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[Yoel Klein]

Hi,
The more I learn the more I find my self knowing so little and knowledge hungry
Lately I’m messing around with Danley Products, and in order to fully utilize the Danley experience, I need to take my knowledge to the next level.  Gone are does days of using a 1642 mixer into a Crown 24 or 36, out to a set of JBL’s or EAW’s slap a sub in the middle, connection via a rane crossover, raise the master fader and your all set!

Then sit at your console and start tweaking the inputs, and “make the best out of it” using my knowledge as a studio engineer.

Now days, With a Danley DNA20K, and a SQ digital console, I have to properly align the speakers, and understanding FIR filters, slopes, phase issues, delay fill speakers, don’t use graphic EQ because it messes up the phase, use instead parametric EQ,  and my head is spinning!!

Ok, I get it, when I use a filter it will change the timing thus creating noice cancelation in certain areas of the music. But why does it change the timing?  The 80hz cut off on the old mixers were also messing with the phase? Why would a 24db slope per octave be worse from a 48db slope?  I need a fundamental lesson or two understanding this all. I would really appreciate if someone can point me to the right direction.

Thanks a ton!

 

[Caleb Dueck]

Hi,
The more I learn the more I find my self knowing so little and knowledge hungry
Lately I’m messing around with Danley Products, and in order to fully utilize the Danley experience, I need to take my knowledge to the next level.  Gone are does days of using a 1642 mixer into a Crown 24 or 36, out to a set of JBL’s or EAW’s slap a sub in the middle, connection via a rane crossover, raise the master fader and your all set!

Then sit at your console and start tweaking the inputs, and “make the best out of it” using my knowledge as a studio engineer.

Now days, With a Danley DNA20K, and a SQ digital console, I have to properly align the speakers, and understanding FIR filters, slopes, phase issues, delay fill speakers, don’t use graphic EQ because it messes up the phase, use instead parametric EQ,  and my head is spinning!!

Ok, I get it, when I use a filter it will change the timing thus creating noice cancelation in certain areas of the music. But why does it change the timing?  The 80hz cut off on the old mixers were also messing with the phase? Why would a 24db slope per octave be worse from a 48db slope?  I need a fundamental lesson or two understanding this all. I would really appreciate if someone can point me to the right direction.

Thanks a ton!

Find a graph of phase and another of group delay, of the same system.  Same info, different visual formats, much easier to wrap your head around the concept. 

I just saw this while reading the Production Partner review of the TW Audio T24N. 

[Tim McCulloch]

Hi,
The more I learn the more I find my self knowing so little and knowledge hungry
Lately I’m messing around with Danley Products, and in order to fully utilize the Danley experience, I need to take my knowledge to the next level.  Gone are does days of using a 1642 mixer into a Crown 24 or 36, out to a set of JBL’s or EAW’s slap a sub in the middle, connection via a rane crossover, raise the master fader and your all set!

Then sit at your console and start tweaking the inputs, and “make the best out of it” using my knowledge as a studio engineer.

Now days, With a Danley DNA20K, and a SQ digital console, I have to properly align the speakers, and understanding FIR filters, slopes, phase issues, delay fill speakers, don’t use graphic EQ because it messes up the phase, use instead parametric EQ,  and my head is spinning!!

Ok, I get it, when I use a filter it will change the timing thus creating noice cancelation in certain areas of the music. But why does it change the timing?  The 80hz cut off on the old mixers were also messing with the phase? Why would a 24db slope per octave be worse from a 48db slope?  I need a fundamental lesson or two understanding this all. I would really appreciate if someone can point me to the right direction.

Thanks a ton!

Hi Yoel-

Everything is about time for our purposes.  Because frequency (time) is a cyclical process it accumulates delay.  When a delayed signal and otherwise identical un-delayed signal are mixed together, cancellations and and additions (infinite and +6dB, respectively) occur at harmonic multiples of the delay time.  Because frequencies have a physical wavelength, time can also measure distance.

Filters are made up of "poles" and each pole requires some rotating around the ol' phase wheel (so search), often more than once.  That takes time in the chronological sense.  The lower the frequency and more poles to the filter, the longer it takes... in the time it takes to go around the phase wheel 360° at 80Hz, an 800Hz wave makes 100 phase rotations, or 36,000°.

See also: summation of sine waves and phase.

[Doug Fowler]

Find a graph of phase and another of group delay, of the same system.  Same info, different visual formats, much easier to wrap your head around the concept. 

I just saw this while reading the Production Partner review of the TW Audio T24N.

Group delay display gets dissed quite a bit but I agree it's much easier to read. Unstable with any noise on the line yes, but with a lot of averaging it's fine in my experience. 

[Ivan Beaver]

Hi,
The more I learn the more I find my self knowing so little and knowledge hungry
Lately I’m messing around with Danley Products, and in order to fully utilize the Danley experience, I need to take my knowledge to the next level.  Gone are does days of using a 1642 mixer into a Crown 24 or 36, out to a set of JBL’s or EAW’s slap a sub in the middle, connection via a rane crossover, raise the master fader and your all set!

Then sit at your console and start tweaking the inputs, and “make the best out of it” using my knowledge as a studio engineer.

Now days, With a Danley DNA20K, and a SQ digital console, I have to properly align the speakers, and understanding FIR filters, slopes, phase issues, delay fill speakers, don’t use graphic EQ because it messes up the phase, use instead parametric EQ,  and my head is spinning!!

Ok, I get it, when I use a filter it will change the timing thus creating noice cancelation in certain areas of the music. But why does it change the timing?  The 80hz cut off on the old mixers were also messing with the phase? Why would a 24db slope per octave be worse from a 48db slope?  I need a fundamental lesson or two understanding this all. I would really appreciate if someone can point me to the right direction.

Thanks a ton!

To answer a few questions (yes there are tons more).

ANY eq (except FIR) parametric, passive, graphic etc will produce a phase shift.

HOWEVER, this DOES NOT mean it is a bad thing.  When that eq is used to "fix" an anomaly in the amplitude response, that is not part of a time issue (meaning around xover points) , it also fixes the phase anomaly as well.

So the associated phase shift is a GOOD thing.  Not all eq is bad-despite what some may think.

The steeper the slope, the greater the phase shift.

One of the things that Danley does in some of the products is to use eq to adjust out of band phase response.  This results in a flatter phase response, even though it has very little effect on the overall amplitude response.

The real problem is that the more you know, the more you realize how little you actually know.  It just gets worse and worse, trust me.

But NEVER stop learning.

The best way to get a handle on what is happening with the phase is to actually look at it, via a real measurement system.  Smaart is one popular one, but REW is free.

You don't need to measure a loudspeaker, in fact it is best not to, for learning purposes.

Just hook the system up to the input and output of a eq or processor.  Then look at the amplitude and phase response as you make various adjustments.  It is not important what they are, this is just to give you an idea of what is happening.  Do it at low freq and high freq, boosting and cutting, eq and bandpass.

Often the best education is what you give yourself, as it tends to stick around longer.

[Mal Brown]

"Just hook the system up to the input and output of a eq or processor.  Then look at the amplitude and phase response as you make various adjustments.  It is not important what they are, this is just to give you an idea of what is happening.  Do it at low freq and high freq, boosting and cutting, eq and bandpass."

Using what tool(s)  to measure / display ?  Is this Smaart and a measurement mic ?

[Chris Grimshaw]

Use REW. It's free and works really well.

Chris

[Ivan Beaver]

"Just hook the system up to the input and output of a eq or processor.  Then look at the amplitude and phase response as you make various adjustments.  It is not important what they are, this is just to give you an idea of what is happening.  Do it at low freq and high freq, boosting and cutting, eq and bandpass."

Using what tool(s)  to measure / display ?  Is this Smaart and a measurement mic ?

If you read the lines above, I say it is best not to use a loudspeaker, so just hook the system to the input and output of a processor and give ideas on a system.

YES, there is a learning curve on all measurement systems, it is NOT something you download and then measure a few minutes later.  You must read and learn the particular system and how it works.

[Luke Geis]

It gets rather tricky when you start trying to factor phase in a quantum manner. It is very difficult to visualize in the real world. The easy way to wrap your head around it is to learn the crossover slopes and the number of poles ( orders ) or filters they use.

1. 6db slope filters are single-pole ( 1st order ) and have a 6db per octave roll-off. They induce a -90* phase shift on the out of band media.

2. 12db slope filters are 2 pole ( 2nd order ) and have a 12db per octave roll-off. They induce a -180* phase shift on the out of band media.

3. 18db slope filters are 3 pole ( 3rd order ) and have an 18db per octave roll-off. They induce a -270* phase shift on the out of band media.

4. 24db slope filters are 4 pole ( 4th order ) and have a 24db per octave roll-off. They induce a -360* phase shift to the out of band media. This is the most popular filter type for that reason.

5. The next typical filter is a 48db filter and is an 8 pole ( 8th order ) filter. It introduces a -720* phase shift to the out of band media.

As you can visualize, the steeper the slope the more phase delay there is. The 24db filter is the most popular because it is obviously the easiest to visualize and figure out how to align in time in the real world. At -360* if you know the wavelength, you can almost just about use a tape measure and figure out how much to either place the speakers, or apply the needed delay to the mains ( in most cases ).

All pass filters also help greatly with helping to align phase. If you were to apply a single order all-pass filter to the mains you would introduce a -180* phase shift to all the media. So stacking two single order all-pass filters would then provide a -360* phase shift to the mains. This would, in theory, allow the mains and the subs to align without using delay ( physical or electrical ) assuming the mains and subs drivers were in line. While it is of course not this simple to actually employ, the only way to truly get alignment is to use measurement software that can detect the phase so can use the appropriate filter type, electronic/physical delay and or all-pass filter combination.

In general, the above information is a way to start wrapping your head around it. As with anything in audio, it is more complicated than it needs to be if we allow it. In order to even get to bothering yourself with phase, you really need a way to detect it. Smaart, and a couple other programs allow that ability. Otherwise it requires math and careful measurement to only get it in the ballpark.

[Mal Brown]

If you read the lines above, I say it is best not to use a loudspeaker, so just hook the system to the input and output of a processor and give ideas on a system.

YES, there is a learning curve on all measurement systems, it is NOT something you download and then measure a few minutes later.  You must read and learn the particular system and how it works.

Sorry, I'm sure I'm being dense but I am curious as to what tool(s) I would use to perform the measure. Chris mentions REW.  That runs on a PC as I recall.  In my case a laptop.  So I have the XLR outputs of my system.  Those go to some external converter and then to the PC and REW?  What is that external converter ?   (off to google.)

[ProSoundWeb Community]