Quick question for John K. On the NaO with the rear mounted tweeter, was it wired in phase(bipole) or out of phase(dipole) to match the rear output of your mids? And did did you pad down the tweeters output compared to the front? Any other thoughts on the matter would be appreciated, I'm working on a similar design idea with my first open back speaker. Thanks.
mige0 said:
The EQ I used for the drivers in the response shaping was done using SoundEasy's digital equalizer. It works by dividing the target response curve by the measured driver response to find the transfer function for the filter. Then the PC processes the whole thing for play back. With the correct sound card up to a stereo 5-way system digitally.
Caferacer said:
I do connect it out of phase with the front and I have the option to switch it on and off. It's not really padded down. Also, it doesn't make a big difference if it's in or out of phase except right around the crossover point. The separation between the front and rear tweeter is such that the front are rear tweeters are uncorrelated which means the operate more like two independent sources.
Minimum phase???
Michael,
I have created a (VERY) short introduction to the difference between minimum and non-minimum phase systems. Please have a look at my five-minute introduction to the minimum-phase concept for a two-way speaker.
Best Regards,
Edward
mige0 said:
Michael,
I have created a (VERY) short introduction to the difference between minimum and non-minimum phase systems. Please have a look at my five-minute introduction to the minimum-phase concept for a two-way speaker.
Best Regards,
Edward
Re: Minimum phase???
Hi
Edward, thanks a lot for your extra work. I'll try to digest it the right way.
I alway tried to visualise " minimum phase " with somthing like " time coherent ". Though I was close its obviously not exactly what describes it completely.
Greetings
Michael
Hi
EdwardWest said:
Edward, thanks a lot for your extra work. I'll try to digest it the right way.
I alway tried to visualise " minimum phase " with somthing like " time coherent ". Though I was close its obviously not exactly what describes it completely.
Greetings
Michael
Hi
JohnK, would you say that a correction of a Q=20 peak is possible that way ( I am aware but don't care of the limitations sound wise in this case ) ?
Did you have to process the wav file by " SoundEasy " and than play it or is it a convolution file that is computed and then can be used with a plugin like SIR_1011 ?
IIRC you mentioned somewhere at the start of this giant thread that you also use a computer based crossover. Is it somthing like this
http://www.thuneau.com/allocator.htm
?
Greetings
Michael
john k... said:
JohnK, would you say that a correction of a Q=20 peak is possible that way ( I am aware but don't care of the limitations sound wise in this case ) ?
Did you have to process the wav file by " SoundEasy " and than play it or is it a convolution file that is computed and then can be used with a plugin like SIR_1011 ?
IIRC you mentioned somewhere at the start of this giant thread that you also use a computer based crossover. Is it somthing like this
http://www.thuneau.com/allocator.htm
?
Greetings
Michael
mige0 said:
SoundEasy does it all. I doesn't create wave files. It generates the required transfer function and emulates then digitally using FIR convolution. Please don't confuse FIR convolution with linear phase crossovers as is commonly done by many. FIR can do any type of causal filter, linear phase, minimum phase, arbitrary phase.
gedlee said:
I bet I'm not the only one trolling this thread that does not fully appreciate the logic of what you are saying. In an electrical system, I can create any arbitrary average DC value I like with a half sine, saw tooth, or whatever. Why would it be impossible to to reflect this acoustically, assuming the room was sealed? Surely the air in the room is compressed for duration t of the impulse, and to amplitude A You can't really do this perfectly, but if the impulse causes the cone to move from rest to position +A, and back to rest, rather than to -A before returning to rest, why would the average area under the curve be zero? I appreciate that bandwidth limitations on the top end will result in undershoot, and on the low end prevent compresion of the air for any usable time, but at the end of the day, didn't we really compress the air in the room briefly? For a continuous function, what you say makes sense, but an impulse is not a continuous function. I am missing something.
Dick
dickmorgan22 said:
Acoustics is NOT electronics - people really need to grasp this as so many misconceptions come from making this mistake.
Lets discuss a free field condition for now since if you don't follow this you would not follow the closed room argument as it is far more complex.
In a free field no acoustic source is large enough to change the static pressure of the planet - nothing short of an atomic bomb which may be able to do that, but your three way won't. Thus no source generating an acoustic wave can net out a change in the static pressure. Thus as the impulse passes the static pressure must return to zero AND the average of that impulse must also go to zero - the DC response.
I explained this before.
There is also a similar argument for any "real" room and the situation is not much different even for a completely closed room - which no human could listen in.
The examples which take out the minimum phase, etc. are all bogus because that is a manipulation just as much as using a pencil. Either the response is correctly done and unmodified or it isn't. If it does not average to zero then one or the other must have occured.
Lynn Olson said:
LOL!
I'm glad I'm not the only one. The longest period of no recorded music for me was 10 years. Just could not stomach it. No CD, no LP, just a clock radio for the news. It was the change to working alone on the night shift in my present job that got me back into "audio." Just wanted to listen to music to pass the hours. Still struggling with it, but having fun.Lynn Olson said:
Oddly enough, one of the best systems I ever heard was purely mechanical - recording and playback. An Edison Disk system. Not cylinder, disk. It was at the little "Museum of Technology" in Palo Alto, California. The player looked like a deluxe version of the typical Victrola, but used Edison's vertical needle movement on a 12" disk.
The owner of the museum saw that I was interested and asked if I cared to listen to it. "Of course!" Well, I was floored. The sound was astonishing.
Not Hi-Fi, no - there was surface noise and limited bandwidth, but it was so lifelike, pure, so good.As the disk played it drew in people from all over the museum. They all had to come see what sounded so good, find the source of that great sound. And they all commented on how good it sounded. "What IS that thing?" was a common refrain.
Don't even remember what the record was. A man singing with a small orchestra behind, IIRC. The amazing thing was how real, how good, how musical this old machine and the purely mechanical recording sounded. How have we strayed so far?
soongsc said:
I understand what you are saying, and I agree it covers most part of the signal regeneration. Preshaping a signal is a very common technique used in control, and I understand that we are not talking about feedback. However, we must realize that presure is generated from different parts of the surface and different time throughout the signal variation.
Well thinking about direct sound, at the first cone mode, I'm visualizing in the VC moving first, part of surface close to where VC and Cone connect move a bit later, then finally the outer edge of the cone follows. Throughout this first 1/4 cycle is the onset, if compensation prevents this part from being faithfully reproduced, then the sound will not be as good as a more well behaved cone. I won't be upset if I cannot let you understand this part.
Now if we wanted to do somthing really exotic using real=time software technology to it's extreme, in might be possible to model the energy storage charateristics of a cone, and depending on what the previous state and new signal state is, more adequately compensate for cone defficiencies. But I don't think anyone is going to spend millions on a project like that.
How the response is generated isn't important. You really caught up in that and it doesn't matter. What matters is what is: a) the characteristic of the response at the the observation point and b) is the generator (predominately) a linear system?
If the generator is a linear system, which drivers basically are (they better be), then we have two options when observing the response at some point. Since we are at some distance from the generator the options are:
1) The response at the observation point is minimum phase plus a linear phase component due to the propagation delay. This is the case if the driver response is purely minimum phase.
2) The response at the observation point is some arbitrary non minimum phase response plus a linear phase component due to the propagation delay. This would be the case, for example, if breakup wasn't minimum phase.
Now, any response can be decomposed into a minimum phase component and an all pass component. That is just mathematics. In case 1 the all pass component is just the linear phase resulting from the propagation delay. In case 2 the all pass component is the propagation delay pulse the deviation of the driver from minimum phase.
I assume we agree that we can forget about the propagation delay, or let us say that at the observation point we want to match our minimum phase target plus the propagation delay. In case 1 all we need do is apply minimum phase equalization and the response at the observation point will then match the target response. In case two all we need to is apply minimum phase eq and, if we add an all pass correction for the deviation from minimum phase we will match the target.
The point here is that at any point in space if you measure the response and have the amplitude and phase, and the system is linear, then you have the impulse response as well. The impulse response defines the transfer function. After all that is why they refer to digital filters and IIR and FIR, because they are emulating the impulse response. Same impulse response = same time response.
Hi
Oh man, what IS a rigid body for you – if you look closer there is no such material nor form. Linear behaviour is not bound to " rigidity ".
When I was at school we were told that anyone that can't imagine a steel plate of several inch thickness being bent by the weight of a fly wouldn't pass the exam.
In fact there IS no such thing than absolute rigidity.
Einstein's theory simply doesn't allow for that.
Greetings
Michael
soongsc said:
Oh man, what IS a rigid body for you – if you look closer there is no such material nor form. Linear behaviour is not bound to " rigidity ".
When I was at school we were told that anyone that can't imagine a steel plate of several inch thickness being bent by the weight of a fly wouldn't pass the exam.
In fact there IS no such thing than absolute rigidity.
Einstein's theory simply doesn't allow for that.
Greetings
Michael
gedlee said:
I fear I am still not quite understanding this. The first sentence above makes perfect sense. The analog is a STEP response, rather than an impulse response. In this case we get a positive impulse (t=0) followed by a negative impulse (t=0+). Since the universe is infinite, its a high pass filter acoustically, and the step is filtered resulting in a typical positive impulse at the t=0 discontinuity, followed by an equal negative impulse at the t=0+ discontinuity, and the area under the curve sums to zero.
An impulse, however, does not have any DC component like a step, and intuitively, its frequency components make no attempt to compress the universe, just propagate a spectrum of waves through it. In the examples posted by John K and Edward W, it looks like a faithful reproduction of the input impulse: the "High Pass" of the universe never comes into play since the lowest frequency component of the impulse present in any magnitude is much higher than the high pass of the speaker/universe combo.
I have always assumed that the value in an impulse response was examining the degree to which the various frequency components of the impulse were shifted in time, resulting in the changing of the shape of the response due to phase errors between the components. The resting static pressure returns to zero in both cases shown....neither makes an attempt to leave a DC offset in the sound field. It is not clear to me why a non zero area under the curve represents an attempt to leave a DC offset in the universe.
Perhaps the truth lies in the dusty pages of a book penned by the dead French mathematician. I am curious, perhaps you have a link I can check out?
Dick
Re: Sure Mongo, ask the question.
The pressure MUST balance in both time and space at any point as time increases. Think again.
Ed LaFontaine said:
Pressure does return to equilibrium. Do the fluctuations necessarily balance in time and space at the microphone? I don't think so. I think in this case gathering data at a sampling point is a limitation.
edit for clarification
The pressure MUST balance in both time and space at any point as time increases. Think again.