"Official" Audyssey thread (FAQ in post #51779) - Page 574 - AVS Forum
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post #17191 of 71784 Old 08-05-2009, 12:17 PM
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Larry,

Thanks for the reply.

Do you think that if I decide to go ahead with the Mythos speakers and Denon receiver I should just set all the receiver setting to default forget about an Audyssey room calibration and hope for the best?

Thank you.
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post #17192 of 71784 Old 08-05-2009, 12:18 PM
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Originally Posted by pepar View Post

Correct major peaks with the PEQ and then let Audyssey finish the job.

So, what are we to make of this scenario? 1/12th octave measurement before EQ looks fairly good although there is a bit of a node (1/3 octave at about 65Hz that is about 6dB down). The choices at that time are to permit MultEQ to correct the rest of the low spectrum to smooth out, or smooth it out pre-MultEQ with a PEQ and let MultEQ do the rest. Playing around, I did the latter. The post Audyssey calibration RTA FR measurements look much less smooth...characterized primarily with hump that is about 1.2 octave wide centered at 36Hz and averages 3dB to 4dB higher than the rest of the low spectrum. Inconsequential, considering the RTA is only looking a the frequency domain?
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post #17193 of 71784 Old 08-05-2009, 12:21 PM
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Originally Posted by pepar View Post

I have found an inline mic isolation mount and will link it soon.
edit: they are nuts and want $75 for it.

And we all thought cables were high-margin items.
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post #17194 of 71784 Old 08-05-2009, 12:25 PM
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Quote:
Originally Posted by Big Picture View Post

Larry,

Thanks for the reply.

Do you think that if I decide to go ahead with the Mythos speakers and Denon receiver I should just set all the receiver setting to default forget about an Audyssey room calibration and hope for the best?

Thank you.

Hi,

If you have already decided to make these purchases, then it makes sense to experiment with Audyssey on and off.

It doesn't hurt to hope for the best with Audyssey on.

Larry
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post #17195 of 71784 Old 08-05-2009, 12:27 PM
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Quote:
Originally Posted by Cam Man View Post

So, what are we to make of this scenario? 1/12th octave measurement before EQ looks fairly good although there is a bit of a node (1/3 octave at about 65Hz that is about 6dB down). The choices at that time are to permit MultEQ to correct the rest of the low spectrum to smooth out, or smooth it out pre-MultEQ with a PEQ and let MultEQ do the rest. Playing around, I did the latter. The post Audyssey calibration RTA FR measurements look much less smooth...characterized primarily with hump that is about 1.2 octave wide centered at 36Hz and averages 3dB to 4dB higher than the rest of the low spectrum. Inconsequential, considering the RTA is only looking a the frequency domain?

I am in the "they-invented-it-they-should-know-how-to-best-use-it" camp and think that I have better things to do with my time than futz around with RTA's and PEQ's.
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post #17196 of 71784 Old 08-05-2009, 12:44 PM
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Originally Posted by pepar View Post

I am in the "they-invented-it-they-should-know-how-to-best-use-it" camp and think that I have better things to do with my time than futz around with RTA's and PEQ's.

There is a good bit of truth to that. Chris does a terrific job of providing information that gives the technology credibility. The government should do such a good job these days. "Trust us on the details in these bills." Chris I trust.
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post #17197 of 71784 Old 08-05-2009, 12:52 PM
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Originally Posted by Big Picture View Post

Larry,

Thanks for the reply.

Do you think that if I decide to go ahead with the Mythos speakers and Denon receiver I should just set all the receiver setting to default forget about an Audyssey room calibration and hope for the best?

Thank you.

If you decide on that Mythos speaker, you should certainly try using Audyssey and decide whether it is an improvement or not. OTOH, I would sooner use Audyssey with more conventional speakers, even little ones, than go for this, or any other, all-in-one.

Kal Rubinson

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http://www.stereophile.com/category/music-round

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post #17198 of 71784 Old 08-05-2009, 12:56 PM
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There's another thread entirely devoted to the question of doing Audyssey or PEQ first, followed by the other. The opinions seem about equally split, with 'Audyssey people' feeling that it (Aud) should be done last, and 'Velodyne (SMS-1) people' feeling that PEQ via the SMS-1 (or BFD, etc.) should be done last.
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post #17199 of 71784 Old 08-05-2009, 12:59 PM
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Originally Posted by millerwill View Post

There's another thread entirely devoted to the question of doing Audyssey or PEQ first, followed by the other. The opinions seem about equally split, with 'Audyssey people' feeling that it (Aud) should be done last, and 'Velodyne (SMS-1) people' feeling that PEQ via the SMS-1 (or BFD, etc.) should be done last.

What a surprise!
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post #17200 of 71784 Old 08-05-2009, 01:03 PM
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Originally Posted by Cam Man View Post

So, what are we to make of this scenario? 1/12th octave measurement before EQ looks fairly good although there is a bit of a node (1/3 octave at about 65Hz that is about 6dB down). The choices at that time are to permit MultEQ to correct the rest of the low spectrum to smooth out, or smooth it out pre-MultEQ with a PEQ and let MultEQ do the rest. Playing around, I did the latter. The post Audyssey calibration RTA FR measurements look much less smooth...characterized primarily with hump that is about 1.2 octave wide centered at 36Hz and averages 3dB to 4dB higher than the rest of the low spectrum. Inconsequential, considering the RTA is only looking a the frequency domain?

This and some recent posts got me thinking that we have a paradox here akin to Gödel's Incompleteness Theorem. Here is my version of that as it applies here: "An audio system measured with any tool whatsoever can be made to look perfect when measured with that same tool afterwards". This is a universal theorem and applies to all measurement systems I know of, including MultEQ.

When I hear about how smooth the graphs look with one method or another I can't help but smile. The person writing that is making an inherent assumption that the method has some correspondence to what we hear. "I know flat is good and I get flat on my beautiful graphs so my method is great".

Many years ago the Audio Engineering Society conducted a bake-off among the leading (at the time) speaker measurement systems. The creators of the software were invited to measure a speaker in the the exact same position in the same room. The results were disheartening: every method produced a completely different result! The theories behind each method appeared correct and speaker manufacturers still use one or another of those tools today and produce beautifully flat measurements. But, when measured with a different tool they look different.

I don't want to turn this into a marketing op-ed for MultEQ. But, I do want people to know that before Audyssey was even a glint in our eye we were performing research at the university to understand why these things happen and how to get around them using an intersection of knowledge from acoustics, adaptive audio signal processing, and psychoacoustics.

Chris

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post #17201 of 71784 Old 08-05-2009, 01:07 PM
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Quote:
Originally Posted by audyssey View Post

Many years ago the Audio Engineering Society conducted a bake-off among the leading (at the time) speaker measurement systems. The creators of the software were invited to measure a speaker in the the exact same position in the same room. The results were disheartening: every method produced a completely different result! The theories behind each method appeared correct and speaker manufacturers still use one or another of those tools today and produce beautifully flat measurements. But, when measured with a different tool they look different.
.


That is really true and very frustrating for hobbyist. That and the fact that different theories exist as to what is/are the most important measurements-polar response, on and off axise F/R, distortion (is or isn't important). Enough to turn you from an objectivist to a subjectivist
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post #17202 of 71784 Old 08-05-2009, 01:08 PM
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Quote:
Originally Posted by audyssey View Post

This and some recent posts got me thinking that we have a paradox here akin to Gödel's Incompleteness Theorem. Here is my version of that as it applies here: "An audio system measured with any tool whatsoever can be made to look perfect when measured with that same tool afterwards". This is a universal theorem and applies to all measurement systems I know of, including MultEQ.

When I hear about how smooth the graphs look with one method or another I can't help but smile. The person writing that is making an inherent assumption that the method has some correspondence to what we hear. "I know flat is good and I get flat on my beautiful graphs so my method is great".

Many years ago the Audio Engineering Society conducted a bake-off among the leading (at the time) speaker measurement systems. The creators of the software were invited to measure a speaker in the the exact same position in the same room. The results were disheartening: every method produced a completely different result! The theories behind each method appeared correct and speaker manufacturers still use one or another of those tools today and produce beautifully flat measurements. But, when measured with a different tool they look different.

I don't want to turn this into a marketing op-ed for MultEQ. But, I do want people to know that before Audyssey was even a glint in our eye we were performing research at the university to understand why these things happen and how to get around them using an intersection of knowledge from acoustics, adaptive audio signal processing, and psychoacoustics.

I am going to link this as required reading in the AS-EQ1 thread . . .
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post #17203 of 71784 Old 08-05-2009, 01:08 PM
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Originally Posted by audyssey View Post

This and some recent posts got me thinking that we have a paradox here akin to Gödel's Incompleteness Theorem. Here is my version of that as it applies here: "An audio system measured with any tool whatsoever can be made to look perfect when measured with that same tool afterwards". This is a universal theorem and applies to all measurement systems I know of, including MultEQ.

When I hear about how smooth the graphs look with one method or another I can't help but smile. The person writing that is making an inherent assumption that the method has some correspondence to what we hear. "I know flat is good and I get flat on my beautiful graphs so my method is great".

Many years ago the Audio Engineering Society conducted a bake-off among the leading (at the time) speaker measurement systems. The creators of the software were invited to measure a speaker in the the exact same position in the same room. The results were disheartening: every method produced a completely different result! The theories behind each method appeared correct and speaker manufacturers still use one or another of those tools today and produce beautifully flat measurements. But, when measured with a different tool they look different.

I don't want to turn this into a marketing op-ed for MultEQ. But, I do want people to know that before Audyssey was even a glint in our eye we were performing research at the university to understand why these things happen and how to get around them using an intersection of knowledge from acoustics, adaptive audio signal processing, and psychoacoustics.

Quote:


Chris I trust.

See what I mean?

I'll confess it was a loaded question, because I knew what the response would be and that it would be detailed, appropriate, and serve to continue to inform/educate.
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post #17204 of 71784 Old 08-05-2009, 01:22 PM
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Originally Posted by Cam Man View Post

See what I mean?

I'll confess it was a loaded question, because I knew what the response would be and that it would be detailed, appropriate, and serve to continue to inform/educate.

It is good to have a definitive statement on this issue. I have found myself getting scooped up in the hubbub of measuring, especially with the AS-EQ1 which generates before and after graphs that, much to Mr. Mullen's chagrin, no one can duplicate with the tools the average enthusiast - and even the not so average acoustician - has available. Perhaps Thomas Haden Church's character in Sideways said it best - paraphrasing here of course - "I like it. Sounds great."

Just my $.02.

- Jeff
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post #17205 of 71784 Old 08-05-2009, 01:25 PM
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Originally Posted by audyssey View Post

It's not that frequency domain analysis and correction doesn't work. One can get reasonable results with such methods. But, not all measurement methods are equal. We spent nearly 8 years in my lab at the university trying to figure out what works and what doesn't before even dreaming of creating an automatic method.

A frequency-domain method that averages spatially and temporally (i.e over multiple measurements at the same spatial location in order to increase signal to noise ratio) has been proven to give good results. The test signal should not be a pure sine because of the extreme sensitivity to room modes. Pink noise typically gives the best results assuming the measurement software can perform the spatial and temporal averaging mentioned above.

Better results can be obtained by using what is called a system identification approach. The idea is to excite the system (in this case the speaker-room system) with a known signal, then measure the response and deconvolve it (a fancy word for dividing in the time domain) with the known signal. The result is called an impulse response and it can be shown that it fully describes the behavior of the system in time and frequency.

The benefit of impulse response data is that they contain information about the direct sound as well as the reflected sound. A pink noise method is "time blind" and just looks at the end result. Measuring the amplitude of sine tones is also a time blind approach... unless you can play a sine sweep that covers all audible frequencies and then deconvolve it to get an impulse response. That's what MultEQ does. The signals you hear are a sweep of sine waves with an additional twist: they are weighted to linger longer in the low frequencies by spacing them logarithmically in frequency. That gives us more low frequency info where the problems are more serious in small rooms.

Now comes the tricky part. If you read almost any signal processing book, you will find that system identification can be done with a single measurement. Not so in room acoustics. Multiple measurements are required to capture the distribution of problems in the listening area. But, we can only make a single filter for each speaker. So, the measurements have to be combined somehow and we came up with a way to do that while providing more weight to the bigger problems first.

Getting back to your room: it sounds like you have length and width almost the same and close to 17'. That is a pretty big problem to overcome because the modes that cause anomalies in the low frequencies are being reinforced by both of your room dimensions. These problems are best addressed by first applying room treatment. A tuned absorber in the corner can do wonders for that 35 Hz peak, particularly because it diminishes its decay time. A parametric filter cut does not do that. It just diminishes the initial energy. The decay time stays the same. But, a time domain FIR filter can in fact diminish the decay as can be seen when you plot a time-frequency spectrogram. Some of the commercial tools out there let you do that.

I love it - not hard to follow and fairly straightforward to understand (not that I'd now or ever be able to develop a system to compete with Audyssey - have no fears). So, in my effort to become better educated about temporal correction a couple of additional questions about the above:
  1. Is a "time domain filter" and an "FIR filter" one and the same?
  2. Either way, simply put, is a time domain FIR filter a "model" based on what is needed to correct a given frequency's peak/dip over time (i.e., it varies the correction over the decay time as detected from the FIR test impulse in order to reduce/increase that decay time to match it against some reference and/or against that of the other frequencies in the system)?
  3. I assume that Audyssey's corrections are not "active" in that it can't dynamically change the correction should the actual peaks/dips and/or decay times vary from that based on the FIR impulses (i.e., like what I assume a feedback destroyer does). Since there is no feedback mechanism (i.e., a microphone) during normal use of the system I guess the answer to this is a "duh"?
  4. In your statement above "Some of the commercial tools out there let you do that." are you saying that there other non-Audyssey products that allow the user to set up time domain corrections (as opposed to what a PEQ does)...not that I am asking you to advertise for your competitors..., or are you simply saying that REW, et.al. can plot a time-frequency spectogram (aka a "waterfall graph")?
Thanks!
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post #17206 of 71784 Old 08-05-2009, 01:37 PM
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Quote:
Originally Posted by pepar View Post

It is good to have a definitive statement on this issue. I have found myself getting scooped up in the hubbub of measuring, especially with the AS-EQ1 which generates before and after graphs that, much to Mr. Mullen's chagrin, no one can duplicate with the tools the average enthusiast - and even the not so average acoustician - has available.

Yes. Experimentation in the pursuit of knowledge, experience, and proficiency is a good thing. The trick is being able to understand when you have reached a level of knowledge that meets your target, and continue moving forward rather than in a never-ending circle. At least that's my objective.
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post #17207 of 71784 Old 08-05-2009, 02:10 PM
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Originally Posted by av-ra View Post

  1. Is a "time domain filter" and an "FIR filter" one and the same?
  2. Either way, simply put, is a time domain FIR filter a "model" based on what is needed to correct a given frequency's peak/dip over time (i.e., it varies the correction over the decay time as detected from the FIR test impulse in order to reduce/increase that decay time to match it against some reference and/or against that of the other frequencies in the system)?
  3. I assume that Audyssey's corrections are not "active" in that it can't dynamically change the correction should the actual peaks/dips and/or decay times vary from that based on the FIR impulses (i.e., like what I assume a feedback destroyer does). Since there is no feedback mechanism (i.e., a microphone) during normal use of the system I guess the answer to this is a "duh"?
  4. In your statement above "Some of the commercial tools out there let you do that." are you saying that there other non-Audyssey products that allow the user to set up time domain corrections (as opposed to what a PEQ does)...not that I am asking you to advertise for your competitors..., or are you simply saying that REW, et.al. can plot a time-frequency spectogram (aka a "waterfall graph")?
Thanks!

[1]No. A "time domain filter" is one that is represented by a time-domain impulse response rather than a frequency domain magnitude response. An IIR filter can also be represented by an impulse response. However, for what I was talking about above I was, indeed, referring to FIR filters. The reason I use that notation is that the room data collection and combination is performed in the time domain.

[2]Sort of. An FIR filter doesn't vary with time, but it has a "time signature". Getting further into this would probably derail the thread and people would start glazing over the equations... There are many basic DSP books and references available online that will give you more background info on FIR filters.

[3]Duh

[4]I was not referring to correction tools. I was referring to measurement tools and the fact that some of them let you plot waterfall (time-frequency) plots that will give you an idea of modal decay.

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post #17208 of 71784 Old 08-05-2009, 02:15 PM
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Quote:
Originally Posted by audyssey View Post

[1]No. A "time domain filter" is one that is represented by a time-domain impulse response rather than a frequency domain magnitude response. An IIR filter can also be represented by an impulse response. However, for what I was talking about above I was, indeed, referring to FIR filters. The reason I use that notation is that the room data collection and combination is performed in the time domain.

[2]Duh

[3]I was not referring to correction tools. I was referring to measurement tools and the fact that some of them let you plot waterfall (time-frequency) plots that will give you an idea of modal decay.

Chris, I think you misnumbered your replies (3->4) and left one out?
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post #17209 of 71784 Old 08-05-2009, 02:24 PM
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Originally Posted by rickardl View Post

Chris, I think you misnumbered your replies (3->4) and left one out?

oops. Corrected above.
Thanks

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post #17210 of 71784 Old 08-05-2009, 02:48 PM
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Hi again Chris, all,

If you'll recall my posts from last week about hearing definite differences with bitstream vs PCM from the PS3 (regular DVDs) to my Denon 1909, thinking it only happened with Dynamic EQ, but then also noticed it without DEQ. So I'd started a separate thread to try figuring out what's going on... But now I'm correctly back here!

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Originally Posted by batpig View Post

you should NOT hear a difference between bitstreamed DD and pre-decoded DD (sent as multich PCM) from the PS3. there is almost certainly some setting somewhere (e.g. DRC may be "on" for DD inputs) that is confounding the test.

I've done the test myself on PS3 + Denon setups (playing a DD track from an SD DVD) and they sounded identical.

either way, it's not really an "Audyssey" issue at all....

Last night I burned the DIY Audio Test DVD from a member here to hopefully get a better idea of what was going on. Played the pink noise from the mains and they were the same, bitstream or PCM, to my ears and the SPL meter. The LFE pink noise was, as I expected, noticeably louder with bitstream, and registered 3-5dB higher than PCM depending on where the meter was -- I found that weird, with the meter sitting in the same place(s), the relative difference varied?? Read on...

Then I played individual tones, and the one from a main speaker still sounded the same to my ears (I think) bitstream or PCM, but the meter showed almost a 10dB difference! Same exact spot, just the bitstream/PCM switch? Something changing with phase...?

Using the LFE tones, I started at 80Hz, and bitstream was 3dB louder than PCM (meter sitting about a foot from sub...). Going down to 75 or 70Hz, and bitstream/PCM were the same. I think somewhere Hz-wise PCM was a couple dB louder. Then at around 45 or 50Hz, bitstream was 10dB louder! Well that explains the difference I'm hearing in the movie tests.

I'm confused wondering what the heck is going on. It's like there was a totally wacky, random EQ being applied differently to bitstream and/or PCM somewhere along the line.

And then I found the cause, by doing something I'd never done: switched MultEQ to Off! Every test tone was then exactly the same, bitstream and PCM. And I went back to all the movie scenes I was checking, DD and DTS, and sure enough, with MultEQ Off, every aspect is exactly the same to my ears, bitstream or PCM. (Even the -4dB Dialnorm offset DD doesn't have an overall level difference AFAICT; wasn't sure what to expect there.) Was glad that I ruled out the PS3 and found the actual cause, but don't know what's next...

So I assume this is pretty messed up and MultEQ shouldn't be changing the sound between different signal types, right? All 3 MultEQ modes do this. Like Off, Manual is fine too, though I don't have any manual adjustments anyway. Ever heard of anything like this Chris?

Think I should do a microprocessor reset? It's hard for me to believe that would fix it, since everything else seems to be working fine. I've never reset before, but I assume I'd lose all settings and have to run Audyssey again -- neither a huge deal, and I just ran it the night before after adding a back speaker... Wanted to get some input here to see how to proceed.

Thanks! (Please don't say this is normal. )
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post #17211 of 71784 Old 08-05-2009, 03:12 PM
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Quote:
Originally Posted by DR_LaRRY_PEpPeR View Post

Hi again Chris, all,

If you'll recall my posts from last week about hearing definite differences with bitstream vs PCM from the PS3 (regular DVDs) to my Denon 1909, thinking it only happened with Dynamic EQ, but then also noticed it without DEQ. So I'd started a separate thread to try figuring out what's going on... But now I'm correctly back here!



Last night I burned the DIY Audio Test DVD from a member here to hopefully get a better idea of what was going on. Played the pink noise from the mains and they were the same, bitstream or PCM, to my ears and the SPL meter. The LFE pink noise was, as I expected, noticeably louder with bitstream, and registered 3-5dB higher than PCM depending on where the meter was -- I found that weird, with the meter sitting in the same place(s), the relative difference varied?? Read on...

Then I played individual tones, and the one from a main speaker still sounded the same to my ears (I think) bitstream or PCM, but the meter showed almost a 10dB difference! Same exact spot, just the bitstream/PCM switch? Something changing with phase...?

Using the LFE tones, I started at 80Hz, and bitstream was 3dB louder than PCM (meter sitting about a foot from sub...). Going down to 75 or 70Hz, and bitstream/PCM were the same. I think somewhere Hz-wise PCM was a couple dB louder. Then at around 45 or 50Hz, bitstream was 10dB louder! Well that explains the difference I'm hearing in the movie tests.

I'm confused wondering what the heck is going on. It's like there was a totally wacky, random EQ being applied differently to bitstream and/or PCM somewhere along the line.

And then I found the cause, by doing something I'd never done: switched MultEQ to Off! Every test tone was then exactly the same, bitstream and PCM. And I went back to all the movie scenes I was checking, DD and DTS, and sure enough, with MultEQ Off, every aspect is exactly the same to my ears, bitstream or PCM. (Even the -4dB Dialnorm offset DD doesn't have an overall level difference AFAICT; wasn't sure what to expect there.) Was glad that I ruled out the PS3 and found the actual cause, but don't know what's next...

So I assume this is pretty messed up and MultEQ shouldn't be changing the sound between different signal types, right? All 3 MultEQ modes do this. Like Off, Manual is fine too, though I don't have any manual adjustments anyway. Ever heard of anything like this Chris?

Think I should do a microprocessor reset? It's hard for me to believe that would fix it, since everything else seems to be working fine. I've never reset before, but I assume I'd lose all settings and have to run Audyssey again -- neither a huge deal, and I just ran it the night before after adding a back speaker... Wanted to get some input here to see how to proceed.

Thanks! (Please don't say this is normal. )

I don't have a PS3 but I know that you can specify the resolution and
I think if you have it more than 96kHz, Audyssey in the Denon will not be operational?
Could that explain the difference between bitstream and PCM with MultEQ On
when it would actually mean bitstream with MultEQ On and PCM with MultEQ implicitly Off?
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post #17212 of 71784 Old 08-05-2009, 03:17 PM
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Dr. Pepper -- that is definitely not "normal"!! I can't speak for Chris but there is no way there should be different EQ filters for PCM vs bitstreamed content.

Something is fishy... Rickard's suggestion is a good one but AFAIK Denon's do not have any limitation with using Audyssey MultEQ at high bitrates (as do some other models e.g. Marantz).

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post #17213 of 71784 Old 08-05-2009, 03:23 PM
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Thanks! (Please don't say this is normal. )

This is a stumper...It's not normal and it's definitely not because of the MultEQ filters because they don't have any gain. The trim levels that MultEQ found remain in effect even when you turn MultEQ off.

I will send the question to Denon and see what they say.

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post #17214 of 71784 Old 08-05-2009, 03:35 PM
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@rickardl -- No, Audyssey is indicated as being on in both cases. And in the Denon's Status it still shows 48kHz sampling rate for PCM too. I thought that PS3 "upsampling" thing is for audio CDs or such, and I assumed MultEQ could still be used above 96kHz... Hmm, well that's not part of my issue anyway.


BTW, I forgot to mention before that MultEQ On (as I always had) with PCM sounds fairly similar to MultEQ Off with either type of signal. Whether MultEQ+PCM is slightly "dulled" a bit (vs MEQ Off) is hard to say, but MultEQ+bitstream definitely has the greater impact as I said previously. Don't know if that's messed up by being too loud, or if everything else is too "quiet"...



Oh, and thanks Chris! For the confirmation and if you contact them. (Yeah, I know the speaker levels are still used with it Off.) Let me know if someone thinks the microprocessor reset might fix things.
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post #17215 of 71784 Old 08-05-2009, 03:50 PM
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...[2]...An FIR filter doesn't vary with time, but it has a "time signature". Getting further into this would probably derail the thread and people would start glazing over the equations... .

Thanks - that's plenty clear enough for my purpose without delving into Laplace transforms (if my attempt at sounding smart is correct) and I doubt I'll be asking for the name of a recommended DSP book/reference anytime soon (but don't hold your breath)....
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post #17216 of 71784 Old 08-05-2009, 04:05 PM
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Using the LFE tones, I started at 80Hz, and bitstream was 3dB louder than PCM (meter sitting about a foot from sub...). Going down to 75 or 70Hz, and bitstream/PCM were the same. I think somewhere Hz-wise PCM was a couple dB louder. Then at around 45 or 50Hz, bitstream was 10dB louder!

What frequency do you have the LPF filter of the LFE channel set to in the 1909?

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post #17217 of 71784 Old 08-05-2009, 04:15 PM
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120. But at any setting, it shouldn't make any relative difference between the 2 formats, right? Nor with MultEQ on or Off?
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post #17218 of 71784 Old 08-05-2009, 04:30 PM
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120. But at any setting, it shouldn't make any relative difference between the 2 formats, right? Nor with MultEQ on or Off?

It depends on how the PCM decoding is implemented.

For example, in the PCM case if the LFE lowpass filter is applied to the LFE track in the PS3 then the decoded LFE track would go straight to the MultEQ subwoofer filter in the 1909. But, in the Bitstream case the LFE track goes to the 1909 LFE LPF filter before passing through the MultEQ subwoofer filter.

It's a long shot, but would be interesting to set the LPF filter in the 1909 to 250 Hz and try the test again.

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post #17219 of 71784 Old 08-05-2009, 04:42 PM
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(Even the -4dB Dialnorm offset DD doesn't have an overall level difference AFAICT; wasn't sure what to expect there.)

They have the same level because whether the DD decoder is in the PS3 or the AVR, the Dialnorm offset is applied the same in either case. The PCM coming from the PS3 should be identical to the PCM coming from the AVR's internal decoder. It appears that's the case. I wish I could help with your actual problem!
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post #17220 of 71784 Old 08-05-2009, 04:45 PM
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Hi Jeff,

I believe that's what I said. Refer to highlighted statement.

It is effecting both the timing, which can be corrected by changing the distance in the AVR, and it is erroneously affecting the filters.

Larry

Larry, damn ... you guys are intense, i need some rest. all i do is test, report, ask questions, read and then repeat ...

so here's a pic of my famous Mic stand (homemade < $4). it is resting on the sofas plush cushion, so shouldnt mic vibrations be isolated or reduced ? followed by REW measurements before/after Audyssey correction.



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