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G2 RACE encoder
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mosc
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PostPosted: Wed Apr 16, 2008 4:37 pm    Post subject: G2 RACE encoder Reply with quote  Mark this post and the followings unread

After a lot of experimenting, I'm posting a pretty deluxe RACE ambiophonic encoder realized on the G2.

See: http://electro-music.com/forum/topic-25669.html

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PostPosted: Wed Apr 16, 2008 4:48 pm    Post subject: Re: G2 RACE encoder Reply with quote  Mark this post and the followings unread

mosc wrote:
After a lot of experimenting, I'm posting a pretty deluxe RACE ambiophonic encoder realized on the G2.

See: http://electro-music.com/forum/topic-25669.html


Ain't got no G2 so I can't tell what's delux about it. I wana know so I can steal it for mine Cool

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PostPosted: Wed Apr 16, 2008 6:13 pm    Post subject: Reply with quote  Mark this post and the followings unread

The deluxe part is you can set the delay in 10 microsec increments from 40 to 110 usec. You can adjust with knobs the filter settings, the feedback, ambio attenuator, and I/O pads. Makes experimenting pretty easy. Not all that is essential, but I get obsessive sometimes, like everyone else around here...


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PostPosted: Wed Apr 16, 2008 6:43 pm    Post subject: Reply with quote  Mark this post and the followings unread

Well mine looks rather simple by comparison. Guess I'm going to have to add some knobbies Laughing


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PostPosted: Wed Apr 16, 2008 6:58 pm    Post subject: Reply with quote  Mark this post and the followings unread

I find that delay line attenuator a very important thing to be able to adjust. The more of the delay the wider the stage but there are some timbre distortion. That knob might be better labeled width. With wide stages you can tweak the feedback knob and it can reduce the timbre distortion.

Adjusting the filters changes things too, but 300 - 5000 with 6 Db/8va seems to make things sound best. Apparently, those are the freqs where ITD works best. If one has full range speakers, it doesn't make sense to feed them out of phase signals. Robin has really thought this out, but I'm sure he'd admit there are many things to discover.

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PostPosted: Wed Apr 16, 2008 7:25 pm    Post subject: Reply with quote  Mark this post and the followings unread

The filter's in my implementation are 12dB/8ve so that gives me an excuse to a button Wink I too have found that one needs to play around to find the settings that minimize "artifacts." To my ear the Carver C9 is virtually free of such, at least compared to my Ambi encoder. Not sure what to make of that.
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PostPosted: Thu Apr 17, 2008 9:32 am    Post subject: Reply with quote  Mark this post and the followings unread

I'd like to see an impulse response of the C9 unit. That might be very informative.
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PostPosted: Thu Apr 17, 2008 10:06 am    Post subject: Reply with quote  Mark this post and the followings unread

mosc wrote:
I'd like to see an impulse response of the C9 unit. That might be very informative.


Let's see what we can do on that.

I'd like to play around with some HRTFs in RACE's delay path but I aint got no stinkin HRTFs, but I'll keep looking.

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PostPosted: Thu Apr 17, 2008 5:06 pm    Post subject: Reply with quote  Mark this post and the followings unread

This is a formula I've found for center channel (common signal) cancelation.
L = 1/2(L+R) + 1/2(L-R)
R = 1/2(L+R) + 1/2(R-L)

When I added it to my encoder it seems to do nothing. Would you mind trying it in yours and see what you get?

Edit: added link to source

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PostPosted: Thu Apr 17, 2008 5:18 pm    Post subject: Reply with quote  Mark this post and the followings unread

bachus wrote:
This is a formula I've found


Wasn't that one used for vocal removal from pop songs? ... like in the 80ties .. pre karaoke sort of thing.

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PostPosted: Thu Apr 17, 2008 5:27 pm    Post subject: Reply with quote  Mark this post and the followings unread

Blue Hell wrote:
bachus wrote:
This is a formula I've found


Wasn't that one used for vocal removal from pop songs? ... like in the 80ties .. pre karaoke sort of thing.


I don't know but that would make sense.
Howard had speculated that removing "center chanel" info might reduce the timbre distortion. That's why I was playing with it.

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PostPosted: Sun Apr 20, 2008 7:30 pm    Post subject: Reply with quote  Mark this post and the followings unread

bachus wrote:
This is a formula I've found for center channel (common signal) cancelation.
L = 1/2(L+R) + 1/2(L-R)
R = 1/2(L+R) + 1/2(R-L)
That's not quite right. To remove center channel information you take the difference between L and R signals. That removes what is common between them, i.e. the center channel and leaves you with a monophonic signal.
D = L - R
where D is the difference signal.

The equations you posted are nearly the ones for encoding a stereo signal for FM Stereo broadcast. The Sum is the mono signal broadcast on the main carrier for compatibility with mono receivers. The Difference (same as center channel removal!) is for broadcast on the subcarrier.
S = L + R
D = L - R
Run these through the same matrix to decode them.
S + D = (L + R) + (L - R) = 2L
D - D = (L + R) - (L - R) = 2R

Cheers,

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PostPosted: Sun Apr 20, 2008 7:40 pm    Post subject: Reply with quote  Mark this post and the followings unread

bachus wrote:
Well mine looks rather simple by comparison. Guess I'm going to have to add some knobbies Laughing
Yours doesn't look like Howard's because Howard's is the actual patch. Yours is simply the GUI in front of some code. Is yours a VST plugin? Will you share it?

Cheers,

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PostPosted: Sun Apr 20, 2008 7:52 pm    Post subject: Reply with quote  Mark this post and the followings unread

djfoxyfox wrote:
bachus wrote:
This is a formula I've found for center channel (common signal) cancelation.
L = 1/2(L+R) + 1/2(L-R)
R = 1/2(L+R) + 1/2(R-L)
That's not quite right. To remove center channel information you take the difference between L and R signals. That removes what is common between them, i.e. the center channel and leaves you with a monophonic signal.
D = L - R
where D is the difference signal.


That is correct but if we want a left and right signal that has reduced "center channel" info, that is not very useful for then we'd have
L-D = L - (L - R) = L - L + R = R so we just end up going in circles.

If we do the same for SRS's algorithm it's not any better though for we get
L = 1/2(L+R) + 1/2(L-R) =>
L = 1/2(L + R + L - R) =>
L = 1/2(2L+ R-R) = L
So we haven't gotten anywhere either. Thus I conclude that stereo signals transcend the common algebra of the reals and await an explanation that I assume will be based on vectors in the complex plane. Can you shed any light on this?

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PostPosted: Sun Apr 20, 2008 7:59 pm    Post subject: Reply with quote  Mark this post and the followings unread

djfoxyfox wrote:
bachus wrote:
Well mine looks rather simple by comparison. Guess I'm going to have to add some knobbies Laughing
Yours doesn't look like Howard's because Howard's is the actual patch. Yours is simply the GUI in front of some code. Is yours a VST plugin? Will you share it?

Cheers,


Mine is done in Reaktor which can be run as a VST. Unfortunately one cannot export a standalone VST from it. one has to own Reaktor Sad

My patch is available here
http://electro-music.com/forum/topic-25269.html

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PostPosted: Mon Apr 21, 2008 5:25 am    Post subject: Reply with quote  Mark this post and the followings unread

bachus wrote:
Mine is done in Reaktor which can be run as a VST. Unfortunately one cannot export a standalone VST from it. one has to own Reaktor.
Thank you for sharing, although I don't have Reaktor. Too bad Reaktor doesn't have a feature to turn code into a real standalone VST. One would think that would highly increase the value of having, using, and buying Reaktor. Does anyone reading this particular thread have MAX/MSP?

Cheers,

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PostPosted: Mon Apr 21, 2008 5:57 am    Post subject: Reply with quote  Mark this post and the followings unread

djfoxyfox wrote:
Too bad Reaktor doesn't have a feature to turn code into a real standalone VST. One would think that would highly increase the value of having, using, and buying Reaktor.


It would, but it's clear they have made the calculation that it would not benefit their bottom line. Apparently a significant number of people buy Reaktor to gain access to the large library of pre-built/user-built modules. Same old story -- sigh.

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PostPosted: Tue Apr 22, 2008 12:19 pm    Post subject: Reply with quote  Mark this post and the followings unread

I was interested in the center channel extraction because I've noticed, and Bill last night independently noticed, that the center vocals sounded a bit down in the mix compared to stereo. It sounds a tiny bit like a karaoke quizmo. If there was a center channel extraction, then maybe mixing that into the final mix stage might counter that effect and also miraculously correct some timbre distortion that occurs.

So, no need for center channel cancellation, but center channel enhancement.

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PostPosted: Tue Apr 22, 2008 1:16 pm    Post subject: Reply with quote  Mark this post and the followings unread

mosc wrote:
So, no need for center channel cancellation, but center channel enhancement.
Sum L + R and band limit it to the voice range (or not since bass guitar, kick and snare, and solos usually live in the center of mixes) and do one of two things:
1) Mix the summed signal to both speakers.
or
2) Send the summed signal to its own amp and speakera dn place the speaker in the center, between the two Ambiophoniclly placed speakers.
Option 1 seems like the more cost effective method. Just program a new variation on your G2 patch.

Cheers,

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PostPosted: Tue Apr 22, 2008 5:29 pm    Post subject: Reply with quote  Mark this post and the followings unread

mosc wrote:
I was interested in the center channel extraction because I've noticed, and Bill last night independently noticed, that the center vocals sounded a bit down in the mix compared to stereo. It sounds a tiny bit like a karaoke quizmo. If there was a center channel extraction, then maybe mixing that into the final mix stage might counter that effect and also miraculously correct some timbre distortion that occurs.

So, no need for center channel cancellation, but center channel enhancement.


At least using my setup, timbre distortion is what I find more of a concern. But there is really no way to a/b the source material without two pairs of identical speakers, two in the ambiophoinics position and two in the std. 60 degree stereo position. I do suspect something like comb filtering or position dependent phase cancellation is going on, again at least in my setup. I do have four halfway decent identical speakers but the logistics of such a test setup are problematic for me.

Also I'm not clear on how important arrival time data is and what kind of spatial effect mixing L+R and "repositioning" that mix is going to have especially on "live recorded" music. I don't have a quantitative grasp of this so am going by intuition at this point. Rolling Eyes

Hopefully I will have my time correct panner done in the next week or so and I can do mixes of a pan positioned signal with and with out time correction and then A/B them to hear if that is relevant.

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PostPosted: Tue Apr 22, 2008 5:41 pm    Post subject: Reply with quote  Mark this post and the followings unread

bachus wrote:
djfoxyfox wrote:
Too bad Reaktor doesn't have a feature to turn code into a real standalone VST. One would think that would highly increase the value of having, using, and buying Reaktor.


It would, but it's clear they have made the calculation that it would not benefit their bottom line. Apparently a significant number of people buy Reaktor to gain access to the large library of pre-built/user-built modules. Same old story -- sigh.


Why not use Plouge Bidule?

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PostPosted: Tue Apr 22, 2008 5:45 pm    Post subject: Reply with quote  Mark this post and the followings unread

Ignore that one. It seems Plogue Bidule is dead. Nothing seems to have happened there since 2005/2006.
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PostPosted: Wed Apr 23, 2008 9:40 am    Post subject: Reply with quote  Mark this post and the followings unread

Let’s say we have a synthetic stereo signal. L is a component panned hard left. R is a component panned hard right. C is a component panned to the center. This (I think) allows us to ignore most of the complexity of time and phase for the relevant issue here.

If all three components are of equal perceived amplitude then we can assume that

left = (L + C/2) , right = (R + C/2)

(L + C/2) + (R + C/2) = ( L + R + C)
ratio of C to L = .5

(L + C/2) + a( L + R + C) = L + aL + C/2 + aC + aR

arbitrarily setting a = 1 : 2L + 3/2C + R

ratio of C to L = .75

So we see that the “center” component fraction does increase relative to left and right by a linear factor of .25. However the left channel now contains the component R and the right L which would necessarily(?) reduce the separation of the left and right channels. This does not seem like a good trade off to me. I will implement it tonight and hopefully Howard and I can compare subjective impressions.

Critiques of my analysis always welcome.

EDIT ====================================

empirically .5 seems the best compromise value for 'a'

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PostPosted: Wed Apr 23, 2008 9:56 am    Post subject: Reply with quote  Mark this post and the followings unread

elektro80 wrote:
Ignore that one. It seems Plogue Bidule is dead. Nothing seems to have happened there since 2005/2006.


Thanks for trying. As HRTFs are so important in location I'd like eventually to include those. I've found some in the form of impulse responses but I'm not up to doing convolution in Reaktor so I'm looking at some C++ VST libs. There is some very good convolution C++ source code out there. Very Happy

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PostPosted: Wed Apr 23, 2008 6:54 pm    Post subject: Reply with quote  Mark this post and the followings unread

Running the panner (see time correct panner) with the encoder makes clear how much the center "channel info" is attenuated when run through the encoder. Which is to say, I'm having the same problem on my setup.
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