Tony, you really should eliminate the possibility that the bug-fix in beta12 has somehow introduced this problem. Do you have a spare running beta11 that you can try?

BTW, I'm going to be incommunicado for the next 9 days, but if you haven't found the cause by then, I'll look into what can be done in the kernel to help debug this.

Tony, you really should eliminate the possibility that the bug-fix in beta12 has somehow introduced this problem.

Agreed. Since everyone else is running Beta 11, I was hoping that posting the settings could kill two birds with one stone by verifying the problem exists at all, as well as checking that it was present on beta11. If Maczrool gets a chance, this will take care of that requirement.

How do I enter "independent" mode with the EQ? I can't find it in the FAQ. I've got the player put back together and the "snippet" loaded. Let me know and I'll run the test.

Stu

I believe that "dependent" and "independent" correspond to "locked" and "unlocked" in the equalizer, respectively. But then, I'm only running 1.03, and don't know if it's different in 2b11.

I guess I need step-by-step instructions on how to get the independent EQ enabled. I never really use the EQ, because the Rio is not my main headunit. If someone knows where they are or could write some that would be helpful.

Stu

I guess I need step-by-step instructions on how to get the independent EQ enabled.

When in the equalizer, press the DOWN button on the front panel to open the menu. There should be a "Set Locks" option on this menu. That lets you choose "Independent" or "Locked L/R". This is in the 2.0x software of course.

I've done the test. The clipping is there on both analog and digital outs. If you want, I can send you a WAV file of the actual output from the digital outs. PM me and I'll send it out to you.

Hope this helps,

Stu

The clipping is there on both analog and digital outs.

Interesting!

And you have verified that it goes away with a flat EQ?

You can e-mail the WAV file to [email protected] at your convenience. I'm now broadband at home so the file attachment size is not a problem. Will this WAV file be some kind of a genuine digital grab from an S/PDIF input?

Thanks for doing this for me. I will run the WAV file through SpectraLAB and see what it tells me about the frequency distribution.

Yes, the clipping mysteriously vanishes with the Flat EQ setting, both on analog and digital outs. Yes, I will send the digital output from the coax S/PDIF to my digital sound card on my Mac and capture the output to hard disk and save as WAV, so it should be a "genuine digital grab."

One thing I must mention about the WAV file I will be sending you is that it may be 1dB quieter than the original MP3 due to something funny with the I2S outputs (the signals feeding my digital out board) on the main board of the Empeg. I have discovered that the volume must be reduced by 1 dB (from 0 dB) to remove some mild clipping during peaks in tracks. I know it's nothing our digital out circuitry is doing, because I've tried I2S (which bypasses everything but the line driver IC) and both optical and coaxial outputs with the same clipping at 0 dB. I've also tried different external DACs, with the same result.

Stu

If I recall correctly, the sample settings I gave you will produce the distortion even at -1db so that's fine.

Did you get a chance to play with the WAV file I sent you?

Stu

Unfortunately, oronet's mail and web servers went Tango Uniform yesterday, so I can't get my email at the moment. If your mail didn't bounce, it's probably sitting in my inbox on their server, and whenever it comes back up again I'll be able to retrieve it.

Okay, got the files onto my hard disk. Haven't listened to them yet. But in your accompanying email you mentioned that you noticed the distortion on only one channel.

Wonder if this has something to do with the Beta11 L/R bug? I think I'm hearing it in only one channel as well, but I'm running a beta that's not supposed to have the bug...

(And I did do all of the corrections in beta12, so in theory I should not have been saving a preset with beta11.)

Okay, yeah, that WAV file clearly shows the distortion on the right channel, encoded right in there.

Analysis of the file shows nothing special, other than that the file has a lot of bass. There's notable peaks at 32hz and 64hz, but nothing special about that. Correcting either of those frequencies downward sharply doesn't fix anything.

Wait a minute. Those are multiples of each other, aren't they? I wonder... Hang on...

WHOA, weird.

Okay, so I have this idea. The bass notes have peaks at 32 and 64 hz. So I think to myself, what if the problem is actually at 16hz? What if it's a harmonic of 32+64 going wonky?

Well, the EQ doesn't go down to 16. But it goes down to 18. So I take one of the two unused bands, set it to 18hz, give it a fairly high Q (20.0), and start correcting downward. At -5db, the music is still distorting, sounding like it always has. At -6db, things go NUTS, the sound distorts TERRIBLY on all frequencies. At -7db to -8db, there is no output except for the occasional bit of distortion/static on the heaviest downbeats. At -10db, there is no output at all from the player.

Wonder if this is related?

How do these peaks in my WAV file compare with the peaks in a WAV of the clip that has not been run through the Empeg's EQ?

Stu

Oh, and by the way, a direct spectral analysis of the player's outputs shows that I'm not being bitten by the L/R EQ bug as the two channels have an essentially the same spectral signature.

How do these peaks in my WAV file compare with the peaks in a WAV of the clip that has not been run through the Empeg's EQ?



Purple=Your supplied WAV file.
Green=Original, unmodified audio file without the Empeg being involved.

Funny how, apart from amplitude, they don't look that different, considering how severe some of the EQ corrections were.

I also don't know why yours has a greater amplitude, when all of the EQ corrections were cuts instead of increases.

INTERESTING...

I think it is some kind of a bug, either in the DSP or in the programming that goes to the DSP.

All right. Start with a flat EQ, any volume setting, any music....

Take the first band and set its frequency to 18hz (not khz) and its Q to 20.0.

Decrease the gain (with a fixed Q) and you will hear:

-6db= SEVERE Distortion in the RIGHT CHANNEL ONLY, while the left channel is fine.
-7db= Right channel gone, left channel fine.
-27db= Both channels fine again.

Of course, no one outside of the Alpha team need mess with this, as it's on internal beta 12 which is the version that's not supposed to have L+R bugs...

/me posts to internal bug list...

/me confused.

Tony, are you saying that this is a beta12 bug only?

I think it's in both beta12 and beta 11, and it's a different bug than the one-channel-only-gets-the-Q bug in beta11.

On a possibly related note. I was hoping to use a 18Hz high Q cut as a "poor mans" subsonic filter. Running pink noise through 2b11. comparing the input (reference) and output using various PC&soundcard 2 channel analizers (SpectroLAB, SmaartLive, & EFT). I noticed that below around 40Hz the EQ stoped working, meaning the output trace just floated back up to match the refence level (averaging on) as I selected a center frequency below that point. It was in the High 30's I think. Can't post results, didn't save em sorry. I say "around 40" only because I don't remember, but there was a consistant frequency below which this behavior occured, and was consistantly reproducable in both 2 and four channel, independant and locked modes. I was using only the first band of the parametric, all others were flat.

This led me to think the only way to get a subsonic High Pass filter would to use an fmod (they come in 20Hz and 50Hz, -12db/octave) into the amp. An unfortunante solution because my starting parameters (which I'd hoped to tweak) were more like -24dB @ 25 Hz.

This was my first pass of playing with the Rio "on the bench" (PC). Then am now embarked on my install (dash, subs, amp, cap, etc.). I plan to retry this again and in the car when thats complete. And will reproduce and post more detailed results.

Michael

Yes, I've used F-mods in the past (when my amps didn't have crossovers built-in, before I bought newer amps) and they worked quite well.

This is an interesting note about the lower limit for the EQ, and would explain why I was never able to make it satisfactorily work as a high pass filter. I wonder if this is a limitation of the DSP, or if it's a bug in the code?

"... after I did my RTA adjustments ..."

What are some of the cheapest ways to RTA a system? Renting an audio analyzer is mucho $$$ (~150 per day from a cursory search) which would limit me to one day. Any software based solutions that aren't very expensive would be nice.

Cory

I did it with a downloadable 30-day trial version of SpectraLAB and a Shure SM-58 microphone being run through an XLR-to-minijack convertor. The SM-58 was borrowed from a musician friend, so I essentially did it for free.

Oh, and DON'T expect one day to be enough. I've gone through several passes on my system, tweaking different things and learning a little more each time.

Thanks for the pointer. I've got an Electro-voice ND257 mic that hasn't seen any use for over a year... I fully expect to spend a couple of months fine tuning it. Funny how high powered and really high quality audio can expose a LOT of issues...

Anyone else interested in this approach should start out at http://www.soundtechnology.com/LAB432v17.EXE.

You might want to check and see if that EV 257 has a pre-defined response curve already built-in to the software. The reason I chose the SM58 was because there was already a template for it in SpectraLAB.

I actually called up my musician friend and read him the list of built-in mic templates, and he stopped me on the SM58 and said, "oh, I've got a bunch of those".

Reminds me-- I gotta return it to him one of these days.

hey tony, did that software to automatically change your EQ with a mic ever get released? i remember you talking about ti around the time of the bay area meet

The idea of writing such a piece of software was discussed, but non one ever wrote anything. Remember that the input sampling rate of the player's mic input would preclude the ability to adjust frequencies over about 11khz, so it would only be about a 50% solution, even after lots of work.