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Author Topic: Intermodulation Test  (Read 10816 times)

InflatableMouse

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Intermodulation Test
« on: September 02, 2013, 01:01:32 pm »

I was reading this article again and figured I would try the intermodulation test with my new DAC (Teac UD-501). The results were quite shocking and I'm glad I did it. If you look at the results below I think I could have damaged my speakers with the warbles on PCM output.



To put things in perspective, 0 dB on the Teac headphone output is about similar to -10 dB on the receivers headphone output. This is loud but not uncomfortable. For the speaker test, -45 dB is normal, radio for instance, -48 dB is background. -20 dB is getting loud and -10 dB is when my wife gets mad and the cupboards are rattling. I never played anything on -10 dB that I can remember (maybe some very quiet classical music).

What is completely beyond me is why there was distortion during the PCM tests when I had the 24 kHz low pass enabled in the PEQ? Shouldn't this completely filter out any ultrasonics?
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6233638

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Re: Intermodulation Test
« Reply #1 on: September 02, 2013, 01:46:27 pm »

Interesting tests. Even though my hardware is supposed to handle 192kHz just fine, and my headphones are specified to 80kHz, at louder volumes I do hear issues with the test tones - but that's several times louder than I would ever listen at.

I have the headphone output padding set to -20dB and only ever go about 1/3 up on the volume control. I don't hear anything until I'm around 3/4 on the volume control.
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InflatableMouse

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Re: Intermodulation Test
« Reply #2 on: September 02, 2013, 01:56:01 pm »

Especially at louder volumes its said to be likely due to nonlinearity in the output, although I'm not sure what that means exactly (Im just quoting here  ;)). I was about to google it :P.

Would you mind to try PCM output with a PEQ for a low pass filter 24 kHz@48db/octave? I would be interested to know if you have the same 'issue'. Likely I'm doing something wrong though but I thought that was strange.
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6233638

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Re: Intermodulation Test
« Reply #3 on: September 02, 2013, 01:58:17 pm »

Would you mind to try PCM output with a PEQ for a low pass filter 24 kHz@48db/octave? I would be interested to know if you have the same 'issue'. Likely I'm doing something wrong though but I thought that was strange.
It's silent with that 24kHz filter.

DSD and PCM sound the same, but I believe the DAC is converting to PCM internally anyway. There might be some variance in where the "warbling" starts, but it seemed about the same. (about 3/4 volume - which would be deafening with music)
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InflatableMouse

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Re: Intermodulation Test
« Reply #4 on: September 03, 2013, 12:49:05 am »

Do you (or someone else) have any idea why the low pass filter wouldn't work for me in those PCM tests? PEQ is at the bottom of the list in DSP and I can see it does something as when I disable it it starts clipping, so it does filter out something.
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6233638

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Re: Intermodulation Test
« Reply #5 on: September 03, 2013, 01:59:09 am »

It probably depends on how your hardware handles IMD?
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InflatableMouse

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Re: Intermodulation Test
« Reply #6 on: September 03, 2013, 02:13:47 am »

Well ... I could accept that if I would get some random noise or some faint monotonous beep like with some other tests. The warbles have a distinct pattern which is audible, it actually sounds like the same waveform in the audible spectrum but distorted. There is no difference to this audible pattern between the filter enabled or disabled.

So if I filter that out before JRiver outputs it, how can that same pattern be heard unless it was already present in the audible spectrum before JRiver filtered it out?

The way I see it (which is probably wrong, I realize that) but with my limited knowledge it seems that either JRiver doesn't filter it out properly or the distortion was already present in the audible spectrum before MC applied the filter. In reality, I don't think either is likely but I can't see any other options?

Do I need to change the order of DSP items? PEQ is currently the last item being processed.
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mwillems

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Re: Intermodulation Test
« Reply #7 on: September 03, 2013, 07:45:17 am »

Well ... I could accept that if I would get some random noise or some faint monotonous beep like with some other tests. The warbles have a distinct pattern which is audible, it actually sounds like the same waveform in the audible spectrum but distorted. There is no difference to this audible pattern between the filter enabled or disabled.

So if I filter that out before JRiver outputs it, how can that same pattern be heard unless it was already present in the audible spectrum before JRiver filtered it out?

The way I see it (which is probably wrong, I realize that) but with my limited knowledge it seems that either JRiver doesn't filter it out properly or the distortion was already present in the audible spectrum before MC applied the filter. In reality, I don't think either is likely but I can't see any other options?

Do I need to change the order of DSP items? PEQ is currently the last item being processed.

I think part of the problem is that by setting the filter that high you're not going to have significant attenuation where you need it.  A 48 dB lowpass at 24 KHz will be 48 dB down at 48 KHz, but at 30 KHz will only be about 13 or 14dB down.  So your first test tone is only receiving about 13 or 14 dB of attenuation.  The Warbles are (according to your source site) oscillating between 26 KHz and either 48 Khz or 96 Khz, respectively.  At the top of their bands they should be well and truly attenuated, but at the 26 KHz end of the warble, they're 3 or 4dB down at most.  

If your DAC is sensitive to ultrasonic noise (and it sounds like it is), I'd recommend setting your filter lower (22KHz, or even 20 KHz depending on how well you hear above 15 KHz) or maybe stacking two filters at 24 KHz to get the equivalent of a 16th Order Linkwitz Riley.  That would be 7 or 8 dB down by 26 KHz and 24 dB down by 30KHz.

But you should be hearing some difference when the filter is engaged, and the fact that you're not seeing some difference in loudness is odd in itself.
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InflatableMouse

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Re: Intermodulation Test
« Reply #8 on: September 03, 2013, 08:11:45 am »

I think part of the problem is that by setting the filter that high you're not going to have significant attenuation where you need it.  A 48 dB lowpass at 24 KHz will be 48 dB down at 48 KHz, but at 30 KHz will only be about 13 or 14dB down.  So your first test tone is only receiving about 13 or 14 dB of attenuation.  The Warbles are (according to your source site) oscillating between 26 KHz and either 48 Khz or 96 Khz, respectively.  At the top of their bands they should be well and truly attenuated, but at the 26 KHz end of the warble, they're 3 or 4dB down at most.  

If your DAC is sensitive to ultrasonic noise (and it sounds like it is), I'd recommend setting your filter lower (22KHz, or even 20 KHz depending on how well you hear above 15 KHz) or maybe stacking two filters at 24 KHz to get the equivalent of a 16th Order Linkwitz Riley.  That would be 7 or 8 dB down by 26 KHz and 24 dB down by 30KHz.

But you should be hearing some difference when the filter is engaged, and the fact that you're not seeing some difference in loudness is odd in itself.

Ah that explains a lot. I didn't realize it wasn't removing it all I forgot to take that into account.

The DAC itself didn't really misbehave and the warbles were audible only at max volume and the filter did make a difference. But when something is so low volume its hard to make up any difference so I described it as less noisy. The receiver however is the real "problem" but the way you explain the attenuation makes sense. I'm going to play with the filter some more.

Concerning the PCM filter on the DAC itself and based on this experience I think 'Slow' would be preferable over Sharp (graphs here).

Great learning experience! Thanks!
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mwillems

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Re: Intermodulation Test
« Reply #9 on: September 03, 2013, 08:35:53 am »

Ah that explains a lot. I didn't realize it wasn't removing it all I forgot to take that into account.

The DAC itself didn't really misbehave and the warbles were audible only at max volume and the filter did make a difference. But when something is so low volume its hard to make up any difference so I described it as less noisy. The receiver however is the real "problem" but the way you explain the attenuation makes sense. I'm going to play with the filter some more.

Concerning the PCM filter on the DAC itself and based on this experience I think 'Slow' would be preferable over Sharp (graphs here).

Great learning experience! Thanks!

If you're interested, I thought I'd throw out a few additional notes about crossover topographies that might help as you fiddle with the filters (forgive me if I'm telling you things you already know).  The JRiver filters are butterworths, which means at the setpoint frequency (in your case 24 KHz) they are -3dB regardless of the slope of the filter.  Stacking two butterworths gives you a Linkwitz Riley filter which will (regardless of slope) be -6dB down at the setpoint frequency.  Any X dB crossover filter will be -X dB at one octave above the setpoint frequency (i.e. double the frequency), so 48 dB down at 48 KHz in your case. 

In order to reach -3dB (or -6 dB) at the setpoint, the low pass filter must begin attenuating at a lower frequency than the setpoint (have a look at the graphs in this wiki article, you'll see what I mean http://en.wikipedia.org/wiki/Audio_crossover).  Generally speaking, higher order, steeper filters of the same topography will need less "lead-in" room.  For example, a second order butterworth filter (12dB) will begin attenuating more than an octave before the setpoint, whereas an eighth order filter (48dB) will begin attenuating much less than an octave before the setpoint.  The disadvantage of steeper filters is that they can create phase or impulse anomalies.

The goal is (of course) to have none of the ultrasonics one doesn't want, while leaving the actually audible material unchanged and not introducing any "ringing" or phase anomalies.  That's hard to do with conventional crossover topographies, because filters that will effectively eliminate all ultrasonics will also affect audible material to some extent (due to the issues outlined above).  So you have to balance whether some slight attenuation/phase weirdness in the upper part of the audio band is worth less ultrasonics, or vice versa.  For my part, I start my filtering at 18 KHz, because I can't hear much above 15 or 16 KHz, and I find a slight roll down in frequency response above 10 KHz to be less fatiguing to my ears anyway.  But that's a personal preference, and you'll need to find the solution that works for you.

On another note, the "trade-offs" I'm describing are less of a concern in the FIR filtering context (i.e. convolution).  If you use convolution, you have access to arbitrarily steep phase adjusted filters, so that would be a theoretically "perfect" solution, but convolution is a bit fiddly, and I wouldn't recommend starting to use it just for this.  If you're already using it though, it offers the next best thing to a theoretically perfect "brick wall filter."
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InflatableMouse

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Re: Intermodulation Test
« Reply #10 on: September 03, 2013, 10:36:26 am »

Thanks for explaning that Mwillems. I'll add that page to my homework list ;). It seems a bit easier to read due to the lack of formulas which is great. I don't need to code them, I just wish to understand more about them.

I just tried and started adjusting the filter down in 2 kHz steps. WHen I arrived at 10 kHz @ 48dB/octave, I still had warble noise as without a filter. The only difference is that gradually the highs were filtered out more but that changed little as the noise is below 10k.

For some reason I think its not working for me, but I don't know why.

I can do the same thing on the Asus Xonar and it works perfectly fine.

When I play the file over HDMI to my receiver, I don't need a low pass filter at all because the Denon's DAC filters it perfectly.
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InflatableMouse

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Re: Intermodulation Test
« Reply #11 on: September 03, 2013, 10:55:58 am »

Could this be a bug?

Asus Xonar Analyzer without PEQ/Low pass:


Asus Xonar PEQ settings:


Asus Xonar Analyzer with PEQ/Low pass:


Teac Analyzer without PEQ/Low pass:


Teac PEQ settings:


Teac Analyzer with PEQ/Low pass:


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InflatableMouse

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Re: Intermodulation Test
« Reply #12 on: September 03, 2013, 11:00:02 am »

Now this is getting really weird.

Figured I'd hook up the DAC to my other PC, its almost silent, even without a low pass filter. With filter its dead silent  ? ? ?.
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mwillems

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Re: Intermodulation Test
« Reply #13 on: September 03, 2013, 11:17:15 am »

Now this is getting really weird.

Figured I'd hook up the DAC to my other PC, its almost silent, even without a low pass filter. With filter its dead silent  ? ? ?.

That really looks like some kind of software bug because the unfiltered TEAC PCM analyzer shows tons and tons of audible sound (below 20 KHz), that the Xonar analyzer doesn't show at all.  Most importantly it's showing it in the software chain before it ever even gets to the DAC, which means that it's probably something gone wrong with the specific output settings being used, and not your hardware.  Your sample is clipping in that picture, so that may be part of it, but in the second low-passed version there's no clipping and analyzer is still showing your audible warble.

Are the JRiver settings on the other (quieter) PC identical to the first PC? 
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InflatableMouse

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Re: Intermodulation Test
« Reply #14 on: September 03, 2013, 12:00:34 pm »

Are the JRiver settings on the other (quieter) PC identical to the first PC? 

That was because on the louder pc process independently from internal volume was ticked, on the other pc it was not. I corrected it and tried again and the analyzer is now the same.

I happen to have the same Asus Xonar in the HTPC (on which this noise is happening) and hooked up my headphones. The noise is happening exactly as with the DAC output.

So:
on my own PC, MC filters fine and even without a filter, noise is very low exactly how it looks on the analyzer screenshots.
On the HTPC, MC has a boatload of noise in the audible range and the filter does not work.

Somehow, MC's configuration on the HTPC is different but I can't see what it would be. I created a brand new zone from scratch, set output to ASIO and tested. There is nothing I can mess up I think?
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InflatableMouse

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Re: Intermodulation Test
« Reply #15 on: September 03, 2013, 12:06:36 pm »

One thing I notice on the HTPC with all the noise, is that when I stop playback and start it again, DSP filter seems to work for about 5 seconds, then the noise comes in, and it won't go away by disabling/enabling the PEQ.
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mwillems

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Re: Intermodulation Test
« Reply #16 on: September 03, 2013, 12:09:43 pm »

That was because on the louder pc process independently from internal volume was ticked, on the other pc it was not. I corrected it and tried again and the analyzer is now the same.

I happen to have the same Asus Xonar in the HTPC (on which this noise is happening) and hooked up my headphones. The noise is happening exactly as with the DAC output.

So:
on my own PC, MC filters fine and even without a filter, noise is very low exactly how it looks on the analyzer screenshots.
On the HTPC, MC has a boatload of noise in the audible range and the filter does not work.

Somehow, MC's configuration on the HTPC is different but I can't see what it would be. I created a brand new zone from scratch, set output to ASIO and tested. There is nothing I can mess up I think?


The only thing I can think of would be to check all of your settings that aren't zone specific.  Otherwise, I have no idea what could be causing it, although it's pretty obviously not the hardware.  I'm as baffled as you are.
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6233638

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Re: Intermodulation Test
« Reply #17 on: September 03, 2013, 01:09:26 pm »

As expected, the filters are working correctly for me when I check the analyzer.

I wonder though, what would be an ideal filter to use? Two 24@48? More? A lower frequency filter?
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InflatableMouse

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Re: Intermodulation Test
« Reply #18 on: September 03, 2013, 01:15:28 pm »

Reinstalled MC, reset all settings, rebooted, uninstalled some other things I didn't need (OpenAL, some sound API? No idea where that came from). But nothing helps. Created new zones as all settings were reset, just a filter nothign else. Noise all over the place.

Something on this HTPC is preventing the low pass filter from doing its work it seems.
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6233638

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Re: Intermodulation Test
« Reply #19 on: September 03, 2013, 01:29:13 pm »

OpenAL, some sound API? No idea where that came from
A lot of games install this. Rapture3D also gets installed by a number of them. (very good HRTF for headphone users... I wish it could be integrated into Media Center)

Not sure what to suggest about your other problems though. :-\
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InflatableMouse

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Re: Intermodulation Test
« Reply #20 on: September 03, 2013, 01:50:33 pm »

Dunno, I don't play games on the HTPC and it doesn't have any games installed, or anything related :s. I have no idea where it came from.

Anyhow, the MC reinstall didn't help so I've restored a backup with settings and thinking about the above where I mentioned HDMI didn't suffer from it because (so I thought) the Denon's DAC filters it out properly? Well, turns out that zone didn't suffer from the noise at all. I reconfigured it to output to the DAC and Voila!, noise is gone.

So I'm happy but this doesn't mean its solved. I have other zones that all suffer from this issue and I don't know where it comes from. I created new zones and they get this issue right off the bat, it's wearing me out :(.

I'll go over the settings of each zone, comparing them with the HDMI zone. I wouldn't know what to look for though, we'll see.
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InflatableMouse

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Re: Intermodulation Test
« Reply #21 on: September 03, 2013, 02:00:02 pm »

When I restore my HTPC library and settings on my own PC, I can recreate the problem there.

This means the problem is definitely in the zones somehow as they transfer with the backup/restore.
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InflatableMouse

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Re: Intermodulation Test
« Reply #22 on: September 03, 2013, 02:03:44 pm »

lol. every new zone I create on my pc suffer the same issue now!

When I base a zone on HDMI it still gets the problem!

I'm beginning to think something has become corrupt.

I'll try restoring my pc's library and settings again, see if that gets rid of it. If it does, I'll have to restore it on my HTPC as well and see if that helps.
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InflatableMouse

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Re: Intermodulation Test
« Reply #23 on: September 03, 2013, 02:22:57 pm »

Complete reinstall on my pc, wiped all settings, removed Teac drivers just in case. First zone, directsound, I changed nothing, default output. Noise.

So I compare all settings one by one from the HDMI zone that doesn't have the issue but the noise remains.

I'm done troubleshooting for the day, this is the weirdest issue I've had in a long time!
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mwillems

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Re: Intermodulation Test
« Reply #24 on: September 03, 2013, 04:00:24 pm »

Dunno, I don't play games on the HTPC and it doesn't have any games installed, or anything related :s. I have no idea where it came from.

Anyhow, the MC reinstall didn't help so I've restored a backup with settings and thinking about the above where I mentioned HDMI didn't suffer from it because (so I thought) the Denon's DAC filters it out properly? Well, turns out that zone didn't suffer from the noise at all. I reconfigured it to output to the DAC and Voila!, noise is gone.

So I'm happy but this doesn't mean its solved. I have other zones that all suffer from this issue and I don't know where it comes from. I created new zones and they get this issue right off the bat, it's wearing me out :(.

I'll go over the settings of each zone, comparing them with the HDMI zone. I wouldn't know what to look for though, we'll see.

I once had a zone that would crash every time it started MadVR.  None of my other zones had the problem, and I never successfully identified what made that zone work badly.  I just copied the working zones :-(


As expected, the filters are working correctly for me when I check the analyzer.

I wonder though, what would be an ideal filter to use? Two 24@48? More? A lower frequency filter?

Like I said above, there are trade offs.  "Ideal" depends on your tolerances and priorities because no filter will solve the problem perfectly without potentially audible consequences.  I prefer a lower frequency filter (at around 18KHz) because I'm not that concerned about audio content in the upper range of the band.  As I recall, you can hear to almost 18 KHz, right, so you might find that unacceptable.  One would have to set a 48 dB filter at 28 or 30 Khz to have no effect in the audible band, and that's too high to be useful (IMO).  It's a matter of taste.
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6233638

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Re: Intermodulation Test
« Reply #25 on: September 03, 2013, 11:12:45 pm »

It would be nice if there was some way to have Media Center plot what the filters were doing.
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InflatableMouse

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Re: Intermodulation Test
« Reply #26 on: September 04, 2013, 12:31:09 am »

I think I once had a spreadsheet that could do that but it got lost some time ago.

It might even be in this sub forum attached to one of Mojave's posts. I'll see if I can find it again.

Edit: Ha! It's right here! Use the LPF tab. It takes slightly different parameters though, it takes a value Q instead of dB/oct.

If you want to use the Parametric EQ the same as the sliders, then enter 4.318 as the Q. The sliders usually have a bandwidth of 1/3 octave which corresponds with a Q of 4.318.

From this post down there's some posts explaining some of it.
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6233638

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Re: Intermodulation Test
« Reply #27 on: September 04, 2013, 12:49:10 am »

Thanks, but this is the problem - I don't know how Q corresponds to the dB/octave options.
And what if you are stacking filters?

It would be nice if the analyzer plotted the filter response.
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InflatableMouse

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Re: Intermodulation Test
« Reply #28 on: September 04, 2013, 12:51:54 am »

Edited post above while you responded but you're right, I don't know how that works either.
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6233638

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Re: Intermodulation Test
« Reply #29 on: September 04, 2013, 01:04:43 am »

Looks like it may not be working correctly in LibreOffice, or I've misunderstood how to use it. The frequency response goes up when I use that.
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InflatableMouse

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Re: Intermodulation Test
« Reply #30 on: September 04, 2013, 01:09:42 am »

Looks like it may not be working correctly in LibreOffice, or I've misunderstood how to use it. The frequency response goes up when I use that.

Try opening it web excel. I uploaded it to my skydrive and opened it, it seems to work fine.
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InflatableMouse

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Re: Intermodulation Test
« Reply #31 on: September 04, 2013, 04:08:02 am »

Posted it in the beta forum as well under the latest build.

I found which registry setting is responsible for the noise problem with the intermod tests.

This value: "Between Tracks Mode"=dword:00000002 is responsible for the distortion. When I set it to "00000000" the distortion disappears.

Full path is: [HKEY_CURRENT_USER\Software\JRiver\Media Center 19\Zones\10004\Audio Settings]

where 10004 stands for the zone. I can perfectly reproduce this issue on every zone, on every computer. When I set it to 0, the problem dissapears.

This leads me to believe that for someone who can't replicate this issue, this value defaults to 0 already. I would appreciate it if someone could set his value to 2 to check if he can reproduce the problem.

Now I need to find out why mine defaults to 2.

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InflatableMouse

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Re: Intermodulation Test
« Reply #32 on: September 04, 2013, 05:16:52 am »

When using gapped fade (I use 0.4s but it doesn't seem to matter) if you pause a track that's currently playing, and start playback of another track, it resumes the current track for a second to fade out and into the new one.
Because nothing is currently playing, it should just fade into the new track immediately.

Well I thought that registry setting was that setting in the GUI, but it doesn't appear so. When I change it in the GUI, that registry setting doesn't change and vice versa.

So I think its another setting. If you want to test it, you're going to have to close Media Center, edit the registry setting for the zone you want to test it with (I'd create a new zone and base it off another one) and restart Media Center. The zone name can be found in the root of a zone entry in the registry.
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InflatableMouse

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Re: Intermodulation Test
« Reply #33 on: September 04, 2013, 05:32:30 am »

Argh I'm being dim.

I was looking at the registry Between Track Mode to change when I changed the time setting, but obviously its the MS value that is changing.

So, crossfade is causing the distortion for the duration of MS. It all makes sense now.

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InflatableMouse

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Re: Intermodulation Test
« Reply #34 on: September 04, 2013, 06:17:22 am »

To replicate:

Make sure volume is set low.

Add warbles_96 to playing now.
Set aggressive crossfade to 1 second.
Set repeat mode to repeat current item or repeat playlist with warbles as the only track.
Start playback.
Watch analyzer.


Note: it doesn't happen on repeat off or with multiple items in the playlist. Basically, crossfade needs to be applied on the same track when on repeat for it to happen.
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6233638

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Re: Intermodulation Test
« Reply #35 on: September 04, 2013, 06:27:46 am »

I can confirm that noise for the first second, when using aggressive crossfading.
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InflatableMouse

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Re: Intermodulation Test
« Reply #36 on: September 04, 2013, 06:44:17 am »

Thanks!

Been chasing this for days, I'm glad its confirmed by someone else too.
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mwillems

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Re: Intermodulation Test
« Reply #37 on: September 04, 2013, 07:45:33 am »

Thanks, but this is the problem - I don't know how Q corresponds to the dB/octave options.
And what if you are stacking filters?

It would be nice if the analyzer plotted the filter response.

Some of us have been asking for a graphical representation of what's happening in PEQ for a little bit now, and I think the response has been that it's on the list.  If you look in the recent feature request thread there have been three or four requests for exactly that feature.  It would make PEQ much more useful: folks without measurement equipment/software have no way of knowing what the EQ will do to the sound (unless they're much better at math than me), and they have no way to detect failure scenarios like this one (where EQ isn't working as expected).  

But, for now, if you're just trying to get a sense of what the filter topographies will look like, silly as it sounds, I use RePhase for that.  It's a free tool for building convolution filters, but it also graphically models everything and will let you dial in low passes and high passes of different types and with different slopes, and has a fully functioning parametric EQ.  It includes some EQ options that JRiver does not (alternative crossover topographies), and lacks one or two that JRiver includes (shelving filters), but it's perfectly good for modelling crossovers and conventional parametric EQ. When I'm considering new PEQ settings in JRiver, I'll usually model them in RePhase first, and then dial them into JRiver once I like the look of them (I prefer to use JRiver's DSP for actual processing when I can because it's much lower latency than trying to accomplish the same thing in convolution).  I'm sure there's much better modelling software out there, but RePhase is free and pretty no frills.
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InflatableMouse

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Re: Intermodulation Test
« Reply #38 on: September 04, 2013, 07:58:37 am »

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mwillems

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Re: Intermodulation Test
« Reply #39 on: September 04, 2013, 09:51:14 am »

Very cool, thanks!

http://sourceforge.net/projects/rephase/?source=dlp



The only caveat I should offer is that RePhase will show you both frequency response and phase response of your proposed filters, but you should ignore the latter.  Because RePhase is a tool for actually doing DSP (not just modeling), its models assume you're using the filters RePhase offers, and RePhase uses phase linear crossovers (which are only possible in FIR/convolution filtering). 

So the frequency line (the solid line) in RePhase will model what the JRiver filters look like in the frequency domain, but the phase line (dotted line) will not resemble what JRiver's filters would do to the phase in most cases. Most people aren't that concerned about phase anyway, but I just wanted to make sure that was clear to anyone who might be using RePhase as an off-label modelling tool like I am.
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Matt

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Re: Intermodulation Test
« Reply #40 on: September 04, 2013, 10:09:22 am »

To replicate:

Make sure volume is set low.

Add warbles_96 to playing now.
Set aggressive crossfade to 1 second.
Set repeat mode to repeat current item or repeat playlist with warbles as the only track.
Start playback.
Watch analyzer.


Note: it doesn't happen on repeat off or with multiple items in the playlist. Basically, crossfade needs to be applied on the same track when on repeat for it to happen.

You're reporting this like a bug, but it's simply a side-effect of mixing signals with artificially perfect periodicity.

When analyzing audio, you need to either disable cross-fading or use clips long enough that you can watch after the fade window.
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Matt Ashland, JRiver Media Center

InflatableMouse

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Re: Intermodulation Test
« Reply #41 on: September 04, 2013, 11:54:18 am »

You're reporting this like a bug, but it's simply a side-effect of mixing signals with artificially perfect periodicity.

When analyzing audio, you need to either disable cross-fading or use clips long enough that you can watch after the fade window.

Interesting. Obviously I don't know enough about it to realize that, which is why I reported it as a bug. I was indeed convinced it was one :). It had me worried it can happen for real when 2 high sample rate files are mixed.

Either way, I'm still happy I figured out what it was. It was quite a hunt!  ;D
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Matt

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Re: Intermodulation Test
« Reply #42 on: September 04, 2013, 03:21:44 pm »

Interesting. Obviously I don't know enough about it to realize that, which is why I reported it as a bug. I was indeed convinced it was one :). It had me worried it can happen for real when 2 high sample rate files are mixed.

To take an extreme example, imagine mixing two sine waves offset from each other so that one peaks when the other hits its low point.  The two signals perfectly offset in the time domain, and you would hear silence.

I don't think there's any answer to whether this is right or wrong.  It's just how it works when you mix signals in the time domain.

I suppose you could mix in the frequency domain instead of the time domain, but I've never seen that done (and I'm not sure that wouldn't just create different types of artifacts).

The good news is that these weird effects from time-based mixing are not really an issue with real world signals.  They only show up with test tones.
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Matt Ashland, JRiver Media Center
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