Question About the Front Channels in the SM...and others

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@Soundfield

I am very aware of such implementations. there is a nice comparison btw at puremix net
I don't want to mess with the original soundfield though (I should have explained better, perhaps). I want to "add" the lost (in crosstalk) ambience information present in the recordings to the sides.
 
I want to "add" the lost (in crosstalk) ambience information present in the recordings to the sides.
Then that's even simpler - that's just a pair of Hafler (L-R) connected speakers. But why you'd then want to stick them to the sides rather than behind you is beyond me.
 
Then that's even simpler - that's just a pair of Hafler (L-R) connected speakers. But why you'd then want to stick them to the sides rather than behind you is beyond me.

must be because I never experienced how this sounds. I am allready eleborating a plan of simulating this on headphones with binaural room impulse responses.
I actualy have no second pair of speakers. I want to know what efect I can get before I buy one.
How does a Hafler setup actualy sound on studio mixed tracks? does the extended ambience come more from the sides or more from behind?
 
Then that's even simpler - that's just a pair of Hafler (L-R) connected speakers. But why you'd then want to stick them to the sides rather than behind you is beyond me.
Wire an appropriate wattage pot ~16 ohm at one end to the wire connecting neg to neg between speakers and the other end to common amp ground. Then you can control variable out of phase blending.
 
But why you'd then want to stick them to the sides rather than behind you is beyond me.

I actualy have a hard time understanding the placement at the back. what is the reasoning behind it? what should determine where the out of phase stuff is coming from should be what is the most dominant reflection captured out of phase, no? this would be without a doubt the reflections coming from the sidewalls. again, I am a total noob at this, but I am trying to aply my general knowledge to the concept
 
I actualy have a hard time understanding the placement at the back. what is the reasoning behind it? what should determine where the out of phase stuff is coming from should be what is the most dominant reflection captured out of phase, no? this would be without a doubt the reflections coming from the sidewalls. again, I am a total noob at this, but I am trying to aply my general knowledge to the concept
Well, with multi-track recorded music, stereo or actual surround sound, it's a lot of fun. To listen to. With music of more "organic" origin such as classical it takes you a step closer to a live performance. In real life music or environmental sounds come at you from a multiplicity of directions, not just from the front. Have you ever been to a cinema theater with surround sound? Besides the movie the music soundtrack is usually in surround sound sound. So consider that.

Ths would be a good time to share Peter Scheiber's Towards a More Spatially Accurate Environment with 3.0. I can only find it for sale at AES. Can anyone point to or share a source?
 
Well, with multi-track recorded music, stereo or actual surround sound, it's a lot of fun. To listen to. With music of more "organic" origin such as classical it takes you a step closer to a live performance. In real life music or environmental sounds come at you from a multiplicity of directions, not just from the front. Have you ever been to a cinema theater with surround sound? Besides the movie the music soundtrack is usually in surround sound sound. So consider that.

Ths would be a good time to share Peter Scheiber's Towards a More Spatially Accurate Environment with 3.0. I can only find it for sale at AES. Can anyone point to or share a source?

I uderstand, but note that I am targeting 2 channel sources with the desire to leave intrument panning where it was placed by the sound engeniers.
I now found this and side placement make more sense to me: "Wendy Carlos Surround 2" (I am not allowed to post links)
what are actualy the angles proposed by Hafler?
however I will simulate all those placements with binaural room impulse responses, which is something I have experience in
 
I may have asked this question before in a general sense, but since we have someone who actually works for a manufacturer on the QQ I figure I might get a better informed answer. The question is: In surround synthesizer mode(s) (or when a stereo signal is run through the SM and it creates additional channels), does it alter in anyway the original front left and right channels?

I ask this because I use separate amps for my front, rear and surround channels. For that reason, I don't run the front L&R feeds from a decoder back to the front channels amp since that amp is already receiving the front L&R signal from the original source. I figure the fewer devices a signal has to run through, the less chance of signal loss, colorization, distortion, etc. Plus, I like being able to control volume and tone for each pair of channels because as we all know surround mixes can vary widely in their balance of front to rear to surround to height/overhead.

OK, so for those of you who have deep knowledge of the legacy decoders - Tate, Sansui QSDs, AS&IC, Lafayette L or W, EVX-44, Sonys, Fosgates, etc: Do any of those change, add or subtract anything from the front channels when processing a stereo signal to "fake surround" ?

I don't think so. Correct?

I use a system similar to yours (preamp/front amp/rear amp/sub). I used to run my SQ-W the way you describe but due to suggestion from members here I tried using all the decoder outputs as described. It didn't take long to get used to it and I've left it that way because it gives a more coherent surroundfield with no apparent lack of seperation.
 
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I use a system similar to yours (preamp/front amp/rear amp/sub). I used to run my SQ-W the way you describe but due to suggestion from members here I tried using all the decoder outputs as described. It didn't take long to get used to it and I've left it that way because it gives a more coherent surroundfield with no apparent lack of seperation.
Every surround decoder that I know of processes both the front and the rear outputs. Years ago I used the original Audionics SQ decoder and reasoning that the fronts outputs were the same as the input signals I briefly tried bypassing the fronts as Wagonmaster_91 suggests, well it just didn't sound right as the front all-pass network was bypassed the front and back outputs were now not in proper phase with each other. I did try that same thing more successfully with a home-brew decoder that was similar to a Dyna and EV-4 decoder, as often I left the fronts undecoded.
 
Keep in mind that my original question was only about the when the decoders were turned to the "synthesized" (4 ch from 2 ch) modes, not decoding of actual quadraphonic sources.
 
Keep in mind that my original question was only about the when the decoders were turned to the "synthesized" (4 ch from 2 ch) modes, not decoding of actual quadraphonic sources.
Yes, and my reply would be the same, possibly with the exception of a more passive decoder as I described. A decoder, processor, synthesizer whatever you want to call it treats a stereo source as if it were an encoded source processing both front and rear outputs.
 
Yes, and my reply would be the same, possibly with the exception of a more passive decoder as I described. A decoder, processor, synthesizer whatever you want to call it treats a stereo source as if it were an encoded source processing both front and rear outputs.

My reply is also the same
 
Hello people,
how do I actualy realize a Hafler circuit digitaly? If I use a convolver with an impulse response on the left channel and send the signal of the right channel through an inverted (of the same) impulse to the same left output I will only have filtered out the center portion of the sound right? But from how I understand this Hafler circuit - Wikipedia it is what the circuit is doing? what is the part I am not understaning here?
 
Hello people,
how do I actualy realize a Hafler circuit digitaly?
I'm afraid I've no idea about how to do it in software, but it is very simple to do the necessary L-R subtraction in hardware:
hafler.jpg
 
TL071, 72 and 74's are pretty much bullet proof. At times I've tried newer higher performance op-amps only to have the circuit become unstable but never had a problem using these.
If I remember correctly (and it was around 25-30 years ago) if they got overdriven you got phase inversion i.e. so the tops of the waveforms folded over. Which messed up an ALC as it turned the gain up as the RMS level dropped, rather than turn it down!
 
ok so I got confused by the wiki article. it doesn't realy filter out out of phase material but simply filters out the phantom center for the rear speakers
Sorta, but you can have identical L and R signals but have a center still if the L and R are phase shifted. So it really is sensitive to out of phase also
 
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