New Surround Master coming! Its a jump to the left and a step to the right

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ar surround

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I seem to be back to about 2 o'clock with most things tonight, as closer to 3 and it starts to clip, so I guess I was about right after all.
Where do others here land at just before you get a little red blinky?
I actually have mine at about 1 o'clock. It's not because of clipping, but rather for level matching when I switch between Involve 4(.1) and Logic7 decoders.
 

M-D-Z

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From Sweetwater.com

When multiple pieces of electronic audio (or video) equipment are used together, the gain structure of the system becomes an important consideration for overall sound quality. This basically refers to which pieces are amplifying or reducing the signal how much. A properly set up gain structure takes maximum advantage of the dynamic range and signal to noise ratio of each piece in the chain. No one piece is doing a disproportionate amount of the amplification unless it is a piece designed for that function (such as a mic preamp). An example of poor gain structure would be a setup where a mixer’s master fader is near the bottom, while all of the individual channel faders are near the top. The resulting level out of the mixer is the same as it would be if all faders were at some mid setting, but the chances of distortion are much higher because of limited available headroom in the circuits preceding the master fader, while the S/N ratio of the final output isn’t as great as it could be were the master fader at a more appropriate level. Part of assembling a system with good gain structure is making sure all the pieces can operate at the same reference level. This is where people go wrong combining -10 dBV equipment and +4 dBu equipment. It can work under the right circumstances, but sometimes the resulting gain structure severely compromises the signal to noise ratio of the final result (or in some cases causes it to be distorted). Gain structure must be considered to optimize any system where levels can be adjusted in more than one place.
 

proufo

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From Sweetwater.com

When multiple pieces of electronic audio (or video) equipment are used together, the gain structure of the system becomes an important consideration for overall sound quality. This basically refers to which pieces are amplifying or reducing the signal how much. A properly set up gain structure takes maximum advantage of the dynamic range and signal to noise ratio of each piece in the chain. No one piece is doing a disproportionate amount of the amplification unless it is a piece designed for that function (such as a mic preamp). An example of poor gain structure would be a setup where a mixer’s master fader is near the bottom, while all of the individual channel faders are near the top. The resulting level out of the mixer is the same as it would be if all faders were at some mid setting, but the chances of distortion are much higher because of limited available headroom in the circuits preceding the master fader, while the S/N ratio of the final output isn’t as great as it could be were the master fader at a more appropriate level. Part of assembling a system with good gain structure is making sure all the pieces can operate at the same reference level. This is where people go wrong combining -10 dBV equipment and +4 dBu equipment. It can work under the right circumstances, but sometimes the resulting gain structure severely compromises the signal to noise ratio of the final result (or in some cases causes it to be distorted). Gain structure must be considered to optimize any system where levels can be adjusted in more than one place.
Add to that the desirability of feeding the ADC a signal that will allow for an optimal (within its limitations) conversion, one that doesn't waste bits/numbers.
 

Sonik Wiz

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Dear all

I have been doing some background investigation on the new SM V2 and I will be advising all of some findings shortly but as a general operational note, it is better to operate the thing with the main input level knob as fully advanced clockwise as possible to just below the onset of the clipping light. If you are dumb like me you will be setting the thing with all knobs at 12 o clock by habit - not right. Doing the fully clockwise thing will optimise the decode operation, distortion and signal to noise ratio.

Regards

Chucky
So last night I decided to recheck my systems overall balance accuracy & the SM v2 in specific. I start by using the internal channel sequence noise from my Anthem pre-pro with a SPL meter to match. Then with my Oppo plugged directly into the SM (no Chase RLC-1 at the front end) I play QS/RM test noise precisely generated by Adobe Audition. Using the meter again I tweak the SM left front to 8o dB, match the right front to left front & match the rears to that. I turn off my power amps for the 4 corner speakers & just leave the center front on with the SM set to Involve 5.1. I adjust that signal with of course center front (mono) noise to -3db the fronts or SPL of 77dB. Just by empirical experience that gives me a decent center front with out overpowering.

I turned on the power amps again & rechecked this time listening for crosstalk from low separation. Really there was a lot of bleed through, quite low separation for any direction. Then I remembered I set the level control to ~ straight up because the Oppo has a fairly high output level. I turned the SM input level to max & reduced the output on the Oppo to just under clipping. I repeated the previous steps & was amazed to hear almost zero bleed through from anywhere. Playing music it sounded maybe even better than I had been used to. The soundfield & direction was expansive & pin point.

Why the difference? My theory (& it's only that. No grief if I'm wrong) is that in a decoder like this are are 2 parallel signal paths. The input is sent to the direction sensing & control signal circuits & meanwhile it is also goes to the variable matrix circuits where the actual decoding & enhancement occurs. I am wondering if the input level control only controls the signal that goes to the matrix & separation enhancement circuits? And the direction sensing signal are not affected by this? This would mean that they would not be at the correct levels for each other unless the input level control is set at max.

Just wondering but of course you get everyone's attention when using big bold scary underlined font about this.

At any rate I'm sure all your customers would like a little more detail from the source why the level control needs to be at max. When then of course it can not be properly utilized as a level control!

Thanks in advance for the reply!!
 

chucky3042

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Hi Sonik
Good questions. Firstly the signal matrix paths and the steering paths go down the same "input level" adjust pipe. The reason I advocate pushing the level up to maximum is that in fact there is a 16 db level drop from the maximum position to the 12 o clock position (not the 6 db as indicated on the panel- yes our stuff up) this means 16 db more noise and less resolution.

I have not observed or expect and separation changes with respect to this level but I will do some tests this week. My only thought is that if the Oppo signal level was too high then something might be clipping or soft clipping - this will throw out the separation detect software severely.

I will check

Regards

Chucky
 

Sonik Wiz

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Hi Sonik
Good questions. Firstly the signal matrix paths and the steering paths go down the same "input level" adjust pipe. The reason I advocate pushing the level up to maximum is that in fact there is a 16 db level drop from the maximum position to the 12 o clock position (not the 6 db as indicated on the panel- yes our stuff up) this means 16 db more noise and less resolution.

I have not observed or expect and separation changes with respect to this level but I will do some tests this week. My only thought is that if the Oppo signal level was too high then something might be clipping or soft clipping - this will throw out the separation detect software severely.

I will check

Regards

Chucky
Thanks for the explanation. That's a much less awful problem then what I came up with.

Many folks here (not me) are used to just having a reciever run it's own built in program to adjust level & tone. Then they have a product like the SM that must be adjusted manually. I wonder if you have any suggestions to them how to accuratley set up the SM v2? Or any improvements to my procedure I metioned?
 

chucky3042

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Thanks for the explanation. That's a much less awful problem then what I came up with.

Many folks here (not me) are used to just having a reciever run it's own built in program to adjust level & tone. Then they have a product like the SM that must be adjusted manually. I wonder if you have any suggestions to them how to accuratley set up the SM v2? Or any improvements to my procedure I metioned?
Yep working on it - one more tip- try to set the output level knobs beyond 1.30 o clock.

The input level knob was a big compromise of cost/ complexity as we wanted to avoid the use of a 6 way digital level control chip on the outputs (as with the Sinn box and new Y4).
 

kfbkfb

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I don't have a Surround Master decoder, however, I was wondering how the
decoding sounds with highly data reduced (perhaps SiriusXM) audio?

(maybe make a test audio program of SQ encoded material with MP3
at a very low data rate)


Kirk Bayne
 

chucky3042

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I don't have a Surround Master decoder, however, I was wondering how the
decoding sounds with highly data reduced (perhaps SiriusXM) audio?

(maybe make a test audio program of SQ encoded material with MP3
at a very low data rate)


Kirk Bayne

No Problemo
In fact one of the test tracks we use in our demo is a compressed mp3 downloaded from Youtube T2 trailer. The resolution of the data is not really a major parameter in the decode matrix.
 

atrocity

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I don't have a Surround Master decoder, however, I was wondering how the
decoding sounds with highly data reduced (perhaps SiriusXM) audio?

(maybe make a test audio program of SQ encoded material with MP3
at a very low data rate)


Kirk Bayne
In my experience, it definitely works in the sense that you get surround music. But if whatever you're playing suffers from compression artifacts, they seem to become even more prominent. Or at least that's been my (limited) experience. Of course, that's not the fault of the SM, it's just doing what it can with what it's been given.

I haven't tried playing lossy SQ or QS encoded material. Would be an interesting test.
 

mixfo

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Excuse me if this is a dumb question, but a number of people have indicated a preference for the 4.1 setting over 5.1 when processing a stereo input using the SM2. Under what circumstances would 4.1 be better than 5.1? Is it a completely different mix, or is the centre channel signal just fed into the front L and R channels for 4.1? Just curious ....
 

sukothai

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It is not just for SM2. Some people prefer the virtual center channel image produced from 4.1 over the discrete center channel of 5.1 even for discrete multichannel recordings. It is really a personal preference. I usually prefer the virtual center channel of 4.1 over 5.1, but 5.1 can be good if done well. It is my belief that it is harder to create a good 5.1 mix than a 4.1 mix because the 5.1 center channel is so prominent and inflexible. With SM2 processing, the center tends to include everything that is common between the left and right front channels and I prefer things a little more spread out.
 

mixfo

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OK, thanks for the explanation. Since the SM2 has a control that alters the centre channel volume, I have been able to use the 5.1 setting without being overwhelmed by it. However, I will experiment with the 4.1 setting to see if it provides better imaging (or maybe it'll be a case by case thing).
 

kfbkfb

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