But I do want to use it for a crossover.
If I understand correctly, when using a 2 channel XLR audio interface one channel can be used as a timing reference (loopback). This way, the delay of the audio signal received by the mic can be measured very precisely, which increases the accuracy of the phase measurement, right?
Is the phase information calculated from the frequency response (without timing reference) not accurate enough for crossover modeling?
It can be done that way and we can discuss options. So to see where I was coming from I'll give you a bit of the back-story. A few years ago it was thought odd that someone would try to measure for a crossover without a timing reference (Ok, in the '90s I used to do it without because I was doing it at home and that was typical back then).
Then a few years ago some outfits started selling equipment with plug-and-play usb mics in the kit to make things simpler. Many people didn't think much of it because signal processors do delay separately.. even though this has nothing to do with anything.
Then a few years ago some outfits started selling equipment with plug-and-play usb mics in the kit to make things simpler. Many people didn't think much of it because signal processors do delay separately.. even though this has nothing to do with anything.
This works for data which doesn't have delayed sound influencing the frequency response, and it doesn't include any overall delay either.jbarelds said:
I have the imm-6 and an extension and plug it into a MacBook pro. I recently got a dats V3 as well to measure the driver specs. Tcp115-4 just arrived, but I have 2 of the 8ohm version as well.
I just recalled - did measure the 8 ohm ones in their test boxes back when I built them. ( 3/8" roof sheathing)
Dayton TCP115-8 small desktop speakers
Dayton TCP115-8 small desktop speakers
Hopefully not too off topic, but it seems that no one here uses the option of a 2nd powered speaker from the second channel output on REW to have a reference timing signal ? (when using a USB mic)
Does this option not work well ?
Thanks,
Rob.
Does this option not work well ?
Thanks,
Rob.
I'm guessing this is the usual hump / dip from baffle diffraction. Simulate your baffle and tweeter placement in the edge
If anyone is reading this and thinking that Allen and I are disagreeing, that's not really true.
I would advise anyone who wants to get into this speaker-design madness (excuse me, hobby) to get the right equipment if you can afford it. A two-channel audio interface with a proper mic preamp, combined with a good omnidirectional measurement microphone, along with REW or ARTA or similar software... this is by far a better way to go than a USB microphone.
But if all you have is a USB mic, and spending $100 - $200 on an audio interface, plus $70+ for a new mic, plus $20 for cables is not in the budget picture... Well as Allen says, there are ways to make a USB mic work.
What do you guys mean when you talk about timing reference? Between drivers or the latency in the sound card?
Between drivers, yes, but more crucially is a reliable reference between measurements. We want to be able to repeatably measure a driver under the same circumstances and get the same phase response each time. This means absolute phase, not just relative phase.
If we can do this, we can capture the FR and phase across many horizontal and vertical axis, and for all of the drivers. A good simulation tool can assemble all this and make accurate simulations of the off-axis response.
With a USB mic, the phase is not consistant, there may be a random delta time error introduced. At low frequencies it may not be important but at high frequencies it is. We are forced to work with the minimum phase response of the drivers.
If we can do this, we can capture the FR and phase across many horizontal and vertical axis, and for all of the drivers. A good simulation tool can assemble all this and make accurate simulations of the off-axis response.
With a USB mic, the phase is not consistant, there may be a random delta time error introduced. At low frequencies it may not be important but at high frequencies it is. We are forced to work with the minimum phase response of the drivers.
Maybe I am confused, or maybe I have a crappy laptop. My machine has a single microphone-in audio jack and a single audio-out jack, both 3.5 mm. I don't see how I could connect a Dayton EMM6 or Audix TM1 mic to the computer... where would the XLR cable go? How would the mic get the phantom power?
I was advised on the ARTA thread and the REW thread to get a 2 channel audio interface, the cheapest workable option was the Behringer UMC202HD. Was this an unnecessary purchase? not that I care, it was only $100...
I have a feeling I am missing your point.
Jim
It can also be done passively with 9V batteries. I had a jiffy box that would fit in the palm of my hand.
If I understand correctly I need a loopback timing reference to measure absolute phase. This is particularly important when doing multiple off-axis measurements, where small changes in mic positioning will impact the phase measurement, especially in the higher frequencies. Loopback timing reference requires a 2 channel audio interface and cannot be done using a USB mic.
On the other hand I see a lot of folks using USB mics and apparently the lack of absolute phase doesn't hinder them from designing good crossovers.
I'm somewhat torn on what to get. The 2ch audio interface + XLR mic solution is a bit more expensive, and also more complicated. But if a USB mic will likely limit me later on, I'd rather get the more complicated setup right away.
On the other hand I see a lot of folks using USB mics and apparently the lack of absolute phase doesn't hinder them from designing good crossovers.
I'm somewhat torn on what to get. The 2ch audio interface + XLR mic solution is a bit more expensive, and also more complicated. But if a USB mic will likely limit me later on, I'd rather get the more complicated setup right away.
I have both and would go with the XLR mic solution, I started off using the USB mic a lot but only tend to use it to verify my other mics and living room EQ now. Most of the time the cable is too short, I tried a 5m cable but it didn't work (and would still be too short a lot of the time). Be careful to check that a cheap interface actually outputs 48V for phantom power as some only do 15V (which may be fine but may also limit your mic choice).
When I make a response measurement with my OmniMic, there is a lot of excess phase in the measurement (i.e. time delay). My first step is to "unwrap" most of that excess phase, which can be done in the omnimic software. If I make repeated measurements of the same tweeter, there are slight differences in the phase response on each one. That is the timing error.
If the recording system can capture the absolute phase of two drivers (woofer and tweeter), then the crossover simulation for those two drivers can use the absolute phase measurement directly. The distance between the acoustic centers of the woofer and tweeter is built into the phase responses.
I am in the same situation as kipman725. I bought OmniMic when I first got started two years ago. Last year I wanted to take advantage of the full capability of VituixCad2 software, which required a 2 channel loop-back rig. I started with a Dayton EMM6, but it had a defect, so I bought an Audix TM1-plus. Now, after a fairly lengthy learning process, I can make full polar measurements of my drivers and completed systems.
I do not regret buying the OmniMic. It is a turn-key system that enabled me to get started right away. There is no ambiguity to it's measurements, 96 dB SPL is for certain 96 dB SPL... no uncertainty about preamp gain or mV/Pa. I also believe that it is my most accurately calibrated microphone. I like the harmonic distortion measurements from OmniMic better than ARTA or REW. If I had tried to get started with an audio interface and an XLR mic, I am sure I would have collected flawed data and I would have spent a LLooonnnggg time trying to understand what was going wrong. I am not an audio professional so everything about the microphone and digital recording process was completely new to me.
j.
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Is there some proper documentation available on how to measure correctly. Something understandable for newbies.
Also some explanations/definitions of the different variables involved when taking measurements. eg. absolute/relative phase, time delay, exess phase, ....
Some pictures/schemes of setup would help a lot too.
Also some explanations/definitions of the different variables involved when taking measurements. eg. absolute/relative phase, time delay, exess phase, ....
Some pictures/schemes of setup would help a lot too.
To me these sources have helped me a lot to get started:
https://www.minidsp.com/applications/acoustic-measurements/loudspeaker-measurementshttps://audiojudgement.com/soundeasy-tutorial/
(The latter uses the SoundEasy application, which I don't (I use REW), but the explanation of the method is still valuable.)
The right equipment is not all that expensive if you have a little time. A tower or desktop PC with internal A/D by Intel and 3.5 stereo jacks for line level in and headphone out is ~$75. The windows XT or 7 is a problem, but can be replaced for free with linux. I'm using lubuntu20.04 op system on my dual core PC, but ubuntu versions back to linux14 (a classic) 11 or 8 can still be downloaded. Audacity can be down loaded separately and comes with ubuntustudio op system flavor. I have gotten REW for linux to download and works with lubuntu20.04. No USB A/D required. The intel A/D won't examine artifacts 80 db down or more like a $200 A/D, but most speaker inaccuracies are more like 10 or 20 db from the mean.
A condensor mike can be had for $35 from newark/farnell, and there is a dynamic omni for $19. I've used the 1" condensor cardiode. Cardiode doesn't work right below 100 hz under 2 m distance but I don't know why one would measure at 1 m anyway. I don't listen to speakers at that distance. Studio monitors only at 1 m in a tiny control room. The $35 cardiode doesn't respond above 15 khz, but my ears cant hear that anyway. Most males over age 12 don't have hearing above 8 khz, in my experience.
A mixer with phantom power can be had used for $30-70. 90's ones from Peavey & Alesis are repairable. Usually the master volume pot is shot. I've replaced master volume pot on a Peavey PV8 for $7. $2 for a smaller 10k slide pot from Bournes and $5 for a box of 2 mm flat head screws to hold it to the front panel. When the Peavey PV8 was stolen, another was $65 without the power supply. It works with a 12 vac supply costing $2 at a charity resale shop. Original PS is 16 vac. There is an untested alesis 4 channel for $40 + freight today on ebay https://www.ebay.com/itm/304318473919?hash=item46dacb66bf:g:NQ8AAOSwYcVheFQF and a tested 6 channel for $60 + freight. Cable from mixer to PC can be 3.5 mm male to dual 1/4 phone male from newark $12 and XLR mike cables can be had from newark or even cheaper, ebay. I got 4 8.3 m XLR male to female cables from ebay for $36.
A stand for the mike can be a $10 camera stand from the charity resale shop. Gets mike 2.5 m off the ground to eliminate floor reflections, cheaply. A metal bar can adapt the 1/4" thread of the camera stand to the shock mount that comes with the newark cardiode microphone.
Set the speaker off the ground on a stand outside for best results. World's cheapest anechoic chamber. Yes, you have to wait until no cars or trucks are driving by. Not a huge problem, you need 2 quiet seconds to take a sample. Probably easier in North America where there is more open space, than in Europe.
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