Jump to content

Variable Time Delay


Kevin Ross

Recommended Posts

Hi,

 

 

 

I’m wondering if anyone has ever done this before. You know if you had 2(or more) subs and you were to try to delay one to the other it would always be a compromise, if it was right in some areas it would be wrong in others and cause destructive interference . Well as bass notes tend to be quite long would it be possible to build a delay line that you could set to constantly vary the delay from say X to Y over time period Z. in a saw tooth pattern so change every ms=(X-Y)/Z it would loop like this

 

 

 

Interval=(Lower_delay-Upper_delay)/Time_per_Sweep

 

 

 

Do

 

Do Until Delay=>Upper_delay

 

Delay=Delay+Interval

 

Loop

 

 

 

Do Until Delay <= Lower_delay

 

Delay=Delay-Interval

 

Loop

 

Loop

 

 

 

I wonder if this would smooth out the destructive interference that is caused when using more than one sub array.

 

 

 

Or am I just talking rubbish and it would not help anyone?!?!?

Link to comment
Share on other sites

What you're referring to might be known as a trombone line - a similar device is used in RF engineering to present variable impedances.

 

Other than introducing a doppler modulation to the sound, I don't see what else it will achieve.

Link to comment
Share on other sites

What you're referring to might be known as a trombone line - a similar device is used in RF engineering to present variable impedances.

 

Other than introducing a doppler modulation to the sound, I don't see what else it will achieve.

 

Accepted that there would be a slight Doppler shift in the sound but if the difference between to upper and lower delays was small (maybe 10ms) would you notice? I guess I’m trying to change the point at which the two wave forms combine constructively and destructively so instead of being a fixed point it is constantly moving so the ear would average out the results and achieve a balanced bass sound

Link to comment
Share on other sites

If you have lots of subs, and in several stacks, I tend to time them with the main PA in an arc, but also diffusely stack the subs (is not in an uber block) so that each subs output reaches each seat at a slighty different time, thus creating a smoother response around the room, and alieviating the distructive filtering you get from room modes with a single, or dual stack of subs, though, if times short I'll just lob them in a block still.

 

I always delay the subs the crossover frequency with the main PA anyway, so any time/phase differences inherent in the box will not matter, as that will be the behaviour of just the one type of speaker, and won't be the summation of sub/low mid for instance. Variable time delay would play merry hell with your phase response, and probably sound like a chavs Nova with a boom box in the boot.

 

Edit:

 

In response to your comment on noticing 10ms, the human ear is massivly sensitive to minute phase/timing issues (it's how you pinpoint a noise sources position exactly useing the time beween your ears) so 10ms, very noticable, I tend to get bothered by anything less than 0.5ms out of time. Unless your useing haas to your advantage, but that is a completely different kettle of squid

Link to comment
Share on other sites

I guess I’m trying to change the point at which the two wave forms combine constructively and destructively so instead of being a fixed point it is constantly moving so the ear would average out the results and achieve a balanced bass sound

 

Firstly, there may be both coherent and non coherent interfefence. The coherent type is the one that either cancels the pressure component or doubles it (0dB or 6dB) at that specific point. If the two sources are non coherent they will always add.

 

Secondly, you are proposing a large, low frequency "phaser" guitar effect.... It's one thing to have poor coverage due to low frequency summation and cancellation. It's quite another to have that continuously changing during the event!

 

Why not investigate cardioid sub arrays to provide improved coverage?

 

Simon

Link to comment
Share on other sites

Firstly, there may be both coherent and non coherent interfefence. The coherent type is the one that either cancels the pressure component or doubles it (0dB or 6dB) at that specific point. If the two sources are non coherent they will always add.

 

:)

At up to +-120 degrees phase shift (time delay), sources will always add constructively.

At 120 degrees, no addition or cancellation occurs.

At >+-120 degrees, cancellation occurs, with the deepest cancellation obviously being at 180 degrees.

Link to comment
Share on other sites

Shez,

 

To add correlated sounds that are in-phase we just add the waveforms, as the pressure will simply be a sum of the individual pressures as no cancellation will occur:

PT(t) = P1(t) + P2(t)… + Pn(t)

 

However, for correlated sounds with different phases we need to do the following:

First consider the pressure of a sine wave at different points. This will be the peak pressure multiplied by a factor which takes into account the changing of the wave, i.e. a sine term:

Pat a point = Psound amplitude sin(360ft)

Where

f = frequency of the wave (Hz)

t = time of point along wave (s)

 

If we know the delay time introduced (say from a reflection) then:

Ptotal = Pdelayed sin(360 f (t+ delayt)) + Pundelayed sin(360ft)

 

This is what you've described - and you are perferctly correct - as long as the sources are coherent.

 

 

For uncorrelated sounds, we do not know the time or phase relationship, therefore to add non correlated sounds we would convert each individual SPL into its equivalent pressure, add the square of the pressures, then take the square root to arrive at the total pressure. We can then convert that to the total SPL.

 

P(total uncorrelated) = Sqrt[(P1^2)+(P2^2)+Pn^2)]

 

Here, uncorrelated soures always show an increase in level as phase and cancellation are not considered. Examples might inlcude addition from late, delayed reflections or sound from a number of different sound sources.

 

I cannot paste my proper equations in this box, so here's a link to another page that doesn't require a student login.

 

Hope this helps...

 

Simon

Link to comment
Share on other sites

Some excellent theories going on here.

 

 

 

I would just like to say that I never thought of this idea to solve a problem of low frequency coverage as I would just use one of the other more proven techniques, but I did what to see what would happen if you think “out side the box”. Could this ever be done? Has this ever been done? and I guess more to the point should this ever be done? If we change our delay time was varied by say +-10ms over a 20ms window would the human ear notice or would it just sum up what it is hearing in that 20ms window to give an average response

Link to comment
Share on other sites

Presumably in this application, the sources will be coherent as subs are almost invariably run in mono?

 

Sound from the subs themselves may well be coherent, but the reflected sound in the room may not be. That's one of the reasons why it's unlikely for someone to hear perfect cancellaton at a given point unless they are in the direct sound field and a single note is playing. Even then, moving your head from side to side tends to reveal areas of increased and decreased sound, rather than doubling and complete cancellation.

I should have highlighted in my first post that 0 degrees and 180 degrees were the two extremes - you've kindly filed in what happens at phase angles in between.

 

 

With regards to the OP, he was trying to achieve more precise delay alignment. Although it's not possible to provide exactly the correct delay time for all listening positions, careful exploitation of both the Haas and Precedence Effect means that there is some leeway available.

 

The reason I suggested cardioid sub arrays was that it is possible to achieve a directional pattern which improves the amount of forward radiated energy, whilst reducing the amount of rear radiated energy. This can give better coverage for a large crowd, reduce environmental noise impact (from the rear!) and reduce or eliminate the "power alley" efftect.

 

HTH,

 

Simon

Link to comment
Share on other sites

If you look at the Haas curve, then 20ms is getting towards the point where two sources can be heard as distinct sounds. I think a variable delay at 50 Hz would be very noticeable from a source location viewpoint.
This thread on PSW is about cardioid sub placement and delays
Phil, Thanks for linking to that - I had lost it and couldn't find the thread! If anyone wishes to learn more about cardioid subs, then go to one of Meyer's live sound seminars - they are very keen on them, plus you can model it for free in MAPP!Simon
Link to comment
Share on other sites

Archived

This topic is now archived and is closed to further replies.

×
×
  • Create New...

Important Information

We have placed cookies on your device to help make this website better. You can adjust your cookie settings, otherwise we'll assume you're okay to continue.