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[mattkhan]

I have a 5.1 setup using acourate to deal with everything involved in bd (audio) playback except for the +10 LFE boost. My ears tell me something is not right with the SW (seems rather thin) but I can't see what I have done wrong. Therefore I thought I'd offer up my config for review, any & all comments much appreciated.

The basic setup is everything (room correction, delays, bass management) done in convolution except for the +10dB to the LFE channel which is done via jriver room correction.

The end result is great (main channel levels are matched nicely & sound v good) except LFE seems weak, there just isn't the impact I'd expect and hence this makes me think I have something wrong on the bass management side of things. This is not a missing +10dB for LFE problem, there is definitely +10 there.

jriver is configured according to the attached pics; output format is 5.1 with no mixing and everything resampled to 96kHz, room correction does nothing except add 10dB to the LFE & then convolution does the rest. The convolver cfg file is listed below, comments added to explain what each path is intended to do.

To make it clear what the WAVs are, a bit of background on (my use of) acourate ....  the acourate UI is designed for stereo use but can be used for multichannell quite easily. This means I have 4 sets of filters; one each for L-R, L-C, L-SL and L-SR. I am running a 2 way (5.1) setup so each pair of channels produces 2 2 channels WAVs containing the correction filter. By default these are named as follows

Cor1S96.wav
Cor2S96.wav

so each wav has 2 channels (L & R for example) where Cor1 is the low pass to the sub and Cor2 is high pass to the main channel. I then copy them all into a single dir and append the channel names so I know what is what, i.e. I end up with;

Code: [Select]

Cor1S96_LR.wav
Cor1S96_LC.wav
Cor1S96_LSR.wav
Cor1S96_LSL.wav
Cor2S96_LR.wav
Cor2S96_LC.wav
Cor2S96_LSR.wav
Cor2S96_LSL.wav

I run a 120Hz XO so I just reuse 1 of the Cor1 filters for the LFE channel.

Finally the cfg file

Code: [Select]

96000 6 6 0
0 0 0 0 0 0
0 0 0 0 0 0
# low pass L to SW
C:\Users\Matt\Documents\Acourate\Filters\Cor1S96_LR.wav
0
0.0
3.0
# low pass R to SW
C:\Users\Matt\Documents\Acourate\Filters\Cor1S96_LR.wav
1
1.0
3.0
# low pass C to SW
C:\Users\Matt\Documents\Acourate\Filters\Cor1S96_LC.wav
1
2.0
3.0
# LFE
C:\Users\Matt\Documents\Acourate\Filters\Cor1S96_LSR.wav
1
3.0
3.0
# low pass SL to SW
C:\Users\Matt\Documents\Acourate\Filters\Cor1S96_LSL.wav
1
4.0
3.0
# low pass SR to SW
C:\Users\Matt\Documents\Acourate\Filters\Cor1S96_LSR.wav
1
5.0
3.0
# high pass L
C:\Users\Matt\Documents\Acourate\Filters\Cor2S96_LR.wav
0
0.0
0.0
# high pass R
C:\Users\Matt\Documents\Acourate\Filters\Cor2S96_LR.wav
1
1.0
1.0
# high pass C
C:\Users\Matt\Documents\Acourate\Filters\Cor2S96_LC.wav
1
2.0
2.0
# high pass SL
C:\Users\Matt\Documents\Acourate\Filters\Cor2S96_LSL.wav
1
4.0
4.0
# high pass SR
C:\Users\Matt\Documents\Acourate\Filters\Cor2S96_LSR.wav
1
5.0
5.0

Thanks
Matt

[mwillems]

The configuration I can see looks good.  It's hard to know without seeing what kind of filtering the convolution filter is doing on the sub, but what you're describing (thin bass despite correct measured volume) is, in my experience, usually a result of poor phase alignment in the crossover region (either due to crossover issues or failing to compensate for the extra delay resulting from the sub's position), and the best way to diagnose that is to run a measurement of the 2-way system (i.e. the sub and main interacting at a listening position).

Three troubleshooting questions:  

1) How are you accounting for the sub's delay based on it's position in the room (time of flight delay)?  It looks like you're reusing a filter for another speaker for the LFE channel, are they the same distance from the listening position?  Or are all of that batch of filter set at the appropriate delay for the sub?
2) What type of crossover slope are you using for the sub and the mains (i.e. is it symmetrical, what orders, linear phase?, what's the delay like, etc.), and
3) How does the system measure?  Since each of your speakers is effectively a 2-way with the sub, you should be able to run (say) a sweep on just the left channel and see how the sub is interacting with the left main. Does everything look good in the sub region through the crossover to the mains when you run a sweep on the left or right speaker?

It's probably some kind of phase alignment issue, an integrated measurement should show you something.

[mattkhan]

The configuration I can see looks good.  It's hard to know without seeing what kind of filtering the convolution filter is doing on the sub, but what you're describing (thin bass despite correct measured volume) is, in my experience, usually a result of poor phase alignment in the crossover region (either due to crossover issues or failing to compensate for the extra delay resulting from the sub's position), and the best way to diagnose that is to run a measurement of the 2-way system (i.e. the sub and main interacting at a listening position).

thanks for looking at it. I'm don't think it's the punchiness I'm missing, music sounds really excellent. It's more a question of a lack of weight at the low end for films.

1) How are you accounting for the sub's delay based on it's position in the room (time of flight delay)?  It looks like you're reusing a filter for another speaker for the LFE channel, are they the same distance from the listening position?  Or are all of that batch of filter set at the appropriate delay for the sub?

the basic approach is as follows

- align the mic precisely between the L & R
- calculate the delay from the SW to the L & R, this shows the SW is about 40 samples further away so I apply the relevant delay to the low pass XO to bring them into line
- run mic alignment for the other channels against L, the other channels are also closer to the listening position by varying amounts so the high pass XO is shifted accordingly to align them with the L & R

i.e. the low pass XO is identical for each channel.

2) What type of crossover slope are you using for the sub and the mains (i.e. is it symmetrical, what orders, linear phase?, what's the delay like, etc.), and

2nd order linear phase Neville-Thiele at 120Hz

3) How does the system measure?  Since each of your speakers is effectively a 2-way with the sub, you should be able to run (say) a sweep on just the left channel and see how the sub is interacting with the left main. Does everything look good in the sub region through the crossover to the mains when you run a sweep on the left or right speaker?

I can't be 100% certain as I haven't worked out a way to get actual timing data yet for individual measurements of each speaker using REW when convolution is involved. I don't think acourate has this feature either as it shifts the peaks to sample = 6000 automatically. I don't think there is an obvious discontinuity though, I've attached an unsmoothed measurement showing 15-300Hz. The dip at 125Hz seems to correspond to a room mode & otherwise it seems ok (given that I have only 1 sub).

[mwillems]

I can't be 100% certain as I haven't worked out a way to get actual timing data yet for individual measurements of each speaker using REW when convolution is involved. I don't think acourate has this feature either as it shifts the peaks to sample = 6000 automatically.

I don't think there is an obvious discontinuity though, I've attached an unsmoothed measurement showing 15-300Hz. The dip at 125Hz seems to correspond to a room mode & otherwise it seems ok (given that I have only 1 sub).

Your methodology seems sound, and the graph generally looks quite good for a 1 sub setup, but that High Q dip at 120/125Hz looks a little suspicious given that's your crossover frequency. One way to confirm is to look at the phase.  

Can you repost that measurement with the "phase" box checked?  A room mode usually looks a little different than a crossover misalignment.  I also have another phase-related hunch as to what could be happening, but it's hard to explain and the phase would confirm or disprove it.

P.S. - Am I reading that right that you have rising frequency response at 15Hz?  Assuming that's not a measurement artifact, that's a pretty nice sub, what kind is it?  

[mattkhan]

Your methodology seems sound, and the graph generally looks quite good for a 1 sub setup, but that High Q dip at 120/125Hz looks a little suspicious given that's your crossover frequency. One way to confirm is to look at the phase.  

Can you repost that measurement with the "phase" box checked?  A room mode usually looks a little different than a crossover misalignment.  I also have another phase-related hunch as to what could be happening, but it's hard to explain and the phase would confirm or disprove it.

I've attached 2 pics, one from REW (via the asio line in method) and one from acourate (showing L & R) using the correction filter directly.

P.S. - Am I reading that right that you have rising frequency response at 15Hz?  Assuming that's not a measurement artifact, that's a pretty nice sub, what kind is it?  

yes that's right though the measurements through REW recently are a bit dodgy below about 30Hz, it really exaggerates the rise at the low end. You can see this by comparing against acourate, it is still rising but only a few dB as opposed to loads! Mind you now I look at it, the shapes of the 2 responses is a bit different anyway. This makes me suspect something is wrong in my jriver setup *or* that asio line in is doing something unexpected. The acourate graph suggests the sub is a few dB down on the mains.

I attached a couple more pics showing the full range measurements, one in REW and one from acourate. The former is 1/6 octave smoothed, the latter is run through acourate's macro 1 (psychoacoustic + frequency dependent windowing).

It's a sealed 65L enclosure with a down firing 15" Fi SP4 driver in it. I need to get a new amp though as my current one can't drive it properly (more money to spend!)

[mwillems]

I've attached 2 pics, one from REW (via the asio line in method) and one from acourate (showing L & R) using the correction filter directly.

Ok your crossover looks fine (from what I can see), you're right, it's probably just a room effect.

yes that's right though the measurements through REW recently are a bit dodgy below about 30Hz, it really exaggerates the rise at the low end. You can see this by comparing against acourate, it is still rising but only a few dB as opposed to loads! Mind you now I look at it, the shapes of the 2 responses is a bit different anyway. This makes me suspect something is wrong in my jriver setup *or* that asio line in is doing something unexpected. The acourate graph suggests the sub is a few dB down on the mains.

It may also be that the two suites are doing different filtering.  My experience is that REW's "unfiltered" response doesn't really look like the raw responses I get in any other measurement suite, which suggests to me that some kind of unusual filtering/smoothing is going on in REW (especially on the low end).  I haven't played around with REW's gating settings, but that may be part of the story.

It's one of the reasons I stopped using it for measurement, I couldn't ever seem to get results that squared with other measurements I took.  For example, raw measurements I take from a given position with the free measurement tool from Audiolense, look almost identical to the measurements I've taken in Holm, etc.  

If the Acourate measurement is correct, the entire sub range looks about 3 or 4 dB down from the mains (more in some places), which would be very noticeable and could well explain your problem. You mentioned in another thread that you might try Holm, I'd be curious if it shows the same thing as Acourate.  If so, then your issue is probably just levels.

I attached a couple more pics showing the full range measurements, one in REW and one from acourate. The former is 1/6 octave smoothed, the latter is run through acourate's macro 1 (psychoacoustic + frequency dependent windowing).

As noted, those Acourate measurements suggest that your levels are off, which (given that your phase looks good through the crossover) could well be the issue.

It's a sealed 65L enclosure with a down firing 15" Fi SP4 driver in it. I need to get a new amp though as my current one can't drive it properly (more money to spend!)

Interesting; I'm surprised that a sealed cabinet with more or less level FR has 360 degrees of phase wrap, but I haven't modeled your driver and enclosure so maybe that's expected in your setup?  It might be worth trying to straighten out that phase wrap a little bit if it turns out it's not a levels issue.  

That FR is darned impressive though;  Do you like the sound of the driver?  I'm potentially in the market for a new 15" for my frankensub.

[mattkhan]

Thanks for going over my results. I have downloaded holm so will give that a whirl when I next get time to measure. In the meantime I might just run the sub a bit hotter & see if it is just that.

Interesting; I'm surprised that a sealed cabinet with more or less level FR has 360 degrees of phase wrap, but I haven't modeled your driver and enclosure so maybe that's expected in your setup?  It might be worth trying to straighten out that phase wrap a little bit if it turns out it's not a levels issue. 

I've been experimenting with LTs recently & with correcting the near field response of the sub. For speed of experimentation I've been doing this via a number of IIR filters as it means I can do it all offline, if I embedded these into the acourate XO then it's quite slow going with all the measurements you need to take. The measurement I attached is one where I have a few filters in place to flatten the (presumably inductance based) hump at ~55Hz. I then add an LT onto that to reduce Q from ~0.711 to 0.500. I suppose the phase wrap comes from those filters? I have to say it's not something I've looked at given the basic problem I have with levels

That FR is darned impressive though;  Do you like the sound of the driver?  I'm potentially in the market for a new 15" for my frankensub.

I really like it. I've listened to a lot of music since installing it and it has lovely control which yields a really nice texture to the music. It drinks power for AV use though. I'm considering importing a speakerpower amp so I can really get it going as decent power amps in the UK are rather expensive unfortunately.

[mwillems]

Thanks for going over my results. I have downloaded holm so will give that a whirl when I next get time to measure. In the meantime I might just run the sub a bit hotter & see if it is just that.

If that partially resolves it, you've probably found your culprit, which is good news.

I've been experimenting with LTs recently & with correcting the near field response of the sub. For speed of experimentation I've been doing this via a number of IIR filters as it means I can do it all offline, if I embedded these into the acourate XO then it's quite slow going with all the measurements you need to take. The measurement I attached is one where I have a few filters in place to flatten the (presumably inductance based) hump at ~55Hz. I then add an LT onto that to reduce Q from ~0.711 to 0.500. I suppose the phase wrap comes from those filters? I have to say it's not something I've looked at given the basic problem I have with levels

A linkwitz transform could definitely introduce that kind of phase shift so that may be it.  Just FYI my experience has been that phase wrap at low frequencies that doesn't  correspond to changes in frequency response can sound mighty weird, and lead to some undesirable bass consequences (ringing, muddling, loss of bass "authority", etc.) as the sound at 30Hz is nearly a full cycle behind the sound at 70Hz. 

Obviously ported subs typically have at least that much wrap, but that's part of why sealed subs often have a better reputation for clean bass.  Just something to think about if the levels turn out to not be the issue (or don't completely resolve the issue).

Phase distortion is not necessarily super audible, but studies (as well as my own anecdotal experience) suggests that it's most audible at low frequencies.

I really like it. I've listened to a lot of music since installing it and it has lovely control which yields a really nice texture to the music. It drinks power for AV use though. I'm considering importing a speakerpower amp so I can really get it going as decent power amps in the UK are rather expensive unfortunately.

Sweet, I'll see if I can arrange to hear one sometime.  Thanks for the info.

[mattkhan]

If that partially resolves it, you've probably found your culprit, which is good news.

I think this was it. I measured in HolmImpulse and see a different low end to that in REW. I therefore upped the gain and all is well in the world again. I verified using REW RTA and the JRiver calibration tones and all looks good. Playback of some more intensive scenes verifies this. I'm glad it was a simple solution for once 

[mwillems]

I think this was it. I measured in HolmImpulse and see a different low end to that in REW. I therefore upped the gain and all is well in the world again. I verified using REW RTA and the JRiver calibration tones and all looks good. Playback of some more intensive scenes verifies this. I'm glad it was a simple solution for once 

Great news!   I'm always thankful when, after spending a lot of time on a tough problem, the solution only takes a few minutes and costs nothing 

I once spent a day and a half trying to figure out why my stereo imaging was poor in an early version of my bi-amp setup.  The two speakers were identical and similarly placed, and had (a week before) measured almost identically, so I kept measuring the left speaker to check the crossover, check the delay, check everything.  I couldn't figure it out.  Until I finally re-measured the right speaker and realized that when I had taken everything apart two days prior I had wired the midbass module backwards so it was inverted with respect to the rest of the system 

But the good news was there wasn't a fatal flaw in my design, and it was a free fix that took 30 seconds.

[BradC]

Hi

Does the Holm measurement look the same as the one in acourate?
I also use acourate, and have finally worked (with a lot of help from Uli) out a procedure for integrating the subs

Basically you need to work out the delay, polarity and gain parameters of the sub that gives the flattest response. I used my sound card set to loopback and plogue bidule with vst plugins to adjust those parameters. I then used the RTA in REW to adjust them in realtime to get the best response. I think you could also use acourate-convolver for realtime parameter adjustment, or maybe the loopback driver in jriver

The next step is to generate the xover in acourate. Then apply the delay, gain and polarity settings to your xover filters. Then make a multiway wav file.

Apply this multiway wav file to the acourate recorder, to basically perform your sub integration measurement

Apply the room macros to your measured pulses, create filters and a convolver cfg file for jriver.

Now your sub will be integrated with your mains.
You can extend this procedure to multiple subs and active speakers

To generate the LFE channel, simply apply 10dB LESS attenuation compared to the filters above. If this is not possible, turn up the amp gain in the step above. Generating the LFE filter this way, instead of adding gain in jriver, will give more headroom in jriver

[mattkhan]

Does the Holm measurement look the same as the one in acourate?

Holm and Acourate seem quite similar except the acourate measurement rolls off sharply at ~15Hz, I've asked Uli if there is a reason to expect this.

I also use acourate, and have finally worked (with a lot of help from Uli) out a procedure for integrating the subs

Basically you need to work out the delay, polarity and gain parameters of the sub that gives the flattest response. I used my sound card set to loopback and plogue bidule with vst plugins to adjust those parameters. I then used the RTA in REW to adjust them in realtime to get the best response. I think you could also use acourate-convolver for realtime parameter adjustment, or maybe the loopback driver in jriver

The next step is to generate the xover in acourate. Then apply the delay, gain and polarity settings to your xover filters. Then make a multiway wav file.

Apply this multiway wav file to the acourate recorder, to basically perform your sub integration measurement

Apply the room macros to your measured pulses, create filters and a convolver cfg file for jriver.

Now your sub will be integrated with your mains.
You can extend this procedure to multiple subs and active speakers

To generate the LFE channel, simply apply 10dB LESS attenuation compared to the filters above. If this is not possible, turn up the amp gain in the step above. Generating the LFE filter this way, instead of adding gain in jriver, will give more headroom in jriver

Are you running multiple subs? ISTR a mail from you on the mailing list talking about correcting multiple subs. I think I have the sub aligned to each main channel correctly but the per channel delays are not working. I have dropped Uli a mail on this so am waiting to hear back from him. How are you handling main channel delays? via rotating the XO or some other way?

Thanks for the tip on the LFE channel.

[BradC]

Yep I use the rotation function to add delays to the main channel xover filters. I am using dual subs, but one at the front and one at the rear.

6dB down at 20kHz compared to  1kHz is pretty standard. Is your rolloff steeper than this?

I also found that pointing the mic at the front of the room (ie horizontal), produced less roff off than a vertical mic

[mattkhan]

Yep I use the rotation function to add delays to the main channel xover filters. I am using dual subs, but one at the front and one at the rear.

6dB down at 20kHz compared to  1kHz is pretty standard. Is your rolloff steeper than this?

I also found that pointing the mic at the front of the room (ie horizontal), produced less roff off than a vertical mic

I have the same slope hinged at 1kHz reducing 6dB by 20kHz.

ok so you rotate the XO2 for the C/SL/SR etc to bring them into alignment with the L and R? do you see that rotation in the peak of the resulting Cor2? not 100% sure if you're meant to see it there as a shift in the peak of the correction filter but, if not there, I don't know where a delay would be visible.

Do you have your approach written up btw? it would make interesting reading if you do.

[mattkhan]

just to add to this, I have verified experimentally that the delay applied to the XO2 is not appearing in the corresponding Cor2. Manually rotating this after the fact & using this in the convolution filter produces expected results when verifying delays in holmimpulse. Curiously it's interesting that the delays are being applied to the low pass. I will report back to Uli to see if this is a bug or some user error on my part.

It's interesting that I had noted the tonal shift (a certain boxiness to the sound) on the centre channel ever since I set this up.

[mattkhan]

To generate the LFE channel, simply apply 10dB LESS attenuation compared to the filters above. If this is not possible, turn up the amp gain in the step above. Generating the LFE filter this way, instead of adding gain in jriver, will give more headroom in jriver

I don't understand how this differs to adding 10dB to the LFE via jriver & appears to involve more work. If I add 10dB to the sub in room correction then, as I understand it, all it does is attenuate all the other channels by 10dB. If I go in and attenuate all the other low passes in acourate then I'm just doing the same thing but n times instead of once aren't I?

[mwillems]

I don't understand how this differs to adding 10dB to the LFE via jriver & appears to involve more work. If I add 10dB to the sub in room correction then, as I understand it, all it does is attenuate all the other channels by 10dB. If I go in and attenuate all the other low passes in acourate then I'm just doing the same thing but n times instead of once aren't I?

Mattkhan, you're correct; adding 10dB to LFE in room correction just attenuates the other channels by 10dB.  Adding 10dB in PEQ (instead of room correction) would not automatically self-offset in the same way, which may be part of the confusion.  The only way there's an advantage to integrating the offset into your convolution filters is if your main channel filters already involve very substantial attenuation; in that case, you might be better off integrating the attenuation into the convolution filters (i.e. you might have less total attenuation that way).  With an ordinary filter, you lose nothing by using room correction instead of offsetting the volume in convolution, and gain convenience.

Turning up the sub amp is still the preferable offset method (when possible) because that prevents the loss of 10dB worth of digital dynamic range in the other channels, but that's a separate issue.

[BradC]

just to add to this, I have verified experimentally that the delay applied to the XO2 is not appearing in the corresponding Cor2. Manually rotating this after the fact & using this in the convolution filter produces expected results when verifying delays in holmimpulse. Curiously it's interesting that the delays are being applied to the low pass. I will report back to Uli to see if this is a bug or some user error on my part.

It's interesting that I had noted the tonal shift (a certain boxiness to the sound) on the centre channel ever since I set this up.

Yes the issue is to avoid jriver attenuating all the other channels by 10dB.

As mwillems also noted, the preferred option if to turn up the gain on your sub amp.

Then you simply make the gain adjustment in your filters 10dB less for the LFE channel, compared to all the other low pass  channels. This only involves generation of 2 different xover filters.

[BradC]

just to add to this, I have verified experimentally that the delay applied to the XO2 is not appearing in the corresponding Cor2. Manually rotating this after the fact & using this in the convolution filter produces expected results when verifying delays in holmimpulse. Curiously it's interesting that the delays are being applied to the low pass. I will report back to Uli to see if this is a bug or some user error on my part.

It's interesting that I had noted the tonal shift (a certain boxiness to the sound) on the centre channel ever since I set this up.

Are you using workspaces in acourate correctly?
You need to set the workspace in acourate to a given directory eg 'front channels'. You then generate the xover and multiwav file. Record the pulses, place the PulseL48 etc in the same directory.
Run all the room macros, and in macro 4 acourate will convolve the CorS48.dbl filters with the xover filters.

If you do not use the workspace in this manner, the convolution in the last step won't happen. You must not rename the xover files either
To me this sounds like your problem.
I can confirm that with this method I do get the delays applied to the Cor2 Cor3 etc. I have recently triamped and time aligned my front speakers. The difference was significant.

I have experienced some speakers sounding 'boxy'. The problem is either a poor measurement or the target curve. Repeating the measurement has often fixed this problem for me.
If that's not your problem, you can try moving the knee in the target curve higher in freq than 1kHz, and/or raising the -6dB @ 20kHz slightly.

[BradC]

Further to the boxiness:

The last time I had really poor sounding speakers after correction, the solution was to record the pulses at 48khz instead of 96kHz.

Uli also demonstrated how to avoid high freq issues by flattening the gain of the inverse file after 20kHz.
After macro3, if the inverse curve is rising after 20kHz, it may lead to problems.
The solution is to use the phase (?) function (can't remember exact name, but press F2 key)
Set the gain parameter to 0dB from the freq where it rises (21kHz in my case). Then select minimum phase

Doing this may also help with the boxiness, as if the inverse curve has too much gain in this high freq region, it may cause attenuation in other lower freq regions.
Note that recording at 48khz, minimises the high freq measurement

[mattkhan]

Thanks for the suggestions. I think I am doing all of those things correctly though.

I use a separate workspace for each pair of channels (named LC, LR, LSL, LSR)
I generate the XO in each workspace & don't rename them or otherwise mess with them
I record at 48kHz
I use phase extract to avoid boosting at the extremities of the frequency range (<14Hz and >20700Hz for me)

The problem is repeatable in my case. I can

create a new workspace
generate an XO
rotate of each of them by some different amount
run through the macros
compare the Cor with the XO and see that Cor1 is shifted but Cor2 is not

[BradC]

Hi

It seems you have something there.
I just had a look at my 5-way setup (2 subs and triamped mains)

There is a timing difference between the correction filter and the XO filter, but it only matters if it changes for each driver.

Here is the difference (in samples) for each driver (between XO and Cor):
sub1: 71
sub2: 71
Mains1: 459
Mains2: 457
Mains33: 457

So it is not maintaining the time delay I embedded for the subs.

My guess is that acourate thinks that it's getting a better correction by adjusting the timing.
But it's worth checking with Uli

[mattkhan]

thanks for checking your setup, at least now Uli has 2 clear bug reports to go on 

It would be good if Uli added the ability to load a correction filter into the mic alignment tool just as you can in the log sweep recorder. It would make double checking mains alignment post filter a trivial task.

[mojave]

yes that's right though the measurements through REW recently are a bit dodgy below about 30Hz, it really exaggerates the rise at the low end. You can see this by comparing against acourate, it is still rising but only a few dB as opposed to loads!

Are you using same mic and soundcard calibration in REW as in Acourate? Some add a soundcard calibration in REW which changes the low frequency response.

I've been experimenting with LTs recently & with correcting the near field response of the sub. For speed of experimentation I've been doing this via a number of IIR filters as it means I can do it all offline, if I embedded these into the acourate XO then it's quite slow going with all the measurements you need to take.

I recently build four sealed subs that needed an LT. You should just be able to do it with your target. That is what I did in Audiolense. Convolution can make a perfect LT because it knows the exact amount of room gain you have to go along with the subwoofer's rolloff.

I'm considering importing a speakerpower amp so I can really get it going as decent power amps in the UK are rather expensive unfortunately.

You can get one from Mark Seaton at Seaton Sound cheaper than buying directly from SpeakerPower. He is a dealer and also uses SpeakerPower amps in his subs and active speakers.

I also found that pointing the mic at the front of the room (ie horizontal), produced less roff off than a vertical mic

The mic pointed at the speakers is a free field measurement and with it pointed vertical it is a diffuse field (or pressure response) measurement. The calibration for each can be quite different. My iSemcon microphone only varies by a few dB between free vs diffuse. I recently received some info from ACOPacific on their reference microphones and the same mic when pointed vertical is -9 dB at 20kHz and -5.1 dB at 10kHz. If you aren't using the proper calibration you can get very different results when trying to correct the full bandwidth.

I've read that a diffuse field measurement with the mic at about 80 degrees is recommend for room correction since it most represents what we hear. A free field measurement is more used for close mic/anechoic type measurements for designing speakers. However, many take room measurements with the mic pointed straight ahead.

Wow, the Acourate measurement and filter technique sounds rather laborious. This weekend I am helping setup a theater with 15 channels including 6 subwoofers, 3 mains, and 6 surrounds. I will be using Audiolense to take measurements at all 7 seating locations of all 15 channels and creating a filter. I only have to take 7 measurements with Audiolense vs 105 with Acourate (I think). Should be fun!

[BradC]

thanks for checking your setup, at least now Uli has 2 clear bug reports to go on 

It would be good if Uli added the ability to load a correction filter into the mic alignment tool just as you can in the log sweep recorder. It would make double checking mains alignment post filter a trivial task.

As you have seen, Uli doesn't think that this behaviour is a bug, but it's the result of optimisation in macro 4.
I will check tonight the delays that a different correction produced, which had the mains' polarity reversed, to see if a different result was produced.

[BradC]

The mic pointed at the speakers is a free field measurement and with it pointed vertical it is a diffuse field (or pressure response) measurement. The calibration for each can be quite different. My iSemcon microphone only varies by a few dB between free vs diffuse. I recently received some info from ACOPacific on their reference microphones and the same mic when pointed vertical is -9 dB at 20kHz and -5.1 dB at 10kHz. If you aren't using the proper calibration you can get very different results when trying to correct the full bandwidth.

I've read that a diffuse field measurement with the mic at about 80 degrees is recommend for room correction since it most represents what we hear. A free field measurement is more used for close mic/anechoic type measurements for designing speakers. However, many take room measurements with the mic pointed straight ahead.

I have an earthworks M30, which only has calibration for free field, and also a calibrated ECM8000, which has calibrations for 3 different angles.
I actually think that the free field measurement is best for the stereo mains, and the diffuse field is better for surround channels.

Acourate is targeted more towards a stereo setup and making active speakers. It can do surround channels too, but the workflow for many channels takes a while. It also doesn't measure multiple locations, which Uli believes gives a more compromised result for stereo.
Audiolense is better suited to surround setups, but several users have compared the two, and audiolense doesn't give quite as good a result for a stereo setup.
So it really is horses for courses.

[mattkhan]

Are you using same mic and soundcard calibration in REW as in Acourate? Some add a soundcard calibration in REW which changes the low frequency response.

yes, it is exactly the same setup.

I recently build four sealed subs that needed an LT. You should just be able to do it with your target. That is what I did in Audiolense. Convolution can make a perfect LT because it knows the exact amount of room gain you have to go along with the subwoofer's rolloff.

I don't follow what you mean exactly, can you expand on that? My understanding is that the LT is applied to give a specific anechoic response which dictates the response characteristics of the sub itself & that impacts the step response (and hence the perception of a tight sub). You clearly need to set that to "agree" with your room but the room correction step has to be distinct to the LT step otherwise you're just correcting the room not the sub.

You can get one from Mark Seaton at Seaton Sound cheaper than buying directly from SpeakerPower. He is a dealer and also uses SpeakerPower amps in his subs and active speakers.

thanks, that's good to know.

I've read that a diffuse field measurement with the mic at about 80 degrees is recommend for room correction since it most represents what we hear. A free field measurement is more used for close mic/anechoic type measurements for designing speakers. However, many take room measurements with the mic pointed straight ahead.

the description I read was by Herb from CSL on HTS here, I think the only correction system I'm aware of that uses that "multiple measurements at random incidence" approach is RoomPerfect. I use the mic at 80 degrees ish approach myself.

Wow, the Acourate measurement and filter technique sounds rather laborious. This weekend I am helping setup a theater with 15 channels including 6 subwoofers, 3 mains, and 6 surrounds. I will be using Audiolense to take measurements at all 7 seating locations of all 15 channels and creating a filter. I only have to take 7 measurements with Audiolense vs 105 with Acourate (I think). Should be fun!

yes it's not the most user friendly product for multichannel use but I don't think it's quite that bad an n.1 setup (where 1 refers to a mono sub not the no of subs) needs n-1 measurements for correction alone & then there are separate measurements for each n way speakers you are sorting out. If you need multi seat correction for a large theatre then I don't think it's not the right tool.

[mattkhan]

As you have seen, Uli doesn't think that this behaviour is a bug, but it's the result of optimisation in macro 4.
I will check tonight the delays that a different correction produced, which had the mains' polarity reversed, to see if a different result was produced.

I will be surprised if this covers my case as each channel (C, SL, SR) is reverted to 0 offset from the L XO. It's not impossible of course but it seems rather improbable.

[BradC]

It does seem like there is something else going on with your filters. Do you have ver 1.8.9 of acourate?

My system is maintaining the time alignment I program into the mains, it's just moving the sub-main time delay, presumably to get a smoother response. I need to measure to confirm this.

Could you post or email some screen grabs of your XO1, Cor1 and XO2, Cor2 impulses on the same plot? Just to confirm we are definitely on the same page

[mattkhan]

It does seem like there is something else going on with your filters. Do you have ver 1.8.9 of acourate?

I am using 1.8.9

Could you post or email some screen grabs of your XO1, Cor1 and XO2, Cor2 impulses on the same plot? Just to confirm we are definitely on the same page

sure, in this case acourate L is really my L and acourate R is really my C

XO1L vs Cor1L, peaks are at samples 32718 vs 32733 (seems like the sort of shift you saw?)
XO2L vs Cor2L, peaks are aligned at 32768
XO1R vs Cor2R, 32718 vs 32740
XO2R vs Cor2R, 32790 vs 32768

the last one seems a bit fishy to me

[BradC]

I think what your graphs show is that acourate aligns the tweeter to 32768 no matter what (as Uli said)

The sub/woofer is then aligned in time to give the best response.

So in summary, acourate is behaving as expected and there is no problem with your filters, but the room correction timing needs to be applied using jriver or something. ie if your centre is 1m further away than your front, you still need to apply 3ms delay to the front speaker using jriver.

The time delay built into XO filters can still be used to time align drivers within a single speaker

[mattkhan]

So in summary, acourate is behaving as expected and there is no problem with your filters, but the room correction timing needs to be applied using jriver or something.

The problem is this completely contradicts what he has advised me to do with respect to multichannel delays. This makes me think it is some sort of bug.

[mattkhan]

The problem is this completely contradicts what he has advised me to do with respect to multichannel delays. This makes me think it is some sort of bug.

OK so it's not a bug, it's a feature This issue is a reflection of acourate's stereo focus so, as Brad says, it aligns the tweeters and then shifts the woofers accordingly. This can be seen in my subwoofer correction for that channel which has been rotated to the left. In contrast, for a multichannel setup where I have a single sub, we want the woofer to stay in a fixed position and the tweeter to be rotated to the right. Therefore the fix is to manually rotate all CorR filters by (Cor1L-Cor1R) samples so that the Cor1R is aligned to the Cor1L and the Cor2R is in the right place given the theoretical delay (as captured by the XO) & the adjustments caused by phase correction.

[mojave]

I don't follow what you mean exactly, can you expand on that? My understanding is that the LT is applied to give a specific anechoic response which dictates the response characteristics of the sub itself & that impacts the step response (and hence the perception of a tight sub). You clearly need to set that to "agree" with your room but the room correction step has to be distinct to the LT step otherwise you're just correcting the room not the sub.

The perception of a tight sub is primarily due to frequency response at the listening position. Once your box size and driver are set, it is the frequency response that you listen to that matters. I built some sealed subs and first added an LT and one filter to them in JRiver's PEQ using a close mic measurement. I then added EQ so that they would be smooth at the listening position. This became one zone. Then I started over and just added an LT and EQ'd using measurements from the listening position. I ended up with fewer filters and the same frequency response. This became a second zone. Switching between zones and listening revealed them to be identical.

Adding filters through PEQ or convolution are cumulative and the final result will be the same regardless how you got there. It is all just math. Using JRiver's PEQ, if you add a dip at 60 Hz of -10 dB with a Q of 2 and then add a peak at 60 Hz of 10 dB with a Q of 2 what do you hear? You won't hear the dip or the peak. It is as if nothing happened. Same thing with adding an LT using a close mic and then pulling down room induced peaks with a filter later. It is as if the LT at the peak location never existed. Depending on how the convolution filters are added from Acourate with the LT, you could be losing digital headroom with the double filters.

The LT is always correcting the room and not the sub because the only way you know what LT parameters to use is to measure the room gain profile. The actual Fz and Qz of the sub stay the same regardless of where you measure, but you only know their actual value through a close mic measurement. However, the desired Fp and Qp are completely dependent on the room and how much gain the room adds and at what frequency the gain is starting to be increased.

[mattkhan]

The perception of a tight sub is primarily due to frequency response at the listening position. Once your box size and driver are set, it is the frequency response that you listen to that matters. I built some sealed subs and first added an LT and one filter to them in JRiver's PEQ using a close mic measurement. I then added EQ so that they would be smooth at the listening position. This became one zone. Then I started over and just added an LT and EQ'd using measurements from the listening position. I ended up with fewer filters and the same frequency response. This became a second zone. Switching between zones and listening revealed them to be identical.

Adding filters through PEQ or convolution are cumulative and the final result will be the same regardless how you got there. It is all just math. Using JRiver's PEQ, if you add a dip at 60 Hz of -10 dB with a Q of 2 and then add a peak at 60 Hz of 10 dB with a Q of 2 what do you hear? You won't hear the dip or the peak. It is as if nothing happened. Same thing with adding an LT using a close mic and then pulling down room induced peaks with a filter later. It is as if the LT at the peak location never existed. Depending on how the convolution filters are added from Acourate with the LT, you could be losing digital headroom with the double filters.

The LT is always correcting the room and not the sub because the only way you know what LT parameters to use is to measure the room gain profile. The actual Fz and Qz of the sub stay the same regardless of where you measure, but you only know their actual value through a close mic measurement. However, the desired Fp and Qp are completely dependent on the room and how much gain the room adds and at what frequency the gain is starting to be increased.

OK thanks for expanding on that. My approach has been to choose an LT that augments room gain to the extent I want (roughly) but that also reduces Q down to the value I'm after, I then let the room correction fine tune to the actual target curve.

Your post sounds like you don't believe the Q of the system itself is relevant to that perception of tightness, is that a fair interpretation? For example, this thread shows how the theoretical step response changes with different values for Q.

[mojave]

A subwoofer's step response can be derived from its impulse response. It is a way of looking at the time domain behavior. Its impulse response is changed through either a Linkwitz Transform or time domain convolution (such as Acourate). It is more accurate to change it with time domain convolution.

Your post sounds like you don't believe the Q of the system itself is relevant to that perception of tightness, is that a fair interpretation? For example, this thread shows how the theoretical step response changes with different values for Q.

From what I read, the thread deals with Qtc which is correctly described in the first post as "a complex mathematical equation derived from driver, electrical and enclosure parameters." Electrical parameters means the T-S parameters of the driver. The Qtc of a sub is set once you build it. Using PEQ filters, a Linkwitz Transform, or convolution affect the signal that you input into the subwoofer. Those changes to the signal will result in a changed system Q which absolutely is relevant to the "perception of tightness." However, there is no such thing a perfect system Q. It changes to what it needs to in order to get a flat frequency response. If you fix the frequency response, you are changing the impulse response. If you change the impulse response, you change the frequency response. They are both linked. So in the end, you just try to get the frequency response you desire and don't worry about what the system Q might be. You also don't worry about some intermediate system Q that happens after the Linkwitz Transform but before convolution. 

Whether you use an LT and then convolution or if you just use convolution, if the final frequency response is the same then the system Q is identical as well because the exact same change to the subwoofer's input signal was done.

[BradC]

For a minimum phase system, the impulse response is directly linked to the frequency (and phase) response.

However, for a non-minimum phase system that is not the case.

Hence with linear phase filters it is possible to improve the impulse response without changing the frequency response (in principle).

ie linear phase filters of acourate and the like do buy you more flexibility to correct problems

[mwillems]

For a minimum phase system, the impulse response is directly linked to the frequency (and phase) response.

However, for a non-minimum phase system that is not the case.

Hence with linear phase filters it is possible to improve the impulse response without changing the frequency response (in principle).

ie linear phase filters of acourate and the like do buy you more flexibility to correct problems

I think mojave's point was that it's the final measured frequency response and phase that matter, regardless of whether you achieve it a) by doing a minimum phase correction (i.e. a conventional LT) followed by a linear phase correction (convolution) or b) just doing the whole thing in convolution from step one.  

And with one caveat, I agree: the FR and phase results are what counts, not the intervening steps.  The only caveat I'll offer (that is specifically applicable to the perception of "tightness" in subs) is that there is an additional measurable time-domain response parameter other than phase, namely, time decay.  Decay is not readily visible in conventional FR and phase measurements, although the decay information is present in the impulse and can be "cooked out" of it. Decay is usually presented as a waterfall plot (which REW does quite nicely).  

You can (at least theoretically) have two systems with nearly identical FR and phase measurements, that nonetheless have very different waterfall plots as a result of a lot of variables (size of the driver, the driver motor, box resonances, room resonances, a port, etc.).  Some speaker elements are just "faster" than others (which you can easily see if you ever look at a set of speaker waterfall plots taken in free air, like Zaph does). But even with a "fast" driver, some boxes and some rooms will just keep ringing forever.  

In my opinion, the waterfall (speed of the decay) of your system  in the relevant frequency band is the single biggest predictor of the perceived tightness of bass.  You can look at your own waterfall measurement and immediately "see" how tight the bass is or isn't at the measurement position.  

That said, I think mojave's original point is correct: in terms of electronic speaker correction, ordinarily, the single best thing you can do for your waterfall is get a flat FR and flat/coherent phase (however you do it) as flattening FR will tend to minimize the excitation of resonances (by reducing the SPL at those places) and flattening/unwrapping phase will reduce certain types of ringing (like port ringing, for example). Once you've done that, every other step you can take to improve perceived "tightness" involves dealing with issues extrinsic to the audio signal (adding bracing or stuffing to your speaker box, treating your room, etc.).

[mattkhan]

Would it be correct to reduce your (mojave/mwillems) posts to "the transient response of a subwoofer is completely described by its frequency response & hence all subwoofers will sound the same if they are equalised to produce the same response in that room & don't produce audible amounts of distortion"?

[mwillems]

Would it be correct to reduce your (mojave/mwillems) posts to "the transient response of a subwoofer is completely described by its frequency response & hence all subwoofers will sound the same if they are equalised to produce the same response in that room & don't produce audible amounts of distortion"?

I wouldn't say that's exactly my position.  My position is that the transient response of the subwoofer is completely described by it's impulse response, which is informed by: frequency response, phase, and decay.  I would say that two subwoofers that are equalized/convolved to produce identical FR and phase response (at comparable distortion) will "sound" as close as signal manipulation can make them, but still not necessarily identical, because of decay.  If you ring two bells, they might have the same pitch (FR) and start ringing at the same split second (phase), but one might keep ringing for much longer after the strike than another based on what it's made of and mounted to (decay).

For those reasons, there will still be a residual difference in the sound between two subs with identical frequency response and phase response based on mechanical factors: the drivers, box construction, and position in the room.  The driver, the sides of the box, or the walls of the room are like the bells, they keep ringing once excited for differing amounts of time.  Those differences would appear in a waterfall plot, but not necessarily in a frequency response and phase measurement.  The only ways to "correct" those residual differences is mechanical (room treatment, box damping, different drivers, etc.).  

The bottom line is that, in my view, two speakers whose outputs have identical FR and phase measurements are as close as signal processing can get them, no matter how you got there, but they won't sound identical in part because of differing time-decay response.  

To provide a sub-specific example: imagine two subs that both start the uphill journey of the transient at the same time, but drop off at different rates. One of the subs may "ring" at a given frequency for a millisecond after the time zero (the transient takes a millisecond to fully drop off), and the other sub may ring for two dozen milliseconds after the time zero.  Those two subs might show identical frequency response and phase response because they both ramped up to the correct volume (FR) at the correct time/part of the sine wave cycle (i.e. phase), but the time it takes them to stop making noise (decay) isn't shown on a normal FR and phase plot.  And that difference will be audible; the sub that's slower to stop ringing will sound more smeared than the other sub; the sub that decays faster will sound tighter.  

It's one of the reasons that vented speakers get a bad rap; a port introduces significant phase wrap and also ringing around the port frequency. In that case, I've found that fixing the phase can fix most of the port ringing, and, more generally, phase manipulation can definitely improve decay measurements to some extent.  But some kinds of ringing or decay problems (i.e. box vibration, "slow" drivers), can't always be fully "resolved" through signal manipulation.

Does that make sense?  

Check out this article for an interesting case study in why time decay can make a big difference in the sound of a speaker: http://www.soundonsound.com/sos/sep08/articles/yamahans10.htm

And check out this page over at Zaph audio: http://www.zaphaudio.com/5.5test/compare.html . He's tested dozens of drivers, and if you compare the Frequency Response graphs with the "CSD" graphs (cumulative spectral decay, aka waterfall plot) you can see how differently different speakers perform on those tests even with fairly similar FR.

I can't speak for mojave's view (although I'm interested to hear his thoughts).

[mojave]

I can't speak for mojave's view

I'll agree with your view so I don't have to type as much. 

If you take a close mic measurement of a subwoofer/enclosure/amplifier, you can look at a waterfall chart to see the decay. You can never improve the native decay. The faster and cleaner the decay, the better the subwoofer can track the signal. It will sound different because there are more nuances that are smeared in a frequency response measurement. I think the main driver parameters that affect decay are motor strength (BL). mass (Mms) and suspension compliance (Cms). You can also affect it with box size and alignment. Finally, if you don't have sufficient power for transient bursts, then the subwoofer will sound less dynamic all things else being the same.

[mattkhan]

Yes this all makes sense, thanks.

If we go back a few posts to something I said earlier

I don't follow what you mean exactly, can you expand on that? My understanding is that the LT is applied to give a specific anechoic response which dictates the response characteristics of the sub itself & that impacts the step response (and hence the perception of a tight sub).

I was referring to that temporal aspect of how long & strong the ringing is after the initial impulse. It seems to me that this information is adequately described by the step response & a spectral decay/waterfall view (which AIUI is created by shifting a window forward over the IR and transforming into a frequency response for that window) is only needed if you want to drill down into how specific frequencies decay (e.g. to look for offensive resonances). Here's an article that seems to make the same point - http://audioxpress.com/files/2008/10/dappolito2960.pdf

To give an example, I have attached the near field frequency & step response of my sub before and after I cut out that (presumably inductance driven) hump. The flat version is clearly more like the theoretical step response as it strength and length of the post step ringing is clearly reduced.

The issue I've been mulling over though is the one mojave pointed out around any intermediate response being irrelevant to what you actually hear. Once I shift the mic to the LP then obviously the response changes and hence further correction is then required to trim the modal ringing down to produce an acceptable frequency response. I was therefore wondering whether that intermediate step is a worthwhile step or not. I haven't got round to a near field measurement of the sub with the final correction filter in place yet. I guess that could be informative (to compare the 2 approaches to correction; room only & room + near field).

[mwillems]

I was referring to that temporal aspect of how long & strong the ringing is after the initial impulse. It seems to me that this information is adequately described by the step response & a spectral decay/waterfall view (which AIUI is created by shifting a window forward over the IR and transforming into a frequency response for that window) is only needed if you want to drill down into how specific frequencies decay (e.g. to look for offensive resonances). Here's an article that seems to make the same point - http://audioxpress.com/files/2008/10/dappolito2960.pdf

They can be useful for a lot of tasks besides resonance hunting (which is plenty useful), like comparing different drivers/speakers, or comparing different room positions for your existing speakers.  Also, waterfalls make it much easier to visualize certain aspects of the response.  For example, I can't tell from looking at an impulse how many milliseconds it takes before the FR output is -40dB across the board.  On a waterfall, that's easy to see. 

You're absolutely right that all the information in a waterfall is in the step response/impulse (the waterfall is derived from it, as you say), but FR and phase are derived from the impulse too.  For me, anyway, it's much easier to see all those things when they're extracted from the impulse. 

To give an example, I have attached the near field frequency & step response of my sub before and after I cut out that (presumably inductance driven) hump. The flat version is clearly more like the theoretical step response as it strength and length of the post step ringing is clearly reduced.

The issue I've been mulling over though is the one mojave pointed out around any intermediate response being irrelevant to what you actually hear. Once I shift the mic to the LP then obviously the response changes and hence further correction is then required to trim the modal ringing down to produce an acceptable frequency response. I was therefore wondering whether that intermediate step is a worthwhile step or not. I haven't got round to a near field measurement of the sub with the final correction filter in place yet. I guess that could be informative (to compare the 2 approaches to correction; room only & room + near field).

I think comparing measurements will ultimately answer your question.  If the correction from both methods is the same, and they measure the same, (for our purposes) they are the same.  The only reason to do a two-step process is if you get different (and better) results doing it that way than doing the one step process.  In my speaker correction guide, I advised correcting the speaker first and then moving onto the room as a second step, because I've gotten objectively better correction doing it that way using the software I used. In that case the two-step method produced measurably different results than the one step method.  

It may be that Audiolense and Acourate are sophisticated enough that the one-step method is every bit as good (or better) than the two step method.  Your comparison plan is the best way to find out  

I'll agree with your view so I don't have to type as much.  

 

[mattkhan]

It may be that Audiolense and Acourate are sophisticated enough that the one-step method is every bit as good (or better) than the two step method.  Your comparison plan is the best way to find out  

acourate is as sophisticated as the operator, this might be where the problems start 

[BradC]

Just to correct a mathematical point:

The complex amplitude response as a function of frequency is related to the complex amplitude response as a function of time via the Fourier Transform.

That is the Freq and phase plots have the same information as the impulse plot. Delay is contained in both.

It is just that some plots such as waterfall, ETC, group delay process the information to make the delay easier to see.

Useful information about how to treat a room or speaker can be gained from all the different plots. In fact, it is good to look at the different plots to make sure you haven't been fooled by a certain representation.

Some people will argue at length on various fora that their approach is best and you don't need (insert a plot).

[BradC]

acourate is as sophisticated as the operator, this might be where the problems start 

I find it  generally the case that the more flexible a solution, the more ways it can go wrong!

However, the flexibility of acourate does allow you to produce a better solution than the more automatic products.
Further, the macros in acourate do produce a pretty good result for a stereo system with the basic settings.
But there is a big learning curve to go further

[BradC]

 I was therefore wondering whether that intermediate step is a worthwhile step or not. I haven't got round to a near field measurement of the sub with the final correction filter in place yet. I guess that could be informative (to compare the 2 approaches to correction; room only & room + near field).

From what I understand, the 2-step process is useful if you use minimum phase filters in step 1. Then you will benefits from the EQ at more than just the one listening position

[mwillems]

That is the Freq and phase plots have the same information as the impulse plot. Delay is contained in both.

Just to make sure we're not talking past each other: I was talking about using waterfalls to observe decay, not delay.  Delay is definitely shown in a conventional phase graph (that's in large part what the graph is showing), but conventional FR and phase graphs don't show spectral decay, which is primarily what a waterfall or CSD shows.  

Of course, all three graphs are derivable from the impulse (the impulse has all the information in it), and all three graphs are important to speaker analysis.  

Useful information about how to treat a room or speaker can be gained from all the different plots. In fact, it is good to look at the different plots to make sure you haven't been fooled by a certain representation.

I agree completely.

[mattkhan]

one last thing to doublecheck on this, am I right in thinking that the following output format settings are the right ones to use for a 5.1 setup

channels: 5.1 channels
mixing: JRSS mixing
subwoofer: silent

i.e. use jrss to mix to 5 main channels where the source channels != 5

or to put it another way, the wiki has option 3 "If you want JRiver to upmix and let the receiver handle speaker setup and bass management, then . . ." which could be written as "and let the [receiver|convolution engine] handle speaker setup and bass management" right?

[mwillems]

one last thing to doublecheck on this, am I right in thinking that the following output format settings are the right ones to use for a 5.1 setup

channels: 5.1 channels
mixing: JRSS mixing
subwoofer: silent

i.e. use jrss to mix to 5 main channels where the source channels != 5

That's correct, assuming that you're still doing your bass management downstream in convolution.  The settings you've chosen will result in 5.1 remaining 5.1, and everything else will get mixed to 5.1 or 5.0 (including stereo audio).  My understanding is that if there's no LFE in the source, JRSS will not create one with those settings, but if there is an LFE channel (e.g. 7.1 source) it will retain it in the mix.

If you don't want stereo audio to get mixed to 5.0 (or 5.1), make sure the box labelled "for stereo sources only mix to 2.1" is checked. That will, with your other settings, result in stereo sources remaining stereo (because you have subwoofer set to silent).