to convert 5.1 to dpl2 there are others methods (AviSynth, BeSweet) with 2 pass mode, really I don't know how implement this with Foobar.

Of course I am interested in writing a cool downmixer. Having a really good "canonical" Downmix/Upmix chain available could make mp3surround or similar formats nearly unnecessary.

Hi,

I'm very intersted in this component.

BTW, Can I decode my SQ quadraphonic records by using this?
I'd like to convert it into ac3 or dts.

Thanks.

Yes, it should decode SQ quadrophonic properly but unfortunately you'll not get stereo surrounds.
Actually, I have no clue how a decoder can distinguish between rear left and rear right (if the wikipedia matrix is correct) because the difference between them is only that both channels are inverted.



My roadmap at the moment looks like this:
First I will revise the steering of FS to make it practically linear over all positions. When this is done, one can downmix a 5.1 file and just get it back when played through FS, as long as the basic assumption of non-overlapping sources in the frequency space is satisfied.

Then I will add the mentioned controls.

Then, I'll release a proper downmixing plugin with all the necessary config options (including phase shift which already works perfectly, apart from a 22ms delay).

[tebasuna51]

1) Hmmm, I think the downmix will sound pretty dull with LFE disabled, don't you?

Is not a problem with my audio equipment, is a Dolby recommendation. From "214_Mixing with Dolby Pro Logic II Technology.pdf":

"There are other concerns when adding an LFE signal to the mix. If the LFE is simply redistributed within the other channels of the mix, they will usually be subject to some low-frequency bandpass filtering. This filtering causes phase shifts of the LFE signal. When they are acoustically added within a room, these phase shifts are fairly subtle and often go unnoticed. However, when they are electronically added together with the five main channels in the encoder, they may produce less than desirable results at certain frequencies. For this reason, it is recommended that the LFE signal not be used in a Dolby Pro Logic II downmix unless it contains unique information that is not repeated in any of the five main channels."

At least we can maintain optional this issue.

4) Hmmmmm... This not easy to solve. Unfortunately there is a tradeoff between delay and fidelity.
The smaller you make the delay, the fewer separate frequency bands FS can use for steering. And this results in more crosstalk, steering glitches etc.

Ok, can be assumed with a warning about the 128 ms delay if needed for video sync.

[pro_optimizer]

Ok, here is the promised new version (0.3.4).

New features:
- A new steering mode (called linear) has been added, which should reconstruct the sources at exactly the same location where they were before the encoding. The previous versions used a few very simple heuristics for this inverse mapping which worked quite well (probably better than Dolby's ) but they were still far from the perfect solution.
The custom mixing coefficients are not yet supported in this mode (because therefore I would have to generate the inverse functions on the fly, which is nontrivial... but I think I'll get this done over the next few days).

- Controls for front and rear stereo separation have been added.
This should be a goodie for the headphone listeners.

- 4 phase shifting modes are supported now: Music Mode (0°/0°), PowerDVD compatibility (0°,+180°), BeSweet compatibility (+180°,0°) and Exact reconstruction (-90°,+90°). It's not unlikely that you don't hear a difference at all but the desire to do The Right Thing was just too strong.

Fixes:
- The dimension slider was accidentally clamped to [0,1] in the decoder core so the negative range wasn't really accessible.


Filefront's database is corrupted right now, so please download it via this link.

This post has been edited by pro_optimizer: Feb 21 2007, 09:43

[tebasuna51]

Using to downmix:
Lt = FL + 0.7071*FC + 0.8165*SL - 0.5774*SR
Rt = FR + 0.7071*FC - 0.5774*SL + 0.8165*SR

And your FS upmix with defaults and:
- Center Image (1.0)
- Simple and Not-Linear (Legacy)
I obtain output channels identical in phase and amplitude than input channels, with very little crosstalk:

CODE

       FL      FR      C       SL      SR
     ------  ------  ------  ------  ------
FL'  1.0000    -       -       -       -
FR'    -     1.0000    -       -       -
C'     -       -     1.0000    -       -
SL'    -       -       -     1.0000  0.0001
SR'    -       -       -     0.0001  1.0000

But changing to:
- Linear (Near Perfect)
The phase is ok but volume and crosstalk:

CODE

       FL      FR      C       SL      SR
     ------  ------  ------  ------  ------
FL'  0.9443    -       -     0.0836    -
FR'    -     0.9443    -       -     0.0836
C'   0.0230  0.0261  0.9995    -       -
SL'  0.0004    -     0.0003  0.9329  0.0063
SR'    -     0.0004  0.0003  0.0063  0.9329

This post has been edited by tebasuna51: Feb 21 2007, 13:42

[pro_optimizer]

Using to downmix:
Lt = FL + 0.7071*FC + 0.8165*SL - 0.5774*SR
Rt = FR + 0.7071*FC - 0.5774*SL + 0.8165*SR

And your FS upmix with defaults and:
- Center Image (1.0)
- Simple and Not-Linear (Legacy)
I obtain output channels identical in phase and amplitude than input channels, with very little crosstalk:

CODE

       FL      FR      C       SL      SR
     ------  ------  ------  ------  ------
FL'  1.0000    -       -       -       -
FR'    -     1.0000    -       -       -
C'     -       -     1.0000    -       -
SL'    -       -       -     1.0000  0.0001
SR'    -       -       -     0.0001  1.0000

But changing to:
- Linear (Near Perfect)
The phase is ok but volume and crosstalk:

CODE

       FL      FR      C       SL      SR
     ------  ------  ------  ------  ------
FL'  0.9443    -       -     0.0836    -
FR'    -     0.9443    -       -     0.0836
C'   0.0230  0.0261  0.9995    -       -
SL'  0.0004    -     0.0003  0.9329  0.0063
SR'    -     0.0004  0.0003  0.0063  0.9329

Yes, the crosstalk in the corner cases could be lower.
I think the worst part is that the surrounds leak a bit into the front channels - probably I can further improve this.

The reason is this:
The simple version is built to have optimal behaviour at the 4 cardinal points (LF,RF,LS,RS) and to behave reasonably well in between.
The linear version is built to have approximately optimal behaviour over all positions, including the cardinal points. So the overall behaviour is much better, but your test results are worse because they test only 4 points which are treated like any other point by the linear steering.
If you tested every point in the soundfield you should see that the crosstalk (i.e. departure from the desired behaviour) is actually much lower.

Unfortunately my maths program cannot not just invert the encoding function but only approximate its inverse. What you see is the approximation error (which is largest at the corner points).

[poisas]

I wonder whats happening ?

WARNING : foo_dsp_fsurround: clipping encountered (-1.124)
WARNING : foo_dsp_fsurround: clipping encountered (-1.111)
WARNING : foo_dsp_fsurround: clipping encountered (1.071)
WARNING : foo_dsp_fsurround: clipping encountered (-1.090)
WARNING : foo_dsp_fsurround: clipping encountered (-1.071)
WARNING : foo_dsp_fsurround: clipping encountered (-1.029)
WARNING : foo_dsp_fsurround: clipping encountered (-1.043)

If i take a look at foobars console i can see theese .

my spec Athlon 1.7ghz, winxp PR/ 766mb ddr/creative audigySE/ got stereo amp for front speakers, and other set from pc speakers for rear surround/ sound card is set for 4.0 speakers/ realy feals when you set from 5.1 that sub and center speaker goes to front speakers
what actualy goes frong when clipping ? because i cannot hear any strange things on my speakers
dsp's ar set : Vlevel> noisesharpening>freesurround

[Chungalin]

Glad to see all the quick improvements in this DSP.
I know my PC is not very up to date in CPU power but, have you checked the CPU usage of FreeSurround? Here it sucks 25% at a constant rate. It's far, far more than the power consumed by the math-intensive Dolby Headphone engine.

[pro_optimizer]

I wonder whats happening ?

WARNING : foo_dsp_fsurround: clipping encountered (-1.124)
WARNING : foo_dsp_fsurround: clipping encountered (-1.111)
WARNING : foo_dsp_fsurround: clipping encountered (1.071)
WARNING : foo_dsp_fsurround: clipping encountered (-1.090)
WARNING : foo_dsp_fsurround: clipping encountered (-1.071)
WARNING : foo_dsp_fsurround: clipping encountered (-1.029)
WARNING : foo_dsp_fsurround: clipping encountered (-1.043)

If i take a look at foobars console i can see theese .

my spec Athlon 1.7ghz, winxp PR/ 766mb ddr/creative audigySE/ got stereo amp for front speakers, and other set from pc speakers for rear surround/ sound card is set for 4.0 speakers/ realy feals when you set from 5.1 that sub and center speaker goes to front speakers
what actualy goes frong when clipping ? because i cannot hear any strange things on my speakers
dsp's ar set : Vlevel> noisesharpening>freesurround

It just displays this message when it produces data that is so loud that it may clip.
Probably it's unnoticable as long as you have the foobar volume a bit below 0db.
You can also put a limiter behind it. And if the message annoys you, you can put an equalizer (@-2db or so) before it.
Unfortunately I have no time right now to change anything because I have to work on a presentation.

Glad to see all the quick improvements in this DSP.
I know my PC is not very up to date in CPU power but, have you checked the CPU usage of FreeSurround? Here it sucks 25% at a constant rate. It's far, far more than the power consumed by the math-intensive Dolby Headphone engine.

Wow! For me it's at 0% and goes up to 4% in regular intervals.
Luckily the code hasn't been optimized yet, so it's likely that it can be sped up.

Is the CPU usage similar for linear and non-linear modes?
If yes, then the FFT is probably the bottleneck.
Which would be bad because it's already the fastest implementation in the world.

[GeSomeone]

Yes, the crosstalk in the corner cases could be lower.
I think the worst part is that the surrounds leak a bit into the front channels - probably I can further improve this.

From the info it looks like almost a non issue, at least with music. You hear the sound from the side that is louder, and there is the legacy option for less crosstalk.

just my 0.02

[poisas]

thats what i thougt what is recomended setings to get most surrounded sound from mp3's ?

[pro_optimizer]

thats what i thougt what is recomended setings to get most surrounded sound from mp3's ?

The default setting is quite good for most music.
If there is surround information, you will hear it - otherwise the music will be mostly like normal stereo.
You can also move the whole soundfield further behind by using the dimension slider, if you want generally more action in the surrounds.
But keep in mind that you get the largest possible front-back range when dimension is 0, it's just the question whether the music actually uses it.
(with dimension=0 you can get everything between "fully infront" and "fully behind" whereas with dimension=1 you have reduced it to everything between "somewhat infront" and "fully behind").
I think one could add a slider which shifts extremely left or right sounds into the rear speakers (I think this is called panorama mode in PL2).

The phase settings don't matter that much for normal use (because they are barely audible), but you can enable
"exact reconstruction" when you listen to music for which you know that it's surround-encoded.

[Rswave2k2]

I had a question about your plugin. When I try to add a center image it seems alot of the sounds leave the front left and right channel. Is there anyway to get the center speaker on without losing sound from the left and right speaker. How does the center image setting work? Right now I have the center image off. Also I love how your plugin works. How it sounds like normal when theres no surround. It works perfect for this dsp I use. I use the winamp dsp bridge plugin and use the srswow winamp plugin and then your plugin and it sounds great.

If you heard of SRS WoW here is a link:
srslabs.com and click on technology.

[pro_optimizer]

I had a question about your plugin. When I try to add a center image it seems alot of the sounds leave the front left and right channel. Is there anyway to get the center speaker on without losing sound from the left and right speaker. How does the center image setting work? Right now I have the center image off. Also I love how your plugin works. How it sounds like normal when theres no surround. It works perfect for this dsp I use. I use the winamp dsp bridge plugin and use the srswow winamp plugin and then your plugin and it sounds great.

If you heard of SRS WoW here is a link:
srslabs.com and click on technology.

Please excuse the delay (I was very busy),
When you want to pan sounds between left front and right front, you can theoretically use a wall of 20 speakers next to each other: Whenever a sound source is close to the position of a speaker X, that speaker is responsible for playing it back. When it is somewhere between 2 of those 20 speakers, both will play it, usually with different volumes.
You can approximate that effect with just 2 speakers but especially the center sources will sound more "fake", because your brain knows the difference between real center and left+right front. With center image you can morph between a 2 speakers and a 3 speakers wall.
Since music is optimized to sound good in the 2 front speakers setting, it may sound worse with 3 speakers. Apart from that, I would not use it if your center speaker is weaker than the left/right speakers.

I've heard of SRS WoW but I didn't try it yet (it seems to be tailored towards TV sets rather than surround systems).
I'd be careful with SRS 3d, since it might destroy the surround information (have you tried channeltest.mp3?).

[Rswave2k2]

Please excuse the delay (I was very busy),
When you want to pan sounds between left front and right front, you can theoretically use a wall of 20 speakers next to each other: Whenever a sound source is close to the position of a speaker X, that speaker is responsible for playing it back. When it is somewhere between 2 of those 20 speakers, both will play it, usually with different volumes.
You can approximate that effect with just 2 speakers but especially the center sources will sound more "fake", because your brain knows the difference between real center and left+right front. With center image you can morph between a 2 speakers and a 3 speakers wall.
Since music is optimized to sound good in the 2 front speakers setting, it may sound worse with 3 speakers. Apart from that, I would not use it if your center speaker is weaker than the left/right speakers.

I've heard of SRS WoW but I didn't try it yet (it seems to be tailored towards TV sets rather than surround systems).
I'd be careful with SRS 3d, since it might destroy the surround information (have you tried channeltest.mp3?).

Thanks for the reply. I tried turning center image to 1.0 and it my foobar use 50% of my cpu. What's a good recommendation setting for center image? So with center it plays more of the "center" sounds of the sound field. And since theres a center one to play between 2 and 20 more if the sounds of like speakers 9-14 get played on the center only. Also for the center image whats the difference between 0.2 or say 0.7 is it volume or what? What setting would be good lets say with the 20 speaker example have left speaker play 1-7 center play 8-13 and right play 14-20. Does having the cetner image at 1.0 make center play 2-19?

This post has been edited by Rswave2k2: Mar 27 2007, 18:33

[morphguy12]

Failed to load DLL: foo_dsp_fsurround.dll
Reason: This component is missing a required dependency, or was made for different version of foobar2000.

any reason as to why this would happen?