BOSSOBASS Raptor system 3

[maxmercy]

AFAIK, all the USB mic options do not have a good, repeatable loopback timing reference option.

 

JSS

[Madaeel]

@Madaeel, your observations from the nearfield experiment are interesting, and matching my own experiences.

 

At very low freqs, the velocity potenial may not be that important, or it is simply that the velocity we are able to create simply is not enough.

 

Interesting discussion, and many good points made in this thread, and so well hidden I nearly missed it all.

 

I have posted on different places about particle velocity, tactile feel and bass, may be we could create a thread on this subject here on data-bass - where there is interest, knowledge and you don't need to waste energy in discussing "you can't hear below 20hz" or "what is this nonsense about bass you can feel".

A lot of the recent talk is over my head but what I do pick up is much appreciated. That's why I love DB. Almost always some good discussions.

 

@Dominguez1 has a ULF thread at AVS. He and @coolrda are starting some tactile tests to find out exactly what goes on there and how we could possibly increase it without causing problems in the FR. We both wish someone could get ahold of a Microflown meter but until then a phone is apparently our best option. Check that out if you get a chance.

 

Myself, PVL is driving me crazy. A lot of guys here make good points about it and why we do, or don't, feel it as much. I'll say IF you have the capability to even play ULF you need a suspended or riser if you're on a slab to appreciate it fully. When I had the Raptors near field I finally felt it clearly and it was amAzing! Unfortunately my FR was shit so it was a no go.

 

I'll have a chance to take my Raptors upstairs when my wife is away for a month or so. Should be fun.

[dominguez1]

@Madaeel,

 

@kvalsvoll is @okv on AVS, so he's already been contributing! 

 

I'm sure we'll get to the bottom of this SIL/PVL stuff and how best to take advantage of it...there will certainly be bumps along the way, but that's the fun part and essential to learning! 

[Madaeel]

Yah I just saw that haha.

 

Now Max' talk of 50hz and up feelin like a mule kick has got me thinking. It never ends.

[Kvalsvoll]

What I run on the big system now is a config with less velocity than what is possible to achieve in the room, but it has what I find the best overall qualities - feels more balanced and even in response, and especially the upper bass is better with exactly that kick, which is at higher frequencies from around 80hz and up.

 

To achieve this I found the integration with the mains to be important - delay on the mains adjusted for best possible impulse response.

Still, I think there is a lot more to gain here, with more defined and more tactile upper bass.

 

Other configurations have more powerful feel in the lower bass, I believe this is due to the particle velocity.

And one had much better punch on mid-sized drums, around 50Hz - on some recordings the drums had very good definition and real physical punch, and the feel of different sized drums was very distinguished, the smaller sharper ones up in the 80-150hz range, then the bigger ones around 50hz, and even the largest around 30hz sounds different, heavier, but still with punch.

 

It is important to note that the measured frequency response of the different set-ups are all similar within say +-1dB.

Which means all the differences must be attributed to impulse response and sound field properties - the particle velocity.

 

It is not very difficult to measure velocity alone, I have done many experiments with this already, with measurements of velocity, so that I can know what is going on, with no measurement it is more of a guesswork as to finding the causes for the perceived differences in sound.

To measure intensity you need to have the phase relation between pressure and velocity, and this is where it gets a lot more tricky.

However, you can tell a lot by looking at the velocity graphs - they contain information about velocity level, and also direction (you need to measure in different directions).

High intensity means high directivity.

 

What is great about data-bass is the genuine entusiasm and curiosity that overpowers all the usual my-approach-is-better often found on the net. 

[SME]

that's not the correct comparison as we're not talking about the difference between a pulse and a tone, we're talking about the difference between a pulse that has a certain magnitude and that then decays to 0 over a given time period vs a pulse that has a lower initial magnitude but rapidly increases to that same magnitude and then decays to 0. The question then is whether or not your auditory system has the sensitivity & temporal resolution to detect that reduced initial magnitude. 

 

Take 40Hz as an example, you'll get a 1st axial mode here with a dimension of 4.2m & a full cycle at 40Hz is 25ms long. Assume the sub is in a corner so the 1st reflection is really off the wall behind you, by the time you've "heard" a full cycle at 40Hz, the leading edge of wave has already travelled the length of the room two more times hence that increasing magnitude caused by the mode is already happening before you've even heard the entire wavelength. 

 

We also know that human hearing can be modelled as a series of filters (e.g. the gamma tone filter bank model) and that those filters get increasingly long as frequency reduces. This means temporal resolution of our hearing reduces as frequency reduces while the frequency range covered by each filter gets smaller and smaller (in linear terms anyway)

 

The combination of these two physical facts (a mode acts fast relative to the period, your hearing has poor temporal resolution at LF) make me quite sceptical that we can distinguish that lower initial magnitude produced by the use of EQ to cut a mode. 

 

When I get time I'm going to try and play with this anyway by comparing correction using an IIR filter (traditional PEQ) vs a VBA based approach (same sub will emit a delayed, inverse signal to provide the cancellation) vs a physical active bass trap (same VBA sort of approach but the cancelling signal reproduced by a different sub). I can do the 1st two easily enough, the 3rd one needs me to build a box for a driver I have handy. 

I believe you are over-thinking this by bringing psychoacoustics to bear unnecessarily.  In your 40 Hz example, "the 40 Hz part" of the direct sound and first (and likely later) reflection(s) can't be distinguished by either your ear or any measurement instrument.  They are physically indistinguishable and appear as a single arrival that's somewhat delayed overall compared to the rest of the sound.  Unless there are other reflections going on, this single mode is very likely to be minimum phase at all listening locations, including the middle of the room where the sound almost (but not quote) vanishes.  Whether you sit in a peak or dip area, you can use a minimum phase EQ filter to correct the sound, at least as long as you aren't too close to the center null where you'll run out of headroom.  If all your seats are in a row, the same distance between front and back, then you can completely eliminate the influence of the mode at those seats with a minimum phase filter.

 

As I said, the idea that applying say a narrow PEQ dip to correct a minimum phase peak puts a "hole" in the direct sound is not physically correct.  Instead, it takes several cycles, after which much reflection has occurred, for that hole to develop in the filter response.  And in fact, if the filter corrects the magnitude response, then it must also correct the temporal response.  In other words, that energy that arrives later without the filter in place is made to arrive with the rest of the energy again.

[3ll3d00d]

this is the physiology of the ear not psychoacoustics anyway my point is the one you're making, it doesn't seem physically possible to detect that cut in the response. 

[Bossobass Dave]

Well, now we've gone to a more specific example: "As I said, the idea that applying say a narrow PEQ dip to correct a minimum phase peak puts a "hole" in the direct sound is not physically correct."

 

How "narrow" and how much of a "dip" is what will determine how audible the distortion is.

[Bossobass Dave]

Take 40Hz as an example, you'll get a 1st axial mode here with a dimension of 4.2m & a full cycle at 40Hz is 25ms long. Assume the sub is in a corner so the 1st reflection is really off the wall behind you, by the time you've "heard" a full cycle at 40Hz, the leading edge of wave has already travelled the length of the room two more times hence that increasing magnitude caused by the mode is already happening before you've even heard the entire wavelength.

 

The reality is that you'll take the higher frequencies as the cue for transient accuracy and the fundamental frequency as the weight, as what you'll hear is the decay of the 40 Hz fundamental after the higher frequency cues. The onset of the 40 Hz fundamental is masked.

 

Here is the low E string of an electric bass with no amplification or processing, played with maximum attack with a nylon pick:

 

 

The harmonics are louder than the fundamental and the spectra goes out beyond 2k Hz.

 

So, I agree that with real source the arrival timing of the 40 Hz fundamental from the subs is masked and I believe that the 40 Hz fundamental taking 2 to 5 times longer than the harmonics to decay adds the weight. EQ less 40 Hz and the bass gets "tighter, faster", EQ less harmonics and the bass gets "muddier, slower, phatter".

[SME]

+1^^^

 

As well as the final response. Looks like a 20 Hz tuned ported system?

 

I believe that on time delivery is important but is most important an octave above and below cross (which is why including the mains in the measurements is a must, IMO). Below that, the decay time tends to swamp the start time.

 

I'm interested in the main seat only. Responses at seats outside the sound field are not relevant to critical listening discussions.

 

I posted results for the LFE channel to illustrate what's possible using independent EQ on multiple subs and EQ in general.  That EQ only applied to the subs, so it doesn't make much sense to show the mains responses there.  In any case, the bass-managed responses of the mains will usually be better when the response of the sub(s) alone is smooth.

 

In so far as getting good "kick" is concerned though, yes I believe looking at the bass-managed mains responses from 40-160 Hz is better than looking at the sub alone.  Here is a spec of my current bass-managed center channel impulse response at my MLP with 50 ms window:

 

 

Here are some important notes:

• I don't choose to optimize exclusively for one seat.  I like to listen socially, and I believe in maintaining a reasonable compromise between a great response at the MLP versus a good response away from the MLP.
• The LFE response I showed previously represents the response with only MiniDSP 2x4 processing.  The LFE response is additionally altered to have a slight (2-3 dB or so) increase in mid and upper bass to provide a more balanced sound outside the MLP.
• The center channel response alone is also filtered with both IIR and FIR filters.  Without these filters the combined response looks much uglier, even well below the crossover frequency.
• The center channel currently crosses over to the subs at 110 Hz.  This crossover was chosen because it gives the smoothest response at each location, not because of any limitation in the speaker.  As I continue to improve my room with treatments, I will likely opt for another crossover choice, depending on what works best.
• There is an intentional rise in the bass relative to the rest of the response, and most of that rise occurs between 100-200 Hz.  For off-center seats, that rise is spread over a somewhat wider range of 50-200 Hz or so.
• In the vicinity of 110-160 Hz, I run into a non-minimum phase problem caused by reflections:  (1) the direct sound output of the center channel is diminished in that range by an untreated reflection on the wall behind it that causes destructive interference;  this reflection is not visible in the spec because its delayed by less than one wavelength; (2) energy arrives later from reflections on both side-walls at the same time.  The double dose of side-wall energy overwhelms the weak direct sound wavefront leading to the non-minimum phase condition.  The only way to correct this electronically is to use a filter that begins injecting energy before time zero.

As stated in my other posts, minimum phase filters can correct both time and frequency aspects of a response at the same time, but only where that response is minimum phase.  In the trouble region, I have corrected the frequency response approximately but the energy still arrives later than it should.  What we should see there instead is the red band sloping down more with increasing frequency and shifting in color to dark orange while remaining continuous with the dark orange band that appears to start at around 130 Hz.

 

Of course, this non-minimum phase problem is with the center channel on its own.  It's not realistic to have center channels in 4 different locations as it is with subs, so this is much harder to fix.  Since this problem also affect off-center seats, I expect to be able to improve things a lot with a mixed phase filter, but I haven't gotten there yet.  In the long-run I plan to address it acoustically by re-arranging my front stage and adding more bass trapping.  I also am planning to linearize the crossovers using mixed phase filters, which should get rid of the slope entirely at least down to 30 Hz or so.

[maxmercy]

Yah I just saw that haha.

 

Now Max' talk of 50hz and up feelin like a mule kick has got me thinking. It never ends.

 

The thing is, I don't know the FR of that TX theater.  But when SideSwipe pole-vaults over the Audi and fires his cannons, it was as if someone was literally slapping my back/chest, and even stronger when the top secret rail gun fires, and also when Megs blasts Prime's chest out with his cannon in the forest.  Most powerful film presentation I have ever heard  They were playing AC/DC's Back in Black Album through the system before the movie started, and it did not seem to have an EQ boost in the low end like most people playback AC/DC, it sounded like a flat FR.

 

On my first DIY system (30Hz horn loaded sub, capable of clean playback down to ~27Hz at -12dBRef), I had large 30Hz and 60Hz peaks in my response due to modes.  Every film would have more 'kick', even when compared to later EQ'ed systems (that sub was before my first EQ).  It may be that TX theater had a peak in response around 50-80Hz, but I don't know for sure, and it may be that some people may prefer more midbass for that reason.  But most rooms have problems right in the kick frequencies (50-500Hz), with huge holes in response taking away from it.  I am currently building bass traps to help with my midbass punch now that I am good to under 5Hz at my playback level.  My center channel speaker has terrific MB response, but my L/R and surrounds could use some help.  My center is essentially in a baffle wall, surrounded by 4" thick OC703 w/ 2" airspace behind the screen.  My L/R are toed in, so they will need the pink fluffy traps I am building flanking them to get a better response and take care of some nulls I have from 80-200Hz.  The 4" OC703 traps that project will free up will go to 1st reflection points and go around the surrounds to tighten them up.  I am a big believer in baffle-wall installations, so many 1/4-wave problems solved all at once simply by being in-wall.  

 

I have also learned that I prefer a slightly rolled off response, and line arrays can provide that automatically.  My next system will likely be floor-to-ceiling lines in-wall or on-wall surrounded by absorbing material.  But that will be several years in the future.  I will likely also add 8 more 15" RefHFs to get to 0dBRef cleanly, and -7dBRef immaculately.  The problem then will be finding unclipped films to playback.

 

 

JSS 

[SME]

The reality is that you'll take the higher frequencies as the cue for transient accuracy and the fundamental frequency as the weight, as what you'll hear is the decay of the 40 Hz fundamental after the higher frequency cues. The onset of the 40 Hz fundamental is masked.

 

Dave makes a good point here.  To have a 40 Hz wave suddenly start playing, you necessarily have other frequencies present too.  While I wouldn't argue that they mask the 40 Hz tone (it really depends on the relative levels), the higher frequencies will affect our perception of the timing of that tone.  When the sound of a bass pluck reflects from the back wall, the 40 Hz part won't reveal the distinct reflection, but the higher frequency portion of the pluck will.  On the bright side, this means that if you put bass traps on that wall, knowing that bass traps rarely work well below 80 Hz or so, you'll still improve the tightness of that bass pluck because you'll be absorbing the > 80 Hz part of the reflection that contributes to our perception of timing.  This only works to a point though.  As long as contribution to the response by the room mode is fairly minor, then it may not be particularly annoying, even though it might sound subtly better to have all the energy arriving at the same time.  However, if the mode is so severe that the 40 Hz tone rings for a long time after the pluck, then you'll still notice that bass overhang, regardless of what the upper bass is doing.  Moreover, plucks for other notes will tend to be colored by that overhang, and you may hear 40 Hz ringing even with notes whose fundamentals aren't at 40 Hz.

[maxmercy]

SME,

 

Did you EQ and time-align each sub independently?  Or just time-align each and EQ as a group?  I use Eight 15" subs, in two groups.  First group of 4 is on the front wall, at the 1/4 points to cancel width modes.  Second group of four is clustered in the center of the room directly behind MLP for nearfield reinforcement as well as because it gave a better simulated response than a DBA arrangement (mirroring the front wall subs on the back wall and applying a delay and inversion to the signal to cancel the front wall wave out).

 

I found that EQ'ing and time aligning both groups of subs independently  vs time-aligning and then EQ'ing globally yielded the same response.

 

JSS

[SME]

I'm interested in seeing the responses of each sub and what correction and delays you applied to each sub to get good results.  

 

JSS

OK.  Here are specs of each sub before EQ windowed at 100 ms.  Refer to this post for spectrograms of the final result using different window lengths:

 

Front left sub:

 

Front right sub:

 

Left mid bass sub:

 

Right mid bass sub:

 

A few notes:

• My room is rather asymmetric and is open to the rest of the house.  Each sub really does have a unique response.
• The front left and front right subs are slightly different models, both with port tunes in the 15-20 Hz area.  The front left sub was a less expensive product (12" woofer instead of 15" and smaller box) than the front right sub, and it appears to use a higher Q tune to hit 16 Hz (a little too) well as compared to the front right sub.  To compensate for its reduced capabilities, the front left sub is gain-matched -3dB from the front right sub.  This gain difference is readily apparent in the specs.
• The left and right mid bass subs are of the same model and are located behind my (newly acquired) curved sofa, about 1 meter directly to the left and right of the MLP.  They have a built-in 4th order HPF, at 50 Hz.  The subs should be approximately gain-matched.  The right mid bass sub has a stronger response than the left mid bass sub because it is surrounded by more wall.
• The left mid bass sub response looks quite smooth, and it is at the center seat, but it's response rather sucks at the other seats in the 70-90 Hz range, so much so that I get the best overall performance by EQing it way down in that range and letting the right mid bass sub handle that content by itself.  I hope to fix this issue by installing bass traps optimized to absorb in this region on the ceiling above.

I also want to post a picture of the frequency response (high resolution and un-smoothed) of each individual sub before EQ and the frequency response of the LFE channel with all four subs after optimization:

 

 

Notes:

• The black line is the combined response.  It is pretty close to fitting in a +/- 3 dB window.  Before I got my new sofa, this window was tighter at the MLP because I was optimizing over a smaller region.  Now I have about +/-4 .5dB across a sofa that curves to the front and seats up to 7 people (albeit very cozily).
• There is small narrow bump and a relatively deep narrow dip in the region from 56-64 Hz.  This is an area I'm having a lot of trouble with.  In retrospect, I believe I was a bit over-aggressive here in my most recent optimization.  Nevertheless, having my side seats be a bit further forward into the room is making it much harder to get a clean response there across those seats.  I'm fairly certain that the room ringing that coincides with that adjacent peak and dip is sometimes audible.  I know because my previous optimization was a bit cleaner overall.
• The dip at 90 Hz is not as severe as it looks.  I did those measurements before I added some more bass trapping that helped here slightly.  I expect this dip to vanish once I get good bass trapping on the ceiling (see above).

Currently, I use the following delays (ms) for each sub:

•   front right: 0.0
•   front left: 0.5
•   left mid bass sub: 6.8
•   right mid bass sub: 4.5

Admittedly, this information isn't very useful without knowing the layout of the room, but the delays are roughly proportionate to the relative distances involved.  Why the left mid bass sub blends so much better with the right mid bass sub when they are about the same distance from the center of the room, even at higher frequencies > 100 Hz, is a mystery, but the asymmetry of the room is surely a big factor here.

 

Now I will copy and paste the filter definitions I used for this run.  They are formatted as source code, but it should be very clear what they do.  Unfortunately, my MiniDSP 2x4 exhibited signs of insufficient precision when rendering filters intended to act on the lowest frequencies.  As such I had to replace some of these with others using trial-and-error and disable a few others entirely.  This is noted below.

 

Front right sub:

• # This was actually programmed to the DSP.
• LowShelfFilter(f0=25.0, gain=-2.0, Q=0.8),
• # EQPeakFilter(f0=27.0, gain=+2.0, Q=5.0),
• # EQPeakFilter(f0=22.0, gain=-1.0, Q=8.0),
•  
• # These were not programmed to the DSP.
• LowShelfFilter(f0=25.0, gain=-3.0, S=3.0),
• EQPeakFilter(f0=16.75, gain=-5.0, Q=12.0),
• EQPeakFilter(f0=17.75, gain=+3.0, Q=16.0),
• EQPeakFilter(f0=20.75, gain=-4.0, Q=6.0),
• EQPeakFilter(f0=24.25, gain=-1.5, Q=8.0),
• EQPeakFilter(f0=27.5, gain=-1.0, Q=12.0),
•  
• # These were programmed to the DSP as-is.
• EQPeakFilter(f0=30.0, gain=-2.0, Q=12.0),
• EQPeakFilter(f0=36.0, gain=-2.5, Q=8.0),
• EQPeakFilter(f0=38.0, gain=+3.0, Q=2.0),
• EQPeakFilter(f0=45.5, gain=+4.5, Q=5.0),
• EQPeakFilter(f0=48.0, gain=+1.5, Q=6.0),
• EQPeakFilter(f0=55.0, gain=-9.0, Q=6.0),
• EQPeakFilter(f0=64.0, gain=-10.0, Q=5.0),
•  
• LowPassFilter(f0=50.0),
• LowPassFilter(f0=50.0),

 

Front left sub:

• # This was actually programmed to the DSP.
• # LowShelfFilter(f0=25.0, gain=-2.0, Q=0.8),
• # EQPeakFilter(f0=27.0, gain=+2.0, Q=5.0),
• # EQPeakFilter(f0=22.0, gain=-1.0, Q=8.0),
•  
• # These were not programmed to the DSP.
• LowShelfFilter(f0=25.0, gain=-3.0, S=3.0),
• EQPeakFilter(f0=16.75, gain=-5.0, Q=12.0),
• EQPeakFilter(f0=17.75, gain=+3.0, Q=16.0),
• EQPeakFilter(f0=20.75, gain=-4.0, Q=6.0),
• #EQPeakFilter(f0=24.25, gain=-1.0, Q=8.0),
• #EQPeakFilter(f0=27.5, gain=-1.0, Q=12.0),
•  
• # These were programmed to the DSP as-is.
• EQPeakFilter(f0=30.0, gain=-1.0, Q=12.0),
• EQPeakFilter(f0=36.0, gain=+0.5, Q=8.0),
• EQPeakFilter(f0=38.0, gain=+5.0, Q=2.0),
• EQPeakFilter(f0=45.5, gain=-0.5, Q=5.0),
• EQPeakFilter(f0=48.0, gain=+1.5, Q=6.0),
• EQPeakFilter(f0=58.0, gain=+18.0, Q=5.0),
• EQPeakFilter(f0=64.0, gain=-12.0, Q=5.0),
•  
• LowPassFilter(f0=50.0),
• LowPassFilter(f0=50.0),

 

Left mid bass sub:

• EQPeakFilter(f0=50.5, gain=-5.0, Q=8.0),
• EQPeakFilter(f0=52.5, gain=+1.5, Q=10.0),
• EQPeakFilter(f0=59.0, gain=+7.0, Q=6.0),
• EQPeakFilter(f0=65.5, gain=-7.0, Q=8.0),
• EQPeakFilter(f0=72.0, gain=-16.0, Q=8.0),
• EQPeakFilter(f0=81.0, gain=-11.5, Q=14.0),
• EQPeakFilter(f0=90.0, gain=-12.0, Q=12.0),
• EQPeakFilter(f0=94.0, gain=+7.0, Q=10.0),
• EQPeakFilter(f0=110.0, gain=+4.0, Q=5.0),
• EQPeakFilter(f0=120.0, gain=+2.0, Q=16.0),

   

Right mid bass sub:

• EQPeakFilter(f0=36.0, gain=-18.0, Q=8.0),
• EQPeakFilter(f0=50.5, gain=-5.0, Q=8.0),
• EQPeakFilter(f0=52.5, gain=+1.5, Q=10.0),
• EQPeakFilter(f0=61.0, gain=+2.0, Q=16.0),
• EQPeakFilter(f0=65.5, gain=-4.0, Q=8.0),
• EQPeakFilter(f0=73.5, gain=-3.0, Q=6.0),
• EQPeakFilter(f0=77.5, gain=-1.0, Q=12.0),
• EQPeakFilter(f0=80.5, gain=+0.5, Q=12.0),
• EQPeakFilter(f0=87.5, gain=-4.5, Q=12.0),
• EQPeakFilter(f0=90.0, gain=+2.0, Q=12.0),
• EQPeakFilter(f0=94.0, gain=+7.0, Q=10.0),
• EQPeakFilter(f0=126.0, gain=+4.5, Q=16.0),

And there you have it.  These filters aren't necessarily the best possible solution.  They are simply one of potentially many good solutions for my space.

[SME]

SME,

 

Did you EQ and time-align each sub independently?  Or just time-align each and EQ as a group?  I use Eight 15" subs, in two groups.  First group of 4 is on the front wall, at the 1/4 points to cancel width modes.  Second group of four is clustered in the center of the room directly behind MLP for nearfield reinforcement as well as because it gave a better simulated response than a DBA arrangement (mirroring the front wall subs on the back wall and applying a delay and inversion to the signal to cancel the front wall wave out).

 

I found that EQ'ing and time aligning both groups of subs independently  vs time-aligning and then EQ'ing globally yielded the same response.

 

JSS

 

I did in fact EQ each sub independently rather than EQ as a group.  Aside from its asymmetry both in the left/right and front/rear dimensions, my room is open to the rest of the house.  With a few exceptions (e.g., 63 Hz), my response is not as heavily influenced by modes as it is by early reflections.  It's possible that my approach is much more useful in my circumstances than it would be for those using a small, sealed shoe-box shaped room.  I don't have the direct experience with this more common case to know for sure.

[3ll3d00d]

FWIW my experience has been that phase correction is the thing that has made the dramatic improvement to perceived SQ (an immersive & cohesive sound field, transients that really attack). AIUI the brain is quite sensitive to interaural phase differences upto ~1500Hz and hence getting consistency in the phase response across each speaker while also correcting any spectral balance issues separately seems like the key step forward to me. From what I've read, most people whose correction includes phase correction tend to report similar sorts of subjective impressions (though it's a relatively small no of people commenting really).

[Infrasonic]

I only use one on my toilet seat. If I'm constipated, 15 Hz usually does the trick.

 

I guess I should have know better than to ask you about it, Dave.

[SME]

FWIW my experience has been that phase correction is the thing that has made the dramatic improvement to perceived SQ (an immersive & cohesive sound field, transients that really attack). AIUI the brain is quite sensitive to interaural phase differences upto ~1500Hz and hence getting consistency in the phase response across each speaker while also correcting any spectral balance issues separately seems like the key step forward to me. From what I've read, most people whose correction includes phase correction tend to report similar sorts of subjective impressions (though it's a relatively small no of people commenting really).

 

I know what you mean, but I think it would be better to use the term "excess phase correction" instead of "phase correction".  Unfortunately, this misuse of language is part of what leads many to believe that minimum phase EQ *always* ignores time domain problems or doesn't correct time domain problems.  As I've been trying to argue here, many room acoustic problems are minimum phase in nature and can be improved upon at one or more listening locations using a minimum phase EQ filter.

 

Mathematically, any system response may be treated as two systems in series, one minimum phase and one all-pass.  An all-pass system has flat magnitude response and varies only in phase.  The all-pass part of the system is often referred to as "the excess phase part" because it represents the phase response that remains after the magnitude response is corrected using minimum phase filters.  Hence, let's talk about "improving the excess phase response" or doing "excess phase optimization" instead of just calling it "phase correction".  Another term that works for me is "mixed phase correction" because it implies that the filter used to improve performance is mixed-phase.

 

Room optimization is a very complex subject, and virtually all solutions I'm aware of that produce at least half-way decent results are commercial and proprietary.  (That doesn't mean all commercial and proprietary solutions sound even half-way decent!)  There are a lot of reasons why one room optimization solution may perform better than another.  Use of mixed-phase filters is only one distinguishing attribute.  Other things to consider are the filter resolution and the algorithm used to determine what to try to correct and what to leave alone, and I believe these latter two attributes are more important.  It's kind of a shame that filter resolution is still an issue given the capabilities of modern hardware, but manufacturers are always keen to save a few bucks here or there.  The algorithm used is probably the most important attribute if the system.  High resolution mixed phase filters are useless if the algorithm makes poor choices about how to use them.

 

With all that said, mixed phase filters can undeniably improve room response beyond what minimum phase filters are capable of in at least most situations.  How much improvement they can offer depends a lot on the room and goals.  For example, I have read that car cabins exhibit many more excess phase problems than listening rooms do, so mixed phase corrections may be much more crucial to get good performance inside of cars.  In my own room, I have noticed that adding room treatments has reduced the excess phase character considerably.  As such, with treatments added, my room is becoming more amenable to minimum phase EQ alone.  This makes a lot of sense, because you usually see non-minimum phase behavior when you have multiple reflections combine.  If the energy of reflections is reduced enough, then you will never have enough combined energy arriving late to create the zero in the response (pardon the math jargon) that causes the excess phase behavior to show up.  The room response will remain minimum phase despite the reflections because they aren't too strong.  So perhaps in general, the improvement that can be realized with mixed phase filters over minimum phase filters is much greater when the room acoustics are very poor to begin with.  Of course, in such rooms it may be very difficult to improve the excess phase aspects of the response much beyond a narrow region of space and only at the cost of performance outside that region.  Of course, that may be fine if that result is sufficient to satisfy the goals of the treatment.

[3ll3d00d]

That is what I mean yes.

 

I can see your argument (with regards to well treated rooms exhibiting fewer excess phase issues) but find it hard to reconcile that with the user community for a product like Acourate (which overwhelmingly seems to consist of boutique systems in custom built rooms). It might be that the improvement is absolutely small but relative differences are easier to detect and hence it is still a critical refinement. Impossible for me to verify that though.

[dominguez1]

Does anyone know if the popular commercial room optimization solutions like audyssey, mccac, etc implement mixed phase and minimum phase filters?

 

Just curious...

[3ll3d00d]

Does anyone know if the popular commercial room optimization solutions like audyssey, mccac, etc implement mixed phase and minimum phase filters?

 

Just curious...

audyssey, arc & jbl arcos are minimum phase

 

mccac pro & dirac are mixed (excess phase correction) 

 

RP was certainly minimum phase at one point but it's not clear whether it still is, lyngdorf are particularly backward at publishing technical data (e.g.  http://www.krksys.com/krk-ergo.html used RP and it says it is non minimum phase)

 

trinnov is all of the above and some more (afaict)

[maxmercy]

Hey folks, 

 

This thread has gotten far away from Bosso's systems.  I am starting a new thread about Bass Setup and EQ/correction in this forum.  Lets continue the setup/EQ conversation there.

 

 

http://data-bass.ipbhost.com/index.php?/topic/401-the-bass-system-setup-eqcorrection-thread/

 

 

JSS

[SME]

I have a question about the a14k.  Does it allow the power state to be triggered?  I don't see evidence for such a feature, so I assume not.  I think this feature would be useful to allow installation in more out of the way locations for noise control.  I have a large vented coat closet directly behind the wall of my front stage, and I would prefer to install it there once everything was configured.

 

I guess another option would be to leave it running all the time, but I'm not a huge fan of this idea even if the idle power draw is low.  I guess the other downside of having it installed farther away from the listening location is the inability to shut things down quickly in the event of a panic scenario.  Anyway, I'm just thinking here about how to make true ULF capability practical in my current space.

[Madaeel]

I have mine in a closet now and just turn it on when I fire everything up. Though for TV and gaming I rarely turn it on just because there isn't much bass in either unless you really crank it up. A lot of guys bought relays for their pro amps. Notnyt made a thread in AVS' DIY section so anyone could buy the parts and build a small box for one and use the 12v triggers on your AVR or prepro for it. My PS4 is in the closet though so I open the door a couple times a day anyway so it's no big deal to me.

 

I know for me unless there's strong content below 10hz I'm pretty much safe on 99% of material. You could always watch something with strong ULF content, someting like HTTYD or WotW, and set it for that and be good with most material if you need to.

 

What are you using for subs now SME?

[chrapladm]

Not much bass in gaming........WHAT ARE YOU PLAYING.?   I have noticed quite a bit of ULF on some of the games I play on PC and PS3.