Not sure of your first sentence - do you mean I have not chosen to copy the preexisting filters (from XSim or the thread), or that I did not show the actual filter? I think that you mean I chose not to copy the existing electronic filters that I had? I tried to approximate a 250 Hz and 2500 Hz cross - maybe it ended up a bit higher. Certainly didn't do math and just clicked on the circuit values until I had something that looked almost there.
1. Do you mean frequency responses from the speakers or comments on the design? Yes nearfield was 11 inches or so from the drivers, this is almost essential in the setup I have in terms of space. I am looking at the cabinets in a separate model and directivity I have estimated through listening as well as estimated from wavelength/cone size. If I set up a sine wave and walk through the room I can tell there are a lot of standing waves. There is just not enough space and too much reflectivity in the room to avoid that. And one look at the speakers and you will see there are diffraction and baffle effects that no doubt are occuring as well. I tried to just simplify for the purposes of getting a start on the crossover design.
2. OK - Mainly I am trying to consolidate these speakers and remove the racks of equipment with a crossover and simpler amp with a minimum of expenditure.
3. How do I do that? Enable on plots? I put that on this image. Attached is with FRD data. I will also post that. It was 11 inches for the woofer/mid, 8 inches for the tweeter (it is on a horn so technically probably 11 from the driver), moderate levels and not normalized. 16 averages. No smoothing and no delays (so I could really properly read below 500 Hz, at the cost of reflections.
Also posted the DXO file - still a work in progress but could probably use some review.
BTW I am not convinced of the absolute magnitude between the three drivers. I didn't check power or voltage on any of them and attempt to normalize them. I assumed this would be able to be adjusted with an L-pad. Maybe, maybe not.
1. Do you mean frequency responses from the speakers or comments on the design? Yes nearfield was 11 inches or so from the drivers, this is almost essential in the setup I have in terms of space. I am looking at the cabinets in a separate model and directivity I have estimated through listening as well as estimated from wavelength/cone size. If I set up a sine wave and walk through the room I can tell there are a lot of standing waves. There is just not enough space and too much reflectivity in the room to avoid that. And one look at the speakers and you will see there are diffraction and baffle effects that no doubt are occuring as well. I tried to just simplify for the purposes of getting a start on the crossover design.
2. OK - Mainly I am trying to consolidate these speakers and remove the racks of equipment with a crossover and simpler amp with a minimum of expenditure.
3. How do I do that? Enable on plots? I put that on this image. Attached is with FRD data. I will also post that. It was 11 inches for the woofer/mid, 8 inches for the tweeter (it is on a horn so technically probably 11 from the driver), moderate levels and not normalized. 16 averages. No smoothing and no delays (so I could really properly read below 500 Hz, at the cost of reflections.
Also posted the DXO file - still a work in progress but could probably use some review.
BTW I am not convinced of the absolute magnitude between the three drivers. I didn't check power or voltage on any of them and attempt to normalize them. I assumed this would be able to be adjusted with an L-pad. Maybe, maybe not.
Let me take a step back to help make this point. A crossover is not only a frequency, it is also the rate of the slope and it has to include the small variations needed to fix the bumps.
You have a choice to make. You can copy the original filters or you can go out on your own. If you go it alone then you'll have to take the longer road and consider everything there is about a crossover.. which is fine if you want to do that.
The easiest way to duplicate an electronic filter is to copy it directly. You don't need a microphone for this and if you use one, you're only making things harder for yourself. You just have to measure the filter responses at the driver terminals and duplicate those responses with your own filters. This automatically includes the correct levels if you do all measurements without changing the volume setting.
You have a choice to make. You can copy the original filters or you can go out on your own. If you go it alone then you'll have to take the longer road and consider everything there is about a crossover.. which is fine if you want to do that.
The easiest way to duplicate an electronic filter is to copy it directly. You don't need a microphone for this and if you use one, you're only making things harder for yourself. You just have to measure the filter responses at the driver terminals and duplicate those responses with your own filters. This automatically includes the correct levels if you do all measurements without changing the volume setting.
I think I understand what you are saying, it is just that I am not quite set up for that at the moment. That is, I am not set up to do a sweep (or continuous sine waves) and measure voltages at the driver terminals. I have a digital crossover filter and an equalizer filter both running on the system. I don't care about the equalizer for the moment ('all the small variations needed to fix the bumps') - the digital crossover had two frequencies I roughly wanted after 10 years of listening tests, and also had a slope that I liked for each driver. I defined those so I am not really sure that measuring them will be any more helpful.
The driver levels are another caveat that I wasn't completely set up to account for in measurement - I have three different amps with three different volume controls. I can go back and make sure they are all the same for a fixed sine wave, but I also know the efficiency of the drivers and I know I can get close with those values as well.
In the end I was looking at a parts cost that was too high to implement the crossover the way I wanted with the slopes I wanted as defined in the digital domain. But I can always go back and see how these new slopes will work for the final version in the digital version as a sanity check, as well as build out the analog version and make sure that it's something that is still listenable.
The driver levels are another caveat that I wasn't completely set up to account for in measurement - I have three different amps with three different volume controls. I can go back and make sure they are all the same for a fixed sine wave, but I also know the efficiency of the drivers and I know I can get close with those values as well.
In the end I was looking at a parts cost that was too high to implement the crossover the way I wanted with the slopes I wanted as defined in the digital domain. But I can always go back and see how these new slopes will work for the final version in the digital version as a sanity check, as well as build out the analog version and make sure that it's something that is still listenable.
It sounds as though you don't need to do any measurements since you've defined your digital filters.
If you can get those definitions into a simulator to use as a target, remove any measured responses and use the flat defaults but use the measured impedances then you can find filters, phase relationships and levels to reproduce it.
If you don't get these the same as before, there will be a difference, and it won't be the difference between analogue and digital, you'll be listening to the difference in response and power due to the filter values.
If you can get those definitions into a simulator to use as a target, remove any measured responses and use the flat defaults but use the measured impedances then you can find filters, phase relationships and levels to reproduce it.
If you don't get these the same as before, there will be a difference, and it won't be the difference between analogue and digital, you'll be listening to the difference in response and power due to the filter values.
Thanks Allen, that's what I did in the first version that did not use the FRD files in XSim. I appreciate your help, if it wasn't for your tutorial I probably would not have ventured into the world of crossover simulations. I had downloaded PCD years ago and never got to using it. I like XSim because it is Spice-like but it gives almost instant feedback to component changes.
Hello. The tutorial in the first several posts is amazing. So clear and logical. It has really helped me understand crossover design.
I am replacing the tweeter on a center speaker.
The 2 woofers are 16ohm versions of the SEAS T14 driver: https://www.seas.no/images/stories/vintage/pdfdataheet/h0822_t14rcy.pdf
The tweeter is the Seas Prestige 27TDC:
https://www.madisoundspeakerstore.c...as-prestige-27tdc-h1149-textile-dome-tweeter/
I am using this crossover simulator:
https://www.micka.de/en/2weg_en.php#ideal
I found .zma and .frd files for the tweeter, but I have to create one for the woofers for the initial modeling. I have a could of questions.
1) I think I understand how to read the graph and write down freq and level vales for the freq response file. But I don't understand how to read the impedance line. What is the vertical scale?
2) once I understand the impedance line (at 8ohm in the chart?), do I then adjust the freq response file by +3db if it is actually two 16ohm drivers in parallel?
I am replacing the tweeter on a center speaker.
The 2 woofers are 16ohm versions of the SEAS T14 driver: https://www.seas.no/images/stories/vintage/pdfdataheet/h0822_t14rcy.pdf
The tweeter is the Seas Prestige 27TDC:
https://www.madisoundspeakerstore.c...as-prestige-27tdc-h1149-textile-dome-tweeter/
I am using this crossover simulator:
https://www.micka.de/en/2weg_en.php#ideal
I found .zma and .frd files for the tweeter, but I have to create one for the woofers for the initial modeling. I have a could of questions.
1) I think I understand how to read the graph and write down freq and level vales for the freq response file. But I don't understand how to read the impedance line. What is the vertical scale?
2) once I understand the impedance line (at 8ohm in the chart?), do I then adjust the freq response file by +3db if it is actually two 16ohm drivers in parallel?
Sometimes the impedance vertical scale is linear and sometimes it is logarithmic. If it is logarithmic, you won’t see a zero and the major gradations are likely to show a value double the previous.
Curve tracing software can often do both, and depends on you telling it which one to use.
If you use two woofers in parallel, halve the impedance and add 6dB. Actually, it’s better if you use two of them in a simulator since it does this for you.
Using double woofers can increase the directivity index a little at higher frequencies. If you’re not sure about this you can ignore it. On the other hand some simulators will take care of it for you.
Curve tracing software can often do both, and depends on you telling it which one to use.
If you use two woofers in parallel, halve the impedance and add 6dB. Actually, it’s better if you use two of them in a simulator since it does this for you.
Using double woofers can increase the directivity index a little at higher frequencies. If you’re not sure about this you can ignore it. On the other hand some simulators will take care of it for you.
Unless I've had a 'brain freeze', 6dB is double the voltage not double the power.
Sensitivity comes in dB/V, and two woofers in parallel is indeed +6dB in sensitivity.
Sound pressure is the equivalent of Voltage. 3dB comes from driving the extra woofer to the same level, and another 3dB is reconciled in the acoustic impedance from driving via two sources.
Your link refers to summing sources which are not correlated. Non correlated sources add to 3dB. In a two woofer configuration, the woofers are correlated.
Take the example of the baffle step. Half the power goes behind the baffle and half stays in front.. yet the pressure drops by 6dB.
Woah, a debate! On what? Superposition Vs summation. As the waves elongate from the membrane (s) the amplitude of the wave increases and according to D'Appolito (!) the reinforcement on axys is due to constructive interference ( central protruding polar in the graph).
The amplitude results in modulus change of the vector. Same happens electrically as the amplifier power stage drives the (voice) coil and a doubling in coils, being paralleled, means half resistance so double current, well, Er... double watt!
The amplitude results in modulus change of the vector. Same happens electrically as the amplifier power stage drives the (voice) coil and a doubling in coils, being paralleled, means half resistance so double current, well, Er... double watt!
They've been doing it all along, it's just that you didn't notice it !!If you could 'pick-up' 6dB by simply doubling your woofers,
I dare say every manufacturer would be doing it
Looks like you've actually had it !!!
The real problem is the double woofer...or midrange. You (would) want a single voice, i.e. pointing to the point source. Which is physically impossibile due to technology limits but with a 3 way you go near perfection, both theoretically and practically (we're in multi-way, safe side....). Yesterday night by reading the forum I imagined a 3 way with a 3" woofer, a 1" (1.5) mid...and the manufacturer's choices are few, then I'd need a 0.5" or less tweeter.....hmmm, that Sony cellphone of 10 years ago had such a wonderful reproduction of the treble...
Sensitivity is given in dB / 2.83V and not dB / 1W. These are the same only for when the impedance is 8 ohms.
If double voltage is applied, double current would also result and the power would be 4x. However, that is not the case when the current alone is doubled. This is because amplifiers are voltage sources (unless otherwise mentioned).