Confusing phase and timing when we say that different size speakers together is bad?

[Foz]

Backwards wiring will instantly produce a 180° misalignment

Not really - a phase rotation of will offset the initial impulse in time.

If you are merely swapping the wires the initial impulse will still be time aligned... what you have changed is the polarity.


===

FWIW, I have measured in-room speaker response many times and I have seen what is referred to as 'comb-filtering' when the path length between two speakers was different or reflections caused this. It's very easy to see when the path differences are increased/decreased and/or reflections are minimized.

Anyone who has been in a room with reflective boundaries and a single sound source and noticed that the nature of the sound changes as they move around has had a similar experience - which is to say everyone. Every reflection from every surface is a virtual sound source and they all interfere. Silly people are always trying to dumb down the reality of how waves propagate in rooms - it changes nothing and leads to a lot of false understandings.

Sound propagation is a 3d phenomena dammit! Below the Schroeder limit sound propagation FROM A SINGLE SOURCE in a room and the constructive and destructive interference patterns created by its reflection are like a series of bubbles that add and subtract from each other creating awesomely complex patterns as they bounce around - like this...

https://www.youtube.com/watch?v=cl2RKmgTMzk

Adding additional sound sources make the pattern even more complex... and that pattern is not a filter and it doesn't look like a comb and it aint helpful to pretend otherwise - it just makes everyone dumber.

 

[Foz]

like they say, "the truth will set you free, but first it will piss you off"

LOL

 

[1958Bassman]

I am watching a video of Alice in Chains live at the Maquinaria Festival in 2011 and they have a set up guaranteed to cause phase issues. Both the guitar and bass players have one row of speaker cabinets on the stage and then another row of cabinets set above and several feet behind the bottom row. The frequency with half a wavelength of that offset difference and its octaves will be lost on the stage. It appears that the speakers are about 1.5m apart in depth and that would be F# if so. Luckily, they all use IEMs and hopefully most of what they hear comes from them.

What makes you think they actually use all of those cabinets? Even if they have a mic in front of each one, they do that for show, not the mix.

 

[1958Bassman]

He is?
901, worst piece of c... ever.

Yeah, he is. When they refer to him as Dr. Amar Bose, it's because he has a PhD from MIT.

http://en.wikipedia.org/wiki/Amar_Bose

 

[JimmyM]

Adding additional sound sources make the pattern even more complex... and that pattern is not a filter and it doesn't look like a comb and it aint helpful to pretend otherwise - it just makes everyone dumber.

So what do we call it? "Power alley creation?"

 

[1958Bassman]

Not really - a phase rotation of will offset the initial impulse in time.

If you are merely swapping the wires the initial impulse will still be time aligned... what you have changed is the polarity.


===

Anyone who has been in a room with reflective boundaries and a single sound source and noticed that the nature of the sound changes as they move around has had a similar experience - which is to say everyone. Every reflection from every surface is a virtual sound source and they all interfere. Silly people are always trying to dumb down the reality of how waves propagate in rooms - it changes nothing and leads to a lot of false understandings.

Sound propagation is a 3d phenomena dammit! Below the Schroeder limit sound propagation FROM A SINGLE SOURCE in a room and the constructive and destructive interference patterns created by its reflection are like a series of bubbles that add and subtract from each other creating awesomely complex patterns as they bounce around - like this...

https://www.youtube.com/watch?v=cl2RKmgTMzk

Adding additional sound sources make the pattern even more complex... and that pattern is not a filter and it doesn't look like a comb and it aint helpful to pretend otherwise - it just makes everyone dumber.

All of this is the reason reverberant rooms usually sound like crap. The easy way for people to see a representation of sound is by using a pan of water, then dipping their finger in to see the waves as they move out from that spot and how they reflect. Then, dip another finger at the same time, the same distance from the side, like a symmetrical speaker setup in a room.

 

[Downunderwonder]

To start, I'm sure this topic has been discussed ad nauseum, and this is in no way an attempt to troll. I'm pretty new to this forum, and have to say in the short time I have been here I have learned and gained a PLETHORA of knowledge that I cannot thank everyone here enough for. This post is merely an attempt to continue that trend and to educate myself even further.

Too much new info for you?

 

[T-Bird]

Hi.

To be honest, I never noticed anything wrong with my 410 + 115 setup until I came here

The key word here being You.

Near-field (ie. on stage), the problems aren't usually that pronounced. On the contrary really, those very problems may reduce mic-bleed on stage and therefore might even be considered as an improvement.

The audience is the party noticing the non-uniform sound.
Well, perhaps not actually noticing it's the bass , but they sure can tell that something's amiss even though a non-acoustical engineer type may not be able to point their finger at the source.


And this quote from skychiefis a prime example of that. Even more so as the cab combination in question is one of the "approved" ones.

Saw some poor guy a couple of weeks ago at a local club with a 210 or 212 sitting atop a 115. I could tell he was a decent player, but the sound? What a mess. I felt compelled to say something to him, but didnt want to come off as being a know-it-all, so i just kept my mouth shut.

The worse problem than the audience noticing directly, may be the sound-person if he/she doesn't wander around the venue for a reason or another and gets either the wave crest or wave through of the combined bass-instrument signals to the FOH console location and as a result reduces or increases the bass level drastically in FOH.

But you cannot acomplish this trick with mids.

Sure You can, cardioid arrangements work well on mid frequencies as well.
No reason why they wouldn't.

Since controlled mid frequency directivity is usually easier and cheaper to accomplish by other means though, cardioid mids are rather rare outside Hi-Fi circles.

Regards
Sam

 

[SGD Lutherie]

Phase:

What You describe as phase, sounds more like a polarity, ie. 180 degrees apart.

Yes, what we call "phase" is usually referring to polarity.

Timing:

OK, real phase is a dime domain affect. If you make two copies of a waveform, and delay one of them, you will get comb filtering as the now mis timed waves interfere with each other. So you get peaks and valleys. This is the effect known as flanging.

With speakers the first thing you don't want is the polarity being wrong. That will remove a lot of low end, just as having the two pickups on your bass "out-of-phase" will. And they aren't really out-of-phase, they have opposite polarity.

So the idea is that if parts of the audio spectrum are coming from two different drivers, you want them to be time aligned so they get to your ear at the same time, as if they were coming from a single full range driver. Otherwise you might get some comb filtering.

Being that most 2 and 3 way loud speakers are not time aligned, it obviously isn't super critical. But it might improve the sound of the speaker system.

The BBE Sonic Maximizer works by delaying the low frequencies in time to phase correct them. They state that the high frequencies often reach your ears first.

 

[Vince Klortho]

So what do we call it? "Power alley creation?"

It is known as wave interference.

 

[fdeck]

 

[Vince Klortho]

Being that most 2 and 3 way loud speakers are not time aligned, it obviously isn't super critical. But it might improve the sound of the speaker system.

No, it is not super critical, but it definitely can improve the sound. The vast majority of people can't tell the difference without A-B comparison. Some of the fancy active crossovers for PA use have time alignment capabilities and it can really work wonders. When I was working in a sound crew we had our system set up for accurate time alignment and I remember pushing the button that enabled it in and out and the effect was startling. Unaligned, the system sounded really good but when it was time-aligned the high end became much clearer. The effect on the low end up to mids was not noticable and this is a known psycho-acoustic phenomenon. This was a festival-sized PA with about 20KW of amplification and the board was about 60 yards away from the stage.

In smaller scale systems and venues it is usually not noticable. I remember a few home stereo speakers advertised time-aligned mounting of their drivers a while ago and it didn't seem to make much of a difference to many people. As I mentioned, unless you hear an A-B comparison virtually nobody can tell the difference.

 

[SGD Lutherie]

No, it is not super critical, but it definitely can improve the sound. The vast majority of people can't tell the difference without A-B comparison. Some of the fancy active crossovers for PA use have time alignment capabilities and it can really work wonders. When I was working in a sound crew we had our system set up for accurate time alignment and I remember pushing the button that enabled it in and out and the effect was startling. Unaligned, the system sounded really good but when it was time-aligned the high end became much clearer. The effect on the low end up to mids was not noticable and this is a known psycho-acoustic phenomenon. This was a festival-sized PA with about 20KW of amplification and the board was about 60 yards away from the stage.

In smaller scale systems and venues it is usually not noticable. I remember a few home stereo speakers advertised time-aligned mounting of their drivers a while ago and it didn't seem to make much of a difference to many people. As I mentioned, unless you hear an A-B comparison virtually nobody can tell the difference.

That was my thinking too. A/B it and you will hear it. I know the first time I heard a Sonic Maximizer back in the 80s I was quite shocked at the improvement. And it was with a set of Bose speakers, so it was a full range system.

I suppose with a home stereo system you would have to be sitting in a good spot to hear it well.

 

[JimmyM]

It is known as wave interference.

I like that. I approve.

 

[Stumbo]

To illustrate what happens when you put e.g. a 15"woofer and an 8" fullrange driver in one cab without crossover.

Blue= 15"
Green= 8"
Black= summed response

As you can see, the summed frequency-response has a severe dip around 2khz. This is because of the different frequency/phase responses of both drivers.
In picture two you see the phaseresponse(degrees y-axis, frequency x-axis) of both drivers. As you can see the phase between these two drivers already starts to shift at 400hz and is very severe around 2khz.

Hope this explains a bit why using different drivers causes phase issues. To make it even more complicated, using multiple of the same drivers in a cab causes phase issues outside the main listening angle....(mostly in the mid/high frequencies, depends on the distance between the drivers, can't explain it all here...)

Note that the phenomena doesn't look anything like a comb, and consider that the described device isn't a filter [that term implies intent].

Ah......yes it does.

Look at the second graph...the spikes look somewhat like the teeth of a comb, yes?

I'm just sayin'....

 

[walterw]

all glory to the hypnotoad!

 

[Stumbo]

all glory to the hypnotoad!

Rod Serling around?

 

[SGD Lutherie]

The term "comb filtering" is not a term of art in physics. Whenever you see it used you should consider the related statements are probably dumbed down from the explicit truth - perhaps with good reason taking into account the audience - but perhaps not; the term is often used to present concepts that just flat wrong. Interference patterns are generally a lot more complex than a comb shape - and such generally aren't considered filters.

It's a perfectly valid term used in acoustics.

http://en.wikipedia.org/wiki/Comb_filter

In acoustics, comb filtering can arise in some unwanted ways. For instance, when two loudspeakers are playing the same signal at different distances from the listener, there is a comb filtering effect on the signal.[1] In any enclosed space, listeners hear a mixture of direct sound and reflected sound. Because the reflected sound takes a longer path, it constitutes a delayed version of the direct sound and a comb filter is created where the two combine at the listener

No, they don't look even like a comb, but who said they would? It's frequency dependent. That's the difference between flanging and phase shifting. Phase shifting (all pass filter) has very evenly spaced notches. Time delayed flanging doesn't. Microphone and speaker interference looks like a time delay comb filter.

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So it's not a literal meaning, but it is commonly used.

More references in pro audio:

http://www.soundonsound.com/sos/feb08/articles/qa0208_5.htm

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This last article points out that it's normal to hear comb filtering.

Invalid Link Removed

How Does the Brain Deal With Comb Effects?
The precedence effect (previously known as the Haas Effect) dictates that our brain and ears pick out the location of a sound source that reaches our ears in the first few milliseconds of a sound’s arrival. The first sound to arrive at the ears enables you to determine the direction of the source. After hearing an initial signal, the brain will suppress any later-arriving signals, up to about 30 milliseconds. These later-arriving signals that show up with steady-state pink noise (within the 30-millisecond window) do not disrupt the brain’s precise localization mechanism. What occurs is that you do not “hear” the contributions of the later-arriving sounds from the adjacent drivers that are responsible for the measurement artifact of comb filtering. Or rather, your brain hears and processes them but disregards them lest they confuse our directional acuity; in fact all they do in the listener is create a sense of added spaciousness. Numerous scientific researchers, including definitive experiments conducted by Dr. Floyd Toole and Dr. Sean Olive, have verified this. Even in a room having lots of reflections, our brains correctly determine the direction of sounds. (By the way, sounds arriving at our ears after a delay of more than 30 milliseconds are perceived as a second sound or echo.)
Critics of comb filtering who believe it to be a big issue in speaker design have the option of listening in mono to avoid the comb filtering. But we all much prefer listening to music and vocalists in stereo–it’s far more spacious and realistic–and the reason is that our brains and two ears simply ignore those canceling signals that on paper show up with a test signal and a single microphone.

Having a speaker cab for an instrument that has its own tone is not necessarily bad, if that's what you want. Most bass cabs are like this. Personally I prefer more accurate cabs.

 

[Vince Klortho]

Sorry for the tangent but this reminds me of something that used to be quite controversial. I learned about it because I am an EE and I read about this in the release notes of a high frequency D/A chip many years ago.

When the first CD players and discs came out some people claimed they sounded really bad. Most complaints were rather vague about why but were, never the less, quite vehement. When the first digital recording technology came out the same complaints were made. Most complaints I read were that it sounded harsh. I remember several musicians refused to use the early digital recording technology like Frank Zappa. Then I saw some phase response charts and the reasons became clear.

Nyquist's theorem states that for a signal to be reproduced accurately it must be sampled at a frequency that is at least twice as high as the signal's frequency. CDs were designed to reproduce up to 20KHz so the sampling rate was chosen to be 44.1KHz. Also, according to the theorem, signals above half of the sampling rate will be inaccurate or "aliased." For this reason, complex filters were used in the early CD players that had a very steep cutoff at 20KHz. A side effect of these steep filters was really weird phase response at high frequencies. Seriously, the phase response was all over the place starting at about 8KHz. This was the cause of the complaints that were made.

The release notes of the D/A chip I mentioned described how this problem has been solved. Now the digital signal of the CD is usually "supersampled" at 8X the base rate or 352.8KHz and a simple second order filter is used to eliminate the aliased signals. This allows for the signal to be accurately reproduced with the correct phase.

 

[SGD Lutherie]

When the first CD players and discs came out some people claimed they sounded really bad.

It was also because they were just taking the master tapes that were mastered for vinyl and just transferring them to digital.