# First measurements I dare to post!



## entrecour (Jun 13, 2007)

I have been playing with REW on and off for quite a while now and I have finally got to the stage where I dare post something. :bigsmile: I am sure I could get some valueable pointers as to wether I am heading in the right direction. 

My setup is a HK AVR330, BDF2496, SVS SB12-Plus, and 5 x PMC DB1+ in a standard 5.1 HT setup based around a HTPC.

I am using an ecm8000 microphone for measurements and feeding the test tones directly into the analogue input of the AVR330 set to stereo, surround off, no DSP.

I have attached the measurements including the waterfall chart. All comments appreciated. I don't understand why the waterfall doesn't extend past 100Hz.

There were several stages where I wasn't quite sure if I was doing the right thing:- :scratchhead:

First of all I played a film with a lot of LFE (U571) to increase the signal level to the BFD from the AVR330 so the BFD signal meter was all green and occasionally orange. That mean increasing the sub channel in output in the AVR330 to max (+10) when in Dolby Digital surround mode. 

However wwith the soundcard output was connected via the amp (stereo, surround off) I had to decrease the subwoofer channel output to stop clipping in the BFD. (With the AVR one has to set speaker levels for each surround mode.)

After taking the measurements, setting the target level, and programming the filters generated I reset the subwoofer channel output to +10 and rebalanced the speakers, using the subwoofer gain control when it came to the sub.

So how did I do?


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## brucek (Apr 11, 2006)

> All comments appreciated. I don't understand why the waterfall doesn't extend past 100Hz.


Because you have the graph axis on it set to LIN (linear) and not LOG (logarithmic). The FreqAxis icon button in the top right corner of REW will change that. Also use the same scale for the waterfall as your frequency response charts for easier comparisons. (45dB-105dB and 15Hz-200Hz)



> So how did I do?


The filter number 3 is really the only one you probably require. Once you're past the crossover area, the filters obviously become less effective since the sub level is dropping and the mains are taking over. The best bet is to equalize the peaks before the crossover (or around 100Hz) and then add the mains and remeasure and see if you really require filters up above 100Hz.

A tip. There is a filter sort button (from low to high) on the filter panel. Very convenient to have your filters go from 1,2,3,4,5 instead of 3,2,1,4,6,5 

brucek


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## entrecour (Jun 13, 2007)

Thanks brucek,

Here are the new graphs, the corrected curve using just filter 3, and the waterfall plotted on a log freq axis. The waterfall looks a lot uglier - but I am not sure how to interpret it. Any pointers?

Thanks.


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## brucek (Apr 11, 2006)

Basically you can consider it a frequency response graph with the added axis of time.

Drag the waterfall slider back to 1 slice. Look familiar? It's the response graph. Now move to 2 slices. You're now looking 10msec later in time at the level of the signal (if window set to 300msec). By the time you get to 30 slices, you're looking at 300msec later. If any frequency is still at an audible level you'll still be hearing it - considered perhaps not good..

brucek


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## thewire (Jun 28, 2007)

That looks pretty good IMO. Perhaps it could be better but still good.


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## entrecour (Jun 13, 2007)

Thanks brucek and thewire for your comments.

This time around I redid all the measurments (so they may not look identical).

I set one filter at 58Hz and then redit the measurement with main speakers and sub (I balanced both to approx 75dB).

I set the target level by setting speaker type to Full Range. 

With my mains I still have a peak at 58Hz so I suppose this is a room effect, or?

Then I have some some sharp dips at 76Hz, 100Hz, and 177Hz. Are these anything to worry about?

Should I filter the peak at 130 Hz?

I have attached the waterfall - to your expert eyes does it indicate problems that I should look to remedy?


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## entrecour (Jun 13, 2007)

Edit: I can't upload the waterfall as at 222K it exceeds the 200K jpeg limit so I have zipped it.


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## brucek (Apr 11, 2006)

> I set the target level by setting speaker type to Full Range.


No, when you add the mains and are trying to examine the crossover, you simply take the measure from 0-200Hz as if it was the sub only, so you use the sub pink noise and type as sub.



> With my mains I still have a peak at 58Hz so I suppose this is a room effect, or?


Well, almost everything is a room effect from modal resonances to reflections etc.....



> Then I have some some sharp dips at 76Hz, 100Hz, and 177Hz. Are these anything to worry about?


Sharp dips are usually left alone (and are generally inaudible). If they're around the crossover you can try and eliminate them with the phase control (when both sub+mains are playing).



> Should I filter the peak at 130 Hz?


No.

Anyway, you have a classic example of how correcting a single peak can really help out. Look at your peak at 58Hz. You have correctly set the level matching between the sub and mains as a result of the peak at 58Hz. It overpowers and results in one-note bass. You can't turn up the sub or it will sound overpowering.

But, if you remove that peak, then you are able to wholesale turn up your bass level, and as a result, the frequencies between 20Hz and 45Hz that were previously hidden, are now revealed and you feel like you just got a new sub......

brucek


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## entrecour (Jun 13, 2007)

Ok, I'll do another measurment with sub pink noise,etc. 

What's the best way for me to remove that peak now? As I have one filter set already at 58Hz if I automatically add another filter at 58Hz will the BFD sum them correctly?


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## brucek (Apr 11, 2006)

> What's the best way for me to remove that peak now? As I have one filter set already at 58Hz if I automatically add another filter at 58Hz will the BFD sum them correctly?


I'm confused, are you saying that the sub+mains graph was taken with a BFD filter in place for the sub and that the peak was gone completely with the sub only and is back just as strong when you took the sub+mains measure? Let me see the sub only with the filter in place (you only show the probable outcome in the graph you have posted).

brucek


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## entrecour (Jun 13, 2007)

Yes, that's what's happened unless I screwed up somewhere!

Here is the sub only plus the one filter


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## brucek (Apr 11, 2006)

Interesting.

What is your crossover frequency?

Can you take a mains only measure from 0-200Hz and post it? (just shut off the sub).

brucek


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## entrecour (Jun 13, 2007)

I normally use small but with this amp when in stereo, no surround, no dsp mode the speakers can only be set to large. However there is a X-Over mode which is set at 80Hz and has the exactly the same measurment signature.

That peak is at exactly the same freq as for the subwoofer i.e. 58Hz.


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## entrecour (Jun 13, 2007)

I redid the mains only measurment with the amp set to stereo, *DSP ON, SMALL*

This produced a very different result... still a peak at 58Hz though


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## brucek (Apr 11, 2006)

OK, you've definitely got a large resonant peak at 58Hz. You may have to over compensate your equalizing a bit with the subs 58Hz filter. Hopefully you can lower it somewhat below the target and then the sum of that and the mains peak will be a better result. I would also increase the level of the sub somewhat in relation to the mains so that the level around the 30Hz region was at least equal to the level of the mains. To get a better feel for the overall level of the mains, take a measure of the mains only from 100Hz-1000Hz (use full range type here) and then turn on smoothing ~1/3 octave and see what the overall level is and then bring the sub up to that level at least. I think your sub is a bit low.

brucek


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## entrecour (Jun 13, 2007)

Thanks for the pointers.

Here are the new measurments 
The first measurment is sub only, no filters.

The 2nd measurement is sub & mains with 1 filter at 58Hz. I ended up using the auto filter generated by REQ (58.35Hz, Gain -12, BW 0.25). Main speakers set to SMALL.


Do you think the sub is high enough now. I have clipping in the measurments if I turn it much higher.

Thanks! :wave:


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## brucek (Apr 11, 2006)

Yeah, that's starting to look a lot better. I might add a filter at ~45Hz and turn the sub up a touch more.

BTW, you won't have clipping in the measurements if you simply go through the Check Levels and Calibrate SPL routine of resetting REW back to 75dB.

brucek


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## entrecour (Jun 13, 2007)

Looking much better me thinks


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## entrecour (Jun 13, 2007)

And here is the corresponding waterfall chart. I increased the Time Range to 900ms, and kept the Window to 300ms. Looks like there are some strange things going on around 25Hz! 

Should I be looking at room treatments (perhaps to address the slight dip at 36Hz) or are these kind of delays normal?


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## brucek (Apr 11, 2006)

> Looking much better me thinks


You're still a bit high at ~50Hz, but yeah, it looks fine now. Just drop the filter(s) in that area about a couple dB.

No amount of treatment will affect as low as a 36Hz dip. That has to be attacked by positioning usually and then sometimes you have to live with it if positioning isn't possible.



> And here is the corresponding waterfall chart. I increased the Time Range to 900ms, and kept the Window to 300ms. Looks like there are some strange things going on around 25Hz!


Well, since the ringing at 25Hz appears to start at a much lower level than your signal, I would suggest it's room noise. This is where it doesn't hurt to use the Spectrum Analyser without a signal and look at your rooms background problems. I can identify my furnace and fridge etc.

I wouldn't look at a waterfall much more than a 600ms time axis. 

Anyway, you haven't said if it sounds any better?

brucek


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## entrecour (Jun 13, 2007)

> Anyway, you haven't said if it sounds any better?


Ha! Good point! It sound a lot better, principally (as you mentioned earlier) it has allowed me to increase the sub level. There may be more room for increase there still. :T More listening required before I can give a final verdict.

However I have noticed some stubborn mechanical vibrations in the room as I am doing the sweeps (one long side wall is virtually all windows with venetian blinds) and I think a wall or the floor is vibrating somewhere. So probably some tweaking to do still - that's what I was hoping the waterfall might pinpoint.

I also have to check what the background room noises might be.


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## brucek (Apr 11, 2006)

> So probably some tweaking to do still - that's what I was hoping the waterfall might pinpoint.


The best way to find those pesky vibrations is to use REW's sine-wave signal generator and checkbox the 'frequency tracks cursor' option and move the cursor slowly around the low frequency area on the REW screen and find the noises - kinda fun. 

Pictures are the worst I find. I put little pads in the bottom corners of the picture where it contacts the wall. Fireplace doors are horribly noisy also.

Hehe, this is what happens when you reveal the super low frequencies. It adds all sorts of vibration problems. It's like when you first get a high end system and then you hear all the horrible recordings you own.



> I also have to check what the background room noises might be.


Yeah, below is a quick spectrum snapshot I took one day to see my furnace noise. I also picked up my PC soundcards 60 cycle hum and its harmonics at 120Hz, 180Hz etc, and also interesting to see my RPTV CRT horizontal oscillator at 15.750KHz in the room. The furnace was the noisy one. Turn off the furnace and it goes away. I was getting a weird ringing out with my waterfalls at about 30Hz. I thought it was my system and the room.


*Spectrum Analyser - no sine wave stimulus*








brucek


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## entrecour (Jun 13, 2007)

Those are great tips. :thankyou:

What settings did you use for the Spectrum Analysis, anything different to the defaults?


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## brucek (Apr 11, 2006)

> What settings did you use for the Spectrum Analysis, anything different to the defaults?


See this post...

brucek


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## entrecour (Jun 13, 2007)

Thanks, that was interesting reading.

I also noticed this 



> The RTA also allows you to change filters and watch the RTA screen for the real time changes. It's also really useful to adjust phase on a subwoofer for the best crossover response. You simply watch the RTA screen as you dial the phase control. A lot better than taking a bunch of measures to accomplish this task.


I only wished I had read this a few days ago when I went through 8 seperate phase measurements! 

I found it very difficult to choose the "best" phase. What should you aim for - the one that has the highest overall volume or the one that is the smoothest?


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## brucek (Apr 11, 2006)

> What should you aim for - the one that has the highest overall volume or the one that is the smoothest?


You attempt to get a smooth transition from the sub to the mains, that's really it.

The RTA will allow you to do it in real time - quite easy when you use the RTA. Be sure to use Periodic Pink noise with the RTA.

brucek


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## entrecour (Jun 13, 2007)

Here is my background noise spectrum measurment (both dB and dB FS). I used an FFT length of 131072, averaging of 32, and a Hann window. I had turned off as much electrical equipment in the room as I could and I moved the measuring PC into an adjacent room. Still a lot of 50Hz noise. My measurment is strangely linear compared to yours. 

By the way does data < 10-15 Hz have any meaning here considering the freq range of the microphone etc?


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## brucek (Apr 11, 2006)

> My measurment is strangely linear compared to yours


Your averaging of 32 does that, I wanted a more dynamic look.....

brucek


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## JohnM (Apr 11, 2006)

That is somewhat unusual. What preamp are you using with the ECM8000?


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## entrecour (Jun 13, 2007)

The Xenyx 802. The sloping line was similar at other averaging settings.


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## brucek (Apr 11, 2006)

> The Xenyx 802. The sloping line was similar at other averaging settings.


Yeah, I was talking about the amount of hash in the signal when I referenced the averaging - I thought that was what you were talking about. 

With respect to your upwardly sloped line (if that's what you mean by linear), then I suspect you have the Xenyx 802's LO EQ dial of the mixer not in its mid detent 'off' position. I can easily recreate what you have there by dialing my LO EQ dial of the 802 clockwise....

brucek


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## entrecour (Jun 13, 2007)

I rechecked my settings and all of the EQ dials were at 0 "off". 

Anyway I redid the measuements, this time using 65536 FFT length, 8 Avg, and Hann. Basically IMO there doesn't appear to be a significant difference between this and my previous measurments.


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## brucek (Apr 11, 2006)

> I rechecked my settings and all of the EQ dials were at 0 "off".


OK, measure your 802 then and see if it's flat.

Here's how. Remove all the cables from line-out and line-in. 

Plug the soundcard line-out into the line-in of the 802 and the line-out of the 802 into the soundcard line-in.

Do the REW Check Levels (you have to play a bit with the 802 level, but no big deal). Do the Calibrate SPL routine and Measure. Be sure to have your soundcard cal file loaded and the Microphone cal file cleared.

You should get a fairly flat line. If not, there's a problem in the 802 (the LO EQ detent may not be right). You might have to find the spot in the mixers EQ dial(s) that produce a flat line.

brucek


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## entrecour (Jun 13, 2007)

Oh great! And typically I don't have a cable with the right plug for the line in connection on he preamp. :mooooh:


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## brucek (Apr 11, 2006)

Just go to Radio Shack and get one of these, and plug an RCA into it..










You must be using one already for the line-out of the 802?

brucek


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## Guest (Jun 18, 2008)

Basically you can consider it a frequency response graph with the added axis of time:rofl2:


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## entrecour (Jun 13, 2007)

brucek, I use cables with the correct plugs so I don't use RCA to 1/4" adapter. However I dug out a 3.5mm to 1/4" adapter I had for headphones way back and together with a 3.5mm to 3.5mm cable that did the trick.

I noticed there is a slight no linearity in the preamp when the TRIM is not at 12 o'clock especialy at low frequencies. In all my prevous measurements I had the TRIM set at +60 (fully right) in order to get the right levels on the preamp and in check levels.









However even if I set the TRIM to the 12 o'clock position and remeasure the room noise spectrum I still have the same downward sloping charecteristic.


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## brucek (Apr 11, 2006)

I am guessing that when you say TRIM, you mean the Main Mix dial? I find the best gain structure for this device is as shown below.

There are several dials that determine the voltage level through the device.

Directly below the mic input is the mic GAIN. It is set to 12 o'clock.

Then there are the three EQ equalizer dials, which are all set to detent.

Then there is the FX dial, which is set to fully CCW.

Then there is PAN, which is set to fully CW, since we are using the number one (right) channel exclusively.

Then there is right channel LEVEL control which is set to 12 o'clock.

Then the MAIN MIX output dial is used for the variable level you want out depending on the sensitivity of your soundcard line-in.








Anyway, it doesn't look like the 802 is the source of the rising level in your measurements. I think it deserves to be found though.

Just as a matter of course, I would include the 802 in the loop (just as you've done) and do a new soundcard calibrate routine and store the file as the one to use when the 802 is being used. It will make your measurements more accurate.

It would be interesting to also do a loop test measurement of your receiver (if it has preamp outputs) to see if it may have an effect turned on that is causing the level to rise as the frequency drops. This eliminates the room and the speakers from affecting the measure.

brucek


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## entrecour (Jun 13, 2007)

> I am guessing that when you say TRIM, you mean the Main Mix dial?


I mean the first dial that is below the MIC 1 input at the top left. It is marked TRIM on my unit. You have a yellow arrow over it in your diagram. 

I had seen you set-up instructions previously in another post you had made and I'd found them very useful. However when it came to the checking the levels I found that with my M-Audio soundcard setup that I needed to change the TRIM to get a reasonable input signal in REW and a good input signal as indicated by the signal strength leds on the 802. If I move the TRIM back to zero I need to crank up both the LEVEL 1 dial and MAIN MIX dials.

Just as a comment in my M-Audio settings I don't use the MIXER - I set the output channel to be "sw rtn".



> It would be interesting to also do a loop test measurement of your receiver (if it has preamp outputs) to see if it may have an effect turned on that is causing the level to rise as the frequency drops.


This statement has got me worried. When I do the background noise measurment I am not using the amp at all. I thought that was the point. Have I misunderstood something here? :scratch:


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## brucek (Apr 11, 2006)

> When I do the background noise measurment I am not using the amp at all


You should be setting everything up with REW and the mic and the amp and the Check levels, etc, etc, as if you were about to push the Measure button. But instead of pushing that button, go to the Spectrum tab and press the RED button to simply listen. Your amp may be outputting background noise as well as other generators such as fridges and air conditioners, etc.

Normally with the Spectrum feature we would set up the levels normally and then start the signal generator and select Periodic White Noise (White PN) and a Rectangular window and look at the spectrum. 

Or alternatively, we might want to see the harmonics generated with a Sine wave from the generator at specific frequencies for THD and THD+N results (using a Hann window, as the Sine wave is not periodic). 

Or you may just look at the noise spectrum without any stimulus. This will reveal any background noise in the room (again use a Hann Window).

brucek


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## entrecour (Jun 13, 2007)

OK I'll recalibrate with the preamp in the loop and redo the measurments over the weekend (if I get the time). Thanks for all the advice.


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## entrecour (Jun 13, 2007)

brucek,

I finally found some time to do further measuements today. However I am none the wiser if I am on the right track after my strange spectrum test results the other day. :wits-end:

I recalibrated my soundcard with the 802 preamp in the loop. This time I had the preamp TRIM set to the center position and all the other controls pretty much identical to your diagram.

Here are the results including the post calibration verification measurment...








These results are similar (in terms of curve shape) to those I had without the 802 in the loop.

You'll notice the large increase in soundcard signal dB with frequency. Is this normal? :scratch:

I took a full freqency measurment of my sub + mains (Amp DSP off and surround off). IMO the results point to a major problem somewhere..








I also noticed that if I use the Sine Wave generator (freq vs cursor) and check the real time dB vs the measured (graphed) result the dB's were completely different. The Sine Wave Genetator was showing approx 70dB or so at around 5KHz whereas the measured (graphed) result was approx 55dB. :scratch:

Any idea what's going on?


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## brucek (Apr 11, 2006)

> Is this normal?


No, it is not..............

Let's start over and find out what's wrong.

First, let me show you the difference between a graph of a soundcard cal file using a simple loopback cable and then one where I add the 802 mixer into it. It is as you would expect. The mixer has a response that drops off at the low end a bit, so it's reflected in the (green) graph line by a dropping off in the response at very low frequencies. Pay mind to the scaling I'm using. It's an expanded scale, so it shows that the addition of the mixer changes the response at 10Hz by about a dB. Not really a big deal. The mixer is basically flat, but it simply makes it more accurate to add the mixer in the file - why not.









In your case, something is amiss. So, let's find out what that is.

The very first thing I want you to do (an understand) is to *clear* the meter calibration file unless you're using the meter in a measurement. It affects all measurements. You *can't* have it loaded when doing loopback tests.

Be sure all other cables and mics are removed and don't have the 'Use left channel as Calibrate Reference' checked. We'll be using the right channel only. The left channel stays open and never used.

The first thing I would like you to do is connect a simple loopback from the line-out to line-in of your soundcard and do a soundcard cal file routine to create a cal file and save it. (have the soundcard cal file and meter cal files cleared before doing this).

Once complete, take a response measure of the looped back cable. It should be flat. Post a graph.

Next, when that is complete, insert the 802 in the loopback. Clear the soundcard cal file and execute the soundcard cal file routine and save the file. Then do a response measure of the 802 in the loop. It should be flat. Post a graph.

brucek


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## entrecour (Jun 13, 2007)

Here is the soundcard loopback, 48KHz sample rate


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## entrecour (Jun 13, 2007)

And this time with the response measurment, all looking good so far...


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## brucek (Apr 11, 2006)

Yep..............


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## entrecour (Jun 13, 2007)

And here is the same measurement with the 802 in the loop, but look at calibration curve for the soundcard (which this time includes the 802). It certainly doesn't seem to have the same marginal effect as yours.
I can't see any controls that are incorrectly set.


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## brucek (Apr 11, 2006)

> And here is the same measurement with the 802 in the loop


Yeah, and if you loaded the soundcard.cal file from the straight loopback and took a measure of the 802, you would find the response is not good (it will look like your rising graph above).

So, there is a problem with the 802.

Re-load your straight cable soundcard.cal file and take successive measures of the 802 in the loop and find out what dial is causing the trouble. I suggest its the EQ. Looks like its the Hi dial isn't zero at 12 o'clock detent.

Or you simply may have a fault.

Try that out and check back and tell me what happenend.... 

brucek


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## JohnM (Apr 11, 2006)

Even the soundcard loopback doesn't look quite right, it should be much smoother. After any of your loopback measurements change to the Scope tab and post an image of what that looks like. Here is a good soundcard loopback:


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## entrecour (Jun 13, 2007)

Here is my scope for the soundcard loopback calibration run (without the 802), sweep level-6dB -> 3.2dB headroom.










The input and output signal seemed to be out of synch suggesting to me there is a delay inserted somewhere. Here is the M-Audio setup panel, ASIO direct monitoring is disabled. I can't reduce the ASIO buffer size to less than 64 samples, in any case it doesn't seem to have any effect on this issue.









:help:


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## JohnM (Apr 11, 2006)

That looks fine. Ignore the apparent violation of causality with the response starting before the stimulus, its just a data plotting artefact with that version which is fixed in the dev build I used for my plot. Be careful with making buffer sizes too small, it can cause dropouts. How does the scope plot look with the 802 in the loop?


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## entrecour (Jun 13, 2007)

OK, I set buffer size back to 256 samples. 

Here are the measurments with the 802 in the loop (NO Calibration files).


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## entrecour (Jun 13, 2007)

Ditto with the soundcard calibration file loaded....

I had to reduce the Sweep Level to -24dB to avoid clipping with the 802 in the loop.


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## brucek (Apr 11, 2006)

Yeah, so you can see that the 802 does not have a flat response as it should.

Have you taken a response measurement of the other channel of the 802 to see if the problem is in both channels?

Have you taken successive response measures and messed with the EQ dials to determine if that could be the cause?

If that doesn't correct it, then you have a fault in the 802. It could be a bad (or originally) incorrect capacitor, or any number of problems, but there sure appears to be a fault.

brucek


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## entrecour (Jun 13, 2007)

Yep, I tried different EQ settings - way too little effect to flatten the response.

I also tried using the left channel instead - no difference.

I even tried using LINE IN 2 - no difference.

Looks like I'll be returning this unit for repair / replacement. 

And of course all my sub measurments are useless. :gah:

Thanks for all your help guys, it's been a learning experience! As soon as I get the 802 fixed I'll post the new results.


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## brucek (Apr 11, 2006)

> Looks like I'll be returning this unit for repair / replacement


It shows what a great testing tool REW is for line level devices. It would have been difficult to find out you had a fault without it.

brucek


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## entrecour (Jun 13, 2007)

Well I returned the 802 and was sent a replacement. :bigsmile:

I did a loopback test on the new unit and imagine my surprise / shock when the results were virtually identical! :gah:

I triple checked my setup with no change. :scratch:

Somewhat out of desperation I then took contact with Behringer technical support in Germany and provided them with my measurments and details of the test setup, REW, etc. Perhaps unsurprisingly they are not prepared to run the REW test setup themselves and question it's (or rather my) results.

They state..


> Every single unit is tested at the factory using Audio Precision System 1, which is an audio industry standard. Anything which doesn't have a flat response between 20Hz and 20kHz (+/- 1dB) is rejected.


and ..


> Well, the test procedure is due to DIN EN 60268, as this is standard.
> Your attachement unfortunately does not show any benchmark ( dB e.g. is a ratio...not a unit of measurement ).


I find it very hard to believe that two 802 units from seperate batches could be faulty. Behringer have offered to test and replace by existing unit if it is faulty, but if it's not the considerable shipping and insurance costs will be down to me (more than the cost of the unit) so I wonder if you folks have any other suggestions as to what I should check.

Just in case it makes any difference the actual 802 model I am using the latest Xenyx802 with Xenyx Mic preamps and British EQ's.


:help:


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## brucek (Apr 11, 2006)

It would seem strange that two units would have the exact same fault.

If you do a response measure test using a simple cable and it returns a flat response, and then you insert the mixer in that same loop and it returns the poor response you have shown us, then I can't see how the fault is anything but the mixer.

I am at a loss to come up with anything else to try.

brucek


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## JohnM (Apr 11, 2006)

Very strange. The measurement looks more like a plot of crosstalk onto the unconnected channel than a loopback measurement, but you already tried with the other channel selected and got the same result. Also strange is that the problem only seems to afflict the loopback measurement, disregarding the effect of the duff calibration file the measurements looked quite reasonable. I'd be tempted to just ignore the soundcard cal and make a few full range measurements to see if they look OK, if so the problem must lay in the loopback setup somehow. Maybe some pics of the connections would show up something.


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## entrecour (Jun 13, 2007)

I am wondering if the problem is caused by the cables I am using.

I am using one of these 3.5mm stereo TRS mini-jack to 2x 1/4" TRS jacks cables

and one of these 3,5mm stereo jack m/m cables  with a 3.5mm to 1/4" adapter.

I know that these are stereo cables but I figured they must work OK also.

What do you say?


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## brucek (Apr 11, 2006)

> but I figured they must work OK


Oops, why are you using TRS plugs? I was quite specific here and here on page 2 of this thread that the 802 requires TS connectors to unbalance the interface.

The soundcard line-out and line-in jacks are stereo (left and right channel). You must break out these two channels into a right channel and left channel with a stereo splitter, so that you have an unbalanced mono right channel that you will use (+ positive and ground shield). This feeds an unbalanced 1/4" TS plug at the 802.

brucek


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## entrecour (Jun 13, 2007)

Maybe I missed something here but I thought that for unbalanced stereo signals it was equivalent...

My soundcard has two RCA outputs that are combined to a 3.5mm stereo mini-jack. Isn't RCA unbalanced, and so therefore isn't the mini-jack unbalanced?

Then the 3.5mm plug to 2 x 1/4" TRS jacks cable has two mono jacks. Isn't that unbalanced also?


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## brucek (Apr 11, 2006)

> Isn't RCA unbalanced


Yes.



> isn't the mini-jack unbalanced?


Yes.



> 1/4" TRS jacks


A TRS plug has a Tip Ring Sleeve that is (+) positive, (-) negative, and ground, and is for balanced connections.

A TS plug has a Tip Sleeve that is (+) positive, and ground, and is for unbalanced connections.

If your soundcard has RCA jacks already available for right and left connections, why not use simple RCA cables and an adapter.

brucek


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## entrecour (Jun 13, 2007)

They are described as TRS plugs in the product blurb but as might be noticeable in the picture they are TS plus (there is no ring). Anyways I'll get some new cables as it's causing uncertainty on my part.


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## brucek (Apr 11, 2006)

> They are described as TRS plugs in the product blurb but as might be noticeable in the picture they are TS plus


Ah OK, your link in your post (#60) was bad, so I just went by your description that it was a TRS. Sorry.

Yeah, it seems like it has to be the cables either way, so buying a couple cheap adapters will solve the mystery..

brucek


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