so my bd-55 DOES not redirect LFE to large front and lefts when sub is set to no. This sucks as I don't have a sub but use large main speakers with built in subs. Am I really missing out big time or is the LFE encoded track 9 times out of 10 duplicated to the front left and rights anyway?Well, most engineers will definitely mix some of the same info that is in the LFE channel into the L/R channels for continuity and localization. Doubtful it goes as low very often. Or has the same energy. But something should usually be there in the fronts. Occasionally there may be something that is totally unique to the LFE channel. But engineers are not supposed to leave anything out of the front channels that is necessary to the conveyance of the movie's content.

How does yours sound?

For the record, I had to make the decision about a subwoofer in the early '90's because I wanted to hear the .1 channel. Smart marketing if u ask me. I own VMPS STIII's which have several 14, 12 and 10" bass Drivers. The system is biampable but I always thought that biamping was a bit silly in the home. All those expensive crossovers were designed very specifically. No external crossover is ever going to replicate these filters.

I bought an 18" Velodyne sub. WOW Once you hear the impact of the .1 channel, you never come back...

Well, in HDMI 1.1, DSD is certainly being manipulated before output -- it's being converted to PCM. HDMI 1.1. can't pass DSD. Once that conversion's done the signal is much more manipulable by the player....and if anything, I'd think it could 'fool' the downstream AVR into thinking the source was not SACD...and thus treat it like DVD-A/ PCM Dolby Digital.

For ilink/Denonlink, who knows?

I know the post I am replying to is 2 years old, but it seems like this is a good enough thread to ask about current experience with DenonLINK. I believe there is an issue with multichannel DSD relating to low LFE/Subwoofer. Any user experience on this would be greatly appreciated.

I am using a 3910 with the DL3 update, and a current 3808ci AVR. I use typical 6.1 settings with all main channels set to small, crossover 80 hz. Simply put, the sub level when playing multichannel SACD's over DL is very low.

Using the same DL connection for DVD-A or plain jane DD/DTS DVD's shows no problems as verified with test discs (Avia, Chesky Ultimate DVD-A) and an SPL meter.

Unfortunately, I have no such test tones recored in DSD/SACD, so I can only rely on my ears, but the sub channel seems very weak.

I experimented with the 3 methods I know of that the 3808 can deal with DSD (DSD Direct, Multi Direct, Multi In) and none seem to present the sub channel at the correct level. I find that I need to run the sub channel trim up to +12 to get the level close to what it sounds like when using the 5.1 analogs when playing the same content.

Anyone familiar with the 3808 will surely agree that it is a train wreck in terms of the manual and the set-up screens, so I will freely admit that there may be something in the set-up that I missed in this regard, or something that maybe I did not miss but perhaps is an unrelated setting that in fact influences the way the 3808 deals with DSD.

Anyway, if anyone has any experience with DSD over DenonLINK, I am all eyes ;-)

Brian

Hey Brian

The perceived SW level on my Denon 5803 is low whenever I playback SACD on the analog multichannel inputs and I need to really crank it up to get it right. If I switch the multichannel inputs to digital processing, the bass management automatically defaults to +15 on the SW channel and all is right again. I don't use the DL. Don't know if that helps.

Hey Brian

The perceived SW level on my Denon 5803 is low whenever I playback SACD on the analog multichannel inputs and I need to really crank it up to get it right. If I switch the multichannel inputs to digital processing, the bass management automatically defaults to +15 on the SW channel and all is right again. I don't use the DL. Don't know if that helps.

Well, anything will help, although it is backwards from what I am hearing. Maybe Denon fixed your issue in the 3808, and at the same time created mine? :D

The 3808 actually has a switchable sub channel boost. The manual suggests +15, which coincides with the recommendation in the original post of this thread when sending bass managed signals via analog 5.1.

As far as the 3808 goes, I think it is missing the sub channel boost when it gets DSD via DenonLINK. That assumes of course that the 3910 is not messing with DSD signals when it sends them via DenonLINK.

Brian

Well, anything will help, although it is backwards from what I am hearing. Maybe Denon fixed your issue in the 3808, and at the same time created mine? :D

The 3808 actually has a switchable sub channel boost. The manual suggests +15, which coincides with the recommendation in the original post of this thread when sending bass managed signals via analog 5.1.

As far as the 3808 goes, I think it is missing the sub channel boost when it gets DSD via DenonLINK. That assumes of course that the 3910 is not messing with DSD signals when it sends them via DenonLINK.

Brian


There are one or two SACD 5.1 test discs out there...I have this one though it's been ages since I used it:

http://www.amazon.com/exec/obidos/ASIN/B00005RTL7/ref=nosim/hometheater05-20


And this fascinating document describing Phillips SACD test discs suggests that SACD LFE is supposed to be recorded +15dB hotter than the other channels THis implies that it is NOT expecting a boost at playback.

https://www.ip.philips.com/download_attachment/1096/sl00342.pdf

However, I suspect what's happening to you is what is noted in the sticky post: your player is cutting LFE level by 10-15 dB to stay within its headroom limits, expecting that the cut will be restored downstream in your system at point where headroom is greater. But your Denon AVR isn't adding the expected boost, so your SACD LFE, which should be +15dB higher than the other channels at output, is actually at or near the same level as them -- not what the engineer intended.

As this 2005 article (http://www.ultimateavmag.com/howto/805bass/index1.html) notes,

...the digital-to-analog (D/A) converters on disc players' subwoofer outputs often do not have enough headroom to output bass at sufficiently high levels. The level of the subwoofer output is usually reduced by either 10dB (for a player without bass management) or 15dB (for a player with bass management), with the expectation that the signal will be boosted by the controller. This is all well and good except that many controllers are not set up to apply that boost to their analog subwoofer inputs!

Which is another way of talking about the LFE Bug subject.

As to what happens over a digital interconnect, I can only offer the experience I had with my Oppo 970 and my Pioneer AVR, using HDMI. The AVR had the LFE bug, and the former cut the LFE down for PCM sources (not sure by how much, but at least 10dB) even via digital connection. But there is a digital slider the user can access in the Oppo, that lets you raise or lower any of the channels by 10 dB. Pushing that up fixed the bug.

Since then the AVR has been firmware-upgraded, and I reset the LFE slider to 0 dB boost/cut. It shouldn't require the slider adjustment...though I have not rigorously checked this. Something to add to my do-list.

...fascinating document describing Phillips SACD test discs suggests that SACD LFE is supposed to be recorded +15dB hotter than the other channels THis implies that it is NOT expecting a boost at playback...
I thought best mastering practice was don't use the LFE channel AT ALL for bass content of multi-channel SACD/DVD-A. Rather the bass content should be in the five main channels, relying on bass management of the playback chain to separate it.

If that had been followed, there would be no ambiguity about LFE recording level vs playback boost.

Does anybody know what the common practice is for multi-channel SACD/DVD-A music material? Do content producers use LFE for bass or put all the bass in the five main channels? Or is it just random -- whatever the individual recording engineer decided that day?

I thought best mastering practice was don't use the LFE channel AT ALL for bass content of multi-channel SACD/DVD-A. Rather the bass content should be in the five main channels, relying on bass management of the playback chain to separate it.

If that had been followed, there would be no ambiguity about LFE recording level vs playback boost.

Does anybody know what the common practice is for multi-channel SACD/DVD-A music material? Do content producers use LFE for bass or put all the bass in the five main channels? Or is it just random -- whatever the individual recording engineer decided that day?

There is so good comment in this thread http://www.avsforum.com/avs-vb/showthread.php?t=1114082 from Roger Dressler of Dolby.

I think the abridged version is "The mixing standard is that there is mixing standard".

Brian

However, I suspect what's happening to you is what is noted in the sticky post: your player is cutting LFE level by 10-15 dB to stay within its headroom limits, expecting that the cut will be restored downstream in your system at point where headroom is greater.

But your Denon AVR isn't adding the expected boost, so your SACD LFE, which should be +15dB higher than the other channels at output, is actually at or near the same level as them -- not what the engineer intended.



I would agree; its most likely a fault in the AVR relative to SACD. Fortunately, the 3808 via DenonLINK seems to deal with DD, DTS, and DVD-A (PCM) just fine, so one would think that if there is a problem, it could be fixed in firmware.

FWIW, I have logged a problem with Denon, although if I do get a reply from someone that has any idea what I am talking about, I will probably have a stroke. :D

Brian

I thought best mastering practice was don't use the LFE channel AT ALL for bass content of multi-channel SACD/DVD-A. Rather the bass content should be in the five main channels, relying on bass management of the playback chain to separate it.As I pointed out in another thread today this is definitely NOT the case. Almost every SACD and DVD-A that I own contains a specifically encoded LFE channel.


If that had been followed, there would be no ambiguity about LFE recording level vs playback boost.Agreed. Unfortunately that is not how the vast majority of SACDs and DVD-As are mixed.


Does anybody know what the common practice is for multi-channel SACD/DVD-A music material? Do content producers use LFE for bass or put all the bass in the five main channels? Or is it just random -- whatever the individual recording engineer decided that day?See answer in that other thread.


I think the abridged version is "The mixing standard is that there is mixing standard".I think that Brian meant to say "The mixing standard is that there is NO mixing standard".

fwiw, i have logged a problem with denon, although if i do get a reply from someone that has any idea what i am talking about, i will probably have a stroke.Good luck with that. More likely that it would cause THEM to have a stroke.

I think that Brian meant to say "The mixing standard is that there is NO mixing standard".

Yes of course! Thanks for the correction.

Brian

This is a table from the Dolby Guidelines:

http://home.comcast.net/~schiz/multi DVD.jpg

I'm still trying to figure out what "* Simplified design option" refers to.It means that the bass management can be of simpler design in a source product than that which is required in an AVR. For example, when "no sub" is chosen in an AVR, the LFE should be remixed to L/R. In a player, it can just be omitted.

Just FYI, DD+ added extra flexibility so that 2-ch downmixes can have LFE mixed in as defined by the downmix parameters. It can also mix to a PLII style Lt/Rt.

Inspired by this thread, I've been trying for the past two days, with limited success, to get Oppo to explain to me how the 970HD treats PCM LFE. To recap, I reported some odd measurements a few years back on this thread in this post (http://www.avsforum.com/avs-vb/showpost.php?p=9739269&postcount=203), like a 20dB disparity between the subwoofer channel output levels in bitsream vs LPCM modes. Maybe you guys can help me parse Oppo's responses -- bear with me if you can, I'm included the entire exchange so far

I initially wrote:
Dear Oppo,
I'm trying to understand how the Oppo handles channels levels for various media under various conditions. Specificaly, how the LFE channel is treated

As a preliminary, I see that the 'sliders' for the six channesl are all defaulted to the middle position , with +/- 10 setting above and below. I presume the equate to dB?

My understanding is that Dolby Digital (and most DTS) LFE is recorded at -10dB relative to its final output level, which is supposed to be +10dB relative to other channels. So in other words, on the disc, the LFE is at 0dB relative to other channels, expecting a +10dB boost before it it finally sent to the sub.

I have measured the levels of LFE from a test disc (Chesky 'Ultimate Surround Test Disc') and found several oddities between LFE decoded in the Oppo (passed as LPCM), versus as a bitstream decoded in my AVR.

In all cases, the Oppo was set to leave all bass management OFF, distances equal, channel level sliders at factory default position, and my AVR was set to 'pure direct' mode (no DSP , bass management, etc); an HDMI connection was used to pass the signal to the AVR. THere was no attempt to balance channels. All readings (in dB) of Left, Right, and Subwoofer outputs were taken from directly in front of the loudspeakers with a radio shack analog SPL meter, C-weighted. My AVR did NOT apply a 10dB boost to PCM LFE input at the time of measurement (that requires a firmware update) - in other words, it had the 'LFE bug'.

condition|left|sub|right|AVR readout

DD(bitstream) |72 |94-96 |78.5 |Dolby Digital
DD(LPCM) |68.5 |74-76 |75 |PCM direct

Things I don't understand:

1) I would have expected that when the Oppo is merely passing a bitstream, the sub output should measure 10dB louder than the other channels; here, the test signal measured ~20dB louder.

2) When decoding the DD signal, and then passing it as PCM, I would expect the Oppo could either
a. boost the LFE signal +10dB before output to the AVR (which would make it +10dB relative to other channels when it left the Oppo), expecting that the AVR would NOT add another 10dB. This does not look to be happening here. I am told that players typically do NOT add 10dB to LFE, as there isn't the headroom in their output or connections for it;
b. leave the LFE signal as is (at 0dB with respect to other channels when it left the Oppo) , expecting that the AVR WOULD add another 10dB to the LFE signal. This is what my results look like (given that my AVR does NOT add that 10dB to a PCM input LFE signal) - the test signal measures at approximately the same level in all channels at output.

3) By the same token , if 2b is correct, I would expect the PCM LFE version would be -10dB compared to the bitstream LFE output. But again I see a 20dB difference instead.

4) Finally the fact that the channel level sliders have at least "+10" levels of positive play, even in the LFE/sub channel , suggests that the Oppo is by default reducing the level for PCM signals; otherwise I don't see how you could let the user add as much as 10dB to the output (unless the slider levels don't stand for dB?).

So, what *is* the Oppo doing to Dolby Digital LFE when passing it as bitstream vs PCM?

Oppo replied (within hours, as per their usual excellent customer support):


With a bit streamed signal the DVD player is doing no audio processing. All processing must be done by the receiving equipment. In a Pure or Pure Direct mode, the receiver will not do any bass management, channel trim adjustments, or channel distance calculations. For this reason, depending on the location of your subwoofer, and the subwooder's native volume control, the volume of the subwoofer is much higher than it normally would be when the receiver is processing the signal.

With the player doing the processing, we do not add any adjustments to the LFE channel. The Channel Trim adjustments allows for each channel to be trimmed -/+10dB. By a default, for Dolby Digital sources, you may have to increase the subwoofer output 10dB to compensate for encoding practices.


(From this I gleaned two bits of confirmation: that the channel sliders were calibrated in dB, and that the Oppo indeed has the 'LFE Bug' -- it does NOT boost the LFE by +10dB. But that is typical of DVD players. However, it didn't answer my main questions, so)

I wrote:
Thanks very much for this information. I'm going to explore whether it's just my volume setting on my subwoofer, or some peculiarity of the Chesky disc, that accounts for the larger-than-expected difference in LFE levels between channels and between output modes. (Sub location shouldn't matter much in these tests because I'm always taking the reading with the SPL meter microphone positioned just an inch or so away from the center of the woofer.)

I'm still unclear though -- how does the 970HD avoid an output 'digital overload' for non-bitstream sources, if it allows users to add as much as +10dB to output of any or all channels? How much headroom does the player have for PCM output and how is this achieved?

Oppo quickly replied:
Optical and Digital Coaxial will not be effected by the overall Channel Trims, as optical and coaxial when transmitting PCM can only do Stereo. The only way to efficently use the channel trims is to take advantage of the multi-channel analog or the HDMI (set to LPCM) outputs.

...which still didn't answer my question, so I rephrased it in a reply:
Sorry, I guess I wasn't clear. I am only using HDMI output, and let's assume that am talking about a 5.1 LPCM signal (e.g., Dolby Digital decoded to LPCM; or DVD-Audio unpacked to LPCM; or SACD DSD converted to LPCM); under these conditions my question is how it is that the 970HD can offer +10dB of channel boost on any or all six channels, without risking digital overload on that channel. For example, if the recording already has peaks near 0dBFS in one or more channels, as is usually the case, wouldn't digitally boosting that channel by +10dB result in signal clipping?


Oppo replied:
The hardware in the player has been designed to digitally alter the output levels. Any digital manipulation of audio will result in some truncation or another, with out without an overload. Much like Sound Fields and Equalizations, we wanted the player to be as robust as possible in terms of user selectable options. It is up to the end user to determine what functionality they will take advantage of.


!!! from which I glean that the Oppo does not employ dither during digital manipulation...

I think I've kind of hit a wall here; I still can't explain my readings from the Chesky disc (although I probably need consider the subwoofer's own 'native' volume setting --its volume knob). I assume my AVR processes bitream DD LFE correctly, so it adds 10dB before output. I read a subwoofer output level of 94dB on my SPL meter with the Chesky DD test tones passed as bitstream, so the LPCM version, from decoding in the player, should have read 84 dB, since both the player and AVR had the PCM LFE bug. But instead I read 74 dB, suggesting that the Oppo attenuates the LPCM signal by -10dB. Now in their emails either Oppo is saying 'yes, you can drive output to distortion if you move those sliders up, but we want to give users that option', which frankly seems bizarre to me, or there is something else going on that for whatever reason, they just don't want to describe (like, the default attenuation I'm suggesting).

What do you guys think? Am I just asking the wrong questions?

From this I gleaned two bits of confirmation: that the channel sliders were calibrated in dB, and that the Oppo indeed has the 'LFE Bug' -- it does NOT boost the LFE by +10dB. But that is typical of DVD players. The Oppo does not have an LFE bug. Players are not supposed to alter the levels of the source at all. If all the sliders are at default, that's exactly what happens. It's perfect.

I think I've kind of hit a wall here; either Oppo is saying 'yes, you will drive output to distortion, but we want to give users that option', or there is something else going on that for whatever reason, they just don't want to describe (like, default digital attenuation of the signal in all channels, to allow users to apply as much as +10dB of boost). And I still can't explain my readings, although clearly I need to factor in the subwoofer's own 'native' volume setting (its volume knob). If you "boost" any of the channel trims, it will apply a global reduction in gain (in the amount of the channel with the highest boost) to all the channels so as to avoid clipping the boosted channels. It would be electrically the same to add 3 dB to C or to apply -3 dB to L, R, Ls, Rs, LFE. It's just easier to boost one control.

What do you guys think? Am I just asking the wrong questions? Depends what you are trying to find out.

If your AVR is able to apply gain trims to the analog MCH inputs, then that is a better option than using the Oppo trims. But some AVRs have no such trims, and in that case, the Oppo can faclitate matching it's outputs to the gain trims in the AVR's digital section.

And this fascinating document describing Phillips SACD test discs suggests that SACD LFE is supposed to be recorded +15dB hotter than the other channels THis implies that it is NOT expecting a boost at playback.

https://www.ip.philips.com/download_attachment/1096/sl00342.pdf
I looked at this doc, and I cannot see anything in there that implies the 15 dB difference is not supposed to be boosted upon playback. I suspect they encoded the LFE 15 dB hotter than L/C/R so that all outputs will have the same voltage in a player that activates bass management. It's not an ideal test disc for all occasions, however, as it has no tests where all 6 channels carry the same levels.

However, I suspect what's happening to you is what is noted in the sticky post: your player is cutting LFE level by 10-15 dB to stay within its headroom limits, expecting that the cut will be restored downstream in your system at point where headroom is greater. But your Denon AVR isn't adding the expected boost, so your SACD LFE, which should be +15dB higher than the other channels at output, is actually at or near the same level as them -- not what the engineer intended. If the player is connected via HDMI, then it should not use bass management, and that will deactivate any attenuation in the sub output. If the connection is via analog, then the -15 dB factor comes into play. But I understood the Denon is able to apply 15 dB for analog sub input. No?

krabapple- OK, I admittedly only skimmed your thread, but I agree with Roger that the Oppo does not have the LFE bug. And ... let me tell you a little story. :) Applicable or not.

Got a 983 a while back. I couldn't get the levels to jive between the SACD test disc I had and the DVD-A test disc I was using. Long story short is that the LFE/sub levels on the Chesky DVD-A/V disc are 10 dB too low. For analog DVD-A output anyway. One DVD-A test disc that has the correct levels is the Denon Sonic Boom disc. The Denon disc matched up with the test signals that Oppo was testing with with a Tchaikovsky 1812 DVD-A (or something) and the SACD I was using (DMP Surround something).

All of this is with DVD-A and SACD though. DVD-V operates as I expect it to even in relation to SACD and DVD-A. But I go bitstream too. (No HDMI for audio, so I've never tried PCM.)

fwiw.

The Oppo does not have an LFE bug. Players are not supposed to alter the levels of the source at all. If all the sliders are at default, that's exactly what happens. It's perfect.

Understood, and that's why I hedged my bets and put LFE Bug in quotes, and described it as typical. It was also common in AVRs, but really a bug, there. (Though if a player is doing bass management , it *should* alter the source levels, as described in the sticky post on this thread.)

If you "boost" any of the channel trims, it will apply a global reduction in gain (in the amount of the channel with the highest boost) to all the channels so as to avoid clipping the boosted channels. It would be electrically the same to add 3 dB to C or to apply -3 dB to L, R, Ls, Rs, LFE. It's just easier to boost one control.

That is the sort of answer I was hoping for from Oppo. To avoid clipping it seemed to me that gain *reduction* is what had to be happening, even though one appeared to be 'increasing' a channel's level.


Depends what you are trying to find out.

Trying to understand the subwoofer output level measurements I reported way back when, with pure LFE as input (no bass management) -- summarized in part in the last paragraph of my last post, but best viewed in that original post from late '07. This led to questions about what different parts of the signal chain are doing ; I started with the Oppo.

If your AVR is able to apply gain trims to the analog MCH inputs, then that is a better option than using the Oppo trims. But some AVRs have no such trims, and in that case, the Oppo can faclitate matching it's outputs to the gain trims in the AVR's digital section.

As I said, analog connections don';t come into the picture here; everything is via HDMI. And in fact, I used the Oppo subwoofer channel slider to compensate for my AVR's LFE bug, before I got the AVR's firmware upgraded. But the 2007 measurements from the Chesky disc still did't make sense to me...there's still 10dB of subwoofer level unaccounted for. Kevin's testimony that the Chesky disc LFE itself isn;'t mixed to DD standard provides an explanation I can test.

krabapple- OK, I admittedly only skimmed your thread, but I agree with Roger that the Oppo does not have the LFE bug. And ... let me tell you a little story. :) Applicable or not.

Got a 983 a while back. I couldn't get the levels to jive between the SACD test disc I had and the DVD-A test disc I was using. Long story short is that the LFE/sub levels on the Chesky DVD-A/V disc are 10 dB too low. For analog DVD-A output anyway. One DVD-A test disc that has the correct levels is the Denon Sonic Boom disc. The Denon disc matched up with the test signals that Oppo was testing with with a Tchaikovsky 1812 DVD-A (or something) and the SACD I was using (DMP Surround something).

All of this is with DVD-A and SACD though. DVD-V operates as I expect it to even in relation to SACD and DVD-A. But I go bitstream too. (No HDMI for audio, so I've never tried PCM.)

fwiw.


Great to know, thanks. I have the DMP disc and possibly have DD test tones somewhere other than the Chesky disc...so hopfully I can check this out on my 970..

...how the 970HD treats PCM LFE...a 20dB disparity between the subwoofer channel output levels in bitsream vs LPCM modes....
My 980H also has very weak bass on SACD and DVD-A when connected to my Denon AVR-3808CI via bitsream over HDMI. When I switch the interface to LCPM, it sounds OK.

All SACD/DVD-A discs display that "weak bistreamed bass" characteristic on my system, so it's not how one particular disc is mixed.

I haven't measured it, but -20db sounds about right.

I just gave up and left the player/AVR interface in LCPM.

My 980H also has very weak bass on SACD and DVD-A when connected to my Denon AVR-3808CI via bitsream over HDMI. When I switch the interface to LCPM, it sounds OK.

All SACD/DVD-A discs display that "weak bistreamed bass" characteristic on my system, so it's not how one particular disc is mixed.

I haven't measured it, but -20db sounds about right.

I just gave up and left the player/AVR interface in LCPM.Hmmmm. That doesn't make too much sense. Have you calibrated the HDMI connection with something you passed to the receiver as bitstream?

That receiver should have no issues with the LFE/subwoofer channel with bitstreamed content. And the player is only bitstreaming, so it's not an issue with the player.

My 980H also has very weak bass on SACD and DVD-A when connected to my Denon AVR-3808CI via bitsream over HDMI. When I switch the interface to LCPM, it sounds OK.

All SACD/DVD-A discs display that "weak bistreamed bass" characteristic on my system, so it's not how one particular disc is mixed.

I haven't measured it, but -20db sounds about right.

I just gave up and left the player/AVR interface in LCPM.

There are separate issues here... the 20dB disparity referred to in the quote you used, was between bitstream and PCM modes of plain old Dolby Digital, which definitely shouldn't be happening in a standard Dolby Digital 5.1 mix; there should be only a 10dB disparity (if the AVR has the LFE bug; no disparity if it doesn't). Based on Kevin's report, the culprit is probably Chesky (their 'bass management' test track never worked right for me either, so I would not be shocked to find that they didn't adhere to DD recommendation for LFE level).

The other disparity was +/-10dB between LFE of DVD-A and LFE of DD, again based on the Chesky disc as test signal. (DVDA LFE was +10dB compared to LPCM DD, and -10dB compared to bitstream DD)

Trying to understand the subwoofer output level measurements I reported way back when, with pure LFE as input (no bass management) -- summarized in part in the last paragraph of my last post, but best viewed in that original post from late '07. This led to questions about what different parts of the signal chain are doing ; I started with the Oppo. I went back to your 07 post and based on your data, I have a lunch in 20 minutes--my hat being the main course!

Specifically, the data that disturbs me is the change in output levels between PCM with Trims 0, and PCM with Trims +10. The outputs all increase exactly 10 dB in SPL. [The gain reduction method I described was based on how AVRs apply gain trims--so it was my mistake to assume Oppo did the same.] Oppo seems to confirm this when they said “The Channel Trim adjustments allows for each channel to be trimmed -/+10dB.”

What they do not tell you is what gain is being applied at the 0 dB point. Is it 0 dB, -10 dB, or something else? Well, there is an indirect way to deduce it. With Trims 0, the PCM L/R levels are 3.5 dB lower than the DD bitstream levels, and with Trims +10, they are 6.5 dB louder (10 dB change as expected). However, it seems odd to me that a PCM signal via HDMI can play louder than the bitstreamed source. I have not seen an AVR apply gain to a PCM source—as it must assume it can use the full range to 0 dBFS. As a result, I have to assume that the Oppo is in fact able to add gain digitally. Looks like they chose 6 dB max gain. That would mean that the 0 dB setting is applying 4 dB attenuation.

If all this conjecture is true, then one could set the Trims to +4 dB and still avoid any clipping. But one issue remains—the LFE levels. Looking at the sub level difference between PCM and DD bitstream, your data shows it to be 16.5 dB pretty consistently (not 20 dB, since the 3.5 dB difference in main channels has to be included). You mentioned the Pioneer had the LFE bug. If so, that would account for 10 dB of the disparity. The remainder is 6.5 dB. Why the Oppo would attenuate the LFE by 6.5 dB relative to the other channels is a mystery. The Panasonic BD30 applied a 5 dB reduction, but eventually issued new software to fix that. So such things can and do happen. I cannot square this data with their assurance that “we do not add any adjustments to the LFE channel.”

And in fact, I used the Oppo subwoofer channel slider to compensate for my AVR's LFE bug, before I got the AVR's firmware upgraded. But the 2007 measurements from the Chesky disc still did't make sense to me...there's still 10dB of subwoofer level unaccounted for. Kevin's testimony that the Chesky disc LFE itself isn;'t mixed to DD standard provides an explanation I can test. Your 07 data also shows that the LFE level is exactly 10 dB lower relative to the main channels in the Chesky DVD-A program than the DD program, so I think Kevin is correct that the disc was recorded with this offset.

You might still need to apply some LFE boost in the Oppo if the strange 6.5 dB factor is real. One way to check this out is to play a DD soundtrack in DD mode. Note the bass levels relative to the overall mix. Now play the same in PCM mode, at the same overall loudness. Does the bass sound the same?

I looked at this doc, and I cannot see anything in there that implies the 15 dB difference is not supposed to be boosted upon playback. I suspect they encoded the LFE 15 dB hotter than L/C/R so that all outputs will have the same voltage in a player that activates bass management. It's not an ideal test disc for all occasions, however, as it has no tests where all 6 channels carry the same levels.

I didn't see this post last night, sorry.

OK, tell me if I where I've gone wrong here: DD standard is to record the LFE such that, upon output to the sub, which is to say after the expected +10dB increase , it is not only +10dB louder than what was printed to the disc, but +10dB compared other channels (given an appropriate test signal). So relative to the other channels, LFE is at +0dB 'on the disc', and +10dB after proper decoding and processing. (When I tested my Oppo970 in 07 with the Chesky disc, the DD LPCM results seemed to accord with this model; since under these conditions neither my Oppo nor my AVR could be expected to add the 10dB that Dolby specifies, the inter-channel levels read about the same -- what's 'on the disc' is what's being output. But this may have been misleading, if the Chesky disc LFE level WASN'T recorded to DD standard; so I really need to try another DD test disc).


The figures given for SACD are -20dB for LCRSr, and -5dB for LFE. I interpret these as reports of what is printed to disc (which might be my mistake). So, if LFE is already @+15dB relative to other channels 'on the disc', if a +10dB boost is applied only to the LFE down the line, as for DD, then the final relative output will be +25dB (assuming no bass management) compared to LCRSr. That seems wrong to me.

Are these levels just a peculiarity of this Philips test disc? Would a more representative disc offer the same printed levels on all channels? If so, it's a situation like the CHesky disc, and I have to ask the gods above, WHAT IS THE POINT of offering these test discs that don't reflect standard practice? I can't afford to buy a copy of the Scarlet Book, so I don't know what the official recommendation for processing SACD LFE is.


If the player is connected via HDMI, then it should not use bass management, and that will deactivate any attenuation in the sub output. If the connection is via analog, then the -15 dB factor comes into play. But I understood the Denon is able to apply 15 dB for analog sub input. No?

Beats me. I use a Pioneer . :D

OK, tell me if I where I've gone wrong here: DD standard is to record the LFE such that, upon output to the sub, which is to say after the expected +10dB increase , it is not only +10dB louder than what was printed to the disc, but +10dB compared other channels (given an appropriate test signal). So relative to the other channels, LFE is at +0dB 'on the disc', and +10dB after proper decoding and processing. Correct.

The figures given for SACD are -20dB for LCRSr, and -5dB for LFE. I interpret these as reports of what is printed to disc (which might be my mistake). So, if LFE is already @+15dB relative to other channels 'on the disc', if a +10dB boost is applied only to the LFE down the line, as for DD, then the final relative output will be +25dB (assuming no bass management) compared to LCRSr. That seems wrong to me.

Are these levels just a peculiarity of this Philips test disc? Yes, they are. Since the Philips disc comes with no instructions on how to use it and what are the expected results, it's hard to know what they had in mind. I think it's a goofy disc.

Would a more representative disc offer the same printed levels on all channels? If so, it's a situation like the CHesky disc, and I have to ask the gods above, WHAT IS THE POINT of offering these test discs that don't reflect standard practice? I can't afford to buy a copy of the Scarlet Book, so I don't know what the official recommendation for processing SACD LFE is. The LFE level relative to the main channel is a matter of studio calibration, and not the delivery format or the audio codec. Dolby did not invent the +10 dB concept, that came from dubbing stage practice as an evolution from optical to mag recording and the transition to "baby booms" in the 70mm days. DD merely respected and maintained the standard. It has been Dolby's stated position that now that the practice has been defined and widely used, it should be applied universally to any 5.1 production and delivery format to ensure uniform interchange of sources and playback. Aside from the silly "music mode" LFE recording practices used by a some studios, that has been the case.

What is the point of these test discs? Good question. If they cannot tell you, it may be better to find an alternative.

I like what the Dolby CP 200 can do with its press the button for optical 35mm twin-track Dolby A/SR sends the lows off to the subs. 70mm mag Dolby and SR-D dts SDDS8 use discrete sub bass track information only in the subs while the large stage channels handle the remaining lower portions down to 30Hz no?

I keep the AVR set with bass manager OFF as lows are handled by the DCX2496 with a little help from the Eltax A 12-R sub 12” while the JBL 4645 handles the LFE.1 track.

In some case for Dolby stereo mixes I select sw-re-mix mode on the AVR to send lows to the JBL 4645, sounds good with Star Trek the motion picture original untouchable mix laserdisc to DVD-RW.

I like what the Dolby CP 200 can do with its press the button for optical 35mm twin-track Dolby A/SR sends the lows off to the subs. 70mm mag Dolby and SR-D dts SDDS8 use discrete sub bass track information only in the subs while the large stage channels handle the remaining lower portions down to 30Hz no?

I keep the AVR set with bass manager OFF as lows are handled by the DCX2496 with a little help from the Eltax A 12-R sub 12” while the JBL 4645 handles the LFE.1 track.

In some case for Dolby stereo mixes I select sw-re-mix mode on the AVR to send lows to the JBL 4645, sounds good with Star Trek the motion picture original untouchable mix laserdisc to DVD-RW.

Perhaps some day your word salad posts will actually add value to what is being discussed. But that day has not yet arrived.

What is the point of these test discs? Good question. If they cannot tell you, it may be better to find an alternative.

Phillips, along with Sony, was the parent of SACD. If they don't offer a proper calibration disc, that's just sad.

Perhaps some day your word salad posts will actually add value to what is being discussed. But that day has not yet arrived.
Cut the man some slack, mate. He's from Bournemouth. :D

Cut the man some slack, mate. He's from Bournemouth. :D

Yeah we speak Bournemouth, Dorset, Jive, over hare.:D Plus it was freezing-cold last-night,:( in the living-room as the power switched caked-up on the night-storage-heater and only just repaired it this afternoon.

I think a good ole few DCX2496 set-up in 19” rack along with the AVR might show a clearer perceptive of input vs output.

I only have one DCX2496 at the moment as I’m not made of money. I use it for LCR LF/HF and last year, patched the LFE.1 into it and tested a 100Hz tone into the unit where the other channels will be 0 and the LFE.1 at -10db was easy to see on the input and its action with (Flight of the Phoenix 2004) was just nutty!

Also it was easier to control and I can’t wait, thou it would appear I would have to wait until I have £160.00 pounds for another DCX2496.

The LFE.1 mostly passes into the BFQ2496 input doesn’t really go above -18db thou that was with (Master And Commander 2002). I’ll have to see where it goes with (Flight of the Phoenix 2004) as the levels where high.

I liked the audio commentary where the director, started raving about the Dolby mix on this insane six-track mix.:)

http://i279.photobucket.com/albums/kk123/IndianaJones34/FlightofthePhoenixboomchannelLFE-19.jpg

http://i279.photobucket.com/albums/kk123/IndianaJones34/FlightofthePhoenixboomchannelLFE-21.jpg

http://i279.photobucket.com/albums/kk123/IndianaJones34/FlightofthePhoenixboomchannelLFE-22.jpg

http://i279.photobucket.com/albums/kk123/IndianaJones34/FlightofthePhoenixboomchannelLFE-25.jpg

http://i279.photobucket.com/albums/kk123/IndianaJones34/FlightofthePhoenixboomchannelLFE-33.jpg

http://i279.photobucket.com/albums/kk123/IndianaJones34/FlightofthePhoenixboomchannelLFE-34.jpg

http://i279.photobucket.com/albums/kk123/IndianaJones34/FlightofthePhoenixboomchannelLFE-35.jpg

At 13 minutes 18 seconds the input on the BFQ2496 channel A is for the LFE.1 channel B is for sub bass extension of LCRS.

Channel A reaches at -5db, while AVR Kenwood KRF-X9050D THX select is at 0db on the fader. Level for the LFE.1 is at 0db. So I dare not rise it to +10db as that would exceed clipping by +5db.

At 14 minutes 37 seconds as the engine explodes under the stresses I see a -2db as the yellow light illuminates.

At 15 minutes 04 seconds as the plane starts to roll and pitch downwards the deep low pulse of pressure inside the cargo hold peaks steadily at -6db.

The Alesis RA300 bridge mode is handling this comfortably thou I wouldn’t mind a larger Beheringer 2.4KW amp for the sakes of headroom. But if I practice common sense there shouldn’t be any reason why the amp shouldn’t clip.

108dbc is plenty over the small JBL Control 5 LCR with a little sub bass extension help and the room buzzes! That black dude is going “whoa!!!!” LOL oh dear best thrilling plane crash since Alive thou that also is grizzly terrifying re-enactment of the real event.

Amazing most of the passengers walked out that hash environment alive.

And as to Hudson River, New York, plane crash my hat is to the captain that was brilliant piloting.:)

Also levels on the DCX2496 where calm around -20db. I might test this over again and patch the LFE.1 lead into DCX2496 and look this one over again just to be sure the levels weren’t being exaggerated by the BFQ2496 and I’ll make minor adjustments if this is the case with the DCX2496 readings with level trim on the output on the AVR.

Should have short 6 minute video uploaded around 6pm it’s a bit on the dark-side but it shows some of the action of the DCX2496 and BFQ2496 as well as snippets from (Flight of the Phoenix 2004).

:rolleyes:

where's the moderator?

:rolleyes:

where's the moderator?

I’m putting you on the ignore list and you do the same I’m done with you, rude person!

...DD standard is to record the LFE such that, upon output to the sub...it is not only +10dB louder...compared other channels (given an appropriate test signal)
Correct.
Roger, Dolby recommends the actual subwoofer SPL should be +10db vs the L/C/R channels, measured with an RTA, or +4 to +6db measured with an SPL. That's when playing pink noise during channel level calibration.

However -- many recommendations say calibrate the sub SPL using to 0db vs the L/C/R channels, or even less to account for roll-off in the SPL meter. Also, many AVR auto-calibration systems set subwoofer SPL to about 0db vs the main channels.

They both can't be correct.

Which is the correct practice when calibrating home equipment using pink noise?

(1) Set the subwoofer SPL for +10db vs the main channels if using an RTA ( +4 to +6db dBC SPL if using a meter), OR

(2) Set the subwoofer SPL for 0db vs the main channels.

I know you already answered this, just looking for further comment, since the "0db subwoofer calibration" method is so common.

Roger, Dolby recommends the actual subwoofer SPL should be +10db vs the L/C/R channels, measured with an RTA, or +4 to +6db measured with an SPL. That's when playing pink noise during channel level calibration.

However -- many recommendations say calibrate the sub SPL using to 0db vs the L/C/R channels, or even less to account for roll-off in the SPL meter. Also, many AVR auto-calibration systems set subwoofer SPL to about 0db vs the main channels.

They both can't be correct.

Which is the correct practice when calibrating home equipment using pink noise?

(1) Set the subwoofer SPL for +10db vs the main channels if using an RTA ( +4 to +6db dBC SPL if using a meter), OR

(2) Set the subwoofer SPL for 0db vs the main channels.

I know you already answered this, just looking for further comment, since the "0db subwoofer calibration" method is so common. In fact these two different concepts can indeed be correct and consistent. It all depends on the nature of the test signal being used.

If the +10 dB cal (method 1) is done correctly, then all frequencies in a wideband channel will have equal response via the bass managed sub/sat speaker system. The +10 issue gets confusing because it is not directly a subwoofer calibration matter, but a gain relationship relative to the main channels. That gain relationship is not a calibration topic for the end user, the +10 dB mixing ratios is preset in the bass management stage, even if the product also offers access to some sort of "+10, 0 dB" menu for dealing with goofy music recordings with non-standard LFE. If you set it to 0 dB, that is precise--there's no calibration other than what it says.

OTOH, the sub output gain is like any other channel trim, so it must be set based on acoustic output, like any other speaker in the system. For convenience, driving the sub with a signal that makes the needle move in the same reange as the main channels makes things easier to keep straight.

If the pink noise is generated in the AVR, or is in a test signal via one of L/C/R/Ls/Rs channels played by means of a bass manager, (with the main speaker muted) then method 2 works. If the test signal uses the LFE channel, and it is modulated to the same degree as the main channels (-x dBFS in every channel), then method 1 works.

Caveat! Both methods work best with an RTA. A similar "correction factor" needs to be applied to method 2 as shown in method 1 when wideband metering is used. If the AVR includes the mic and test signals, then all this may have been accounted for, so it can shoot for 0 dB across the board.

Here is an example of the FR of the calibration signals that are used in my receiver. The subwoofer test signal is injected via the LFE circuit in my receiver. If the subwoofer calibration signal is via LFE, if you lower the level of the LFE trim (not subwoofer trim) the subwoofer calibration signal will drop by the same amount.


http://i297.photobucket.com/albums/mm232/Red_Foreman/STR-DA4ES_Calibration_Signals-1.jpg




This following chart shows the in room RTA measurement of the speaker's frequency response when the receiver's internal test noise (shown above) was used. The subwoofer was calibrated at + 3 dB above the main speakers level (SPL C scale slow). The room has a peak in the 50 Hz area as can be seen in both FR curves. There is also a room dip in the 35 Hz area.


The SPL meter measures the subwoofer as 3 dB "hot". The RTA FR tells you a different story. The moral to the story is get close with the SPL meter, and then adjust the subwoofer level to suit your ears. Better yet, measure the FR with a software package.


Now my question about all of this is if you connect pretty much any type of player to this receiver, can I assume that this relative speaker level calibration will be maintained if the player in question is set up properly?



http://i297.photobucket.com/albums/mm232/Red_Foreman/STR-DA4ES_Calibration_Signals_In-1.jpg

Here is an example of the FR of the calibration signals that are used in my receiver. The subwoofer test signal is injected via the LFE circuit in my receiver. If the subwoofer calibration signal is via LFE, if you lower the level of the LFE trim (not subwoofer trim) the subwoofer calibration signal will drop by the same amount. Yes, this is exactly how LFE trim should work.

Just curious, is this posted in support or rebuttal of any specific prior post? There was no reference link to help my feeble brain.

Just curious, is this posted in support or rebuttal of any specific prior post? There was no reference link to help my feeble brain.He just likes to post crap. ;)

...If the test signal uses the LFE channel, and it is modulated to the same degree as the main channels (-x dBFS in every channel), then method 1 works...
IMO this is yet another reason why this area is so confusing. Beyond the complexity of just obtaining and using a calibration disc and meter (or RTA), the user must also know how that disc was mastered. Yet that is rarely known.

E.g, was the LFE channel used, or just broad-band 5-chan pink noise, assuming end-user bass management? If LFE is used, what level was printed on the disc? What is the player doing? Does the player LFE behavior change between bitstream vs LCPM? Does the amp change LFE level? What if the player-generated pink tones differ from the amp-generated pink tones, which in turn differ from the disc-generated pink tones? Which do you trust? It goes on and on.

It is incredible that something this simple in concept is so complicated in reality.

If you inspected many residential high-end multi-ch music systems, I wonder what % would have subwoofer level calibrated within +/- 30% of the proper value on SACD and DVA-A material? I'll bet a very low % would be correct.

Besides the format conflict between SACD and DVD-A, the configuration and calibration complexity was an extreme barrier to wide adoption.

I just hope any future Blu-Ray multi-ch music doesn't make the same mistakes.

IMO this is yet another reason why this area is so confusing. Beyond the complexity of just obtaining and using a calibration disc and meter (or RTA), the user must also know how that disc was mastered. Yet that is rarely known.

E.g, was the LFE channel used, or just broad-band 5-chan pink noise, assuming end-user bass management? If LFE is used, what level was printed on the disc? What is the player doing? Does the player LFE behavior change between bitstream vs LCPM? Does the amp change LFE level? What if the player-generated pink tones differ from the amp-generated pink tones, which in turn differ from the disc-generated pink tones? Which do you trust? It goes on and on.This is the reason AVIA uses bass managed bass to calibrate the subwoofer. Sidesteps any LFE issues. Of course, it assumes that the processor bass manages properly. Most do. So, if you get the level of the bass-managed bass properly, the LFE will follow.

When you use the disc, and pass it as both bitstream and LPCM (and/or analog, if applicable), you can see for yourself, first-hand, if there is a discrepancy between how your processor bass manages bitstreamed info vs. LPCM (and/or vs. analog).


If you inspected many residential high-end multi-ch music systems, I wonder what % would have subwoofer level calibrated within +/- 30% of the proper value on SACD and DVA-A material? I'll bet a very low % would be correct.There IS variability between the way that different processors may handle bass management and LFE. But a particular processor does not treat SACD or DVD-A any differently than any other codec. So, if you calibrate a particular processor with a disc like AVIA, you have calibrated the processor to the "proper value". The variability with SACD/DVD-A is in how the discs are encoded. And, as you are lamenting, there is NOTHING you can do about that. If there are differences between how your bitstreamed vs. LPCM'd vs. analog'd info is bass/LFE managed, these can at least be ascertained and even addressed with some degree of confidence. But the variability in the way that any individual SACD/DVD-A is encoded will always be there.

IMO this is yet another reason why this area is so confusing. Beyond the complexity of just obtaining and using a calibration disc and meter (or RTA), the user must also know how that disc was mastered. Yet that is rarely known.

E.g, was the LFE channel used, or just broad-band 5-chan pink noise, assuming end-user bass management?

well, you can usually determine if the LFE channel is used, by setting your system to 'pass through' mode (called "Pure Direct' in my AVR) -- in other words, turning all bass management off. If there's subwoofer output then the LFE channel contains a test signal...of some sort. ;)

well, you can usually determine if the LFE channel is used, by setting your system to 'pass through' mode (called "Pure Direct' in my AVR) -- in other words, turning all bass management off. If there's subwoofer output then the LFE channel contains a test signal...of some sort. ;)I think simply setting all the speakers to LARGE is probably a better way to initially ascertain this (if there even IS some question). What a processor does with digitally passed (bitstream or LPCM) info when in a "pure direct"-type mode can sometimes be tricky.

...a particular processor does not treat SACD or DVD-A any differently than any other codec. So, if you calibrate a particular processor with a disc like AVIA, you have calibrated the processor to the "proper value". The variability with SACD/DVD-A is in how the discs are encoded....
It's more than just the encoding. The signal path from player to output can differ between (say) SACD and a DD 5.1 disc, even using the sample player and processor. E.g, up until recently SACD generally required 6-ch analog interconnects, player did bass mgmt not the processor, etc.

In that paradigm a perfect calibration using AVIA didn't guarantee the proper calibration NOR the proper bass mgmt configuration for SACD/DVD-A.

With HDMI at least that's partially simplified, but SACD over HDMI came along long after the formats were essentially doomed.

In hindsight it was impossible for those formats to ever succeed, with so many barriers to wide acceptance. Very sad, considering how good well-mixed, well-engineered multi-ch music can sound on a properly-calibrated system.

It's more than just the encoding. The signal path from player to output can differ between (say) SACD and a DD 5.1 disc, even using the sample player and processor. E.g, up until recently SACD generally required 6-ch analog interconnects, player did bass mgmt not the processor, etc.If you pass, for example, SACD as multichannel PCM, the processor doesn't know that it is SACD. Same with anything else. So, if you calibrate an HDMI connection with AVIA passed as LPCM, you will calibrate for SACD, too. Again, if there is an issue, the problem is still in how the SACD may be encoded. There is not a decoding problem (see below).

Sure, switching between bitstream and an analog connection can be problematic, especially if one uses their AVR's internal tones to calibrate. But, with my receiver and the adjustability it provides (namely, a 15dB external in sub boost), and a calibration DVD, I CAN at least calibrate both my players digital connection and analog connection identically.


In that paradigm a perfect calibration using AVIA didn't guarantee the proper calibration NOR the proper bass mgmt configuration for SACD/DVD-A.There is no issue with DVD-A decoding. Any issues there are encoding issues. There MAY be an issue with SACD decoding. But this is where I take exception to the way that the problem with SACD is presented in the initial post in this thread. Even if there is a discrepancy between the way that SACD and DD.5.1 are handled by a processor, the issue is still rooted in the way that SACD is encoded. The decoding discrepancy will be consistent. The way that SACDs may be encoded is not. I still use a multichannel analog connection. Some of my SACDs sound fine. Others sound thin. Which are "correct"? I am not confident enough in my own hearing to say. One thing I CAN do is compare, for example, the redbook CD layer to the 2-channel SACD layer of any individual hybrid disc. And when I have done this, I have not noticed a discrepancy.

With both SACD and DVD-A, it still comes down to the encoding. Did the engineer do it correctly or not? With a connection that is calibrated with AVIA, if a DVD-A "sounds wrong" (and this is pretty indefinable) it IS because it was improperly encoded. With SACD, if it "sounds wrong" it MAY be because of some global issue your processor/pathway has with all SACDs, but any variability from SACD to SACD is still due to encoding differences.

Yes, this is exactly how LFE trim should work.

Just curious, is this posted in support or rebuttal of any specific prior post? There was no reference link to help my feeble brain.



I was only half way done. My basic question was after you calibrate the system levels with the receivers internal test tones, do you really have to worry about unbalanced levels from external sources like a blue ray player?

So, if you calibrate a particular processor with a disc like AVIA, you have calibrated the processor to the "proper value".




Are you sure? What value for Dialnorm was used during encoding on the Avia test disk? How about the other test disks?

THX receivers and non THX receivers do not handle Dialnorm exactly the same. THX receivers in general will playback DD material 4 dB louder than a non THX receiver does. Yet DTS encoded material will playback at the same levels when played back on both types of receivers.

I was only half way done. My basic question was after you calibrate the system levels with the receivers internal test tones, do you really have to worry about unbalanced levels from external sources like a blue ray player? Sometimes, yes. If your AVR has analog inputs with no bass management, then the player has to do that, and the issue of the "LFE bug" amongst others come into play. Even with HDMI inputs playing decoded PCM, there may be issues to check and adjust, but I think these are less frequent.

In hindsight it was impossible for those formats to ever succeed, with so many barriers to wide acceptance. Very sad, considering how good well-mixed, well-engineered multi-ch music can sound on a properly-calibrated system.

Well, we can still get that via DTS or DD (or their newer flavors)...if studios choose to do the mixes. But they mostly don't, unless it's a soundtrack to a video. Music-only multichannel seems to be mostly dead in the water.

Sometimes, yes. If your AVR has analog inputs with no bass management, then the player has to do that, and the issue of the "LFE bug" amongst others come into play. Even with HDMI inputs playing decoded PCM, there may be issues to check and adjust, but I think these are less frequent.



I guess that is why I always waited a few years before I "upgraded" to newer technology. I only own standard CD's for audio. For DVDs DD and DTS sound just fine with good source material. I never cared about high rez audio.

I guess I will wait a bit longer for blue ray hardware "issues" to be cleaned up a little more.

What my primary concern here is how was the original JBL 13KW THX sound system set-up at the Empire Leicester Square, screen 1 London, because that was more than impressive to bring subtle soft lows that was easily felt and its hard to tell where the subwoofer boom track came in, with the eight clusters of JBL 4645 housed in the THX baffle wall.

The opening of Indiana Jones and the Last Crusade (Wednesday September 13th 1989) has mild low organ music tone that slowly changes with pitch while getting deeper and pressing on my body from distance of 40 45 feet from screen midway point sweet spot or VIP seating!

70mm six-track Dolby stereo magnetic SR type

Empire Leicester Square screen 1 THX Chief projectionist Fred, tests Dolby tone reference level! Dolby CP200 priceless!
http://www.youtube.com/watch?v=PxrF5MwFd70&feature=channel_page

Sometimes, yes. If your AVR has analog inputs with no bass management, then the player has to do that, and the issue of the "LFE bug" amongst others come into play. Even with HDMI inputs playing decoded PCM, there may be issues to check and adjust, but I think these are less frequent.

Roger

I think AVR should be designed like Dolby CP200 I think most AVR are joke in design and cost. The CP200 has lasted quite a long time in the industry and has been since superseded by CP500, and now with CP650. Haven’t looked at the CP6950 manual yet, not sure if supports Dolby forma code 43, like the CP500 that only supports format code 42.

I like something that has visual reference I like barograph displays like on the consumer professional Dolby DP564, which is stylish looking.

Wow! This thread has taught me alot.

I have a Yamaha HTR 6060 receiver & it does have the issue of having the LFE channel outputting 10db less when using PCM audio over HDMI. I just bought the Playstation 3 & so this problem has just come up for me. I have an SPL meter & so I tested this by using HDMI audio (in dolby digital mode) & switching between bistream & PCM audio output on my playstation 3.

The receiver displays the word “MPCM” and all of the 7.1 speaker indicators light up. So, I know I am doing everything correctly.

The problem for me is it is annoying for me to readjust my subwoofer volume every time I switch from my PS3 to another input source such as watching regular TV programming.

Is there a fix for this? Maybe a firmware upgrade?

Don't know about a Yamaha firmware upgrade (it's worth checking for), but if the PS3 allows adjustment of individual channels, you could try boosting the LFE by +10 (or lowering the other channels) . This should only affect PCM output, not bitstream.

Unfortunately, the PS3 does not have any way of boosting any channels internally.

ok

I have had HDMI with my Rotel 1069 in my system for a while now. I am actually considering moving away from HDMI (for audio only) as a deal on a Classe SSP 600 which I can just not pass up, besides I have a Classe amp which is begging for a companion.

I own a huge BD library, SACDs and SACDs, so my obvious concern is base management. I ordered the BDP 83 as a universal player which can work with the Classe so now I just have to figure a few things out. I feel like I am going back in time.

Hopefully my questions/answer will be shot.

The Oppo allows "trim" for all of its channels. Since I cannot boost the SW signal with the Classe (it a pure bypass) then can I not just "trim" all channels by -10dB except for the SW to compensate?


Somehow I fear 1 of 2 things will will with this post (or both).

1.) the answer will be long and be a "yes and no" scenerio

2.) the question will be so obvious people will say use the search function. I have used it, and I have read a lot of this thread. The problem is I dont really know what to search for or even ask.

thanks

I have had HDMI with my Rotel 1069 in my system for a while now. I am actually considering moving away from HDMI (for audio only) as a deal on a Classe SSP 600 which I can just not pass up, besides I have a Classe amp which is begging for a companion.

I own a huge BD library, SACDs and SACDs, so my obvious concern is base management. I ordered the BDP 83 as a universal player which can work with the Classe so now I just have to figure a few things out. I feel like I am going back in time.

Hopefully my questions/answer will be shot.

The Oppo allows "trim" for all of its channels. Since I cannot boost the SW signal with the Classe (it a pure bypass) then can I not just "trim" all channels by -10dB except for the SW to compensate?The answer is yes, but it may not be as simple as you think. And if you are going to run your speakers as SMALL, you will need to find 15dB, not just 10dB. Will the Classe not also allow adjustment of the individual level trims of the multichannel external inputs? And the Classe may also allow you to boost the external inputs' sub channel. Do you have a link to the Classe?


Somehow I fear 1 of 2 things will will with this post (or both).

1.) the answer will be long and be a "yes and no" scenerio

2.) the question will be so obvious people will say use the search function. I have used it, and I have read a lot of this thread. The problem is I dont really know what to search for or even ask.You're paranoid.



My only concern with the BDP83's multichannel analog outputs would be whether or not the player is completely free of any of the bass management bugs (or similar) that the other OPPO players suffered from.

The answer is yes, but it may not be as simple as you think. And if you are going to run your speakers as SMALL, you will need to find 15dB, not just 10dB. Will the Classe not also allow adjustment of the individual level trims of the multichannel external inputs? And the Classe may also allow you to boost the external inputs' sub channel. Do you have a link to the Classe?


You're paranoid.



My only concern with the BDP83's multichannel analog outputs would be whether or not the player is completely free of any of the bass management bugs (or similar) that the other OPPO players suffered from.


http://www.classeaudio.com

im not paranoid, just hung out in the Oppo BD 83 to long where any question no directly related to the player pulls out self proclaimed mods.


So I will be -5dB short in the end because I will run the speakers as small.. I am also wondering if I can just boost the sub (at the sub level) and calibrate it to the analog bypass and Oppo, then just lower the base digitally for the other inputs. Might me a bit hard to determine the level of the sub.


*** the only other signal the Classe will get is Dolby Digital form my Xbox, PS3, TV and Apply TV, and PCM from my music server. I am guessing I can adjust base for those inputs if I raise the actual sub itself.

So I will be -5dB short in the end because I will run the speakers as small.Per pg29 in the manual:
18 7.1 Channel Analog Audio Input

A 7.1 channel analog input is provided for use with a multichannel SACD
and/or DVD-Audio player. For reasons of copy protection, most such
players do not provide a high-resolution digital output. Instead, they use
high quality analog outputs.

In the SSP-600, these signals are passed through to the speakers with no
further processing, in order to preserve the purity of the signal. When this
input is selected, the SSP-600 provides only volume control (including the
volume offsets used to balance all your loudspeakers to one another).


It is not entirely clear, but it seems you will have the ability to separately adjust the level trims of the Classe's multichannnel analog inputs. If so, then between the player's trims and the processors trims, you should have no problem calibrating to the appropriate levels. What you will obviously not be able to do is calibrate the multichannel analog connection using the Classe's auto-cal. I would recommend you use a good calibration DVD such as AVIA to calibrate the multichannel analog connection. All that is important for the multichannel analog connection is that it be calibrated. The player's bass and time (distance) management settings and its individual level trims are what will be applied to your analog connection. And if the Classe DOES allow you to separately adjust the multichannel analog level trims, you will have that at your disposal, too.

But 15dB is doable even without the ability to adjust the Classe's input level trims. For example, front left speaker (or whichever speaker is your reference speaker) trim in the player at -10dB, subwoofer at +5dB. Or -8dB and +7dB. Etc.. All that is important is that the calibration be correct. If you can calibrate the analog connection's channel levels properly with a disc like AVIA, no matter how you have to do it, then you are set. All that is important is what is coming out of the speakers at the end.

What is important to understand is that the analog connection and the digital connection are separately adjusted and calibrated. The player's bass, time (distance), and level trim settings should only apply to your analog connection; and possibly the Classe's multichannel input level trims. For the digital connections, all that will be applied is the processor's settings.

It is not clear whether you will be using a digital connection with the BDP83 or not and if so, whether it will be S/PDIF or HDMI. The BDP83's bass, time, and level trims settings should not be applied to its optical or coax outputs. With some players, though, some or all of these settings are applied to the HDMI connection if you are passing MPCM. So, you might want to look into this.


I am also wondering if I can just boost the sub (at the sub level) and calibrate it to the analog bypass and Oppo, then just lower the base digitally for the other inputs. Might me a bit hard to determine the level of the sub.You shouldn't have to resort to this. I see no reason why you should.


*** the only other signal the Classe will get is Dolby Digital form my Xbox, PS3, TV and Apply TV, and PCM from my music server. I am guessing I can adjust base for those inputs if I raise the actual sub itself.:confused: You will simply use the processor's settings for the digital connections. You can calibrate the processor with its auto-cal capability, or you can calibrate using the same disc you calibrate the analog connection with.

^^^ I really appreciate your reply and time on this. You have given me a lot of food for thought and I will refer back to this as soon as everything arrives. Ill post my findings in case anyone else with the same setup finds the same issue.

So I will be -5dB short in the end because I will run the speakers as small.. I am also wondering if I can just boost the sub (at the sub level) and calibrate it to the analog bypass and Oppo, then just lower the base digitally for the other inputs. Might me a bit hard to determine the level of the sub.

I would do that. An SPL meter would be handy. Changing channel trims would only bring up the noise floor.

Changing channel trims would only bring up the noise floor.:confused: :rolleyes:

I would do that. An SPL meter would be handy. Changing channel trims would only bring up the noise floor.

can you explain this a little bit more?

I would do that. An SPL meter would be handy. Changing channel trims would only bring up the noise floor.

can you explain this a little bit more?

Sure. Although adjusting channel trims in the player is one way of getting your LFE deficit back, it is not preferred and here's why.

Say you have specified some speakers as SMALL so there will be redirected bass. As this thread notes, the LFE channel will be -15dB after the proper mixing is done. Let's assume that your downstream equipment merely acts as a straight bypass and has no LFE boosting capbility so you do the following setup in your player:

Left channel -10dB
Right channel -10dB
Center -10dB
Left Surround -10dB
Right Surround -10dB
Subwoofer +5dB

Directionally, this should get you your 15dB back. However, the speakers are now artificially lower in output so that forces your downstream AVR/amplifier to be at a higher volume than if all the channels were coming in correctly (ie. LFE -15dB). With this higher volume comes a higher noise floor (ie. things in the background that you couldn't hear before and were not meant to be heard are now audible).

I offer you this tidbit from the Oppo BDP-83 manual on page 62:

"The speaker trim level parameters sets the volume of each individual channel. Channel trim is generally not required since most A/V receivers have the capability to adjust channel trim and compensate for speaker sensitivity differences. However, if your receiver does not have such functions (many receivers do not support channel trim for their multi-channel analog inputs), you may adjust channel trim through the player."

As this thread tries to point out, it is really the job of the downstream equipment to boost the LFE correctly. Compensations such as playing with channel trim levels are really a band aid. Remember, I'm talking about channel trim to compensate for LFE deficit. I have no problem with channel trim under "typical setup" for fine tuning audio calibrations.

Edit: I would follow through on your own suggestion and get your missing 5dB back by increasing the gain on the sub (if possible). Then you could lower the digital LFE setup by 5dB for all your other sources. Let the amplification be done downstream (amp, sub gain, etc.), not upstream (BD player).

Sure. Although adjusting channel trims in the player is one way of getting your LFE deficit back, it is not preferred and here's why.

Say you have specified some speakers as SMALL so there will be redirected bass. As this thread notes, the LFE channel will be -15dB after the proper mixing is done. Let's assume that your downstream equipment merely acts as a straight bypass and has no LFE boosting capbility so you do the following setup in your player:

Left channel -10dB
Right channel -10dB
Center -10dB
Left Surround -10dB
Right Surround -10dB
Subwoofer +5dB

Directionally, this should get you your 15dB back. However, the speakers are now artificially lower in output so that forces your downstream AVR/amplifier to be at a higher volume than if all the channels were coming in correctly (ie. LFE -15dB). With this higher volume comes a higher noise floor (ie. things in the background that you couldn't hear before and were not meant to be heard are now audible).Malarkey! (sorry ;))

Reducing the main channels' output from the player by 10dB is not going to result in that significant an increase in the noise floor. And the same 15dB can be obtained by setting the speakers to, for example, -5dB with the sub at +10dB. The best way would be to simply split the difference as evenly as possible; no trim will be either too negative or too positive. Either way, though, within that sort of range, the noise floor concern is not significant (if at all).



"The speaker trim level parameters sets the volume of each individual channel. Channel trim is generally not required since most A/V receivers have the capability to adjust channel trim and compensate for speaker sensitivity differences. However, if your receiver does not have such functions (many receivers do not support channel trim for their multi-channel analog inputs), you may adjust channel trim through the player."right


Edit: I would follow through on your own suggestion and get your missing 5dB back by increasing the gain on the sub (if possible). Then you could lower the digital LFE setup by 5dB for all your other sources. Let the amplification be done downstream (amp, sub gain, etc.), not upstream (BD player).Sure, you can do that, but most processors adjust the channel trims relatively. (Whether players do the same, I am not certain.)

Malarkey! (sorry ;))

Reducing the main channels' output from the player by 10dB is not going to result in that significant an increase in the noise floor. And the same 15dB can be obtained by setting the speakers to, for example, -5dB with the sub at +10dB. The best way would be to simply split the difference as evenly as possible; no trim will be either too negative or too positive. Either way, though, within that sort of range, the noise floor concern is not significant (if at all). Just to add that even though your player may offer + gain trim settings, if it's smart, it will not actually add any gain, as that would defeat the purpose for the -15 dB scaling in the first place--to avoid clipping. (This assumes the max output is 2Vrms. If it can output 6Vrms, then boost is OK as long as it doesn't smoke the input of the SSP-600.)

Might be worth checking the player to see ow it behaves. For example, set all trims to 0 dB. Play a steady signal from the L channel, and take a loudness reading. Do the same for another channel--say C. Now turn up the C trim +10 dB. Play the signals again. Did the C go up, or the L go down?

If C went up, then you will run into clipping after just a few dB on loud soundtracks. In that case, you should avoid applying boost--maybe you'll get away with +5 on the Sub and -10 on the mains. There's not much headroom left unusued on many action movies.

If, OTOH, C stayed the same and L went down, your player is smart enough to avoid clipping problems. Wish I could say the same for my iPod player--none of the tone boost presets do this right--they cause clipping when activated.

Just to add that even though your player may offer + gain trim settings, if it's smart, it will not actually add any gain, as that would defeat the purpose for the -15 dB scaling in the first place--to avoid clipping. (This assumes the max output is 2Vrms. If it can output 6Vrms, then boost is OK as long as it doesn't smoke the input of the SSP-600.)

Might be worth checking the player to see ow it behaves. For example, set all trims to 0 dB. Play a steady signal from the L channel, and take a loudness reading. Do the same for another channel--say C. Now turn up the C trim +10 dB. Play the signals again. Did the C go up, or the L go down?

If C went up, then you will run into clipping after just a few dB on loud soundtracks. In that case, you should avoid applying boost--maybe you'll get away with +5 on the Sub and -10 on the mains. There's not much headroom left unusued on many action movies.

If, OTOH, C stayed the same and L went down, your player is smart enough to avoid clipping problems. Wish I could say the same for my iPod player--none of the tone boost presets do this right--they cause clipping when activated.

Roger, I forgot to mention the clipping possibility when adjusting player channel trims but thanks for the reminder.

You need a barograph display to see the differences directly before passing it onto the amps.
Get simple affordable DCX2496 at least you can see what is happening on input to output. Also a voltmeter will come in handy.

Might be worth checking the player to see ow it behaves. For example, set all trims to 0 dB. Play a steady signal from the L channel, and take a loudness reading. Do the same for another channel--say C. Now turn up the C trim +10 dB. Play the signals again. Did the C go up, or the L go down?
Most likely the C will go up. I do not think that players (but admittedly am not certain how different players may do it) adjust the trims relatively. I will check mine with an SPL meter, but 'ear-balling' it just now, it seemed not to adjust the trims relatively.


If C went up, then you will run into clipping after just a few dB on loud soundtracks. In that case, you should avoid applying boost--maybe you'll get away with +5 on the Sub and -10 on the mains. There's not much headroom left unusued on many action movies.Of course, this is an idealized situation. It could be that the surrounds are the loudest, in which case the mains and center may have to be adjusted louder which would in turn force the sub trim even higher. For example:

front L = -5dB
front R = -5dB
center = -5dB
surround right = -10dB
surround left = -10dB
sub = +10dB

Whether this 10dB sub boost would be detrimental or not, I do not know. If the player's trims go to +/-12dB, this would provide an additional -2dB that the loudest channels could be adjusted downward.

But I would agree that it would probably be best to avoid + trims as opposed to - ones. Yes, it could cause one to have to use a significantly higher master volume setting on the AVR in order to achieve reference level with their external inputs. In vancouver's case, I think his processor has an input level balancing feature. Whether it can cover a 10dB difference or whether it can be applied to the external inputs, I do not know.

Listen to Cape Fear (1991) on DVD and you’ll have hard time hearing the soft sounds of Max Caddy blowing smoke over left/right front and some mild background cinema sound from the film Problem Child (1991). You’ll hear competition from left/right front with some people laughing in the cinema over the distant echo from the cinema.

single driver closed box vs Bass reflex; or
KEF TDM45B vs KEF PSW4000 Subwoofer:

Okay, so I have a KEF TDM45B Subwoofer which I liked and I figured anyway... Let's place a low ball offer on a PSW4000 on eBay so guess what. I won! So I'm expecting a PSW4000 in the new couple of days so I guess I have to decide which one to keep ( I'm matching these with Kef Ref 3 fronts). What is your opinion about different designs? Does a closed box require a bigger driver?

PSW 4000 specifications
Design Bass reflex, downward firing
Drive unit LF: 300 mm (12") long-throw
Frequency response at 15°
horizontally off axis ±3 dB 25 Hz - 250 Hz
LF Corner -6 dB 22 Hz
Amplifiers 500 W built in
Sensitivity (2.83V/1m) Active system
Maximum output 113 dB
Impedance Active system
Weight 38 kg (83.6 lbs)
Dimensions
(H x W x D) 465 x 495 x 495 mm 18.3 x 19.5 x 19.5 in.

TDM45B Specifications
Design single driver closed box
Frequency response at 15°
horizontally off axis ±3 dB 35 Hz - 150 kHz
LF Corner -6 dB 28 kHz
Crossover frequency 50 Hz - 150 Hz, 80 Hz (THX)
Amplifier requirements 300 W built in
Sensitivity (2.83V/1m) active system
Maximum output 115 dB
Impedance active system
Weight 28 kg (62 lbs)
Dimensions
(H x W x D) 455 x 455 x 567 mm 18.2 x 18.2 x 22.7 in.

single driver closed box vs Bass reflex; or
KEF TDM45B vs KEF PSW4000 Subwoofer:

Okay, so I have a KEF TDM45B Subwoofer which I liked and I figured anyway... Let's place a low ball offer on a PSW4000 on eBay so guess what. I won! So I'm expecting a PSW4000 in the new couple of days so I guess I have to decide which one to keep ( I'm matching these with Kef Ref 3 fronts). What is your opinion about different designs?This is not really the correct place to post this. I saw that you had asked something similar elsewhere (perhaps in the KEF thread?). The subwoofer forum would be the correct place. Very few people here (if any) will have had any real-life experience with either of those subs. If you want to discuss, generically, sealed versus ported subs, then there is already PLENTY of this sort of discussion in the subwoofer forums. And the DIY forums, too.

What is it you want to know, exactly? You'll soon have the new sub so you can determine which you prefer. My inclination would be to say that you'll prefer the PSW4000 over the TDM45B. But I've never heard a TDM45B. I have heard the PSW4000 and it is a fine subwoofer. Gorgeous, too. The 45B may be capable of slightly more output, but based upon the specs you post here, the PSW4000 should outperform it in almost every way. The 45B is down 3dB at 35Hz. Granted, its roll-off from its -3dB point to its -6dB point of 28Hz is not as steep as the PSW4000's is from 25Hz to 22Hz, but with a -3dB point of 35Hz, the 45B is not much of a subwoofer.


Does a closed box require a bigger driver?Require a bigger driver to do what? That question is too vague. Are you talking about output? Depth? SQ? You can't simply independently single out the size of the driver as any sort of sole determinant of performance. There are many more factors involved than just simply the size of the driver. Among many other things, sealed or ported, there is the size of the box and the size of the amp. Again, the DIY and subwoofer forums are going to be the places to discuss this.

sivadselim: thanks for your input... I will go to the correct forum!

Just to add that even though your player may offer + gain trim settings, if it's smart, it will not actually add any gain, as that would defeat the purpose for the -15 dB scaling in the first place--to avoid clipping. (This assumes the max output is 2Vrms. If it can output 6Vrms, then boost is OK as long as it doesn't smoke the input of the SSP-600.)

Might be worth checking the player to see ow it behaves. For example, set all trims to 0 dB. Play a steady signal from the L channel, and take a loudness reading. Do the same for another channel--say C. Now turn up the C trim +10 dB. Play the signals again. Did the C go up, or the L go down?

If C went up, then you will run into clipping after just a few dB on loud soundtracks. In that case, you should avoid applying boost--maybe you'll get away with +5 on the Sub and -10 on the mains. There's not much headroom left unusued on many action movies.

If, OTOH, C stayed the same and L went down, your player is smart enough to avoid clipping problems. Wish I could say the same for my iPod player--none of the tone boost presets do this right--they cause clipping when activated.

I dont know for sure yet, but I am pretty sure there are no options to add any "+" to trims, only reduce each channel hence the reason I suggested reducing each channel by 10dB but the LFE.

This was the same with my HD DVD A1 if I am not mistaken.

Panasonic's SA-BX500 product page

http://www.panasonic.net/avc/blu-ray/sa-bx500/specifications/us_features.html

says it has "Subwoofer Output for Extended Bass Reproduction", and in the user manual says

When you set the front speakers (“L ”, “R ”) as “LARGE ”
The subwoofer also makes audio output in the bass range when you perform stereo playback of analog and PCM sources.
The subwoofer sends out only the LFE (low frequency effect channel) signal contained in 2-channel Dolby Digital or DTS sources when you play them in stereo.

but they don't state any crossover frequency for this. Sounds like a nice feature, but it must do this by crossing over into part of the low range of the LRs. I'm wondering how low the LRs should go to avoid there being a frequency "hole".

Anyone heard of this kind of thing, and can maybe speculate on what they're doing?

I’m stumped by something about the new Oppo BDP-83 player. Oppo says the analog sub channel only needs a 10db boost, even when the player is handling bass management. Here’s what Oppo said when asked whether a 15db boost is needed when speakers are set to small:

When the player internally decodes Dolby or DTS audio, the LFE channel is already 10dB lower than the other channels. This is not attenuation by the player, but a requirement in the original encoding of the audio.

When the player does bass management and redirects bass to the subwoofer, it attenuates the redirected bass to match the -10dB LFE. This way the subwoofer output of the 7.1ch analog output is consistent with the Dolby and DTS requirements.

Under this condition, a simple 10dB boost in either the amplifier or the Subwoofer should be sufficient to handle the level alignment. It is best not to use the speaker level trim settings in the player, as the trim settings are applied in the DSP before the audio signal gets converted to analog. This digital manipulation could result in reduced dynamic range and audio resolution.

Best Regards,

Customer Service
OPPO Digital, Inc.

Everything I’ve read about analog bass management, starting with the first post in this thread, says the subwoofer channel (LFE + redirected bass) needs to be dropped another 5db prior to transmission. As I understand it, if LFE is already at maximum capacity, the addition of redirected bass means the output will exceed the channel’s capacity. Hence, the need to lower the output another 5db and then boost it 15db in the processor or at the sub.

But, as you can see, Oppo says it doesn’t drop the sub channel an extra 5db when bass management is engaged. Is that possible without running the risk of clipping during very loud passages?

Oppo says it doesn’t drop the sub channel an extra 5db when bass management is engaged. Is that possible without running the risk of clipping during very loud passages? No. But I suspect that the rather obscure 15 dB issue is not well understood by most support people, so the reply you received does not surprise me, nor would I read it to mean the player is not designed correctly.

Some of their reply is perfectly correct: >>When the player does bass management and redirects bass to the subwoofer, it attenuates the redirected bass to match the -10dB LFE. This way the subwoofer output of the 7.1ch analog output is consistent with the Dolby and DTS requirements.<<

If the player is compliant with Dolby specs, then the additional 5 dB attn is applied.

Maybe someone in the EAP thread can run a test signal, like the THX Optimode channel cals and see if the LFE level shifts -5 dB when any of the speakers are set to Small as compared to when all are Large.

Panasonic's SA-BX500 product page says it has "Subwoofer Output for Extended Bass Reproduction", but they don't state any crossover frequency for this. Sounds like a nice feature, but it must do this by crossing over into part of the low range of the LRs. I'm wondering how low the LRs should go to avoid there being a frequency "hole".

Anyone heard of this kind of thing, and can maybe speculate on what they're doing? They are doing this because folks have complained that their subwoofers go silent when L/R are set to large and 2-ch sources are played. As you point out, this causes bass duplication so it is not optimal. The specs state the subwoofer output response as 7-200 Hz, so it's possible the sub could be duplicating to that extent. Or they might be defaulting to wherever the crossover frequency is set. No way to tell without measuring.

Since there is apparently no opt-out choice like Denon offers, the only way to avoid it in this case is to set L/R to Small.

I think there is a small error in their explanation: >>The subwoofer sends out only the LFE (low frequency effect channel) signal contained in 2-channel Dolby Digital or DTS sources when you play them in stereo.<< There is no LFE in a 2.0 DD signal, so I think they mean when 5.1 signals are played back in stereo mode.

Roger, thanks for your comments!

No. But I suspect that the rather obscure 15 dB issue is not well understood by most support people, so the reply you received does not surprise me, nor would I read it to mean the player is not designed correctly.

Some of their reply is perfectly correct: >>When the player does bass management and redirects bass to the subwoofer, it attenuates the redirected bass to match the -10dB LFE. This way the subwoofer output of the 7.1ch analog output is consistent with the Dolby and DTS requirements.<<

If the player is compliant with Dolby specs, then the additional 5 dB attn is applied.

Maybe someone in the EAP thread can run a test signal, like the THX Optimode channel cals and see if the LFE level shifts -5 dB when any of the speakers are set to Small as compared to when all are Large.Thank you, Roger. I posted your suggestion in the EAP thread. When this issue came up in the last few days, a couple of Oppo owners have already noted that they have to apply 13-15db of boost to get their sub channels properly calibrated when speakers are set to small.

When this issue came up in the last few days, a couple of Oppo owners have already noted that they have to apply 13-15db of boost to get their sub channels properly calibrated when speakers are set to small.Yeah, I think that (for whatever reasons) this issue is so nebulous that the best thing to do is calibrate the multichannel analog connection properly with a calibration disc such as AVIA. If you do that, you are good to go no matter what settings are required to achieve the proper calibration. Now, in doing so, you should be able to discern what boost was required on the subwoofer channel.

One thing that I have noticed with a few Denon players that I have used is that a 10dB or 15dB boost at my AVR has never solely provided a proper calibration. With a 10dB boost I have to run the external inputs' sub-in level trim 2-3dBs higher relative to the other ext. inputs' channels' level trims which are all adjusted to the same relative values as those of my digital connection's calibration. With a 15dB boost, the sub-in level trim has to be reduced by 2-3 dB relative to the other channels' level trims.

For example:

http://home.comcast.net/~schiz/levels.jpg


Note that in order to level match the ext.in's main speaker channels requires an identical +0.5dB bump to all the main channels. But the subwoofer channel is different.

I have a question if someone wants to address it. I think I know the answer, but that would be an assumption and we all know about them.

I found this thread fascinating. It is over-my-head at the moment, but fascinating.

I have an Onkyo TX-SR806 arriving this week. (Which is more receiver than I would have normally purchased, but the price was too good to ignore.) I also have an Infinity TSS-1100 speaker system that has a 12" powered subwoofer.

If I:

1. Set the powered subwoofer to it's suggested crossover and set the volume for a mid-position; and,

2. Then hook it into the Onkyo and let it run Audyssey MultEQ that program and then run the Onkyo with Dynamic EQ on,

Then that should make whatever corrections should be made in order to provide the proper amount of energy/sound from the subwoofer, correct?

(I know that the speakers are the weak link, but to me they provide good sound for the conditions and budget in my house.)

Thanks.

One thing that I have noticed with a few Denon players that I have used is that a 10dB or 15dB boost at my AVR has never solely provided a proper calibration. With a 10dB boost I have to run the external inputs' sub-in level trim 2-3dBs higher relative to the other ext. inputs' channels' level trims which are all adjusted to the same relative values as those of my digital connection's calibration. With a 15dB boost, the sub-in level trim has to be reduced by 2-3 dB relative to the other channels' level trims. If the bandwidth or the rolloff slopes of the lowpass filters in the player are a little different than those of your AVR, it can look like a level difference. Your data suggests that BW in the player is wider--if this theory is correct. :D

................I also have an Infinity TSS-1100 speaker system that has a 12" powered subwoofer.

If I:

1. Set the powered subwoofer to it's suggested crossover and set the volume for a mid-position; and,

2. Then hook it into the Onkyo and let it run Audyssey MultEQ that program and then run the Onkyo with Dynamic EQ on,

Then that should make whatever corrections should be made in order to provide the proper amount of energy/sound from the subwoofer, correct?

(I know that the speakers are the weak link, but to me they provide good sound for the conditions and budget in my house.)

Thanks.It is unclear how that speaker system is connected. Do those satellites simply connect to the AVR just like any other speakers would be connected? If so, I am not really sure what you are asking about. Why would calibration of that setup be any different than any other?

Are you asking about the subwoofer's own crossover setting? If so, the sub's own crossover (it's really just a low-pass filter, btw) should be set as high as it can possibly be set (or bypassed if that is possible).

As far as what starting volume the subwoofer should be set to prior to running Audyssey, you can sort of figure out what is appropriate. If you set it too low or too high, the subwoofer channel's individual level trim will be adjusted as needed by the AVR. Ideally, you want the AVR to have to apply as little adjustment as possible, here. So, you can use the sub's own volume control to affect the adjustment the AVR has to make to the subwoofer level trim. Or not. That is up to you. You can simply set the sub's volume knob to, as you say, "a mid-position". If the sub's starting level is too high or too low for your AVR to compensate, Audyssey will tell you that you need to turn it up or down.

Does that help? If not, could you be more specific about what you are getting at?

If the bandwidth or the rolloff slopes of the lowpass filters in the player are a little different than those of your AVR, it can look like a level difference. Your data suggests that BW in the player is wider--if this theory is correct. :DYes, I agree that that is one very possible explanation. Or at least a part of the explanation. In trying to measure the player's and AVR's high and low-pass filters' individual and combined behaviors in the past I had observed some interesting and peculiar things that I was never able to fully reconcile.

But I think it can also be explained on the basis that there is something different somewhere in the chain that causes the level of the subwoofer channel of the multichannel analog connection to differ in a different way than the other speaker channels' levels differ, relatively. In other words, assuming the +15dB boost is the "correct" boost that I need to be applying, it would appear that, relative to the main channels, somewhere in the chain, the sub's level is 2dB hotter than the speaker channels. I've never found this to necessarily be at all odd. There's really no reason to assume that the signal level should be the same throughout the whole chain from player through AVR. Whether that difference has already occurred before or as the signal leaves the player or it occurs once the signal enters the AVR is really not important. Only that there is a difference somewhere. (Does that make sense?)

Does that help? If not, could you be more specific about what you are getting at?

Yes, that helped. I have never actually calibrated a receiver before using automatic adjustment hardware/software, although I have used a sound meter to try and even out the perceived volumes. So I apologize if the question wasn't worded well. I do think you responded to what my question really was, for which I thank you.

I'm using the analog inputs of a 5.1 Yamaha RX-V457 pre-HDMI receiver, which are shared via a passive switchbox (no electronics, just ganged groups of RCA jacks connected by pushbutton switches), by:

1) a Panasonic DMP-BD50 Blu-ray player that decodes all the lossless formats to 5.1 analog (the first Blu-ray player to do so)

and

2) a Denon 2910 "Universal" DVD player that I use to listen to my collection of SACD and DVD-A disks.

The Yamaha RX-V457 is an odd duck of a surround receiver that does make its channel level and distance settings applicable to the multichannel input, which makes life easier, since I only have to calibrate one set of controls, instead of calibrating each player and the receiver (for the cable tv input). I verified this experimentally by alternately cutting and then boosting a channel in the Yamaha's setup - it did indeed affect a signal coming in through the multichannel analog input.

All my speakers other than the sub are full-range (Paradigm Studio/40 for LF and RF, Paradigm/CC, and Studio/20 LS and RS), so I keep all devices set to sub present, other speakers Large. The players' channel level and distance or delay settings are all set to 0 to make them inactive. The Yamaha is sent to "double" the mains' bass to the sub as well as the mains.

The Denon 2910 provides for a switchable +10db bump for the LFE channel - accessible, unfortunately, only by going into setup, not on-the-fly.

Since SACD does not cut the LFE level and expect a corresponding bump on playback, if those were the only disks I ever put into the 2910, I would turn off the LFE boost in the 2910 and all would be, as they say, "Copa-setic."

However, I also play DVD-Audio disks with the 2910. Is their LFE behavior like SACD's or like movie soundtracks? If they act like SACDs I'm all set, but if they act like movie soundtracks, I'd have to go into setup and turn on the LFE bump to play them - unless the 2910 is smart enough to apply different treatment to the LFE on those formats on its own - which is probably not the case, since that would make that setup option unnecessary.

Since SACD does not cut the level and expect a corresponding bump in the amplifier, if those were the only disks I ever put into the 2910, I would turn off the LFE boost in the 2910 and all would be, as they say, "Copa-setic."

However, I also play DVD-Audio disks with the 2910. Is their LFE behavior like SACD's or like movie soundtracks? If they act like SACDs I'm all set, but if they act like movie soundtracks, I'd have to go into setup and turn on the LFE bump to play them - unless the 2910 is smart enough to apply different treatment to the LFE on those formats on its own - which is probably not the case, since that would make that setup option unnecessary.DVD-A is identical to DVD-V. But I would make certain that your 2910 doesn't accommodate the SACD difference on the front-end. The Denons I have experience with do not require a different setting for SACD playback. At least not as far as I could tell. Of course, making that assessment is difficult as there is so much variation in the bass level from recording to recording. I think that there are some SACDs which are recorded "incorrectly" in an effort to "correct" them, but when played in a player that does have an SACD 10 or 15dB accommodation, they will have a bass level that is 10-15dB low.



Per the initial post in this thread:

"DVD-Audio/HD DVD/Blu-ray's bass is too quiet"
No. They're mixed exactly the same way as Dolby Digital or DTS on DVD-Video. The LFE track is recorded 10dB low. People are only noticing a problem because they've switched from a DD/DTS bitstream link which works to a multichannel interconnect lacking the necessary 10dB-15dB boost. If they had been listening to DD or DTS decoded in the player through the multichannel interconnect they'd have seen the same problem. And the problem is that their receiver isn't boosting its SW/LFE input sufficiently.

It is not really an option for the player to boost its analogue SW output, as it would be in danger of overloading a receiver's input circuitry when a maximum volume LFE signal appeared - feeding a 6 volt signal into a nominally 2 volt input. You might get away with it if the amplifier was purely doing an analogue passthrough, but it would overload any receiver with multichannel ADCs.

And similarly the player absolutely cannot boost its digital LFE output. There's no headroom to do this.

"Super Audio CD's bass is too quiet"
Same basic answer as the previous section, except for one wrinkle: SACD doesn't actually use a 10dB boost for its LFE channel (which poses the question - why have it at all?)

To maintain compatibility, some multi-format SACD players apparently lower the LFE internally by 10dB, then carry on the rest of their processing as normal. This then leads to the output:

SW = Lower10dB(music LFE)

or

SW = Lower15dB(music LFE + Redirected bass)

So the net boost of 10dB or 15dB is still needed in the receiver to achieve correct playback, consistent with other formats.

Some players, usually SACD-only, do not do this 10dB adjustment, leading to the output

SW = music LFE

or

SW = Lower5dB(music LFE + Redirected bass)

This could arguably result in better quality, by making better use of the range on that input, but means the receiver has to have its SW input switched to +0dB or +5dB respectively just for SACD with that player - inconvenient if it's multiformat.

Over the past weekend, I substantially boosted the level of the signal output to my subwoofer by my receiver. My Radio Shack sound pressure level meter has never really "heard" my subwoofer, so I've always adjusted it subjectively.

A few days ago I boosted it to the point where it felt as loud as - or louder than - the other channels. I also raised the crossover frequency for doubling the low bass from the mains to the sub as well. The Yamaha allows a crossover frequency of as high as 200 Hz. I expect that I'll get feedback from the experts here that this is too high, but I'm not having any directionality problems using it.

What this all has done is lead me to lower my master volume control significantly. Where in the past I would have it at -4 or -5db (according to the amplifier's dashboard), I'm now running it at -10 or -11db, without losing excitement.

If feels like I've increased the effective dynamic range of the system. I've certainly increased the perceived dynamic range!

But I would make certain that your 2910 doesn't accommodate the SACD difference on the front-end. The Denons I have experience with do not require a different setting for SACD playback.

I've been playing DVD-As, SACDs, and Dolby and DTS DVDs with the Denon 2910 for years through these speakers, always using the multichannel analog outputs for everything. Never bothered to switch the 10db setting back and forth, since everything has always sounded fine.

The only significant changes to my system have been the addition of the BD50 and substituting the Yamaha for an NAD receiver this spring - at first temporarily because of a problem with the NAD and then permanently because of the greater convenience of the Yamaha and its more transparent high frequency handling - I hear things with the Yamaha I never noticed with the NAD.

Good to know that I don't have to mess with the SW boost setting.

PS One advantage of my pushbutton analog kludge is that I can listen to a source different than what I have on screen. (I select video inputs using the projector's remote.) This setup makes it easy to watch The Wizard of Oz while listening to the SACD of Dark Side of the Moon - try that with HDMI! (Yeah, I know, you can probably define a custom setting for that with HDMI, but it's easier to do it on-the-fly - and I frequently listen to the radio while cataloging video disks, which would require yet another custom setting with HDMI.)

New information today from Oppo re: how much boost is needed for the sub when the player is doing bass management. Here's the background:

I’m stumped by something about the new Oppo BDP-83 player. Oppo says the analog sub channel only needs a 10db boost, even when the player is handling bass management.

I suspect that the rather obscure 15 dB issue is not well understood by most support people, so the reply you received does not surprise me, nor would I read it to mean the player is not designed correctly.

If the player is compliant with Dolby specs, then the additional 5 dB attn is applied.

This afternoon, Oppo said its previous statements were incorrect - that the BDP-83 actually does properly reduce the sub output by another 5db when doing bass management:
We would like to apologize for some misinformation which has been communicated not only to our customers, but to our beta testers as well. We had incorrectly stated that the player would always be -10dB for the subwoofer output, regardless of the player performing bass management. This is incorrect. The player applies an additional 5dB attenuation to LFE and redirected bass when bass management is in use. This is based on a requirement for Dolby certification.

So the the persistence of members of the community and our beta testing group were correct in their statements and observations. The subwoofer output is attenuated by 10dB when there is no bass management being done by the player, while being attenuated 15dB when performing bass management over the multi-channel analog.

Best Regards,

Customer Service
OPPO Digital, Inc.

In an earlier post, I said that my Radio Shack Sound Pressure Level meter couldn't hear the subwoofer, so I was going by "feel" in setting the subwoofer level.

Tonight I found that if I changed the switch setting on the SPL meter from "A" to "B" it does hear the subwoofer as well as the mains, so I recalibrated the levels again using that setting. That called for me to drastically reduce the subwoofer level, which led to very wimpy sound.

Then I remembered that since I'm using the analog multichannel output of my Blu-ray player, the LFE channel was still at -10db, since, according to what I've read here, the boost has to happen in the amplifier. (My bass management is done in my amp, which miraculously applies all of such settings to the multichannel analog input - remember that I called it an "odd duck" in that respect. Since bass management isn't done in the disk player, the cut is not -15db but only -10db.).

Accordingly, since my SPL meter doesn't go to =+10db (much less 11) but only to +6, I set my mains by the meter to -5db and the subwoofer to +5db using the amp's test tone.

I also set the bass management crossover to the THX standard of 80 Hz.

The system now sounds much better.

Two things do stand out as exceptions, and I'm not really sure how to handle them:

The receiver does the Dolby decoding for my cable box, so it is presumably giving the LFE the customary boost in that process - in addition to the channel balance tweak. It still sounds good, and I don't want to have to recalibrate the subwoofer to watch cable!

My SACD/DVD-A player, the Denon 2910's multichannel analog channel balance screen and its HDMI channel balance screen each provide for a 10 db subwoofer boost, which shouldn't exist at all, from what I've read here. The analog output's SW boost is quite audible, and definitely adds oomph to the sound. Testing with both the DVD-A and SACD versions of Steely Dan's Gaucho (their first digitally-recorded album, which was the immediate follow-on to Aja, and their last album for many years), shows that the boost affects both versions - the SACD more strongly (presumably since it wasn't cut to start with). I suppose that I should turn it off, but it sounds great with it on. "Musical MSG" ?

PS I also experimented with the amp's subwoofer phasing selection and found it sounded slightly stronger in the reverse setting. Test track: "Three times it drops" - the over the waterfall sequence in Indy Jones IV.

Turned off the 10db "Subwoofer" boost in the Denon 2910 SACD/DVD-A player's Speaker Level setup because it sounded much too hot, since I'm already boosting it by 10db in the amp.

LFE probably still ends up a little hot for SACDs, which don't attenuate it to start with, but that's fine with me.

With respect to the receiver's decoding of SPDIF Dolby 5.1 signals from my high def cable box, it doesn't sound like the LFE is too hot, so I'm assuming that the decoder isn't boosting LFE but leaving that to the amplifier section's speaker level setting.

I'm now running my master gain much lower than before I boosted the subwoofer output. Where I ran it at -4 or -5 in the past, I'm running it at -10 or -11 now, for the same perceived loudness.

My assumption is that the greater punch in the low end makes the overall sound more satisfying even with less power in the midrange and treble. This is what I've come to think of as greater apparent dynamic range.

I've found that in my room, which is 15' square, with a 5'x7' entry area on the left side of the room, the phase reversal feature for the subwoofer in my amp makes very little difference. The sub is under the pole for the right front speaker. The front speakers are each about 5 feet away from the ends of the 20' wide front wall, flanking the 9 1/2' wide image. I sit on a couch about 9 feet back from the screen. I have a second sub that I could hook up and put under the left front speaker, but I don't feel the need right now. (For pictures of my speaker setup, which I posted in another thread here, go to http://www.avsforum.com/avs-vb/showthread.php?p=16552610#post16552610.)

I've also kludged together a decent HD radio setup by getting Radio Shack's inexpensive "Accurian" HD table radio and plugging its earphone output into a line input on the receiver. The subwoofer sounds good with that signal as well - there's no LFE, but it is doubling the low bass.

One nice feature of the AV receiver is that you can change the text shown in its display for any of the inputs, so that input now displays "HD Radio" when it's selected.

Trivia question: What does the "HD" stand for? Answer - not High Definition - which is what they hope you'll think, and not "nothing" - which is what they'll say when asked, but "Hybrid Digital." The way it works is by adding an MP3-like signal on a subcarrier, the same way that FM Stereo adds a "L-R" signal on a subcarrier to be added in and out of phase with the mono (L+R) main carrier to derive the two analog channels (the algebra is left to the reader).

In fact, the HD system allows for multiple HD subcarriers, which is why many FM stations have more than one HD channel. However, they all have to share transmitter power with each other and the main carrier, so it's a bit of a zero-sum situation. Devote too much power and bandwidth to the HD system and the analog stereo signal suffers.

And 10dB difference is quite a lot - it means a signal over 3 times the amplitude.

GREAT summary, KMO. I haven't read the whole thread, but the problem is even greater than you state! A 10dB difference is even more than you claim. It's 10 times the amplitude. The first 3dB is 2 times power, the second 3dB is 2 times that (total of 6dB) and takes it to 4 times (2x2), the next 3dB is twice THAT (total of 9dB) and takes it to 8 (2x4) times. The last single dB takes it from 8x to 10x! (I'll spare you the log functions.)

GREAT summary, KMO. I haven't read the whole thread, but the problem is even greater than you state! A 10dB difference is even more than you claim. It's 10 times the amplitude. The first 3dB is 2 times power, the second 3dB is 2 times that (total of 6dB) and takes it to 4 times (2x2), the next 3dB is twice THAT (total of 9dB) and takes it to 8 (2x4) times. The last single dB takes it from 8x to 10x! (I'll spare you the log functions.) KMO's reference to amplitude means signal level, not power. He has it right. 10 times the amplitude is actually 20 dB. Amplitude is 20log(A1/A2). Power is 10log(P1/P2). 10 dB is 3.16 times the amplitude (signal level). It is also a 10dB increase in power, so on that we agree.

KMO's reference to amplitude means signal level, not power. He has it right. 10 times the amplitude is actually 20 dB. Amplitude is 20log(A1/A2). Power is 10log(P1/P2). 10 dB is 3.16 times the amplitude (signal level). It is also a 10dB increase in power, so on that we agree.

Yes, got it, thanks. I forget the power/amplitude relationship.

KMO's reference to amplitude means signal level, not power. He has it right. 10 times the amplitude is actually 20 dB. Amplitude is 20log(A1/A2). Power is 10log(P1/P2). 10 dB is 3.16 times the amplitude (signal level). It is also a 10dB increase in power, so on that we agree.

Current is a linear function of voltage divided by the resistance of the load. Twice the voltage through the same load produces twice the current.

Since power is the product of voltage and current, the effect on power of any increase or decrease in voltage is squared: doubling the voltage produces four times the power; halving the voltage produces one-fourth the power.

Since doubling a number's log and then finding the antilog is yields the number's square, shouldn't the equation be "log(P1/P2)=2log(V1/V2)" ?

It's been decades since I studied this stuff in school, so please tell me where I'm mistaken.

Current is a linear function of voltage divided by the resistance of the load. Twice the voltage through the same load produces twice the current.

Since power is the product of voltage and current, the effect on power of any increase or decrease in voltage is squared: doubling the voltage produces four times the power; halving the voltage produces one-fourth the power.

Since doubling a number's log and then finding the antilog is yields the number's square, shouldn't the equation be "log(P1/P2)=2log(V1/V2)" ?

It's been decades since I studied this stuff in school, so please tell me where I'm mistaken. Your equation is perfectly correct mathematically. It's simply not adhering to the definition of a decibel, which derives from signal loss design criteria in early telephone transmission systems, as Wiki explains (http://en.wikipedia.org/wiki/Decibel).

Hi there,

After reading the topic tonight, I still have a simple question :

If i use a SPL-metter to calibrate the channels
A) With the test tone of my receiver, I should have 75db for every channel.
B) With the test tone DVE in bitstream, I should have 75db for the five channels, and 85db for the LFE.

Am I correct ?

Hi there,

After reading the topic tonight, I still have a simple question :

If i use a SPL-metter to calibrate the channels
A) With the test tone of my receiver, I should have 75db for every channel.
B) With the test tone DVE in bitstream, I should have 75db for the five channels, and 85db for the LFE.

Am I correct ?

Actually, it's exactly the other way around! If there is an LFE test tone on a disk, it will be sent to the receiver 10db quieter than the other channels, since the LFE signal is recorded on the disk and sent from the player to the receiver at -10db. (If the player does bass management as well, the LFE signal is sent at -15db, but very few players do bass management, so I'll use just the 10db figure here for convenience.)

The purpose of this whole discussion is to get that LFE up to the same level as the other channels, not to make it louder.

Unless your receiver has a way to switch a 10db boost on or off - or you know for a fact that it will be automatically compensating for the cut put into the recording - the way to counteract the cut in the signal coming from the player is to use the receiver's test tones to make the subwoofer test tone sound 10db louder than the other channels, so that when you play a disk all channels come out at the same level.

If your test disk has a true LFE channel output (I don't think that either AVIA or DVE has one), after making all adjustments you should hear all channels at the same level.

PS If your sound pressure level meter gives you a choice of weighting filters, use the C setting, not the A setting. The A setting sharply rolls off response at the bottom and top of the range to emphasize 3-6kHz, where the ear is supposed to be most sensitive to noise. The C setting is much more linear, and is what you should use to calibrate your speaker levels, particularly when matching a subwoofer to the other speakers. Here's a link to the Wikipedia article about the different weighting filters used in measuring sound levels: http://en.wikipedia.org/wiki/Weighting_filter

Actually, it's exactly the other way around! If there is an LFE test tone on a disk, it will be sent to the receiver 10db quieter than the other channels, since the LFE signal is recorded on the disk and sent from the player to the receiver at -10db. (If the player does bass management as well, the LFE signal is sent at -15db, but very few players do bass management, so I'll use just the 10db figure here for convenience.)

The purpose of this whole discussion is to get that LFE up to the same level as the other channels, not to make it louder.

Unless your receiver has a way to switch a 10db boost on or off - or you know for a fact that it will be automatically compensating for the cut put into the recording - the way to counteract the cut in the signal coming from the player is to use the receiver's test tones to make the subwoofer test tone sound 10db louder than the other channels, so that when you play a disk all channels come out at the same level.:confused:

With a disc that has a tone encoded in the LFE channel at the proper level you would simply adjust the LFE channel to the same level as the other speaker channels. If the signal is being passed as bitstream, the AVR will automatically apply the 10dB boost to the LFE channel. No need for any compensation. If the signal is being passed from a universal player as multichannel analog, then the proper boost (10dB if the front speakers are set to LARGE in the player, 15dB if they're set to SMALL) must be applied somewhere in order to get the LFE channel up to the proper level. Either way, though, provided the LFE channel is encoded identically to the way it is encoded on a DVD, the LFE channel must be adjusted to the same level as the other speaker channels.

I think the OP is alluding to the fact that the LFE channel tones on DVE, depending upon which version is used, MAY be encoded at the wrong level. Whether the tone is encoded at the proper level or not has always been somewhat controversial. That AVIA calibrates the sub out with bass-managed rerouted bass and not an LFE-channel-encoded tone is an advantage that AVIA has over DVE.

:confused:

With a disc that has a tone encoded in the LFE channel at the proper level you would simply adjust the LFE channel to the same level as the other speaker channels. If the signal is being passed as bitstream, the AVR will automatically apply the 10dB boost to the LFE channel. No need for any compensation. If the signal is being passed from a universal player as multichannel analog, then the proper boost (10dB if the front speakers are set to LARGE in the player, 15dB if they're set to SMALL) must be applied somewhere in order to get the LFE channel up to the proper level. Either way, though, provided the LFE channel is encoded identically to the way it is encoded on a DVD, the LFE channel must be adjusted to the same level as the other speaker channels.

I think the OP is alluding to the fact that the LFE channel tones on DVE, depending upon which version is used, MAY be encoded at the wrong level. Whether the tone is encoded at the proper level or not has always been somewhat controversial. That AVIA calibrates the sub out with bass-managed rerouted bass and not an LFE-channel-encoded tone is an advantage that AVIA has over DVE.

I agree that a user bitstreaming to an AVR can probably assume that the audio decoder in the AVR properly boosts the LFE channel to compensate for the cut made in creating the disk.

That would mean that all channels should be set to equal volume using the test tones and would still be equal during actual playback - the LFE would not be louder. That was the key point I was trying to make, though it got buried under the rest of what I said.

Being a multichannel analog interconnect user (my AVR is so old it not only doesn't know the new lossless codecs, it doesn't even have an HDMI input), I have to apply the 10db boost manually. (Fortunately, my AVR does apply its channel level and delay settings to the multichannel analog input, making life a little simpler.)

As far as the test disks are concerned, using only bass-managed re-routed bass is exactly the wrong way to set channel levels. What about those of us who don't - or can't - bass manage? Why not just give us a raw LFE tone to match against the other channels?

As an analog interconnect guy I couldn't bass-manage if I wanted to, since my player doesn't do bass management and my amp does not provide that service to the multichannel analog input.

Since I use full-range speakers in all positions, the only bass-management I would even consider using would be if it doubled the bass instead of completely re-routing it. That's what I do with my other inputs - but would that even yield the right results with these test disks?

My copy of DVE nowhere gives discrete tones for the channels. Instead it does a pan around the room to many points, including "ghost" locations where there aren't any speakers. That may be fun as a final check, but first please give separate - and clearly labeled - signals for each of the channels - since those are what have corresponding controls that can be adjusted!

Thanks for the answers, even if I'm not sure to understand :D

I thought the LFE channel in dobly digital and DTS, was encoded 10db lower than it should be (and no 10db lower than others channels).
My mistake.

So in short, for a perfect calibration with a SPL-metter :

- If I make a test tone directly from receiver, I should try to get 75db on all channels.

- If I make a test tone DD 5.1 or DTS from DVE disc in my player connected in bitstream to the receiver, I should try to get 75db on all channels.
(it means the LFE channel should be at 65db, but with the 10db boost in the receiver, it should finish to 75db)

- If I make a test tone PCM 5.1 from DVE disc in my player connected in analog to the receiver, I should try to get 75db on all channels.
(By adding 10db manually to the LFE).

So in every case, at the end, it's 75db on all channel.

Thanks for the answers, even if I'm not sure to understand :D

I thought the LFE channel in dobly digital and DTS, was encoded 10db lower than it should be (and no 10db lower than others channels).
My mistake.

So in short, for a perfect calibration with a SPL-metter :

- If I make a test tone directly from receiver, I should try to get 75db on all channels.

- If I make a test tone DD 5.1 or DTS from DVE disc in my player connected in bitstream to the receiver, I should try to get 75db on all channels.
(it means the LFE channel should be at 65db, but with the 10db boost in the receiver, it should finish to 75db)

- If I make a test tone PCM 5.1 from DVE disc in my player connected in analog to the receiver, I should try to get 75db on all channels.
(By adding 10db manually to the LFE).

So in every case, at the end, it's 75db on all channel.

You are correct, sir, since you are bitstreaming to an AVR that does your decoding and should know to do the 10db compensation.

Just be sure to set your SPL meter to use the flat "C" weighting - not the "A" weighting, which sharply rolls off frequencies below a few thousand Herz - making it barely register the subwoofer's output, which is centered around the hundred Herz region.

Thank you Phil, it really helped me ;)

So in theory, the LFE channel in DD, DTS, and PCM 5.1 is always 10db lower.
If you analyse the level without the 10db boost from the receiver, you should see 65db on the SPL (if the others channels are at 75db).

It took me the night, but now I get it :D

Thanks again.

Thank you Phil, it really helped me ;)

So in theory, the LFE channel in DD, DTS, and PCM 5.1 is always 10db lower.
If you analyse the level without the 10db boost from the receiver, you should see 65db on the SPL (if the others channels are at 75db).

It took me the night, but now I get it :D

Thanks again.Provided the calibration disc you are using is encoded properly, the bottom line is that you should calibrate the sub channel to the same level as the other channels. Whether you are using a digital connection or a multichannel analog connection. Forget the 10dB difference. It is irrelevant. The only time it is an issue is in a situation where you are unable to compensate for the 10dBs (or 15dBs, as the case may be) when calibrating a multichannel analog connection.

Now, whether your DVE disc is really encoded correctly is another question.

Provided the calibration disc you are using is encoded properly, the bottom line is that you should calibrate the sub channel to the same level as the other channels. Whether you are using a digital connection or a multichannel analog connection. Forget the 10dB difference. It is irrelevant. The only time it is an issue is in a situation where you are unable to compensate for the 10dBs (or 15dBs, as the case may be) when calibrating a multichannel analog connection.

Now, whether your DVE disc is really encoded correctly is another question. Yes. There's the question of the disc's accuracy, but it's almost impossible to use an SPL meter to check the LFE vs main channel gain unless the mains and LFE are carrying exactly the same shaped noise spectrum. That's pretty rare in test discs--they usually have LF noise for the LFE and mid-freq noise for the mains. It can be measured with reasonable accuracy--using a 1/3-octave RTA, but even then, if the freq response of the speakers is not perfectly flat, there will still be errors.

Dolby made a set of test signals on one of its demo DVDs (Explore Our World) where a low-frequency signal is stepped thru each of the 5.1 channels in a continuous loop (with the -10dB offset in LFE so it plays back at the same loudness). When played, as long as the noise level remains constant as it steps thru all the channels, you know the bass management is working correctly. An SPL meter works well in this case--but so do one's ears.

.................but it's almost impossible to use an SPL meter to check the LFE vs main channel gain unless the mains and LFE are carrying exactly the same shaped noise spectrum. That's pretty rare in test discs--they usually have LF noise for the LFE and mid-freq noise for the mains. It can be measured with reasonable accuracy--using a 1/3-octave RTA, but even then, if the freq response of the speakers is not perfectly flat, there will still be errors.Right. But that's what I like about AVIA's tones which are encoded in the main channels, only. Same problem is there, probably, but at least it is somewhat ameliorated. I don't know exactly what or how the subwoofer calibration tones are encoded on AVIA. All I know is that they "warble" back and forth between the main channel(s) and the sub and the goal is for the meter not to change as the tone moves back and forth. And that this is pretty easy to achieve. Whether they have encoded things such that all (or some) of the issues you mention (and some others) are accounted or compensated for I do not know. But I always say that the discs are OK to get you in the ballpark but that most people end up adjusting their sub a few dBs here or there to taste, anyway.


Dolby made a set of test signals on one of its demo DVDs (Explore Our World) where a low-frequency signal is stepped thru each of the 5.1 channels in a continuous loop (with the -10dB offset in LFE so it plays back at the same loudness). When played, as long as the noise level remains constant as it steps thru all the channels, you know the bass management is working correctly. An SPL meter works well in this case--but so do one's ears.Bass management? Or just the calibration levels? I would think that the introduction of bass management might obscure the results with such tones. You'd leave all your channels set to LARGE, right?

As far as the test disks are concerned, using only bass-managed re-routed bass is exactly the wrong way to set channel levels. What about those of us who don't - or can't - bass manage? Why not just give us a raw LFE tone to match against the other channels?Well, there is very good logic behind using bass-managed tones encoded in the main channels. But, yes, obviously this won't work if your channels are set to LARGE. Supposedly, though, you should be able to set your speakers (or a channel) to SMALL to calibrate the sub with AVIA and then switch it/them back to LARGE and the calibration should still be valid. Now, that assumes your player's bass management works properly.

Right. But that's what I like about AVIA's tones which are encoded in the main channels, only. Same problem is there, probably, but at least it is somewhat ameliorated. I don't know exactly what or how the subwoofer calibration tones are encoded on AVIA. All I know is that they "warble" back and forth between the main channel(s) and the sub and the goal is for the meter not to change as the tone moves back and forth. And that this is pretty easy to achieve. Whether they have encoded things such that all (or some) of the issues you mention (and some others) are accounted or compensated for I do not know. But I always say that the discs are OK to get you in the ballpark but that most people end up adjusting their sub a few dBs here or there to taste, anyway. From your description it appears AVIA has done something very similar to the Dolby test--the LF signal pans from main to LFE and back. It adds a bit of uncertainty as it occupies the pair in the middle of the pan, but ought to work fine.

Bass management? Or just the calibration levels? I would think that the introduction of bass management might obscure the results with such tones. You'd leave all your channels set to LARGE, right?These "equal loudness" bass test tones are useful for calibrating full range mains to the sub, or making sure the bass management in a decoding disc player is set to work correctly with the analog bypass in the AV system. It also sometimes leads folks who choose "full range" to find out their mains aren't really as full as they hoped.

From your description it appears AVIA has done something very similar to the Dolby test--the LF signal pans from main to LFE and back. It adds a bit of uncertainty as it occupies the pair in the middle of the pan, but ought to work fine.They're just tones encoded in the main channels that go up and down in frequency and are routed properly by the processor's bass management. So as the tone goes up and down in pitch it moves back and forth between the speaker and sub. Of course, this absolutely requires setting the speakers to SMALL, at least during the calibration, in order for it to work properly. There is a tone included for each speaker channel. What you find out if you try calibrating the sub to each speaker is that, presumably due to the effects of the room and phasing, the proper level for the sub is different for each channel. And they don't really explain to which channel you should calibrate the sub (at least not in my version). It's a bit amorphous.

I don't know any of the details or rationale (if any) used in encoding the tone; i.e. how (or if) it might be weighted differently at the low end. Etc..

They're just tones encoded in the main channels that go up and down in frequency and are routed properly by the processor's bass management. So as the tone goes up and down in pitch it moves back and forth between the speaker and sub. Of course, this absolutely requires setting the speakers to SMALL, at least during the calibration, in order for it to work properly. There is a tone included for each speaker channel. What you find out if you try calibrating the sub to each speaker is that, presumably due to the effects of the room and phasing, the proper level for the sub is different for each channel. And they don't really explain to which channel you should calibrate the sub (at least not in my version). It's a bit amorphous.

I don't know any of the details or rationale (if any) used in encoding the tone; i.e. how (or if) it might be weighted differently at the low end. Etc..Thanks for the further details. Now I understand. I was also confusing the issue of the 10 dB gain offset vs Frenshprince's original question about how to set the gain of his sub wrt the main speakers.

Let me take a different tack toward a reply. Forget about the cal discs for the moment. Because if you were to cal the sub gain using the disc and then you say "everything sounds too a) boomy or b) weak in the bass" then what would you do--a) live with it because it is "accurate" or b) adjust the sub level until it sounds better? Methinks b) is the right answer.

Now it just so happens that with either the Dolby test disc or DVE (I have the earlier VE disc which I assume has the same siganls as DVE), the noise signals are correctly recorded at -30 dBFS mains, -40 dBFS LFE, and when the mains read "0" on the SPL meter, the LFE signals will read in the vicinity of 0 also. In my case, it's +2 or +3, it bobbles around. But the reading will be greatly affected by the sub's response, the meter's response, yada yada. So it is nothing more than a basic sanity check when using the LFE test signal. If you are getting bass readings like +/- 6dB or more vs the mains, some investigating needs to be done.

Barring that, just set the sub gain to blend well with the mains. After you have done that, it is good to take a reading with the test disc for future ref, in case someone messes with the sub gain, it can be brought back to your "house cal" rather easily with the test disc, rather than playing your various "fave bass discs" all over again.

Question for KMO or Roger or anyone.

Roger will know i have been tinkering with my Oppo Blu ay player painstakingly trying to get the bass levels deep and controlled with some nice output.

I managed to get it sounding as good as the coaxial output for bass although it makes me wonder why i'm bothering with analog but i guess i'm hoping lossless has some benefits at the higher frequency ranges and offers something richer and more detailed but i'm thinking the difference to my ears will be minimal and i'm just buying into the lossless is better because it's lossless scene. If nothing else the placebo effect will be nice and i'll probably think it's better even if it's not.

My subwoofer is currently around 8db hot. Ideally speaking i want it within 3db of the main speakers but would probably live with 5db.

Can i add 5db to the mains and center and surrounds in the Oppo Blu Ray player speaker configuration or is this clipping issue going to rear it's head even though playback volume will likely never exceed 95db and most likely 90db.

My Sony amp says the analog input stage is 2.5v. Could i damage anything by boosting channels by 5db and if so then how about 3db.

Lowering the subwoofer isn't an option since i love the sound of the bass now.

What other options exist for getting the mains and center and surrounds up and remember i have no receiver control of these features when using analog in.

I thought i'd post this here since i have posted a lot in the Oppo thread already and this is more an audio question.

Question for KMO or Roger or anyone.

Roger will know i have been tinkering with my Oppo Blu ay player painstakingly trying to get the bass levels deep and controlled with some nice output.

I managed to get it sounding as good as the coaxial output for bass although it makes me wonder why i'm bothering with analog but i guess i'm hoping lossless has some benefits at the higher frequency ranges and offers something richer and more detailed but i'm thinking the difference to my ears will be minimal and i'm just buying into the lossless is better because it's lossless scene. If nothing else the placebo effect will be nice and i'll probably think it's better even if it's not.

My subwoofer is currently around 8db hot. Ideally speaking i want it within 3db of the main speakers but would probably live with 5db.

Can i add 5db to the mains and center and surrounds in the Oppo Blu Ray player speaker configuration or is this clipping issue going to rear it's head even though playback volume will likely never exceed 95db and most likely 90db.

My Sony amp says the analog input stage is 2.5v. Could i damage anything by boosting channels by 5db and if so then how about 3db.

Lowering the subwoofer isn't an option since i love the sound of the bass now.

What other options exist for getting the mains and center and surrounds up and remember i have no receiver control of these features when using analog in.

I thought i'd post this here since i have posted a lot in the Oppo thread already and this is more an audio question.

The backward-compatible Dolby or DTS soundtracks that fit over a coaxial or optical SPDIF connection are comparable to a high bitrate mp3: to fit in a confined data space, fine detail in the higher harmonics is thrown away. They're probably at a higher bitrate than on a DVD, but they're still lossy compression.

The easiest way to hear the difference between lossless and lossy audio on a Blu-ray setup is to put on a disk with well-recorded small ensemble music, or quiet dialog with ambient sound. What you should listen for is the realism conveyed by high-frequency ambience, what I call "air."

Find the button on your remote that lets you cycle through and change soundtracks without re-starting the disk. Hearing the difference is hard if several minutes and several movie promos come in between. Pirates of the Caribbean I has an excellent soundtrack, but unless you can easily toggle between the different versions, it's hard to use for this purpose.

My favorite music disk for this purpose is the Legends of Jazz Showcase, a highlights disk from that PBS hi-def series recorded in studio by a top-notch crew. I'm sure that a good classical small ensemble disk would show this difference as well.

Spiderman II has a scene near the beginning with Peter Parker in J. Jonah Jameson's office that has dialog and ambient sound, but if you put on Stranger than Fiction, you'll be in for a treat in terms of a subtle soundscape: early in the film, when the main character is brushing his teeth, you can just hear water running in another appointment. When he meets the young baker who has withheld a part of her taxes as a war protest, and begins to imagine her at a demonstration, the surrounds carry a quiet sigh and - for a few seconds - his heartbeat, as he begins to fall in love with her. All the acoustic environments in that film sound real. (And the film is a little-recognized masterpiece that walks the line between comedy and tragedy - and reality and fantasy - with a sure touch and a lot of heart.)

By the way - unless your receiver's manual says that its channel level and delay controls don't affect the multichannel input, it's possible that they do - with my receiver, a pre-HDMI Yamaha RX-V457, the manual doesn't say that they don't, and in fact they do.

It's easy to do a simple test to see what the situation really is. Go into the receiver's channel level controls and turn one front channel all the way down and one all the way up, and see if that affects what happens when you play something through the multichannel input jacks.

If that does affect the channel levels, you've got the additional control you need.

Also test whether the delay controls work for the multi's: using two speakers that are the same distance from you, set one channel to maximum distance (minimum added delay) and one to minimum distance (maximum added delay) and see if there's any time lag between them. If there's a time lag, you only have to configure speaker distances/delays in the receiver, which will work for all sources.

The easiest way to hear the difference between lossless and lossy audio on a Blu-ray setup is to put on a disk with well-recorded small ensemble music, or quiet dialog with ambient sound. What you should listen for is the realism conveyed by high-frequency ambience, what I call "air."

Find the button on your remote that lets you cycle through and change soundtracks without re-starting the disk. Hearing the difference is hard if several minutes and several movie promos come in between. Pirates of the Caribbean I has an excellent soundtrack, but unless you can easily toggle between the different versions, it's hard to use for this purpose.



This may be an easy way to hear a difference, but it's not a good way to determine if the difference is real, or whether it is really due to an inherent sound of the format. "Sighted' tests never are. And it's hard to know if the mastering, and playback processing (level-matching) are the same for both versions. Any of these confounders -- 'sighted' bias, different mastering, level mismatch -- can lead one to a false conclusion about the sound of the *formats*.

This may be an easy way to hear a difference, but it's not a good way to determine if the difference is real, or whether it is really due to an inherent sound of the format. "Sighted' tests never are. And it's hard to know if the mastering, and playback processing (level-matching) are the same for both versions. Any of these confounders -- 'sighted' bias, different mastering, level mismatch -- can lead one to a false conclusion about the sound of the *formats*.

With my Panasonic DMP-50, switching between soundtracks with the Audio button does not lead to level mismatches - all versions are at the same volume level through the multichannel analog outputs.

I understand the value of doing this blind, so if you can recruit a friend to do the switching for you, do so by all means. However, if you can't get a friend to help, it's better to do it yourself than not do it at all.

Beyond that, I'm not sure what krabapple is recommending.

If there is a difference in mastering due to the greater bandwith constraint on a lossy track, it's part of the difference between the formats, not an extraneous "confounder" making comparison impossible.

If you're comparing different models of encoder of the same type, you might be concerned about differences between the mastering done by different engineers for different films making it hard to compare.

However, what I'm advocating comparing are different end-to-end processes on the same film by the same engineering team - to decide whether it's worth spending extra to hear the lossless tracks versus the lossy tracks.

If the difference is worth spending money on, it should be audible in an A/B test switching between soundtracks on a Blu-ray with the levels the same, and should be clear enough that there's no worry about "sight bias."

If the difference is so slight that only a blindfold test can be relied on to find it, it's not worth spending extra money to get a player that decodes the lossless formats or a receiver that does so.

The only alternative is not to do a test and simply take it on faith, relying on the word of those in the industry who say that it's better.

Or is krabapple saying that lossless really isn't better, and those of us who can hear a difference are fooling ourselves?

With my Panasonic DMP-50, switching between soundtracks with the Audio button does not lead to level mismatches - all versions are at the same volume level through the multichannel analog outputs.

This assumes the two versions are mastered at the same level, and that things like dialnorm don't make a difference.


I understand the value of doing this blind, so if you can recruit a friend to do the switching for you, do so by all means. However, if you can't get a friend to help, it's better to do it yourself than not do it at all.


It's better still to not make definitive claims about format audibility from sighted comparisons, or from poorly controlled tests generally. That's what krabapple is recommending.

Beyond that, I'm not sure what krabapple is recommending.

If there is a difference in mastering due to the greater bandwith constraint on a lossy track, it's part of the difference between the formats, not an extraneous "confounder" making comparison impossible.

A difference in mastering need not be due to the greater bandwidth constraints.


If you're comparing different models of encoder of the same type, you might be concerned about differences between the mastering done by different engineers for different films making it hard to compare.

Yes, you should certainly be concerned about that, under those conditions.


However, what I'm advocating comparing are different end-to-end processes on the same film by the same engineering team - to decide whether it's worth spending extra to hear the lossless tracks versus the lossy tracks.

And I'm advocating not making conclusions from poorly controlled tests of this idea, where there are still variables unaccounted for.


If the difference is worth spending money on, it should be audible in an A/B test switching between soundtracks on a Blu-ray with the levels the same, and should be clear enough that there's no worry about "sight bias."

Unfortunately for that idea, you always have to worry about 'sighted bias'.

If the difference is so slight that only a blindfold test can be relied on to find it, it's not worth spending extra money to get a player that decodes the lossless formats or a receiver that does so.


A blinded test is not a 'blindfold test'. A blind test in this case means you have no knowledge beforehand of which format you are listening to right now.
Thus your only means of differentiating them is by hearing alone.


The only alternative is not to do a test and simply take it on faith, relying on the word of those in the industry who say that it's better.

Or maintaining a healthy skepticism; one could even hold the hypothesis that two are unlikely to sound different under typical circumstances.

Or is krabapple saying that lossless really isn't better, and those of us who can hear a difference are fooling ourselves?

Krabapple says that is a possibility.

Posted by Philnick:

Or is krabapple saying that lossless really isn't better, and those of us who can hear a difference are fooling ourselves?


Krabapple says that is a possibility.

Smoked you out!

Dear Krabapple (I will admit that your nom de plume did give me a hint of which category you are in of those who say that ordinary mortals can't determine for themselves whether lossless is better: the experts or the deniers),

Please get yourself to some place where you can do a real-world A/B on an actual Blu-ray disk so you can hear what we're talking about. The difference is not theoretical. Unless you're suffering from hearing loss that prevents you from hearing high frequency harmonics and the separation of layers that the absence of volume compression makes possible, the difference is clear.

If you can't hear the difference you can certainly save a lot of money on your setup and can be happy with low-bandwidth mp3s, but don't begrudge those of us who can hear the difference the pleasure of doing so. We're not crazy or deluded.

Posted by Philnick:

Or is krabapple saying that lossless really isn't better, and those of us who can hear a difference are fooling ourselves?



Smoked you out!


er..right, Sherlock. :rolleyes:

Next time, just look up my posting history if you're that interested.


Dear Krabapple (I will admit that your nom de plume did give me a hint of which category you are in of those who say that ordinary mortals can't determine for themselves whether lossless is better: the experts or the deniers), Please get yourself to some place where you can do a real-world A/B on an actual Blu-ray disk so you can hear what we're talking about. The difference is not theoretical. Unless you're suffering from hearing loss that prevents you from hearing high frequency harmonics and the separation of layers that the absence of volume compression makes possible, the difference is clear.


OK, if this is the best case you can make -- "I hear it in a A/B, therefore it's real' -- you can go away now, because you can't be taken seriously.

Btw, lossy compression does not entail 'volume compression'.


If you can't hear the difference you can certainly save a lot of money on your setup and can be happy with low-bandwidth mp3s, but don't begrudge those of us who can hear the difference the pleasure of doing so. We're not crazy or deluded.

Some audiophiles in my experience qualify as 'deluded'. But even if you're not, you might still just be *wrong*. Can you admit that possibility?

And speaking of real-world experience, have you done ABX tests of high-quality mp3s vs lossless, for example, like I have? What one finds when one does that, is that the vast majority of times, no statistically significant difference is the result. (There are, however, some 'killer samples' that can defeat even the best mp3 codec, but these are rare and often the 'tipoff' is just a small section of a much longer track.)

And if that's the case for mp3, a perceptual encoder whose bitrate is at best 320 kbps, what does that suggest for the perceptual encoders used by Dolby Digital or DTS?

(Btw, your profile says you're 59; making you about a decade older than me. There's a good chance your high-frequency hearing is no better than mine.)

Btw, lossy compression does not entail 'volume compression'.

I'm quite aware of that - in fact I edited out a section of my response talking about the dual meanings of the word compression in connection with audio encoding to avoid being too pedantic.

With respect to lossless compression, there's a claim of there being no volume compression in the decoded file. Is there any such claim for DD or DTS?


Some audiophiles in my experience qualify as 'deluded'. But even if you're not, you might still just be *wrong*. Can you admit that possibility?

Can you admit the possibility that there is a difference?

This is a debate that goes back to SACD and DVD-A. Your requirements for deciding rule out any practical way to determine whether there is a difference without specialized test equipment.

You are too insistent on excluding the most important test: whether over a range of actual availiable disks, there is a consistent perceivable improvement from lossless.

If there isn't, a lab test saying there is wouldn't matter.

If there is, a lab test can provide useful confirmation, but cannot, logically, disconfirm it. Even if a particular test doesn't confirm it, there remains the possibility that the criteria for the test are too narrow, missing potential differences that our senses can detect. Tests only measure what we know to look for.

Perhaps the lab test uses too narrow a frequency bandwidth, by assuming that human audio perception cuts off earlier than it really does - we may "feel" sounds that we can't "hear" and thus haven't reported to audiologists that they can perceive sounds that high.

(Btw, your profile says you're 59; making you about a decade older than me. There's a good chance your high-frequency hearing is no better than mine.)

When I was young, I could hear a CRT television set being turned on, with the sound off, several rooms away through walls - I could hear the whine of the 19 kHz "flyback coil" used to create the high voltage needed to make the electron beam jump from the cathode to the screen.

I can't hear that any more, but I still hear some things before those around me do.

The more important question is not whether my eardrums have hardened, but whether your worldview has. I keep my ears - and my mind - open. Please try to do the same.

krabapple/Philnick,

Thread is getting off topic. This debate will never end. There are better threads where you can discuss this, including this fine one (shameless plug):

http://www.avsforum.com/avs-vb/showthread.php?t=1066645

For the record, I can still hear the CRT whine. ;)

That's all great Philnick but the relevant issue (which has unfortunately been pushed aside) is whether or not it is worth struggling with the difficulties and shortcomings of a multichannel analog connection in order to pursue the lossless codecs. Does the (purported) benefit outweigh the downside? FoxyMulder has made it pretty clear that he has no good way to adjust for the 10dB (or 15dB if his speakers are set to SMALL) deficit in his player's LFE/subwoofer channel. He claims that he cannot adjust the individual level trims of his processor's external inputs and we have to accept that he knows what he is talking about as it seems from his post that he is familiar with his particular equipments' limitations and the issues he is having. In addition to this issue, his player probably offers very poor and/or limited bass and time management capabilities relative to what he may be capable of achieving with his processor. So, everything has to be considered.

Just curious, did you ever complain about the sound quality of your DVDs' soundtracks prior to the advent of the higher resolution codecs?

You guys are sullying a stickied thread.

Question for KMO or Roger or anyone.

Roger will know i have been tinkering with my Oppo Blu ay player painstakingly trying to get the bass levels deep and controlled with some nice output.

I managed to get it sounding as good as the coaxial output for bass although it makes me wonder why i'm bothering with analog but i guess i'm hoping lossless has some benefits at the higher frequency ranges and offers something richer and more detailed but i'm thinking the difference to my ears will be minimal and i'm just buying into the lossless is better because it's lossless scene. If nothing else the placebo effect will be nice and i'll probably think it's better even if it's not.

My subwoofer is currently around 8db hot. Ideally speaking i want it within 3db of the main speakers but would probably live with 5db.

Can i add 5db to the mains and center and surrounds in the Oppo Blu Ray player speaker configuration or is this clipping issue going to rear it's head even though playback volume will likely never exceed 95db and most likely 90db.

My Sony amp says the analog input stage is 2.5v. Could i damage anything by boosting channels by 5db and if so then how about 3db.

Lowering the subwoofer isn't an option since i love the sound of the bass now.

What other options exist for getting the mains and center and surrounds up and remember i have no receiver control of these features when using analog in.

I thought i'd post this here since i have posted a lot in the Oppo thread already and this is more an audio question.

If you really can't get the bass levels over analog to match digital, I say just use optical/coaxial for lossless and call it a day. The full spec lossy codecs (ie. 640kbps DD and 1.5Mbps DTS) are close enough to the lossless where you'll only be missing the discrete RS of a 7.1 lossless track. The traditional DD and DTS tracks on DVD were not so good...

I'm quite aware of that - in fact I edited out a section of my response talking about the dual meanings of the word compression in connection with audio encoding to avoid being too pedantic.

With respect to lossless compression, there's a claim of there being no volume compression in the decoded file. Is there any such claim for DD or DTS?

There is no particular reason for there to be. I have DTS and DD audio files, ripped directly from surround audio discs, that are probably compressed in terms in dynamic range, and others that are not, simply based on their extremely variable loudness. It's a mastering choice.



Can you admit the possibility that there is a difference?

This is a debate that goes back to SACD and DVD-A. Your requirements for deciding rule out any practical way to determine whether there is a difference without specialized test equipment.

So what? It might be that that's the only way to find out the truth of the matter.

But in fact, for mp3s, it's not hard at all for someone with a PC and an internet connection to set up a good ABX test. And if you find that it's hard to tell mp3s from source, what would that say about these other formats?


You are too insistent on excluding the most important test: whether over a range of actual availiable disks, there is a consistent perceivable improvement from lossless.

I'm happy to accept objective measurement data that indicates likely audible difference, but without it I'm insistent on recognizing the power of confounding factors that can lead us to erroneous conclusions about difference. It's not just me. It's why scientific work in psychoacoustics always incorporates controls. It's why hydrogenaudio.org, another audio forum, insists on DBT evidence for claims of differences. It's just good science. Routine 'audiophile' practice is rarely good science.


If there isn't, a lab test saying there is wouldn't matter.

That depends entirely on the conditions used for the two 'tests'.

If there is, a lab test can provide useful confirmation, but cannot, logically, disconfirm it. Even if a particular test doesn't confirm it, there remains the possibility that the criteria for the test are too narrow, missing potential differences that our senses can detect. Tests only measure what we know to look for.

That's the 'god in the gaps' argument. But fortunately, science doesn't work that way. It acknowledges its statistical nature -- there can be no 'absolute disproof' --but also acknowledges likelihood and accumulated evidence. We know that we are not perfect, infallible perceivers of reality. We know we are prone to 'confounders'. The burden is on the claimant to show that the difference they heard is likely to be real, either by comparison of measured difference, or listening tests.

As has been hinted, this is an old, tedious argument, and you certainly aren't bringing anything to it that hasn't been addressed dozens of times already. Please desist here and start a new thread if you really need to hash this out

The full spec lossy codecs (ie. 640kbps DD and 1.5Mbps DTS) are close enough to the lossless where you'll only be missing the discrete RS of a 7.1 lossless track. The traditional DD and DTS tracks on DVD were not so good...

Again, if 320 kbs LAME encoded mp3 is extremely hard for people to tell from lossless...and all evidence to date is that , it is -- and granting that bitrate is not the *only* determinant of perceptual codec performance -- why would a highly-developed codec like DD at the 'standard' 384-448 kbs be predicted to routinely sound different? Or DTS at the standard 1500 kbps?

One *can* always come up with a sample that produces audible artifacts with any lossy codec. But this does not mean such artifacts occur routinely.

The fact is that well-controlled comparisons are DD, DTS and various newer codecs to lossless are just not common in the scientific literature. There is little financial incentive, apparently, to perform them (it's unlikely that Dolby or DTS, who are now also pushing lossless codecs, are going to popularize tests that indicate it makes little difference). This leaves the field open to all sorts of marketing hype that plays on common consumer biases.

My car stereo can play mp3 disks. When I first got it, I compared the sound quality of a CD against an mp3 made from it at Winamp's highest bit rate, its "insane" setting (their term) of up to 320 kbps variable bit rate. I could hear the difference in the ambience, but the mp3 sounded good enough that the convenience of fitting a lot of material on a CD and being able to see the track info on the dash has led me down that path for listening in the car. After all, road noise in a moving car would drown out the difference I could hear standing still.

Dolby at up to 640 kbps will sound better than mp3 at 320, but the comparison is not against a standard CD but against the original source, so it's still playing catch up.

I grant that at 1.5 Mbps, Blu-ray's lossy DTS would sound very good and if every disk had standard DTS on it, an SPDIF connection carrying standard DTS would be very good.

But Dolby has a very strong hold on the movie industry, and many disks have only Dolby TrueHD or LPCM available as the high quality choice. An SPDIF optical or coax connection can only carry 2 channels of PCM, not six or eight.

I'm going to withdraw from this field of battle.

I made my first post in this current debate here only to offer FoxyMulder a way to listen for the improvement that lossless offers. I didn't intend to start a debate on the subject, since I didn't realize that it was still a live controversy.

I have no economic interest in this - I set up lossless by the least expensive method I could, by buying a player that could decode the new codecs and feed them as analog to my old 5.1 receiver, rather than spending much more money on a new 7.1 receiver and additional speakers. (Again, the available disks played a deciding role - most available material is only 5.1.)

Let the disbelievers think what they want - all I'm saying is that it's not "received truth" that everyone who thinks that lossless sounds better is kidding themselves.

Let the disbelievers think what they want - all I'm saying is that it's not "received truth" that everyone who thinks that lossless sounds better is kidding themselves.

The 'received truth' for consumers is quite the opposite and would go something like this: 'obviously lossless sounds better than lossy'. They would be surprised to hear that there is any controversy about it. In other words, it's what *you* believe.

It's very likely that the vast majority of consumers who believe lossless sounds better than lossy, have never properly tested whether they can hear the difference. Going by your testimony, that would include you.

If you really can't get the bass levels over analog to match digital, I say just use optical/coaxial for lossless and call it a day. The full spec lossy codecs (ie. 640kbps DD and 1.5Mbps DTS) are close enough to the lossless where you'll only be missing the discrete RS of a 7.1 lossless track. The traditional DD and DTS tracks on DVD were not so good...

I have been able to match the coaxial output.

What i have found is it's all dependent on master volume level as to whether you get the very deep bass slam or a whimper.

Film soundtracks are mixed that way so if you get your subwoofer to within 3db of your main speakers and then lower the volume to 85db at peaks of the soundtrack the bass will be a whimper instead of deep slam. This is not an issue with the soundtrack it's how it was recorded and it's probably why some like running their subwoofers hot.

Now if your subwoofer is within 3db of your other speakers as mine now going through the analog multichannel setup and you increase the master volume so that for example the explosion scene at the beginning of Star Wars Attack oF The Clones now registers 100db to 105db during those peaks then the slam factor returns.

Now raise again to reference levels and get the best possible deep bass for the soundtrack while keeping levels within 3db of each other.

It's all relative to how the film soundtrack was mixed and recorded and most are recorded so mains have peaks of 105db and sub has a peak of 115db and i think someone replied to me in the subwoofer forum and said when all speakers including subwoofer peak at the same time its 116.5db for film soundtracks.

Thanks to Rodger i have adjusted the actual subwoofer volume or rather decreased the subwoofer volume so the level mismatch between it and front speaker is no longer there and it's within 3db of them.

I was complaining that if i lowered the sub to get within 3db of the mains that i missed the slam factor and deep bass. Obviously thats because of the way the film soundtrack is recorded and the fact i was listening at too low a volume. Increase the volume as i said in the first paragraph and you get the slam and deep bass back and keep all speakers inc subwoofer to within 3db.

No mucking about with any controls on the receiver.

All i then needed to do was slightly adjust the speaker levels in the trim settings of the Oppo and i trimmed the left and center speakers by 0.5db and the surrounds by 3db ( after further testing today )

Situation is it's now all fantastic.

I can't put my finger on it but lossless tracks seem to have a more airy breathy quality about them. It's like tiny little bits of information which were muffled in the compressed sound of a DVD have returned. I checked this using a few discs and maybe it is just a placebo effect. Who knows for sure since level matching coaxial and analog is extremely hard to do but i think i managed it which is what i was trying to do and there is no harm in having lossless and at least thinking it's better. I think it's better.

Since this is audio theory and chat one other thing i noticed while using a number of calibration discs from AIX to Avia to THX test tones found on DVD's and checking all of them while trying to level match and of course finding all showed different readings.

When outputting THX test tones using the Oppo through multichannel analog in i noticed the calibration settings i had set were identical for each front, center and surround channel. Now switching to coaxial i noticed all channels were identical except the center channel. It was 4db higher now. I believe this is dialogue norm at work.

This actually got me thinking.

I have just been re-calibrating my whole system since getting the Oppo and i originally had calibrated settings for coaxial using Avia. If the THX test tones on Star Wars Attack Of The Clones were 4db higher for the center channel does this also mean when watching the movie using coaxial the center channel was 4db higher in volume or does the receiver actually adjust this. If the receiver adjusts it so why not for the test tones.

This also got me thinking. So now i have calibrated and everything is set up to level match. What happens if i play a movie with dialogue norm and output to the multichannel analog in's ?

I have to admit i can't understand why THX test tones sent a signal which was 4db higher to the center channel but only when using the coaxial input. Anyone know the logic behind that and is information being lost when outputting via analog in as the center was not raised by 4db when tested using analog in. Whats happening here ?

Incidentally Phil i think you will find with regards Blu Ray releases i think DTS is pretty equal if not overtaking Dolby and really making inroads with this new format ( 3 year old ) and i hear Sony may make future releases all DTS but i'm not sure when or if it's totally accurate but there is a thread in the Blu Ray software forum about it. I think at the moment Fox, Lionsgate and Universal tend to use DTS while Warner, Paramount and Sony tend to use Dolby True HD for most releases and Disney may well be going DTS too for most of it's releases.

DTS had a much harder time getting their sound codec onto DVD due to space limits. Thats no longer an issue with Blu Ray.

When outputting THX test tones using the Oppo through multichannel analog in i noticed the calibration settings i had set were identical for each front, center and surround channel. Now switching to coaxial i noticed all channels were identical except the center channel. It was 4db higher now. I believe this is dialogue norm at work. I can assure you, it has nothing to do with DialNorm, since that is a global level shift across all channels--same as tweaking the master volume.

I have just been re-calibrating my whole system since getting the Oppo and i originally had calibrated settings for coaxial using Avia. If the THX test tones on Star Wars Attack Of The Clones were 4db higher for the center channel does this also mean when watching the movie using coaxial the center channel was 4db higher in volume or does the receiver actually adjust this. If the receiver adjusts it so why not for the test tones. When using test signals from a given disc it is possible to make an accurate comparison between the analog and digital inputs. If one uses the available trimmers to make them read the same, and of course the bass management settings are also matched, then the sound will be remarkably similar between the two types of connections--as I think you have found.

The problem with comparing levels from two different test discs is that the noise spectrum of the test signals may be different. If there are any freq response irregularities across the speakers, for any reason (like the room acoustics or a TV near the C speaker) then the levels from each speaker may vary from one disc as compared to another. Only if the speaker responses resulting at the SPL meter's location are very flat in the region spanning all the noise test signals can we expect the different test discs to agree.

If you are seeing such disagreements among test discs (in the main channels--ignore the sub for now) look into ways to reduce the response variations across the speakers. That will of course improve not only testing uniformity, but overall sound quality as well. It's often not easy to get the C speaker to match the L/R, but the effort is worth it.

I can assure you, it has nothing to do with DialNorm, since that is a global level shift across all channels--same as tweaking the master volume.



If you are seeing such disagreements among test discs (in the main channels--ignore the sub for now) look into ways to reduce the response variations across the speakers. That will of course improve not only testing uniformity, but overall sound quality as well. It's often not easy to get the C speaker to match the L/R, but the effort is worth it.

All the same sound levels when i used the analog ins.

The only difference i heard and recorded using the RS sound level meter was when i switched to coaxial the center channel was 4db higher in volume and even without the meter it could be heard as a volume change.

Switching back to analog in and all speakers were identical again.

It's possible a trim i applied in the Oppo to match the left and center channel to the front right which has a slight difference in sound levels due to being placed near a room boundary but i didn't apply a 4db cut so that was odd.

I'll run a few tests again tomorrow by putting the trim back to it's 0 setting in the Oppo for analog and running the test against coaxial.

I am very pleased though with the sound levels now. I also wish i did not have neighbours. Thanks for all your help.

Another issue here, FoxyMulder, is that there may be differences in the bass management properties of your processor versus the player. If the crossovers are a little bit different, have different slopes, etc., etc., you are going to get slight variations in the overall sound due to interactions with your room at different frequencies which may or may not be being reinforced or canceled.

All the same sound levels when i used the analog ins.

The only difference i heard and recorded using the RS sound level meter was when i switched to coaxial the center channel was 4db higher in volume and even without the meter it could be heard as a volume change.

Switching back to analog in and all speakers were identical again.

It's possible a trim i applied in the Oppo to match the left and center channel to the front right which has a slight difference in sound levels due to being placed near a room boundary but i didn't apply a 4db cut so that was odd.

I'll run a few tests again tomorrow by putting the trim back to it's 0 setting in the Oppo for analog and running the test against coaxial.

I am very pleased though with the sound levels now. I also wish i did not have neighbours. Thanks for all your help.

Just double checking:

For audio over analog, the Oppo determines the bass management, level, delay, etc. For audio over digital (coaxial/optical), settings in the receiver control.

It's my mistake i actually had the center channel raised in the receiver for the coaxial input and of course a slight trim of the center in the Oppo for the analog ins and this is why there was a 4db difference recorded when switching between inputs.

Hello,

I'm the owner of a Denon AVR 2105 and an Panasonic DMP-BD80.
I use Coax and Analog for the connection.
Because when i watch dvd's, I want to let the Denon handle al the decoding
And when i watch a BR disc, i want to enjoy HD-audio.

I have also added a boost of 15DB for the analog connection.
But the bass on the Blu-ray is mutch to loud.
And when i watch a dvd the bass is normal again.
What to do?

I have also added a boost of 15DB for the analog connection.
But the bass on the Blu-ray is mutch to loud.
And when i watch a dvd the bass is normal again.
What to do?Calibrate both connections with a calibration DVD. If you don't have that, calibrate the BD player's analog connection with the player's internal tones.

Do you have your speakers set to SMALL in the player or not? Where are you applying the 15dB boost; player or AVR?

I have also added a boost of 15DB for the analog connection.
But the bass on the Blu-ray is mutch to loud.
And when i watch a dvd the bass is normal again.
What to do?You should do a proper calibration for each source, rather than simply selecting the +15dB setting in your Denon. I have speakers set to small on a BD55 and the player output levels at 0. I find that the +10dB setting for the analog SW in my Denon 3805 is plenty. I did the calibration tweaks for analog in my Denon, not the player, because the Panasonic user interface is so wretched.

The bass is correct when using the digital connection because the +15dB boost only applies to the Ext Inputs.

Apologies guys if this is answered elsewhere.

I have an ageing Denon AVC-11SR (UK model), THX Ultra Amp etc.

When setting up all speakers I set the test tone to 75db (Small) including the normal subwoofer output.

When I put through an LFE test tone do I also set this to 75db or do I set it at 85db thus giving it a 10 db boost manually?

My denon amp looks at the LFE as -10 or 0. So I set the amplified to Zero.

I am thinking that if I set all channels (including the .1) to 75 db the processor will automatically apply a 10db boost gain to the LFE?

Thanks

Stu

When setting up all speakers I set the test tone to 75db (Small) including the normal subwoofer output.

When I put through an LFE test tone do I also set this to 75db or do I set it at 85db thus giving it a 10 db boost manually?

My denon amp looks at the LFE as -10 or 0. So I set the amplified to Zero.

I am thinking that if I set all channels (including the .1) to 75 db the processor will automatically apply a 10db boost gain to the LFE?Yes, the AVR automatically applies the +10dB to the LFE channel in the bass management process. The 0/-10dB LFE gain choice is there because certain old 5.1 movie mixes, and certain silly 5.1 music mixes, did not properly calibrate their subs in the recording studio, so overdrove the LFE channel in the content. This lets you temporarily fix that without going to your main calibrations page. Leave it at 0 as you have done unless called for by whumping bass.:eek:

As for calibrating the sub to 75 dB, it's a kind of cr*p shoot since the sub is playing a vastly different frequency range than the cal noise in the other channels. All sorts of things can conspire to make the readings inaccurate--including how the test signal was designed. Best to take some familiar music CDs, play them and adjust the sub cal so the spectrum splices together smoothly.

Once you have dialed in the sub, and sounds right on a variety of programs, then it's a good idea to run the 75dB test signals and see where the sub level sits wrt to the mains. Might be rather close, actually--but if not, no matter. Write down the level for future ref. If someone messes with the sub gain later, or you change or add another sub, you can dial it back in from the test tones very easily.

Here's the fundamental question I have. When I set up my first "home theater" (for lack of a better word) in 2000, I set up the whole audio system (Yamaha HTR-5150 AVR; Phase Technology center, bookshelf, and rear speakers) when it was just Dolby Digital 5.1 and DTS. So the AVR did all the heavy lifting in terms of decoding the DVDs played on a Pioneer player. It did great. Now I'm replacing the DVD player with a Panasonic BD80 player and the 53" Hitachi Ultravision rear projection TV with a Samsung PN50B650.

Here's the difference, as far as I see: the BD player will now handle the decoding of the audio tracks, be they DD 5.1 or DTS from by DVDs, or the new HD audio tracks from BDs. The only other difference is that the monitor is superior and can display high definition.

The audio components are all the same. NO CHANGE. So why would I touch my receiver settings at all? It is perfect now, and presumably will continue to be perfect playing DVDs via a coaxial digital cable. Any calibration for BDs would be in the BD player itself, right? If I mess with the AVR, I mess it up for DVD play, right?

Here's the fundamental question I have. When I set up my first "home theater" (for lack of a better word) in 2000, I set up the whole audio system (Yamaha HTR-5150 AVR; Phase Technology center, bookshelf, and rear speakers) when it was just Dolby Digital 5.1 and DTS. So the AVR did all the heavy lifting in terms of decoding the DVDs played on a Pioneer player. It did great. Now I'm replacing the DVD player with a Panasonic BD80 player and the 53" Hitachi Ultravision rear projection TV with a Samsung PN50B650.

Here's the difference, as far as I see: the BD player will now handle the decoding of the audio tracks, be they DD 5.1 or DTS from by DVDs, or the new HD audio tracks from BDs. The only other difference is that the monitor is superior and can display high definition.

The audio components are all the same. NO CHANGE. So why would I touch my receiver settings at all? It is perfect now, and presumably will continue to be perfect playing DVDs via a coaxial digital cable. Any calibration for BDs would be in the BD player itself, right? If I mess with the AVR, I mess it up for DVD play, right?Not clear what you are asking. You can't get the new HD audio codecs via a digital coax connection. If you want to decode the HD codecs at the BD player you will need to use a multichannel analog connection to your AVR (if it has multichannel analog inputs). And you would need to calibrate the BD's multichanel analog connection. Otherwise, all you are doing is passing legacy audio to the AVR via the digital connection. The AVR is still doing the decoding.

Don't know if this is the right place for this but here goes.

My Aperion S-10 Sub's Amp stopped working, no output whatsoever.

Aperion's new digital amp was explained to me by Aperion that it wouldn't be an exact fit, not all the screws would align correctly.

So I took it to a local audio electronic repair shop.

They said it was the circuit board and I had too much dust inside the amp.

They used an off the shelf circuit board and "fixed it".

The Sub has a-lot less power than it used to have.

Before, the room would rumble at half volume / -10db. You just didn't hear it, you felt it.

Now, at full volume and +2db and all I get is a decent sound from it but no rumble, don't feel it at all.

Not even close what it used to be.

They asked me to bring it back.

A friend of mine said that more than likely they will never get that sub back to original output because of the circuit board they used but inform them to play with the "gain" to see if it helps.

Any thoughts?

Thank you

Here's the fundamental question I have. When I set up my first "home theater" (for lack of a better word) in 2000, I set up the whole audio system (Yamaha HTR-5150 AVR; Phase Technology center, bookshelf, and rear speakers) when it was just Dolby Digital 5.1 and DTS. So the AVR did all the heavy lifting in terms of decoding the DVDs played on a Pioneer player. It did great. Now I'm replacing the DVD player with a Panasonic BD80 player and the 53" Hitachi Ultravision rear projection TV with a Samsung PN50B650.

Here's the difference, as far as I see: the BD player will now handle the decoding of the audio tracks, be they DD 5.1 or DTS from by DVDs, or the new HD audio tracks from BDs. The only other difference is that the monitor is superior and can display high definition.

The audio components are all the same. NO CHANGE. So why would I touch my receiver settings at all? It is perfect now, and presumably will continue to be perfect playing DVDs via a coaxial digital cable. Any calibration for BDs would be in the BD player itself, right? If I mess with the AVR, I mess it up for DVD play, right?

You should connect the BD80 to the Yamaha via six analog audio cables into what are referred to in your Yamaha's manual as the "External Decoder" jacks. (I downloaded it for reference in answering your question.)

No coax or optical audio connection is really necessary. The BD-80 will decode and output the soundtrack of a standard DVD - or a CD, for that matter - through the multichannel analog output, just as it will do for Blu-ray disks.

Neither coaxial or optical cables are capable of carrying the multichannel signals after being decoded to PCM by the player - that requires the greater carrying capacity of an HDMI cable, which your amp doesn't have a jack for.

Since the HTR-5150 is a 5.1 device, you should set the BD-80 to output in 5.1, not 7.1. This is set in the BD-80 under the "TV/Device Connection" setup menu (see the bottom of page 34 of the BD-80 manual). Page 37 explains how to set relative channel balance and delay times inside the BD-80.

Get yourself a Radio Shack Sound Pressure Level meter for this - the analog kind, with a moving needle. Use the "B" setting, or whatever setting it has other than the "A" setting (the "A" setting listens only to 1kHz, and ignores the subwoofer completely).

One thing you should try is using the HTR-5150's output level controls (discussed on page 40 of the HTR-5150 manual) - which do affect the External Decoder input - instead of the BD-80's level controls (discussed on page 37 of the BD-80 manual). You could have the BD-80 play its test tones and pan them around the room - while you tweak the level controls on the amplifier, with an eye on your sound pressure level meter.

I'm recommending this because - if the BD-80 behaves like my BD-50 - it's a pain in the ass to actually make the adjustments in the BD-80, since nothing you do on that screen takes effect immediately. You have to quit and save out of that screen and then dive back into it to evaluate the effect of the changes you made on the previous trip to that screen - you can't work truly interactively.

You'll still have to adjust delay times in the BD-80, since the HTR-5150 doesn't offer that kind of control for the External Decoder input. That's not so bad, however, since delay times are not something that you can adjust by ear (or meter) in any case. (That's a "measure the distance and use the table in the manual to set the delay" step.)

Subwoofer volume adjustment will be difficult in any case, since the BD-80, like prior models, doesn't play a test tone for the subwoofer. That'll have to be adjusted to taste using a movie disk - since neither of the leading calibration disks, the AVIA DVD and the Digital Video Essentials Blu-ray disk, provide a discrete subwoofer signal. Instead, both of those disks simply mix low bass into the main signals and rely on bass management to redirect it to the subwoofer - and I'm not sure that the BD-80 even does bass management!

You should connect the BD80 to the Yamaha via six analog audio cables into what are referred to in your Yamaha's manual as the "External Decoder" jacks. (I downloaded it for reference in answering your question.)

No coax or optical audio connection is really necessary. The BD-80 will decode and output the soundtrack of a standard DVD - or a CD, for that matter - through the multichannel analog output, just as it will do for Blu-ray disks.

Neither coaxial or optical cables are capable of carrying the multichannel signals after being decoded to PCM by the player - that requires the greater carrying capacity of an HDMI cable, which your amp doesn't have a jack for.

Since the HTR-5150 is a 5.1 device, you should set the BD-80 to output in 5.1, not 7.1. This is set in the BD-80 under the "TV/Device Connection" setup menu (see the bottom of page 34 of the BD-80 manual). Page 37 explains how to set relative channel balance and delay times inside the BD-80.

Get yourself a Radio Shack Sound Pressure Level meter for this - the analog kind, with a moving needle. Use the "B" setting, or whatever setting it has other than the "A" setting (the "A" setting listens only to 1kHz, and ignores the subwoofer completely).

One thing you should try is using the HTR-5150's output level controls (discussed on page 40 of the HTR-5150 manual) - which do affect the External Decoder input - instead of the BD-80's level controls (discussed on page 37 of the BD-80 manual). You could have the BD-80 play its test tones and pan them around the room - while you tweak the level controls on the amplifier, with an eye on your sound pressure level meter.

I'm recommending this because - if the BD-80 behaves like my BD-50 - it's a pain in the ass to actually make the adjustments in the BD-80, since nothing you do on that screen takes effect immediately. You have to quit and save out of that screen and then dive back into it to evaluate the effect of the changes you made on the previous trip to that screen - you can't work truly interactively.

You'll still have to adjust delay times in the BD-80, since the HTR-5150 doesn't offer that kind of control for the External Decoder input. That's not so bad, however, since delay times are not something that you can adjust by ear (or meter) in any case. (That's a "measure the distance and use the table in the manual to set the delay" step.)

Subwoofer volume adjustment will be difficult in any case, since the BD-80, like prior models, doesn't play a test tone for the subwoofer. That'll have to be adjusted to taste using a movie disk - since neither of the leading calibration disks, the AVIA DVD and the Digital Video Essentials Blu-ray disk, provide a discrete subwoofer signal. Instead, both of those disks simply mix low bass into the main signals and rely on bass management to redirect it to the subwoofer - and I'm not sure that the BD-80 even does bass management!

wow! thanks for all the help! I'm hoping you have the 5150, or you are really going over and above the normal forum. It will be a few months before I do this, but I'll let you know how it goes. Thanks again!:D

dmoney24, you have a couple of choices here. You can use a digital connection for DVDs and CDs as you do now and limit the BD80 analog output to lossless playback. Or, you can let the BD80 handle everything.

Using digital for DVDs and CDs is often preferable because AVRs tend to have better processing tools (bass management and EQ), better DACs, and they allow you to apply DSPs such as ProLogic II to stereo sources. But, depending on your room and equipment, those factors may not matter and letting the BD80 do all of the decoding and processing may be just fine.

Analog requires a separate calibration from your current AVR digital setup. You should definitely get an SPL meter if you don't have one. And a calibration disc such as Avia is also essential if you want to do this right. Avia has the SW tones that are lacking on the player.

The biggest problem with analog is the sub. LFE is designed to play 10dB louder than the other channels. But, it is recorded at the same level in order to prevent clipping during transmission. That means it arrives at the AVR 10dB low and must be boosted in the receiver or at the sub itself. If you set any speakers to small in the player, the sub channel (LFE + redirected bass) is dropped by another 5dB and must be boosted 15dB. This is no problem if your receiver has a setting to boost the sub output for the external inputs. My Denon AVR offers that feature in 5dB increments. The analog subwoofer boost on my Denon does not affect digital sources. That's important because adding 10-15dB to the digital sub output would mean way too much bass on DVDs decoded by the receiver. (With digital, the software applies the needed boost on its own. So, you don't want to boost the sub output a second time on top of the one that's already been done.) I don't know about your AVR, but from what I have seen, many Yamaha receivers don't have a setting to boost the analog sub output. Many also lack the capability to adjust levels for analog independent of digital. If your Yamaha has those limitations, it is not possible to calibrate the right bass levels for both analog and digital. You are left needing to adjust the gain on the sub itself on the fly when switching between sources. In that case, it would likely be best to abandon digital and let the BD80 decode and process everything.

wow! thanks for all the help! I'm hoping you have the 5150, or you are really going over and above the normal forum. It will be a few months before I do this, but I'll let you know how it goes. Thanks again!:D

I actually have a different Yamaha AVR, the RX-V457, which is why I downloaded the manual for the HTR-5150: the RX-V457 - also a 5.1 pre-HDMI receiver - is unusual in that it extends the effect of its channel gain and distance settings to the multichannel analog input, and plays test tones for all speakers, including the subwoofer, as part of that process. The only thing that it does not extend to the multichannel input is the digital signal distortion used to simulate different soundfields - which no one should use anyway!

If I didn't have the RX-V457, I would have assumed that - like most AVRs - not even the channel gain controls affected the multichannel input (called the External Decoder input on your 5150). Yamaha appears to "get it" more than most about the needs of the multichannel input user.

By the way, Hi BIslander! We've corresponded here at AVS before. If there's a discrete LFE tone on the AVIA disk, I'd love to know where to find it - my impression is that, like DVE, it uses bass management for that tone, putting it into one of the other channels instead.

Since the DVE disk is so much better for video calibration, my copy of AVIA has been on the shelf for quite a while.

Because neither setup disk provides a discrete subwoofer tone, I made the discovery that I could use the RX-V457's test tones to calibrate the system. Poor me! I only have to use one set of controls to set up all inputs.

Yeah, that does mean that my sub is set too hot on cable tv and SACDs, but that's a sacrifice I'll willingly make. I like low bass. Many years ago - before anyone had subwoofers - I modified my old Dynaco amplifier to boost the bottom octave - over time, that contributed to shredding the foam surround on my 1970 original Advent speakers, but that eventually happens to them anyway, which is why there's a cottage industry rebuilding them with new foam.

If there's a discrete LFE tone on the AVIA disk, I'd love to know where to find it - my impression is that, like DVE, it uses bass management for that tone, putting it into one of the other channels instead. I don't know how the disc is encoded. My system is bass managed, as is the OP's (presumably, given the speakers he is using). Under those circumstances, Avia produces perfectly fine tones for calibrating a sub.

I don't know how the disc is encoded. My system is bass managed, as is the OP's (presumably, given the speakers he is using). Under those circumstances, Avia produces perfectly fine tones for calibrating a sub.

My system is not bass-managed, since all five main channels have full-range speakers. In that situation, AVIA produces silence through the subwoofer, as does DVE. Why neither of them includes a real .1 signal is a mystery to me.

I suspect they didn't want to try to explain how to set the sub's level relative to the other speakers in all the possible configurations of systems: (1) decoding by the amp or the disk player, which impacts whether or not the sub signal needs a manual 10db boost by the user and (2) with and without bass management, since bass management would make the boost - if needed - 15db instead of 10db.

This might also be why the player doesn't supply a subwoofer tone - maybe no one in the whole industry wants to take on the task of explaining this issue! I guess I should count my blessings that my oddball Yamaha RX-V457 does supply a subwoofer tone and makes its level and distance adjustment controls work on all inputs, including the analog multichannel input.

dmoney24, you have a couple of choices here. You can use a digital connection for DVDs and CDs as you do now and limit the BD80 analog output to lossless playback. Or, you can let the BD80 handle everything.

Using digital for DVDs and CDs is often preferable because AVRs tend to have better processing tools (bass management and EQ), better DACs, and they allow you to apply DSPs such as ProLogic II to stereo sources. But, depending on your room and equipment, those factors may not matter and letting the BD80 do all of the decoding and processing may be just fine.

Analog requires a separate calibration from your current AVR digital setup. You should definitely get an SPL meter if you don't have one. And a calibration disc such as Avia is also essential if you want to do this right. Avia has the SW tones that are lacking on the player.

The biggest problem with analog is the sub. LFE is designed to play 10dB louder than the other channels. But, it is recorded at the same level in order to prevent clipping during transmission. That means it arrives at the AVR 10dB low and must be boosted in the receiver or at the sub itself. If you set any speakers to small in the player, the sub channel (LFE + redirected bass) is dropped by another 5dB and must be boosted 15dB. This is no problem if your receiver has a setting to boost the sub output for the external inputs. My Denon AVR offers that feature in 5dB increments. The analog subwoofer boost on my Denon does not affect digital sources. That's important because adding 10-15dB to the digital sub output would mean way too much bass on DVDs decoded by the receiver. (With digital, the software applies the needed boost on its own. So, you don't want to boost the sub output a second time on top of the one that's already been done.) I don't know about your AVR, but from what I have seen, many Yamaha receivers don't have a setting to boost the analog sub output. Many also lack the capability to adjust levels for analog independent of digital. If your Yamaha has those limitations, it is not possible to calibrate the right bass levels for both analog and digital. You are left needing to adjust the gain on the sub itself on the fly when switching between sources. In that case, it would likely be best to abandon digital and let the BD80 decode and process everything.

will using this make much difference? any objections? It would be nice to put the center and two bookshelves in here.