# AVR Frequency Response Graphs



## MACCA350 (Apr 25, 2006)

This started from this thread.

I Thought we'd start a thread to compare the Frequency Response of different receivers/pre-amps.

Method: Basically its taking the Mic out of the loop and connecting the preamp outputs of your receiver to the inputs of your soundcard. 

There are 2 ways we need to connect so we can get the measurements we are after.
1) RED line = this is for the full spectrum sweep and Mains when crossover is engaged
2) GREEN line = this is for testing what is actually getting to the sub when crossover is engaged

Note: do not connect both at the same time

Thanks to Brucek for the graph:bigsmile: 









There are 3 measurements that we can look at
1) Full spectrum sweep 2Hz -->20kHz]
2) Mains with the crossover engaged
3) Subwoofer with crossover engaged

A couple of things I did to keep things simple.
On the receiver
1) Set volume trims to 0.0
2) Set main volume to 0.0
3) Do not adjust volumes between readings(this will keep a reference for comparing)

On REW
1) Under 'Settings' Make sure No Mic calibration file is loaded
2) Under 'Settings' create a soundcard calibration file (include all cables you will use in the loop) load the file and run a sweep and make sure you get a dead flat line.
3) Under 'Measure', set measuring level(dB FS) to -10
4) Under 'Measure', set the End Freq(Hz) to 24,000
5) After the full spectrum sweep, set the 'Trace Offset'(under 'Trace Adjustments) so that the line around 1kHz is as close as possable to 0db on the graph(I had to use -84.8) This value will need to be input to the other measurements, do not change this number even if the other graphs don't line up with 0db(remember we measured at the same level so these differences are due to the crossover being in chain of processing) You can click the 'Add offset to Data' button if you like.
6) Under 'Graph Limits', set to:
Top = 12
Bottom = -12
Left = 2
Right = 20,000
7) After measurements and adjustments are taken click on the 'All Measured' tab and your ready to post

When posting add the Receiver/Pre-amp model information.
You can zoom into the graph to find -1db and -3db points.
You should end up with something like my next post.

Any suggestions are welcome,
Cheers


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## MACCA350 (Apr 25, 2006)

Denon AVR-3805








Full (Red)
-1db at 4.5Hz 
-3db at 2.2Hz

Sub channel(blue)
-1db at 6.9Hz 
-3db at 4.4Hz

cheers


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## JohnM (Apr 11, 2006)

The slight ripple you see at the low end is caused by a narrower impulse response window than is needed. If you look at the Impulse Response graph you'll see a section of the response that is not being included, so extend the window duration to cover all the area above the noise floor and the low end should show a smoother curve. If you post a plot of the impulse response (with the Y axis set to dB) I can indicate exactly what should be included.


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## Otto (May 18, 2006)

Hi there,

I would expect the sub and main signals to intersect at -6 dB from their nominal levels, but it appears that they intersect more around 4 or 5 dB down (yes, I see that their nominal levels are _above_ 0 dB slightly).

If I look at where each signal actually _does_ cross at -6 dB, it appears that the sub's -6 dB point is about 80 Hz, while the main signal's -6 dB point is about 60 Hz.

Is it possible that there are two different crossovers set up? Perhaps something else going on?


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## brucek (Apr 11, 2006)

> Is it possible that there are two different crossovers set up? Perhaps something else going on?


Of course. Most systems use a 2nd order HPF (x5) and a 4th order LPF. This then accounts for the 2nd order roll off of the mains to result in an effective 4th order.



> Any suggestions are welcome


My feeling is that most receivers/processors will be perfect in their crossover execution and will have a flat line out to 20KHz without fail (your green line). The differences we're most interested in would be the low end response effect on our subwoofers. This is where you would find differences between products. I would rather see the graphs from 2Hz to 200Hz (or at least that in addition to what you're requesting), to better examine the drop below 20Hz. Certainly all the crazies aficionados of bass down to 5Hz would be extremely interested in the constraint their equipment is placing on the final sound.









brucek


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## MACCA350 (Apr 25, 2006)

brucek said:


> My feeling is that most receivers/processors will be perfect in their crossover execution and will have a flat line out to 20KHz without fail (your green line). The differences we're most interested in would be the low end response effect on our subwoofers. This is where you would find differences between products. I would rather see the graphs from 2Hz to 200Hz (or at least that in addition to what you're requesting), to better examine the drop below 20Hz. Certainly all the crazies aficionados of bass down to 5Hz would be extremely interested in the constraint their equipment is placing on the final sound.
> 
> brucek


I agree that the majority of differences will be below 20Hz, but I would like to keep the full sweep also because you never know, we may just find some manufactures altering their response to have a "signature sound". That would be interesting:sneeky: 

The other question is, do we adjust the second sub only graph so that the highest point is at the 0db line, even though it read higher under identical conditions(I'm leaning towards yes). I'm really wondering why the inclusion of the crossover increased the line by about 0.5db:scratchhead: 

The other thing that was interesting was the noisefloor as shown in the impulse response graphs. Both Full and Mains was -105db, but Sub was -60. I think this was due to signal being passed through above the crossover, I've zoomed out the graph, see here:









cheers


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## brucek (Apr 11, 2006)

> we may just find some manufactures altering their response to have a "signature sound".


doubtful........



> The other question is, do we adjust the second sub only graph so that the highest point is at the 0db line


I would say no. You'll see in my graph that I am below 0dB since the crossover at 60Hz never hits 0dB before the response starts to drag the line back down. The proof is in the fact that at 60Hz it is exactly at its proper -6dB target.... if I had used 80hz cross, then it would have hit zero....

brucek


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## MACCA350 (Apr 25, 2006)

JohnM said:


> The slight ripple you see at the low end is caused by a narrower impulse response window than is needed. If you look at the Impulse Response graph you'll see a section of the response that is not being included, so extend the window duration to cover all the area above the noise floor and the low end should show a smoother curve. If you post a plot of the impulse response (with the Y axis set to dB) I can indicate exactly what should be included.


I'm not sure I understand how to make those changes. Here are the Impulse Response graphs.

Full









Main









Sub









cheers


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## MACCA350 (Apr 25, 2006)

brucek said:


> doubtful........
> 
> 
> I would say no. You'll see in my graph that I am below 0dB since the crossover at 60Hz never hits 0dB before the response starts to drag the line back down. The proof is in the fact that at 60Hz it is exactly at its proper -6dB target.... if I had used 80hz cross, then it would have hit zero....
> ...


This doesn't explain my graph, since mine reached over 0db and the -6db line crosses at about 84Hz instead of the 80Hz crossover as set in the AVR. adjusting to 0db brings the -6db point to 80.9Hz. Any ideas why there was an overall increase when the crossover was engaged?

I'll hookup my Denon 1603 in the morning and see what that does.

cheers


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## JohnM (Apr 11, 2006)

Macca,

Click the IR Windows button in the REW toolbar then adjust the Right Window value until the dotted line is out where the impulse response dispappears into the noise level. For the blue trace that is about 650ms for the response you captured (but see the noise floor comment that follows), for the red trace about 1200ms. After you adjust the value, hit the Apply Windows button to apply it. You will then find the responses roll off more smoothly at low frequencies. 

The noise floor is high for the sub (blue) trace because you did a full range sweep but there is no useful output above a few hundred Hz, so the captured signal just has noise for much of the swept range. Do a sweep to 500Hz or so for the sub and you will see a much lower noise floor.


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## MACCA350 (Apr 25, 2006)

JohnM said:


> Macca,
> 
> Click the IR Windows button in the REW toolbar then adjust the Right Window value until the dotted line is out where the impulse response dispappears into the noise level. For the blue trace that is about 650ms for the response you captured (but see the noise floor comment that follows), for the red trace about 1200ms. After you adjust the value, hit the Apply Windows button to apply it. You will then find the responses roll off more smoothly at low frequencies.
> 
> The noise floor is high for the sub (blue) trace because you did a full range sweep but there is no useful output above a few hundred Hz, so the captured signal just has noise for much of the swept range. Do a sweep to 500Hz or so for the sub and you will see a much lower noise floor.


Understood, and adjusted, thanks. I also noticed that increasing the window lowered the -1db and -3db points on the graph for the Full range plot. I'll remember to cut the sweep for the Sub plot next time, I may redo the above Sub plot in this way and see if it makes any difference to the -1db and -3db points.

Do you opinion on the issue of the different trace levels as stated above. I'll add a second graph for the Sub plot, but I'm still not sure whether to set the highest point to 0db for reference purposes, because this will affect the -1db and -3db points(I think these points should be compared to the highest point on the plot, but then again:huh: )

I'm about to run some tests for the Denon AVR-1603, so those graphs should be up soon. 

cheers


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## MACCA350 (Apr 25, 2006)

Ok I've got an interesting one here.
This is the Denon AVR-1603 with 80Hz crossover








Red = V.AUX input and CDR output
Purple = EXT. input and SUB output
Green = V.AUX input and SUB output with speakers set to LARGE(LFE+MAIN on/pff didn't make any difference) set to both STEREO/DIRECT(this trace is also the same with speakers set to SMALL but with prologic cinema or DIRECT enguaged)
Blue = V.AUX input and SUB output with speakers set to SMALL set to STEREO

Something really screwy going on there:scratchhead: I even reset the microprocessor but it didn't make any difference.

Any ideas?

cheers


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## MakeFlat (Mar 30, 2007)

MACCA350 said:


> Blue = V.AUX input and SUB output with speakers set to SMALL set to both STEREO/DIRECT
> 
> Something really screwy going on there:scratchhead: I even reset the microprocessor but it didn't make any difference.
> 
> ...


I'll comment on the blue curve. Your result is consistent with my REW runs for the Stereo Mode on the Denon 1604. There is one difference: In the Direct Mode, the SUB response appears to roll off at 8Hz on my 1604's SUB output. So I guess Denon changed the design. Why they decided to roll off Stereo at 40Hz on the Denon 160x, it stumps me:scratchhead:

Also, on my REW runs, in Stereo Mode, the mains and sub appear to be quite a bit lower in level when compared to all other modes: PL2, 5 Ch, Direct. The level is consistent with what you find on the sub's level.


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## MACCA350 (Apr 25, 2006)

MakeFlat said:


> I'll comment on the blue curve. Your result is consistent with my REW runs for the Stereo Mode on the Denon 1604. There is one difference: In the Direct Mode, the SUB response appears to roll off at 8Hz on my 1604's SUB output. So I guess Denon changed the design. Why they decided to roll off Stereo at 40Hz on the Denon 160x, it stumps me:scratchhead:
> 
> Also, on my REW runs, in Stereo Mode, the mains and sub appear to be quite a bit lower in level when compared to all other modes: PL2, 5 Ch, Direct. The level is consistent with what you find on the sub's level.


You're right. I just redid the DIRECT and STEREO measurements with speakers set to SMALL, and the DIRECT line is identical to the green plot(I've changed the previous post to add these changes).

So it seems that with speakers set to SMALL and in STEREO mode the SUB channel is seriously attenuated, yet setting the speakers to LARGE gives you full output on the SUB channel(and the LFE+MAIN feature makes no difference), to me this is backwards.

It seems to be only the STEREO mode when speakers are set to SMALL that cuts the SUB channel output. I ran a sweep with the crossover set to 120Hz and comparing both it looks like there is a subsonic filter engaged in this mode around 40Hz or so(as you suggested)

But the question is why, If you tell the receiver your mains are small and you have a sub, why is it cutting off the sub channel like this ONLY in stereo mode, really doesn't make sense:coocoo: 

cheers


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## Otto (May 18, 2006)

Hi MACCA,

I will agree that that seems strange in both cases. The way I _think_ it should work is that there wouldn't be a difference between the LPF crossover between small and large -- the small and large settings should only affect the mains. I guess I should that that this is true as long as you have the "LFE+Main" feature set. But you already know that, which is why you're complaining about the seriously attenuated sub output in the "Small & Stereo" case above.

Unfortunately, I don't have any reasoning for its behavior, but I can offer a few other comments.

While I tend to agree with brucek that we are frequently concerned with the subwoofer band, I do believe there's some value in keeping the mains sweep in there. I would imagine that most of the time it will be correct. However, as you may have seen in my thread here, I was able to detect a problem with the HPF of my Outlaw 990 (I notified Outlaw, and after a bunch of discussion, they have indeed acknowledged that this is likely a real problem). I only measured the listening position response, but you can clearly see a HPF engaged on the mains when it should not be (in DIRECT mode, of all things!?!?!). So, I'd like to see the mains signal in there in addition to the others. I think that the upper frequency of 200 Hz is good.

Here's another thing. I don't know how many receivers you have access to, or how much trouble it might be, but the Outlaw discrepancy taught me something: the behavior of the different "modes" may be different, even though one would expect it to be the same. I don't think you started this thread/project as an effort to find bugs in these products, but as long as you have one "on the bench" you might consider scrolling through as many modes as possible. I didn't go through _all_ the modes on the 990, but I did hit all the 2-channel modes. Anyway, just a suggestion.

Wow! All this talk of modes reminds me of "Direct Mode" a.k.a. "Bypass Mode". I tend to think of this mode as applying zero change to the original signal, so you might see some different behavior here. It was "Bypass Mode" in which I found the Outlaw was applying a rolloff to the mains when set to Small (though it had nothing to do with the actual crossover frequency selected). Do you know how the Denon is supposed to behave? I would somewhat expect that there would be no LPF on the sub out in Direct/Bypass mode. I do believe the implementation of "Direct Mode" could vary from manufacturer to manufacturer.

So, MACCA, I just want to say that this is an awesome thread -- very educational. Keep up the good work! I will be working to post this information for my Outlaw 990.

Thanks!!!


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## clubfoot (Apr 12, 2007)

MACCA350 said:


> You're right. I just redid the DIRECT and STEREO measurements with speakers set to SMALL, and the DIRECT line is identical to the green plot(I've changed the previous post to add these changes).
> 
> So it seems that with speakers set to SMALL and in STEREO mode the SUB channel is seriously attenuated, yet setting the speakers to LARGE gives you full output on the SUB channel(and the LFE+MAIN feature makes no difference), to me this is backwards.
> 
> ...


Could it be that in LFE+large mains, the design thinking is that you are driving mains that are wired to the sub and using the sub xover instead of the processor xover? And when small is selected the processor is "thinking" it has to protect the small speakers from damage so it adds subsonic protection!.....regardless of a sub being connected?!

This could also be designed to not over drive the power amp section of receivers.

My question is, I set my processor to stereo, small + sub and use REW to do calibrations,...wouldn't this mode introduce "errors" in what I see on plots without knowing there is a roll off to compensate for, I will have to try it with mains set to large.


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## brucek (Apr 11, 2006)

> Any ideas?


That drop off in the response in stereo doesn't make a lot of sense. Try just feeding one channel and not two channels from REW into the receiver. The REW signal is mono, and should only be sent to one channel to do the testing....

brucek


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## Otto (May 18, 2006)

I had an opportunity to measure my Outlaw 990 today, and there are some interesting results. I know that I used a -30 to +12 range on the X axis, but I liked it because I find it gives better resolution as to crossover slope (I went back and forth for a while, and settled on this). My apologies that it's different than previous posts. 

Here is "Stereo Mode". I believe that this is the most correct one. However, I believe that the mains are employing a 4th order filter where it should only be a 2nd order filter. I determined this by looking at the mains signal from 80 Hz down to 40 Hz. In that range, it has dropped 24 dB. Each "order" of filter should cause a drop of 6 dB per octave, so 24 dB per octave is a 4th order filter. I think this is wrong to have a 4th order filter on the mains, as described by brucek above, and also because the 990 is spec'd in its manual as having a 2nd order filter on the mains. 

As I understand it per brucek, the mains, with their 2nd order filter, should intersect the crossover point(in this case 80 Hz) at -3 dB. In this case, the mains having a 4th order filter causes them to intersect with 80 Hz at approximately -6 dB. I believe this is incorrect.

The sub signal crosses 80 Hz at -6dB with what appears to be a 4th order filter. This seems correct, although the response itself does hump up a bit from 20 to 50 Hz, likely as a result of imperfect filters (none of them are "perfect"). All in all, the sub response seems OK.

The "Large" signal for the mains seems OK.









I also measured "Upsample Mode". From time to time, I get a very annoying static/crackling sound when using "Upsample Mode". This weekend is "one of those times" (started on Thursday night). I can't seem to get rid of it at this point, so I thought I'd measure it. Yes, it sounds as good as it measures. Interesting that the noise doesn't seem to affect the sub. "Upsample Mode" also seems to be afflicted by the 4th order filter on the mains. I opened a trouble ticket with Outlaw yesterday about the crackling, but have not yet received a response (understandable since it was submitted on a Saturday -- The Outlaws are generally very responsive to user issues, and have excellent customer service, product support, and warranties). EDIT: Two things -- first, the graph should be labeled "Upsample" mode at the top, not "Bypass". Second, this phenomenon of crackling in "Upsample" mode has gone away after a week or so. The Outlaw was out of my system (unplugged from all AC power) for about a week, and when reinserted, it didn't do this anymore. I had unplugged it for ~30 seconds in the past to try to get rid of this problem, and it had persisted across at least a couple power cycles.









Here is a measurement of "Bypass Mode," which is intended to be just the true analog signal as it comes from the source. No digital to analog conversion, no bass management on the mains or the sub, regardless of speaker "size" (large/small) or crossover frequency; none of those controls should make a difference to the output signal. Note that some type of filtering is applied to the mains signal when the mains are set to "small". This measurement was the same as shown, regardless of crossover frequency, as long as the mains were set to small. This is unacceptable in my opinion. It also leads me to ask: is this filtering going on in the digital domain? If so, why are we sampling (performing an analog to digital conversion) on bypass mode? And finally: if the Model 990 is doing those types of operations on the simplest input "mode" what else is going on that we still don't know about? I have opened a problem ticket with Outlaw, and I do acknowledge that they are considering this an actual problem. Kudos to Outlaw in that regard. My other thread is here. 

Also notice the slight discrepancy between main and sub levels in the full range measurements. I would think that a true bypass mode would be able to get those levels dead on.









Finally, there have been some complaints on the Outlaw Forum ("The Saloon") about the Model 990 applying bass management to the LFE signal. Indeed, Outlaw has described the bass management's crossover points such that they define the transition point between the bass material in a given speaker group ("fronts", "center", "surrounds" and "rears") and the subwoofer. The LFE channel is an independent channel as defined by the DD/DTS specs, and should not be filtered. If it _is_ filtered, where does that information go? Apparently in the trash can... The LFE channel is spec'd to 120 Hz (for DD, at least, IIRC), and should be allowed its full bandwidth through the system. Since REW doesn't have the ability to output a discrete 5.1 signal, I opted to use the 7.1 channel analog inputs on the Model 990. Actually, I only used the "sub" of those 7.1 inputs, but that's all I'm really interested in for this test. Now this is where I make a small leap -- I'm going to assume that just like the discrete LFE track coming down a digital connection, the Model 990 should NOT apply bass management to any "sub" channel that comes in via the 7.1 inputs. My reasoning is twofold. First, some (all?) DVD players implement their own bass management. Therefore, any subsequent bass managment in the processor would not only be undesirable, it would be mucking with an already filtered signal. Second, the same reasoning as before -- why apply bass management to a signal that is, by definition, passed to 120 Hz and has no where else to go when filtered? You're just dumping content down the drain. This is unacceptable to me as well, and another member at the Outlaw Forum has submitted a trouble ticket for this issue. I believe that Outlaw will fix this issue as well, as theyhave been very helpful in the past, and have already provided at least two firmware upgrades. Anyway, here's the trace (sorry about the X-axis range, I didn't notice that till I was done):


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## MACCA350 (Apr 25, 2006)

brucek said:


> That drop off in the response in stereo doesn't make a lot of sense. Try just feeding one channel and not two channels from REW into the receiver. The REW signal is mono, and should only be sent to one channel to do the testing....
> 
> brucek


Yeah, no sense at all. All testing was with REW feeding the Right channel input only. Maybe I'll try it with REW driving the Left channel only.

I'll have another play shortly. I also want to recheck with the Large setting engaged because that should cut the sub channel completely.

Found some info in the Denon AVR-3805 manual about the Direct and Pure-Direct functionality(I couldn't find much in the 1603 manual)



> *DIRECT mode*
> Use this mode to achieve good quality 2-channel sound while
> watching images. *In this mode, the audio signals bypass
> such circuits as the tone circuit and are transmitted directly,
> ...


Explains why bass management is occurring in Direct and Pure Direct mode with Denon models, maybe we should take our own advice......RTFM:rofl: 

Otto, Interesting stuff there especially what's happening with the upsampling, and you're External inputs, as you can see the 1603 is not modifying it's External inputs(although it has other issues). Was there anything abnormal above 200Hz? Good to hear Outlaw are taking things on board.

cheers


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## Chrisbee (Apr 20, 2006)

The effect you see in "stereo" may be simply be a rumble filter for turntables.
Most music (except organ and electronic) lacks anything useful under (say) 40Hz anyway. 
Though a bypass option would be nice.
I'm glad I don't use an AVR for my stereo organ music listening!
In fact I still don't own an AVR. :T


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## MACCA350 (Apr 25, 2006)

Chrisbee said:


> The effect you see in "stereo" may be simply be a rumble filter for turntables.


That would be fine except that this happens on all inputs, eg. DVD, CD, etc.



> Most music (except organ and electronic) lacks anything useful under (say) 40Hz anyway.
> Though a bypass option would be nice.


Thats beside the point, and I've got plenty of music that digs well below 40Hz.

Basic stereo mode with bass management applied should operate correctly and it's not, to make it work correctly I need to put it into direct mode.

The other thing to note is that this attenuated sub output(blue line)only occurs when in STEREO mode, speakers set to SMALL and subwoofer set to YES. And when you change the speakers to LARGE(and keep everything else the same) you end up with full sub output(green line), in fact every mode with speakers set to LARGE gives you full output on the sub channel. 

This is definatly not right because the signal is being input on the Main L or R channel and if speakers are set to LARGE and LFE set to NORM you should get nothing on the sub channel. Only when LFE is set to +MAIN should you get output on the sub channel in this case, and with the 1603(and 1604, thanks to MakeFlat) this is not the case.


cheers


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## MACCA350 (Apr 25, 2006)

I re-measured everything trying both V.AUX and DVD Left channel input and came up with the same results.

I might have another play with the 3805 and see if it does anything funny(although I highly doubt it, but you never know)

Cheers


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## Otto (May 18, 2006)

MACCA said:


> Was there anything abnormal above 200Hz? Good to hear Outlaw are taking things on board.


Didn't notice anything weird above 200 Hz. Yeah, I really tried to credit the Outlaws because they do such a good job -- my commentary is somewhat critical, but I don't want to bash the Outlaws, really. I'm just reporting what I'm finding...


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## MACCA350 (Apr 25, 2006)

Ok, I did some more testing on the 3805. Here's a couple of things to note.

1)the sub output is correct for all modes. eg. no attenuated funny stuff like the 1603.

2)the sub output can be turned off in the volume trims(when DIRECT or PURE DIRECT is engaged) as noted in the manual. This setting is global for DIRECT and PURE DIRECT. eg for all inputs(not sure about DD or DTS signals) but does not affect any other modes(eg STEREO, etc). so you can have it either way for DIRECT and PURE DIRECT.

3)the LFE - LFE+MAIN setting makes no difference for 2channel material in STEREO, DIRECT, PURE DIRECT and some DSP simulated modes including 5/7.1 STEREO mode.

4)the LFE - LFE+MAIN setting only operates on 2channel material when PLIIx, DTS NEO:6 and most DSP simulated modes. Going on this I'd assume proper operation on DD and DTS multichannel tracks.

The manual explains 3) and 4) on page 37


> Selection of the “LFE ” play mode will play the low frequency signal range of the channel selected with “LARGE” from that channel only.
> Therefore, the low frequency signal range that are played from the subwoofer channel are only the low frequency signal range of LFE (*only
> during Dolby Digital or DTS signal playback*) and the channel specified as “SMALL” in the setup menu.
> • Select the play mode that provides bass reproduction with quantity.
> ...


BTW, I suggest disconnecting *ALL* speakers when testing DSP simulations, I near [email protected] myself on one of those:raped: 

cheers


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## Hakka (Sep 10, 2006)

Denon 3803

First graph is Stereo

Red = small/main
green = large/main
purple = small/sub

My sub trim was set at +0.5 for the test, not sure why the level is up that much over the mains.

Second graph is Pure Direct

Green = Sub
Brown = main large/small

The mains show the same signal regardless of the speaker size setting.

The third graph shows stereo(green) vs pure direct (blue) main/large.

The fourth graph shows stereo(purple) vs pure direct(green).

Notice the high freq boost in stereo mode.


I tried to do a measurement using the optical output of my soundcard but my receiver shutdown as soon as I started the sweep. :holycow: 

Harry.


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## MACCA350 (Apr 25, 2006)

Hakka, did you set your global channel trim levels, or just the mode you were in at the time? maybe the particular mode you were in for testing had different trim levels for the sub, because that is a pretty big boost on the sub channel. It seems that you may have a tone control active in the top end for stereo mode, since the direct mode trace was flat.

cheers


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## MACCA350 (Apr 25, 2006)

Here's a couple of things I've been thinking about, I know, I think too much:nerd: 

I'm thinking that we can set the 0db reference level when we loop the soundcard, this way we can see any attenuation from input to output(with the receivers at 0.0db) and have a reference point between receivers. Or maybe set the main fullrange signal at 0db(at the 1kHz point) but also keep a soundcard trace for reference(I like this idea better). 

I'm also thinking of changing the graph limits to 
*Top = 6
bottom =-18*
left = 2
right = 20,000(fullrange graph)
right = 200(sub graph)
This will give us a vertical scale with multiples of 3db

Maybe we can set out some colors for particular traces. The first four that REW use are Red, Green, Blue, Purple. We could set them to something like this.

Red = Soundcard reference trace
green = Mains fullrange trace
Blue = Mains with crossover engaged trace
Purple = Sub with crossover engaged trace
And keep all colors and traces for the sub graph so it will basically be a zoom-in of the fullrange graph. Also make sure we use no filtering in the traces.

If we can pick a common mode for the tests to be performed in, or just say whichever one performs the correct bass management for the tests.

What I'm hoping for is identical looking graphs and identical testing methods(atleast for the first two graphs) so we can accurately and easily compare those receivers and pre/amps tested. 

Once we've worked out the best and easiest method we can write up some instructions so that anyone can perform the test and post accurate and comparable results. We may need to start a new _clean_ thread for just the results(kind of like a reference thread to post just the comparable graphs and keep this thread for all the discussion and other graphs), once we've nutted all these things out.

Anyway its some food for thought, I'm really interested to see what kinds of variations we get, It just makes it hard to compare the different graphs.

cheers


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## Otto (May 18, 2006)

MACCA350 said:


> Here's a couple of things I've been thinking about, I know, I think too much:nerd:
> 
> I'm thinking that we can set the 0db reference level when we loop the soundcard, this way we can see any attenuation from input to output(with the receivers at 0.0db) and have a reference point between receivers. Or maybe set the main fullrange signal at 0db(at the 1kHz point) but also keep a soundcard trace for reference(I like this idea better).


Yeah, I was thinking the same thing, namely becuase I screwed up my mesaurements slightly. I did not normalize all my plots from any initial measurement, I normalized each one to a relative point within _each measurement_. It's not much of a big deal that it's wrong, as they are still relevant because each is relative to itself, but you're right: all these measurements should be shown with respect to the soundcard loopback measurement after the soundcard cal is done (which should be a flat line). We can leave the cable measurement in each plot for reference, or just "know" that the flat line would be a 0 dB if it were in the picture. Perhaps leaving it in is better, then we somewhat know for sure. Everyone just has to be careful to always enter the same "offset" to all measured data.



> I'm also thinking of changing the graph limits to
> *Top = 6
> bottom =-18*
> left = 2
> ...


I like the idea of 3 dB increments on the X-axis, but I think that -18 is too shallow to show us all the effects of the slopes. I'm not trying to push what I did just because I did it, but I like the way that my 12 to -30 dB plots came out. They are in 6 dB increments, which, to me, are mostly as usable as 3 dB increments. As long as it's not 4 dB or 8 dB increments, since we are so interested in 3 dB and 6 dB points (as you know).

I think I'm going to move this thread from the BFD forum to the Receiver|Processor forum.


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## MACCA350 (Apr 25, 2006)

Otto said:


> Yeah, I was thinking the same thing, namely becuase I screwed up my mesaurements slightly. I did not normalize all my plots from any initial measurement, I normalized each one to a relative point within _each measurement_. It's not much of a big deal that it's wrong, as they are still relevant because each is relative to itself, but you're right: all these measurements should be shown with respect to the soundcard loopback measurement after the soundcard cal is done (which should be a flat line). We can leave the cable measurement in each plot for reference, or just "know" that the flat line would be a 0 dB if it were in the picture. Perhaps leaving it in is better, then we somewhat know for sure. Everyone just has to be careful to always enter the same "offset" to all measured data.


Ok, so we agree to keep a 'soundcard loop' for reference. With this I see two options:

1) *Set the 0db line to the 'soundcard loop trace'* (in this case we shouldn't need to have a trace for it since the 0db line is the trace, but then again:dontknow: ) 

2) *Set the 0db line to the 'main fullrange trace' and keep a 'soundcard loop trace' for model to model comparison purposes* (this will make finding -3/-6/etc points on these traces easier to see(atleast the 'main fullrange trace'). Since we all calibrate our systems, the difference between the 'soundcard loop trace' and 'main fullrange trace' is irrelevant but it will still be interesting to see variations. 

What would be handy is if REW was able to set an offset automatically based on a 'zero set' trace, like running a trace and then tell REW to 'set this one to zero' and all the rest will automatically be set with the same offset



> I like the idea of 3 dB increments on the X-axis, but I think that -18 is too shallow to show us all the effects of the slopes. I'm not trying to push what I did just because I did it, but I like the way that my 12 to -30 dB plots came out. They are in 6 dB increments, which, to me, are mostly as usable as 3 dB increments. As long as it's not 4 dB or 8 dB increments, since we are so interested in 3 dB and 6 dB points (as you know).


Yeah, I'll agree to that:T I'd love to keep the -3db line, but I can see that we may need to keep more range in the graph.

Top = 12
bottom = -30
left = 2
right = 20,000(fullrange graph)
right = 200(sub graph)

Any other suggestions?

cheers


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## Otto (May 18, 2006)

MACCA350 said:


> Ok, so we agree to keep a 'soundcard loop' for reference. With this I see two options:
> 
> 1) *Set the 0db line to the 'soundcard loop trace'* (in this case we shouldn't need to have a trace for it since the 0db line is the trace, but then again:dontknow: )
> 
> 2) *Set the 0db line to the 'main fullrange trace' and keep a 'soundcard loop trace' for model to model comparison purposes* (this will make finding -3/-6/etc points on these traces easier to see(atleast the 'main fullrange trace'). Since we all calibrate our systems, the difference between the 'soundcard loop trace' and 'main fullrange trace' is irrelevant but it will still be interesting to see variations.


I _think_ option 2 is better for the reasons you cite (being able to easily see the cutoff frequencies).



> What would be handy is if REW was able to set an offset automatically based on a 'zero set' trace, like running a trace and then tell REW to 'set this one to zero' and all the rest will automatically be set with the same offset


Good idea.



> Any other ideas?


Not at the moment. I should be measuring the AudioControl Maestro M2 sometime within the next week or so. I'll see if anything comes to me as I go through the process again, and I'll stick to the stuff we've agreed to so far.

Thanks.


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## brucek (Apr 11, 2006)

> What would be handy is if REW was able to set an offset automatically based on a 'zero set' trace, like running a trace and then tell REW to 'set this one to zero' and all the rest will automatically be set with the same offset


When I do this type of measurement I first set up the Check Levels routine and then execute the Calibrate SPL meter routine to 75.0 dB and then execute the Set Target Level routine on the looped cable itself.

If it returns (for example) +75.6dB as the target, then I do a measurement with the cable to ensure the flat line at +75.6dB. Then I simply enter -75.6 into the Trace Offset feature and click the Add Offset to Data feature and then Save the measurement. This creates the cable reference and the measurement at 0dB for future reference and comparison.

Then I insert the UUT (unit under test) and do a measurement (with multiple sweeps x8 to obtain the best S/N ratio) and go through the calibrate routine as described above to get the zero reference and then save.

Every time I change an end frequency, cutoff or pink noise reference, etc, I redo the Check Levels and all the other items mentioned above.

brucek


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## Hakka (Sep 10, 2006)

MACCA350 said:


> Hakka, did you set your global channel trim levels, or just the mode you were in at the time? maybe the particular mode you were in for testing had different trim levels for the sub, because that is a pretty big boost on the sub channel. It seems that you may have a tone control active in the top end for stereo mode, since the direct mode trace was flat.
> 
> cheers



Stereo mode had the sub channel at 0db, pure direct was set at +0.5db, all other trims were the same for all tests. I will check the tone controls but I'm pretty sure they are set flat and tone bypass is engaged.

I have a Y cable coming out of my sub preout into the soundcard but REW should only use one channel so I doubt that has anything to do with the boost.

Hakka.


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## Otto (May 18, 2006)

I got up very early this morning and measured the AudioControl Maestro M2 pre/pro. I borrowed this pre/pro from another Shack member to see if it could work in my system. The Maestro M2 has a good sound, but I found that its feature set is lacking, especially on Room 2 controls, so I won't be able seriously consider it for permanent use in my system.

Anyway, here are some quick measurements. It seems that they are using a 4th order crossover for both mains and sub.


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## MACCA350 (Apr 25, 2006)

Otto, did you work out why there is a 6db boost on the sub channel when compared to the mains?

cheers


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## Hakka (Sep 10, 2006)

If anyone has a 380x sereis denon I would be interested to see the results of a stereo mode full range measurement.

I double checked all tone controls and they are not engaged.

Hakka.


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## brucek (Apr 11, 2006)

> Otto, did you work out why there is a 6db boost on the sub channel when compared to the mains?


I would think that the check-levels etc has to be reset every time a new measurement is taken to account for the different response to the band limited pink noise used. 

Barring that, I see little value in being concerned about the differing levels. Is it not the response that we're really concerned about here? Why not normalize everything to 0dB.

brucek


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## MACCA350 (Apr 25, 2006)

Hakka said:


> If anyone has a 380x sereis denon I would be interested to see the results of a stereo mode full range measurement.
> 
> I double checked all tone controls and they are not engaged.
> 
> Hakka.


For the Denon 3805, See post no.2
The red line is the 'stereo mode full range measurement'

cheers


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## Otto (May 18, 2006)

brucek said:


> I would think that the check-levels etc has to be reset every time a new measurement is taken to account for the different response to the band limited pink noise used.


Really? I'd think that it wouldn't matter much what the target was, or the calibration with respect to an SPL meter. All REW is doing at this point is sending a frquency sweep to a device and reading the results. There's no clipping or other malfunction occurring that I know of.



> Barring that, I see little value in being concerned about the differing levels. Is it not the response that we're really concerned about here? Why not normalize everything to 0dB.


Agree.

I think that the 6 dB increase in the sub may be caused by the fact that I'm using both left and right inputs to the preamp. They are summed and filtered to create the sub signal. Since I measured only one of the L/R outputs, we're really missing half of the response of the mains. If both L and R were present, they would combine to be in line with the sub. The sub's output should be considered with respect to both left and right mains. MACCA, I think you have a similar thing going on in this post.

brucek is right in that it doesn't matter much, as we all normalize our signals by using an SPL meter anyway. One thing we are learning here -- there's definitely some liberty taken on the implementation of bass managment from manufacturer to manufacturer. I'm sure some of these differences would be audible. Also, it's obvious that implementations are sometimes just plain wrong. Interesting stuff...


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## MACCA350 (Apr 25, 2006)

brucek said:


> I would think that the check-levels etc has to be reset every time a new measurement is taken to account for the different response to the band limited pink noise used.
> 
> Barring that, I see little value in being concerned about the differing levels. Is it not the response that we're really concerned about here? Why not normalize everything to 0dB.
> 
> brucek


My original intention was to see how the receiver modifies the input signal.

It was decided to use the main fullrange signal as the zero point(whereas Otto used the soundcard loop as zero piont:scratch: BTW did you have a 20kHz graph for the Maestro:bigsmile: ).

If you look at post no.2 the only difference between the main fullrange(Red) and main with crossover(green) was setting the mains to small to engage the crossover, so the difference in level is a byproduct of the 3805's processing.

I understand that this is not critical information because in the end we calibrate the system within a room, but it was discussed earlier and we thought it was interesting enough to keep in in the graph, but all this is still up for debate.

It has brought up a question of why is there a difference between receivers level, compare the AudioControl Maestro M2 pre/pro(#33) to the Denon 3805(#2) and the Denon 3803(#25)

But if everyone wants all traces normalized then the only difference we'll see is bass roll-off and crossover operation.

But maybe we can set the 'main's crossover' trace at 0db, to make the identification of crossover order bass roll-off easier(since this is the most common setup mode)

cheers


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## brucek (Apr 11, 2006)

> I'd think that it wouldn't matter much what the target was, or the calibration with respect to an SPL meter. All REW is doing at this point is sending a frquency sweep to a device and reading the results. There's no clipping or other malfunction occurring that I know of.


But if I send a band limited pink noise signal for a subwoofer (with low cut of 30Hz and high cut of 80Hz) and then set up the levels at the bass managed sub out, the level at the sub out and mains out will not be the same. The mains pink noise has a different set of cuts and would require a new level check.

Either way, I would just normalize everything to 0dB so the response is revealed...

brucek


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## MACCA350 (Apr 25, 2006)

Otto said:


> Really? I'd think that it wouldn't matter much what the target was, or the calibration with respect to an SPL meter. All REW is doing at this point is sending a frquency sweep to a device and reading the results. There's no clipping or other malfunction occurring that I know of.
> 
> 
> 
> ...


That was Hakka's graph, and I think you are right.

Once a well defined method is set out these setup differences will not be there.



> brucek is right in that it doesn't matter much, as we all normalize our signals by using an SPL meter anyway. One thing we are learning here -- there's definitely some liberty taken on the implementation of bass managment from manufacturer to manufacturer. I'm sure some of these differences would be audible. Also, it's obvious that implementations are sometimes just plain wrong. Interesting stuff...


It is something to consider, I took to long to post my previous post, and didn't see this one, where I mention this: 'But maybe we can set the 'main's crossover' trace at 0db, to make the identification of crossover order bass roll-off easier(since this is the most common setup mode)'.

I'm not sure which way to go on normalizing, but we'll see what everyone thinks.

cheers


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## MACCA350 (Apr 25, 2006)

brucek said:


> But if I send a band limited pink noise signal for a subwoofer (with low cut of 30Hz and high cut of 80Hz) and then set up the levels at the bass managed sub out, the level at the sub out and mains out will not be the same. The mains pink noise has a different set of cuts and would require a new level check.
> 
> Either way, I would just normalize everything to 0dB so the response is revealed...
> 
> brucek


But thats why all levels and trims etc, in the AVR are set to 0db, so we have a level field.

cheers


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## Otto (May 18, 2006)

MACCA350 said:


> But thats why all levels and trims etc, in the AVR are set to 0db, so we have a level field.
> 
> cheers


Correct. I didn't try to set up any levels in the "channel trims" in the pre/pro. All levels were at 0, and I didn't really do anything with the band limited pink noise. Just in and out, through the pre/pro.

I don't think I swept up to 20 KHz. I was having some trouble with my soundcard cal at high frequencies. I'm not sure if there's actually a problem, or if it's just some noise that I've been hearing in the analog outs of my cheap soundcard. I'm considering going to a better sound card.

Sorry if I didn't set up correctly, I didn't go back and read the setup stuff. I like MACCA's idea of seeing what the actual contribution of the system is, though, so I think it's a good idea to set one reference and then adjust all traces by that offset only. Obviously, I think there's also benefit in normalizing to 0 db, as I did just that for my second plot yesterday. We appear to still be going through some set up trials, and that's OK. I think we'll continue through this for a while, and then perhaps MACCA can edit his initial post to summarize all the set up stuff (there's some already there, but I think it doesn't include some of the stuff we talked about after that first post).

That's it for now. Have a good day.


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## MakeFlat (Mar 30, 2007)

Retracted by me.


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## Otto (May 18, 2006)

I think if I just leave all channel trims at +/- 0 dB and don't adjust anything else, it should make no difference. All I'm doing is just sending a sweep signal in and measuring what comes out the other end of the receiver/pre/pro. There's no BFD in the circuit, and there's no house curve applied.


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## MakeFlat (Mar 30, 2007)

Otto said:


> I think if I just leave all channel trims at +/- 0 dB and don't adjust anything else, it should make no difference. All I'm doing is just sending a sweep signal in and measuring what comes out the other end of the receiver/pre/pro. There's no BFD in the circuit, and there's no house curve applied.


Okay - cancel my previous post. Now I see that FR=0dB, FL=0dB and SW=0dB on the trims. It is now looking like the 6dB boost on the sub channel is there by design. So now I can see why normalizing the sub level is the right thing to do.


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## brucek (Apr 11, 2006)

> But thats why all levels and trims etc, in the AVR are set to 0db, so we have a level field.


I don't think I made my point very well. 

The sub cal pink noise used in the Check Levels routine is band limited from 30Hz to 80Hz. If I set up my levels out of the sub port with that as the source to 0dB, then I measure from 2Hz to 300Hz for example, the level will drop. I suppose if you used Full Range pink noise to set the levels it would work, but that isn't the case. My feeling is to reset all the measurements to a 0db level with the offset feature and then compare responses.

brucek


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## Otto (May 18, 2006)

I think I understand what you're saying... 

I've never used pink noise for any set up when measuring the pre/pros directly. I just set everything to 0 dB and let the signal go straight through the system.


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## brucek (Apr 11, 2006)

But how do you set the levels. Don't you use the Check Levels routine?


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## Otto (May 18, 2006)

I didn't set the levels per se. I left REW as it was set up previously, ran the loopback, saw reasonable results and there was no clipping. From there, I just sent the signal through the pre/pro. The response of the pre/pro shouldn't change based on the levels I set in REW. Am I missing something?


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## brucek (Apr 11, 2006)

> Am I missing something?


When you connect a cable or a cable with a device in the loop, it's no different than taking a measurement with a speaker and microphone as far as setting up REW. It doesn't know the difference and expects a proper setup to get the best response and lowest noise floor etc.

You need to do a Check Levels routine, and then a Calibrate SPL routine to 75dB and then Set Target Level routine and then a Measure routine. The only difference is that when you do the Check level routine you may need to adjust the receiver volume along with the REW input level to get the right level.

brucek


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## Otto (May 18, 2006)

I don't recall specifically doing any set level stuff, perhaps it was close enough. Loopback measured flat, headroom existed. Set preamp to unity gain and let it go...


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## fireanimal (Dec 21, 2006)

Here are the results from my new Sherbourne PT-7010A

AT 60HZ CROSSOVER










AT 80HZ CROSSOVER










8-CHANNEL DIRECT MODE










The purple line is the Sub In/Out
The red line is the Front Left In/Out
The green line is the same input as previous graphs for comparison.

I really like the Sub extension in the 8-Channel mode. It seems to be boosting the Sub channel in all modes.


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## Otto (May 18, 2006)

Hi fireanimal,

Some of those signals look kinda lumpy. Do you get that same phenomenon when using just a loopback cable?


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## MACCA350 (Apr 25, 2006)

Otto said:


> Hi fireanimal,
> 
> Some of those signals look kinda lumpy. Do you get that same phenomenon when using just a loopback cable?


This is a good example of where a loopback trace in the graph is a good idea, because we'd already know the answer.

cheers


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## fireanimal (Dec 21, 2006)

I did, sorry I should of included the feedback line on the graph as well. I don't know where that signal is coming from, could be my hundred year old laptop.:duh:


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## MACCA350 (Apr 25, 2006)

fireanimal said:


> I did, sorry I should of included the feedback line on the graph as well. I don't know where that signal is coming from, could be my hundred year old laptop.:duh:


did you run create a soundcard calibration file with the feedback loop in place?

cheers


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## fireanimal (Dec 21, 2006)

I had some input/output volume problems causing distortion in the signal path. Here are some new graphs. The red line in all the graphs is the reference line from the feedback loop.

Crossed at 60HZ










8-Channel Direct










Here is the graph out to 22,000. Notice how the Channel Input in Small and Large both taper at the end, as well as the direct sub input. Do you think the Channel Input tapering would cause many problems.










I haven't hooked the preamp up to listen to it yet, just picked it up used, for less than the Outlaw 990 I was looking at.
Does it seem strange that the sub output is hotter than the rest. Also notice on the sub the peaks/valleys until it hits 10hz...

Do you guys notice anything else strange, or does everything look pretty good.

Thanks.


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## Hakka (Sep 10, 2006)

I am a moron! :coocoo:

I thought the tone controls on my 3803 were global but it seems they can be set independently for each surround mode. Stereo mode is now flat.

Hakka. :coocoo:


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## brucek (Apr 11, 2006)

> Does it seem strange that the sub output is hotter than the rest.


Did you reset the levels with each measure? You should......


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## Otto (May 18, 2006)

brucek said:


> Did you reset the levels with each measure? You should......


Back to this...

I'll give you this much -- your levels should indeed be reasonable. They shouldn't be so low that your measurements are in the noise, and they shouldn't be so hot as to cause clipping. But I don't really see how resetting levels between measurements does anything but add confusion to the plots by adding another variable between measurements. I think it's interesting to see the actual change to the signal that's imparted by the processor.

If I have a loopback signal that measures flat from 10 Hz to 20 kHz, and I send that through a processor that has a presumed unity gain, and everything is "ideal" and "perfect", I expect to see that same exact signal at the output of the processor. Now, if I change the input used on this "ideal" preamp to a different input (say "aux" instead of "CD"), and I remeasure, I expect to again see the same measurement as 1) the loopback cable and 2) the input I'd used previously.

Of course, no preamp is "ideal" or "perfect", so any difference between the loopback measurement and the measured output of the processor has, by definition, been caused by the processor.

I find it very relevant to compare whatever inputs and outputs of the processor with _precisely_ the same input signal. We can then see differences between processing modes and such. Yes, I realize that some of this isn't important, and we all normalize our signals with respect to each other.



brucek said:


> But if I send a band limited pink noise signal for a subwoofer (with low cut of 30Hz and high cut of 80Hz) and then set up the levels at the bass managed sub out, the level at the sub out and mains out will not be the same. The mains pink noise has a different set of cuts and would require a new level check.


I would submit that if you set up your levels with the sub out, with band-limited pink noise, your levels will be set to X, whatever that may be. If you do a loopback measurement from 10 Hz to 20 kHz, you should measure a flat line (assuming your soundcard cal is in place, etc.). 

Furthermore, if you set your levels using full-band pink noise from 20 to 20 kHz, your levels will be set to Y, whatever that may be. Again, measure the loopback, and you should get a flat line. You absolute measurement between the two will be different, but that doesn't matter (again, unless your levels are noisy or clipping).

Now, if I send either of those two signals through the preamp (and there's no clipping), the output measured is the response of the preamp. I cannot see why the amplitude of either the bass-managed sub signal will be greater than that of the bass-managed mains signal.



brucek said:


> The sub cal pink noise used in the Check Levels routine is band limited from 30Hz to 80Hz. If I set up my levels out of the sub port with that as the source to 0dB, then I measure from 2Hz to 300Hz for example, the level will drop.


How will it look if you do a loopback test? Flat?

Now, before you pop off a terse one-liner to all of this, re-read my post, and try to understand what I'm saying. I know you gave up on this discussion with me before, but it's back, and I think it's adding confusion to this thread (for me, if no one else). If I'm really making a mistake in my measurements, I'll be happy to hear about it. But, so far, the thing that makes the most sense to me is to send a constant signal through the system, and measure the output. If I know what I have going in (the loopback test from 10 to 20 kHz), and I can see what I measure coming out the other end, then the difference was created by the processor, not the level set up.

Anyone else have any opinion on this?

Also, I offered this response as a reasonable explanation to some of the difference in the sub measurements (the sub being 6 dB hot). Is this at all plausible in your opinion?



Otto said:


> I think that the 6 dB increase in the sub may be caused by the fact that I'm using both left and right inputs to the preamp. They are summed and filtered to create the sub signal. Since I measured only one of the L/R outputs, we're really missing half of the response of the mains. If both L and R were present, they would combine to be in line with the sub. The sub's output should be considered with respect to both left and right mains. MACCA, I think you have a similar thing going on in this post.


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## MACCA350 (Apr 25, 2006)

fireanimal said:


> I had some input/output volume problems causing distortion in the signal path.


What was the problem, too high or too low? Did you set the Sherbourne's main volume(and all trims) to 0db for any of tests? 



Otto said:


> How will it look if you do a loopback test? Flat?
> 
> Now, before you pop off a terse one-liner to all of this, re-read my post, and try to understand what I'm saying. I know you gave up on this discussion with me before, but it's back, and I think it's adding confusion to this thread (for me, if no one else). If I'm really making a mistake in my measurements, I'll be happy to hear about it. But, so far, the thing that makes the most sense to me is to send a constant signal through the system, and measure the output. If I know what I have going in (the loopback test from 10 to 20 kHz), and I can see what I measure coming out the other end, then the difference was created by the processor, not the level set up.
> 
> Anyone else have any opinion on this?


Yes, I agree. checking levels between traces will invalidate any differences in processor output levels, again this is only of importance if we decide not to normalize all the traces



> Also, I offered this response as a reasonable explanation to some of the difference in the sub measurements (the sub being 6 dB hot). Is this at all plausible in your opinion?
> 
> "Otto wrote:
> I think that the 6 dB increase in the sub may be caused by the fact that I'm using both left and right inputs to the preamp. They are summed and filtered to create the sub signal. Since I measured only one of the L/R outputs, we're really missing half of the response of the mains. If both L and R were present, they would combine to be in line with the sub. The sub's output should be considered with respect to both left and right mains. "Hakka", I think you have a similar thing going on in this post."


It seems possible, I didn't notice that in the hookup graph, I used just one channel all the way through the chain, I'll change the OP hookup graph to show this. I'll have a play later and see if the different hookup makes a similar difference.

cheers


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## MakeFlat (Mar 30, 2007)

Pink noise contains an equal amount of energy in each octave band. Therefore, using pink noise to calibrate level assures that the calibrated level is weighted equally for the entire range of the test. Thus if you use sub pink noise to calibrate, it will mean that the sub's entire frequency range is taken into account. As you can see, it is significant in that the calibrated level is based on peaks and valleys and anything in between. In contrast, if you were to calibrate based on an x hz tone, the resulting level could be either too high or too low overall. We can now see the advantage of using pink noise to test. 

Now if we were to use the sub cal pink noise to calibrate the level of full range measurement, it would mean that the mid range and above were not given any weight in the determination of level. If the signals over the entire frequency range stay mostly constant, except at the extreme ends, then whatever level we use to calibrate would be good enough, as long as we use the same calibration level throughout the entire set of tests.

The significance of the test level to which we calibrate REW, is that the transmitting and receiving ends are set to optimum levels for signal detection and measurement. It assures that the test system operates at a level that is within the normal operating bounds of the devices being tested.


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## Otto (May 18, 2006)

Makeflat said:


> If the signals over the entire frequency range stay mostly constant, except at the extreme ends, then whatever level we use to calibrate would be good enough, as long as we use the same calibration level throughout the entire set of tests.


Exactly. I totally agree that there's value in setting up your levels correctly -- you don't want your measurements in the noise or clipping. But once you've got that across whatever bandwidth of interest, you don't need to recal it between measurements.


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## fireanimal (Dec 21, 2006)

Quote:
fireanimal wrote: 
I had some input/output volume problems causing distortion in the signal path. 

What was the problem, too high or too low? Did you set the Sherbourne's main volume(and all trims) to 0db for any of tests?


The Windows mixer output level was very low, I reset it and used the volume controls on the USB SB Live to set the levels.

All measurments were done with no alteration to the levels. All the trims on the sherbourne were at 0, but I had to set the volume at -15.0, to be at an even level with the -20.0 signal that I was using through REW. If I set the volume at 0, then the input signal would clip. I even did a master reset on the Sherbourne, and everthing remained the same as before.


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## brucek (Apr 11, 2006)

> Now, before you pop off a terse one-liner to all of this, re-read my post, and try to understand what I'm saying.


Yeah, I'm sorry if a lot of my answers are short, it's just that I don't have a lot of time.

It's hard to add to MakeFlat's excellent response. REW has only two levels of level checking pink noise that it uses. They are either a (low and high cut of 30Hz to 80Hz) or (low and high cut of 500Hz and 2000Hz) respectively . The assumption being that you are checking either the mains or the sub channel and would use the appropriate band limited signal that matched. 

As a double check that you aren't mis-using these Check Level pink noise limits, the Measurement routines Check Level button introduces pink noise using a fixed low cut of 20Hz with a variable high cut of the End Frequency chosen in the Measurement panel itself. 

This final double check can catch a lot of mis-set levels and errors with reference to the actual measured bandwidth and so I will acquiesce my point of setting the levels irregardless of the UUT bandwidth. Since we have agreed to normalize the end result, who am I to argue...... 

brucek


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