# Frequency Response in the car: FFT vs RTA



## ErinH (Feb 14, 2007)

To jog my brain a bit, as it's been a while since I've looked at software with FFT allowable measurements (I typically only use trueRTA), I wanted to bring up the topic of FFT vs RTA measurements and their use in the car audio world and home audio world.

It is my understanding that FFT measurements should be taken to get proper frequency response of a system or driver while 'disregarding' the effect the environment (room) has. Is this correct?

I see home audio folks often using FFT analysis for speaker measurements, and then RTA analysis for room measurements. I believe the RTA is used in the car environment rather than FFT simply due to the reflective nature of the car itself, thus resulting in near similar impulse response arrival times of both direct and reflected sound to the listener, and therefore defeating the usefulness of an FFT measurement (in the car). Of course, I'm not 100% sure of this. 
IOW, if both the direct sound and the reflected sound are hitting the listener at dang near the exact same time, is the FFT analysis of any use here, and is that the reason we simply jump straight to RTA?
Going beyond, wouldn't we still benefit from FFT by means of singling out the environment (ie: gating) and 'looking' only at the speakers' response? 
Furthermore, how can one even guarantee the environment isn't playing a role with FFT without gating the initial response? I suppose that goes back to why I'm thinking we don't even bother with FFT to begin with, in the car.

In my older testing of speakers, I used FFT as a means to find fault with the speaker itself and then RTA to find fault with the response curve due to the combination of speaker + room. It's been a while, though, so I wanted to see if my understanding of the usefulness of the two is accurate.

Anyone with knowledge on the subject care to chime in and let me know if my thought process is right or wrong? I'm familiar with RTA, and familiar with FFT but it's been a while since I've touched anything other than a sole-use RTA program...


Thanks,
Erin


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## CraigE (Jun 10, 2008)

Interesting topic.
I'll throw a couple of things out there.

Aren't many of the computer based "RTA" programs actually using FFT based calculations to show the familiar RTA bars ? 
Are the "RTA" bars more of a display type, that requires little setup, and is therefore easier to use? The FFT has more parameters to setup, such as FFT windowing and FFT type, which can be somewhat intimidating. 


From Wikipedia- Real Time Analizer; 
"Digital RTAs use digital sampling technology and microprocessor based digital signal processing to perform necessary calculations, such as Fast Fourier Transforms, to perform the measurements and thus do not need analog hardware filters to isolate each frequency band. The digital approach to signal analysis generally yields much higher accuracy and resolution and thus most RTAs currently in production use digital signal processing technology"

Here is a comparison of RTA vs FFT from Studio Six Digital. It's based on their products, but it's pretty basic and easy to understand. This may be helpful for readers that are less familiar with FFT.
FFT or RTA? | Studio Six Digital


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## ErinH (Feb 14, 2007)

I agree, some of the programs certainly make them _seem_ the same. 

What I'm getting stuck on is how a smoothed FFT is different from RTA. Specificaly with TrueRTA, where one can have more data points than standard 1/3rd octave (which seems to be primary indicator of the program's use). So, maybe in this case, it is an FFT, just in a more user friendly package? However, you are not allowed to change the sampling rate like you do with most FFT software.
Furthermore, when you read the description of the differences, TrueRTA certainly seems much more like FFT than RTA (ie: generated tones).


With that said, I'm back to the original question: does it really even matter for us?
Is the benefit with FFT to get driver characteristics via impulse response and the benefit for RTA to get all sound characteristics? That's what I'm leaning toward...
Personally, I've used both impulse and pink noise generated tones through trueRTA, and IIRC, the differences were minimal at most. Heck, that's probably why I just started sticking with RTA.
However, I'm not the only one who uses pink noise for car measurements... most every 'test cd' for car audio includes pink noise much moreso than it does swept sine.


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## CraigE (Jun 10, 2008)

bikinpunk said:


> What I'm getting stuck on is how a smoothed FFT is different from RTA.


This helped me.
Scroll down to #10 

"RTA Bar Levels vs. Smoothed FFT Levels.;

FFT or RTA? | Studio Six Digital


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## SSSnake (Mar 8, 2007)

Erin, 

I think you probably are aware of Andy's position on this but for the thread...

Paraphrased - We are in such a small highly reflective environment that we hear the direct and refelcted sounds as one sound. 

This would lead me to believe that gated FFT analyses would be less beneficial than RTA approaches. I am in the habit at looking at both. Particularly the differences in the two measurements. This should allow us to locate some of the reflection issues and perhaps treat them (difficult to do in such a small environment). Directivity control would seem to be a key to success but is tough to do at lower freqs. That is the reason that I am looking into waveguides for tweets...


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## 14642 (May 19, 2008)

There's one big difference between a true RTA that uses bandass filters for each portion of the resolution (like an old Audio Control 3055) and an FFT. The resolution of the RTA is by design a log scale and an FFT is a linear scale. The octave smoothing is designed to make the FFT appear to be the same as an RTA, and from my experience, it similar enough that it doesn't really matter, so long as the FFT size is large enough to provide good resolution.

A 1024 point FFT provides about 20Hz resolution and that resolution is spread evenly from 20-20k. That means there are about 512 points above 10k and about 512 below. It's not a useful distribution. Who needs that kind of resolution above 10k? 20Hz resolution isn't good enough in the bass. The fix is to increase the FFT size to have better resolution in the bass, but that requires more processing power. For a PC, that's probably no big deal, but for signal processors that use FIR filters, that costs more money. 

The other benefit with FFT measurements (which is really an impulse response measurement that's converted to the frequency domain using the Fast Fourier Transform) is that it's possible to remove reflections from the measurement so you can measure the "anechoic" response of the driver without having to build an expensive anechoic chamber. 

There's one HUGE caveat, though, and that is that there's one additional resolution buster when you do this. The window for the impulse response is the time that the measurement runs, and the samples are spread over that window. When you gate the measurement to remove the reflection, you throw away part of the samples that contain the reflection. In a really small room, the reflections come pretty quickly, so you end up throwing away a bunch of samples and spreading the few that are left over the full frequency range. This puts us back in the previous spot, where there isn't enough resolution to see what's going on at low frequencies. 

gated measurements are great for making anechoic measurements outside or in a big room, or if you only care about what's going on at high frequencies. in small rooms, it's often not possible to make useful measurements. cars are small rooms.

In cars, we don't need to fix the speaker and then fix the room, we can just fix the whole thing together, so long as the speaker design isn't terrible. There isn't much you can do to the speaker outside the crossover region, and it's probably possible to see what's going on at 3.5k with a gated in-car measurement. Using a gated measurement in a car to figure out what's going on in the midbass is a mistake.

The gated measurement in a room can be pretty useful. You can also move the window to capture ONLY the reflection and convert it into a frequency response measurement with an FFT to partly determine the effect that the room has one the response. However, that window will be really small and you won't have much resolution. Trying to isloate reflections this way in cars is nearly impossible--make an impulse response measurement to see this. The measurement you see will almost prove why there's no need for speaker correction and separate room correction.


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## ErinH (Feb 14, 2007)

Andy Wehmeyer said:


> Trying to isloate reflections this way in cars is nearly impossible--make an impulse response measurement to see this. The measurement you see will almost prove why there's no need for speaker correction and separate room correction.


This is what I was leaning towards in the original post. Gating (or looking at a limited impulse response) is meaningless as the environment itself is highly reflective (in a car). 

*** Going a bit off toic regarding the measured environment...
I do, however, get tripped up how one can properly measure a speaker, say, in the home. You can't surpass baffle diffraction, so how do you do anything about it? Is this the point at which you make both nearfield and farfield measurements (ie: 1cm, and 1 meter away, respectively)?
So, for instance, if you measure a 3" driver from 20-20k to see the FR and IMD, you put the mic right in front of the driver in order to pick up low end response but also remove artificats of baffle step, and farfield + gating to get higher frequency response, then 'stitch' the two FRs together. However, this seems like a total PITA, to me. I thought the purpose of gating was to get non-reflective response, so I'm not sure I see the benefit of stitching two different responses anyway.
For some reason, I have the hardest time seeing why the two different methods (nearfield/farfield) are needed.

A lot of these questions are based on my desire to start testing drivers again, and I'm trying to refresh my memory. I used to know how things needed to be done, but never quite grasped why they needed to be done that way. Quite honestly, this thread and my research on the topic will drive how I start testing drivers and the data that will be posted here...
********

- Erin

Sources:

http://www.kirchner-elektronik.de/~kirchner/DIPOL-CARDIOIDeng.pdf

http://www.fesb.hr/~mateljan/arta/AppNotes/AP4_FreeField-Rev03eng.pdf


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## masswork (Feb 23, 2009)

Now i am confused.

My understanding is FFT (impulse response) is more powerful than RTA. 
By looking at the impulse response itself already tell us many things like time arrival and polarity - which is not there in RTA.
And when we're done with them we can just take another impulse response, and convert it into frequency response - and equalize it if we want to (or needed).


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## SSSnake (Mar 8, 2007)

Mass,

As with any tool the "power" or usefulness is specific to the application. Most would consider a CNC machine a more "powerful" tool than a hammer. However, if you are trying to drive nails I would pick the hammer every time. 

Erin's post was focused on freq response in the car (at least initially). For that specific application IMO - RTA is more useful for many of the reasons that Andy mentioned. However, there was a good thread about using impulse functions to determine necessary TA values. In this case and if used correctly (short paraphrased summary - turn off your low pass filters) using impulse functions can work fairly well (read the discussion as it is very good - at least IMO).


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## masswork (Feb 23, 2009)

I was thinking that Andy prefers doing "FFT" rather than RTA.
At least that's how MS8 works, right? Chirps? Impulse - TA - FR - EQ ?

IMHO getting repeatable FR using RTA is quite difficult. The bar is always moving 
Not sure when to stop 
And it's prone to external noise as well... so...


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## ErinH (Feb 14, 2007)

masswork said:


> I was thinking that Andy prefers doing "FFT" rather than RTA.
> At least that's how MS8 works, right? Chirps? Impulse - TA - FR - EQ ?
> 
> IMHO getting repeatable FR using RTA is quite difficult. The bar is always moving
> ...


I may be wrong here, but I think Andy is saying that for _us_ and our measuring of response, RTA makes the most sense.
The ms-8 uses FFT for impulse driven adjustments.

IOW, I think Charles is answering your question: it depends on what you're trying to do.
For a piece of equipment such as the ms-8, maybe the FFT version makes more sense, while for the average user looking to tweak an EQ, the RTA makes more sense.


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## michaelsil1 (May 24, 2007)

I think in a car it is more important to get an average of several readings. I've heard this over and over again and I've found that it works very well.


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## highly (Jan 30, 2007)

I think we are possibly confusing terms a bit in this discussion. These are the terms *as I understand them*. I'm very interested to hear differing views here, of course.

FFT: Fast Fourier transform - the math used to get the 1/3 (or 1/8 or 1/24) octave amplitude measurements used in the RTA display by providing a frequency domain representation of a signal.

RTA: Realtime Analysis - a tool that uses FFTs to provide a visual representation of the amplitudes over a given frequency range and aperture (the 1/3 octave part). This measurement includes the speaker and its environment in the frequency domain. Can't tell you how far away a speaker is, just the relative volume of one frequency band to the next measured band.

IRF: Impulse Response- A plotted time-domain response to an impulse used for phase and time measurement. This can measure both time to the source (loudspeaker) and time to reflections arriving after the source. It is a measure of the environment's response to the stimulus applied by the impulse leaving the speaker.

So an application like SMART will provide the ability to determine distance to the driver, distance to the first reflection, and ringing due to resonance and reflection using an Impulse Response measurement. That measurement will not tell you what the speaker will sound like frequency-wise in a given installation. It's a measurement of the space the speaker is in. 

<dons flamesuit for what is to follow...>

-Todd

P.S: Erin...+1 for the Mod status, bro!


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## ErinH (Feb 14, 2007)

Todd, I agree with your post. That's essentially the way I've come to understand things. It's also my understanding that FFT and impulse are brothers, but that understanding may be flawed.

I think what's tripped me up is trueRTA, as it seems more like an FFT program but in RTA clothing because it allows for a variety of resolution. However, that can't be the sole factor when making the dilineation between FFT or RTA. TrueRTA doesn't allow for time based measurements, so must still fall on the side of a true RTA (no pun intended), but with varying resolutions.


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## ErinH (Feb 14, 2007)

take 2:

Ever heard of writer's block? Well, I have RTA'rs block. 



I found this article explaining the differences and I think this does really well to sum things up:


> An RTA (Real-Time Analyzer) is a two-dimensional measurement system that displays energy in dB SPL or volts versus frequency in hertz.
> 
> Meanwhile, TEF, Smaart, SIM and the like are all 3-dimensional (3-D) measurement systems that display energy vs. frequency vs. time.
> 
> Therefore an RTA, unlike FFT (Fast Fourier Transform) based 3-D measurement systems, is time blind and lumps all energy occurring within a fraction of a second together. A fraction of a second is an eternity to a 3-D measurement system.


An article based on FFT tuning by the same author:
http://www.prosoundweb.com/article/tuning_a_system_with_fft_getting_it_close_without_listening/P1/


So, to sum it up, it seems to me that FFT is time based and RTA is simply a measurement of cummulative energy.


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## Patrick Bateman (Sep 11, 2006)

Andy Wehmeyer said:


> There's one big difference between a true RTA that uses bandass filters for each portion of the resolution (like an old Audio Control 3055) and an FFT. The resolution of the RTA is by design a log scale and an FFT is a linear scale. The octave smoothing is designed to make the FFT appear to be the same as an RTA, and from my experience, it similar enough that it doesn't really matter, so long as the FFT size is large enough to provide good resolution.
> 
> A 1024 point FFT provides about 20Hz resolution and that resolution is spread evenly from 20-20k. That means there are about 512 points above 10k and about 512 below. It's not a useful distribution. Who needs that kind of resolution above 10k? 20Hz resolution isn't good enough in the bass. The fix is to increase the FFT size to have better resolution in the bass, but that requires more processing power. For a PC, that's probably no big deal, but for signal processors that use FIR filters, that costs more money.
> 
> ...


I didn't agree with this line of thought initially, and did a lot of gated measurements in the car. But for the most part, I just try to get the speaker to sound right anechoically now, then put it in the car.

The reason that I changed my tune on this one is that it's hard to get a clean gated measurement in the car, because the reflections pollute the data unless you use an insanely short gate.

And then if you do that, your data doesn't have any resolution.

So either way, you're kinda screwed.

To make a long story short, I think Andy is right (as usual.)

I still do tons of gated measurements before the speakers go in the car, and use polar and power response a great deal.


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## ErinH (Feb 14, 2007)

^ yes, I agree. Based off my own experiences.

I must apologize as I was essentially trying to ask two questions but formed the thread title as one which resulted in a mess of a thread.

The initial questions were:
1) What is the difference between FFT and RTA
2) Why do we not bother with FFT in the car


I'm quite certain #2 has now been answered, and #1 is much more apparent to me now.



so, next up, I think Todd has a series of questions that I'm interested in as well. Todd?.....


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## highly (Jan 30, 2007)

Well, all (that I know of) digital spectrum analyzers use fourier analysis; either DFT (Discrete Fourier Transforms) or FFTs to produce the measurement. So, in the digital world, you rarely have one without the other. The FFT is basically a chain of bandpass filters set at 1/whatever octaves, each computing the amplitude at that point. If you were to try to calculate every amplitude of every sample over the required bandwidth in realtime you'd need LOTS of processing power. The difference between a DFT and FFT performing the spectral analysis is roughly a 100-fold speed increase (~80Mflops v 880Kflops)

You would also use fourier analysis to plot an impulse response. The FFT is the means, the plots are the end.

Did that make any sense?


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## SSSnake (Mar 8, 2007)

In an effort to ensure we are all talking about the same thing let's try to get a common vocabulary... (and someone please correct me if I misspeak)

FFT - Fast Fourier Transform is an algorithm that converts information in the tme domain to the frequency domain.

The FFT works out the amplitudes and phases of a set of cosine waves that, when added together, would give the same set of measurement values as the time signal. The amplitudes and phases of those cosine waves are a different way of representing the time signal, in terms of the frequencies that make it up rather than its individual measurement values. The amplitudes are easy to understand, a larger amplitude means a bigger cosine wave. The phases indicate the starting value for the cosine waves at the time of the first sample in the sequence that was measured. A phase of zero degrees would mean the starting value was amplitude*cos(0) = amplitude. A phase of 90 degrees would mean a starting value of amplitude*cos(90) = 0. We are more often interested in the amplitudes than the phases, but we shouldn't forget about the phases entirely - they contain half the information about the shape of the original time signal. 

When an FFT is used to calculate the spectrum it uses a set of frequencies that are evenly spaced from DC (zero frequency) up to half the sample rate (the maximum that can be properly represented). The spacing depends on the length of signal we analyse in the FFT. FFT calculations are most efficient when the signal lengths are powers of two, such as 16k (16,384), 32k (32768) or 64k (65536). To calculate a 64k FFT from a signal that is sampled at 48kHz we need 65536/48000 seconds of the signal, or 1.365s. The frequencies would be spaced at 24000/65536 = 0.366Hz. If the FFT were generated from 16k samples the frequencies would be 1.465Hz apart. The fewer samples used to generate the FFT, the further apart the freqencies are so the lower the frequency resolution. For high frequency resolution we need to analyse long time periods of signals. 

RTA
A common way of viewing the spectrum of a time signal is to use a Real Time Analyser or RTA. The RTA shows a plot of the amplitudes of the frequencies that make up the signals it is analysing. However, whereas the FFT produces signals that are at uniformly spaced frequencies, an RTA groups them together in fractions of an octave. An octave is a doubling of frequency, so the span from 100Hz to 200Hz is one octave. So is the span from 1kHz to 2kHz - the actual freqeuncy span of an octave fraction is more the higher the frequency gets. For a 1/3 octave RTA the span is about 4.6Hz at 20Hz, but is 4.6kHz at 20kHz. For a 1/24 octave RTA the spans are 1/8th as wide. Within the span of an octave fraction many individual FFT values may be used to produce the single value the RTA assigns to that band of frequencies. Below is an image of the REW RTA displaying the spectrum of a 1kHz tone and its distortion harmonics. - From REW V5 manual - not to proud to plaugerize 

So in practical application what does this mean...

With RTA measurements we really don't care much about the measurement signal in terms of timing. You can use a built in pink noise (or other signal) or a completely seperate source such as a CD with the measurement signal. There is very little synchronization between the input signal and the measured output.

With what we call FFT measurements timing and signal content directly affect the results. The measurement signal and acoustic measurement must be synchronized so that you can extract timing information. The ability to gate the measurement allows to remove FR contributions from things such as reflections. However, in a car the reflection is so closely spaced to the original signal that removing it from the measurement is ineffective except at higher freqs. So what happens when the gate on an FFT measurement gets very large? The response curve should be nearly identical to the response curve you get from an RTA measurement (someone correct me if I'm wrong here).

As far as the MS-8 using FFT measurements... Andy will have to be the defninitive word but since the MS-8 addresses TA I believe that some form of FFT analysis is employed. If memory serves from my last calibration it uses a form of swept log function (just a guess from listening to the signal).

One thing that I think must be addressed is spatial averaging (which BTW the MS-8 apprently does - binaural mics with multiple head positions). With a highly reflective environment like a car constructive and destructive interference patterns are liberally created within the listening environment. While we don't move around much compared to a home environment the reflections are much worse. Spatial averaging helps the mitigate some of these issues.


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## Patrick Bateman (Sep 11, 2006)

bikinpunk said:


> ^ yes, I agree. Based off my own experiences.
> 
> I must apologize as I was essentially trying to ask two questions but formed the thread title as one which resulted in a mess of a thread.
> 
> ...


Tom Danley talks about another factor which makes FFT very nice. Tom does TEF measurements, as did Richard Clark with the GN back in the day.

In a nutshell, he talks about how woofers have a built in "delay", and how he factors that in to his passive crossover and the physical location of the woofer and the tweeters. Here's the quote, from Audioasylum:

http://www.audioasylum.com/cgi/t.mpl?f=hug&m=51313

"_Every driver has a time delay, an amount of time between the microphone and speaker radiator, in addition, each driver has an additional time delay component (normally much smaller) between the instant the signal is applied and when the radiator produces sound. 
This delay is also frequency dependant so a driver has a different apparent location “front to back” with frequency. A well-loaded horn and electrostatic speaker have the smallest delay and variability of delay of this type in this regard. 
Richard Heyser was the first to devise an explanation for the delays and a way of measuring them fwiw and a number of his landmark papers are in the AES loudspeaker anthologies. 
Every filter also has a time delay as well as the more commonly referred to amplitude and phase effects.

The first delay is a fixed value, proportional to the distance from the mic to the radiator, the second delay is one which changes according to the driver size, implementation and so on. 
The third is dependent on the filter’s frequency corner shape and order. 
All of these delays can be seen as a changing “distance” if one looks at the group delay for any of them and thinks in terms of there being .883 feet distance separation per millisecond of GD. 
Because a given time is not frequency dependant (why a 1Ms GD at 20 Hz is nothing but 1 ms at 20 KHz is a shift of MANY wavelengths) Group Delay is not the most illustrative measure of what one is trying to accomplish. 
In all of these cases, time delay is related to frequency, the lower one is in frequency, the larger time delay any of these factors produce. 
It is this, which must be remembered when looking at the GD plots for a low frequency loudspeaker too, one cannot make a low frequency filter or speaker without having a shocking amount of group delay. Good thing that is not a good indicator.

As far as I see it, the driver’s frequency dependant delay, once converted to driver acoustic phase is the most informative domain to look in, here, one WL is always 360 degrees of phase shift no matter what frequency. 
It is the acoustic phase which governs if two driver signals will add completely or cancel completely or anything in between, all within the 0 to 180 phase difference they can have. 
Adding the outputs from two drivers that are “in phase” produces an addition where the result is greater than either. In a loudspeaker crossover, this bump is why the Linkwitz Xover format was devised, it spaces the corner frequency’s of the hp and lp farther apart so that the sum has a flat amplitude response. 
On the other hand, adding two signals that are 90 degrees apart also results in a constant amplitude but phase that is 45 degrees from either. 
A 180 degree difference results in cancellation, if the sources are acoustically close enough together to combine fully (less than about ¼ wl) they also cancel fully in this condition. 
A larger separation results in the formation of frequency dependant lobes of energy at odd directions and incomplete cancellation (or addition if in phase).

In a speaker like the one shown, time correction would involve first defining the listening position and placing the microphone there. 
Then, measure the time delays for each horn AT / NEAR the respective Xover F’s. 
Then set the time delays in a loudspeaker controller like a BSS366 so that the Low frequency section has No delay. 
Delay the mid so that it is “in time” with the low section at crossover and the acoustic phase is as seamless as possible. 
Then proceed upward in F with the higher sections getting progressively more delay. 
A unit like the BSS366 is really quite good at this job and as well as time delay adjustments, also has the ability to phase shift individual sections.

*While this does put the drivers in the proper “front to back relationship” so far as time to compensate the driver /filter delays, it can only do so at the position the microphone was in. All other locations have a different path length (from the driver to your ear to the other driver) so what ever delays were added, become progressively “off” as one moves further from the measurement position. 
Also, if the drivers for each range are not within about ¼ wl of each other, they produce directivity patterns in the range where they are both on (the lobes going out to the sides).*


My approach in the Unity Horn was to arrange the drivers “front to back” physically to offset the driver delays and passive crossover delays so that no DSP etc was needed. 
Then, at the same time, have the output of each range added to its neighbor at a point where the dimensions were less than ¼ wl so that there is complete addition (or cancellation if wrong) and no directivity or lobes from driver interaction. 
Because I wanted constant directivity vs frequency, I used a conical horn. 
Conical horns have a variable expansion rate, they become more of a low frequency horn the further one gets from the apex. 
The approach was to use drivers suitable for horn loading at the frequency range where the expansion rate made that section a suitable horn. 
For example a compression driver at the apex may have a 900Hz low Frequency corner but driven 3 inches further down the horn a driver couples into the horn to 300Hz. 
In other words, a conical horn (or any horn really) is in effect a “high pass” filter, the idea is to tap in to the “variable filter” at various points appropriate for the frequency in question. This way instead of 3 different horns say, one has one horn covers a much wider range and because the low cutoff of the low section sets the size of the mouth, one has the largest mouth possible at the high frequency end which produces the most constant directivity. 
As the different ranges are within a small space at xover, there are no directivity artifacts like the lobes in the previous case, polar plots do indeed show the pattern of a single driver / horn even when 7 drivers in 3 ranges are used on one horn..

A further refinement is to not only sync the drivers in time and acoustic phase but then to use another approach to un-shift the drivers acoustic phase, making each section and summation as close to zero group delay as possible. This last thing is what Tom and Kurt heard the last time they were out.

If this was interesting, there was a great deal more written explanation on it back when I tried to explain its workings to Wayne so one can do a search I think and find that stuff.


Tom _"

#########################################################

For me, it's a bit of revelation when you see how many problems are solved if you can physically get the wavefront of the tweeter and the woofer in sync. It simplifies the crossover, improves intelligibility, and improves imaging. And it doesn't cost a thing!

And this is something you could never do with an RTA, because you're dealing with a problem in the time domain, a problem where you need resolution on the order of a millisecond to solve the problem.


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## highly (Jan 30, 2007)

On the Todd's Questions topic...

I've always wondered why we don't test speakers in-car using a waterfall plot initiated by a swept impulse. Would that not show both _frequency over time_ *and *_reflection by frequency_ in the environment? We still have comb filtering issues to deal with, but on a per-driver basis it seems pretty comprehensive if more difficult to read.

With all of the measuring I've tried to do so far I have to say that I don't know how much difference it really makes. Assume you start out with a driver that on a baffle in free air in an open field has reasonably good power response, acceptably low levels of distortion, and relatively flat frequency response. Put it in the car and everything changes for the worse. Now, try as you might, you are not going to get back to the measurements you just made. It's 'nice' to see what a speaker CAN do in ideal conditions, but how much of that ideal response is really valid in-car? I would definitely say that it would be near worthless to try to compare in-car measurements between drivers. There are just too many overwhelming environmental variables to make any sense of it for comparison's sake. I don't know of any way to take the 'car' out of the measurement and end up with something meaningful.

Even if I knew how, as Andy infers it just doesn't matter.

On comb filtering - that's the one place I have seen a valid use of measuring in-car. Use the RTA to measure the effects of driver position and azimuth to correct as much as possible during the install. Once the location is finalized there will be no _correcting _those combing issues, just pushing them around with T/A to try to make them less problematic.


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## SSSnake (Mar 8, 2007)

Todd,

It seems you and I were headed off in the same direction... Lunch interrupted my post but I do agree with your comments except for possibly one area (and I think this is one of the things that contributes to the miscommunications). An FFT analysis approach does NOT have to use an impulse function as the measurement signal and therefore I stayed away from the impulse response term. I believe (someone pelase correct me if I am wrong) that the important difference is the synchronization of the stimulus and measurement. For example I believe that the MS-8 uses some type of FFT analysis but I have not heard impulse functions being emitted from my speakers during acoustic cal. Again, I'm not sure what it is using but it sounds more like log sweep.


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## highly (Jan 30, 2007)

SSSnake said:


> Todd,
> 
> It seems you and I were headed off in the same direction... Lunch interrupted my post but I do agree with your comments except for possibly one area (and I think this is one of the things that contributes to the miscommunications). An FFT analysis approach does NOT have to use an impulse function as the measurement signal and therefore I stayed away from the impulse response term. I believe (someone pelase correct me if I am wrong) that the important difference is the synchronization of the stimulus and measurement. For example I believe that the MS-8 uses some type of FFT analysis but I have not heard impulse functions being emitted from my speakers during acoustic cal. Again, I'm not sure what it is using but it sounds more like log sweep.


I agree, and didn't realize that I had insinuated that an FFT analysis was forced to be impulse-response based. I agree also that the MS-8 appears to use a log sweep. It's my personal supposition that the initial sweep on the MS-8 is performing a timing analysis of the sweep as an impulse and that it uses this initial sweep to determine the relative time alignment on a per-side basis. As Patrick says, above; 'aligning the initial wavefronts' of the drivers on a side. Frequency correction likely happens after that. Just my guesses and not entirely relevant


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## SSSnake (Mar 8, 2007)

Nods head in violent agreement


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## highly (Jan 30, 2007)

Patrick Bateman said:


> *
> Also, if the drivers for each range are not within about ¼ wl of each other, they produce directivity patterns in the range where they are both on (the lobes going out to the sides).*


While I agree with this statement, and with the choices that Tom made in designing the Unity Horn (not to insinuate that he even cares what I may or may not think) I can't help but wonder how much effect it has in the nearfield environment in the car? In a car, I've found that a number of things that I have read as repeatable auditory phenomena in the farfield become overshadowed by overriding phenomena in the nearfield. The effect of the massive reflections are simply too strong. Does wavefront alignment have as pronounced an effect in the car given the other overriding concerns?

Oh, and great to have you aboard the discussion, Patrick! Your input never fails to expand my understanding of the tiger whose tail I've got ahold of. Even if it means I spend the next month and a half in research trying to understand the implications of what you write!


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## Patrick Bateman (Sep 11, 2006)

highly said:


> I agree, and didn't realize that I had insinuated that an FFT analysis was forced to be impulse-response based. I agree also that the MS-8 appears to use a log sweep. It's my personal supposition that the initial sweep on the MS-8 is performing a timing analysis of the sweep as an impulse and that it uses this initial sweep to determine the relative time alignment on a per-side basis. As Patrick says, above; 'aligning the initial wavefronts' of the drivers on a side. Frequency correction likely happens after that. Just my guesses and not entirely relevant


A couple of comments:


As noted by Danley, digital delay won't do you many favors in the polar response, since it only fixes the problem where the microphone happens to be measuring. Many would argue that's "good enough" because we know where the listeners will be seated in a car. But if you think that's the case, try using a speaker with really well behaved polar response, and you'll notice that it sounds much more natural in a reflective environment like a car. I believe this is because we're able to perceive if the reflected energy is consistent with the initial wavefront.
TEF is a bit of a mystery to me. It seems like one could achieve similar results in four steps. First, get your two drivers as close together as you can. Second, record the impulse of the woofer. Third, record the impulse of the tweeter. Now *physically* move the tweeter forward or backward to align the impulse of the woofer and the tweeter. *Admittedly the crossover will affect the impulse* so you'll need to tweak the crossover and the tweeter location in tandem.


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## ErinH (Feb 14, 2007)

regarding Todd's waterfall post:
My initial reply would be something along the lines of what I posted in the OP: an initial response (possibly via gated measurement) would help to show you driver related issues, where the rest of the data (past the initial gating) would show you environment based issues. But, to me, this would have to be done outside the car for reasons stated before (hard to gate measurements when reflected sound is coming at you just behing, or even before, direct sound).

The one thing is that we can't really do much about comb filtering, etc, in the car... if we have a null, we have null. So, we'd use that kind of information to make the install the best we absolutely can... which gets back to my thread a few months ago regarding "how can we use science to make the best sounding car possible".



Oh, and +1 for you guys getting involved. I honestly thought this thread was going to turn into a 2 reply thread and get pushed off the cliff by a "what midbasses should I buy?" thread.


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## highly (Jan 30, 2007)

This is one of those 'meat' posts that I've been pondering starting since spending the winter reading up on acoustics. There may not be as many of these as there once were, but they are no less valuable for it!

My head is full of all sorts of things that I am trying to learn to balance in the real world. Theory gets squirmy when the rubber meets the road on much of this for me, so getting a better understanding of the relationships between the problems we face certainly helps when it comes time to make install decisions. What really matters, and how much influence does it have on the final product?

And, where this tread is concerned, what measurement tools do we have, what do they measure, and how do we interpret those measurements to better the end result?


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## ErinH (Feb 14, 2007)

SSSnake said:


> The fewer samples used to generate the FFT, the further apart the freqencies are so the lower the frequency resolution. For high frequency resolution we need to analyse long time periods of signals.



just a couple things...

So, how does the sample window fall in line here?
aRTA allows the user to set both sampling rate and window time, IIRC and I never quite understood (for lack of effort) why/how it mattered. It seemed to me like you'd want a longer window for low frequency measurements, and smaller window for high frequency content. Is that even valid?
Furthermore, how exactly does the FFT sampling rate play into this? Is it just another way of communicating the samples per second the system is collecting and you need more samples/second for higher frequency content than you do for lower? 
personally, this has never really "clicked" for me. You may have to give me a walk through at the GTG. 




SSSnake said:


> With what we call FFT measurements timing and signal content directly affect the results. The measurement signal and acoustic measurement must be synchronized so that you can extract timing information. The ability to gate the measurement allows to remove FR contributions from things such as reflections.


What befuddles me is the dayton omnimic setup doesn't include a way to loopback, so there's no way in extracting latency of the software/hardware and subtracting that from the response.
But, furthermore, does proper FFT really require a hardware loopback? If the software (and in this case, the dayton software does) shows an impulse, can't you simply gate your measurement (extracting anything before and after that isn't applicable to you) based off the impulse? 

Additionally, does one even have to worry about this hardware loopback if you want to get impulse response of each driver as a means to time align or check phase? 

IOW, why is that we need to have a loopback system for FFT measurements? This is another particular subject that doesn't quite make sense to me.





PS: If you guys don't stop with the info, I'm going to go into sensory overload.


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## michaelsil1 (May 24, 2007)

bikinpunk said:


> regarding Todd's waterfall post:
> My initial reply would be something along the lines of what I posted in the OP: an initial response (possibly via gated measurement) would help to show you driver related issues, where the rest of the data (past the initial gating) *would show you environment based issues. *


How would we get around environment based issues?


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## Patrick Bateman (Sep 11, 2006)

bikinpunk said:


> just a couple things...
> 
> So, how does the sample window fall in line here?
> aRTA allows the user to set both sampling rate and window time, IIRC and I never quite understood (for lack of effort) why/how it mattered. It seemed to me like you'd want a longer window for low frequency measurements, and smaller window for high frequency content. Is that even valid?
> ...


The loopback can help you pinpoint if the garbage that you are seeing on the screen is legitimate. For instance, the loopback is how I discovered that Windows was arbitrarily changing the sample rate of my measurements, which turns them into complete garbage if you don't catch it. (IE, I'd try to record a 44.1khz impulse and Windows would resample it to 48khz.)

It can also help you detect if there's clipping or weird delays and such.


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## highly (Jan 30, 2007)

> How would we get around environment based issues?


Just a few possibilities:

* Relocate the driver
* Change the polar dispersion characteristics of the driver
* Change to a driver with more favorable characteristics for the intended location
* Change the reflective properties of nearby environment
* Change the reflective properties of the distant environment
* Change the physical relationship between driver pairs


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## ErinH (Feb 14, 2007)

michaelsil1 said:


> How would we get around environment based issues?


Installation.

To me, this is prime area where one can test different install methods, rather than just throwing stuff in the doors and wondering why it doesn't work.

Edit: Todd hit it. I lump it all together into a single word. lol.


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## michaelsil1 (May 24, 2007)

bikinpunk said:


> Installation.
> 
> To me, this is prime area where one can test different install methods, rather than just throwing stuff in the doors and wondering why it doesn't work.
> 
> Edit: Todd hit it. I lump it all together into a single word. lol.


Oh!

Now that I've done all that what now! 


BTW

I did the throw it in and wonder why method; it was expensive.


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## ErinH (Feb 14, 2007)

michaelsil1 said:


> I did the throw it in and wonder why method; it was expensive.


I've done that about 5 times now. lol!


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## masswork (Feb 23, 2009)

bikinpunk said:


> just a couple things...
> 
> So, how does the sample window fall in line here?
> aRTA allows the user to set both sampling rate and window time, IIRC and I never quite understood (for lack of effort) why/how it mattered. It seemed to me like you'd want a longer window for low frequency measurements, and smaller window for high frequency content. Is that even valid?
> ...


Creating a loopback is very easy.
Just take the line out of the sound card, and assign the left channel to the DUT, and the right output goes straight to the right line input of the sound card.

DUT in car should be an AUX input, or processor input.
So, the left line out needs to be spliited (parallel) into 2 for left and right AUX in.

The left line input channel should be connected to the output of the mic, or pre-amp mic.

The software then compare the time arrival in right signal (that is latency in the laptop itself), and the time arrival in the left signal which is the DUT.
This way the software knows the actual delay in DUT.


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## ErinH (Feb 14, 2007)

I know what a loopback is. I just don't necessarily see the use for our case. 
I've been using loopbacks for a while, thanks to Chad making me a loopback cable for my mobile pre.


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## masswork (Feb 23, 2009)

Ah i see.
My experience is without loopback ARTA shows weird numbers that's always in the range of 6ms regardless of mic position (and distance to driver).

But with a loopback, everything looks perfect. The time arrival shows exactly like what we see, ie: there's a delay from every speakers.

Right, loopback isn't really necessary if we only want to convert impulse into frequency response.


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## masswork (Feb 23, 2009)

michaelsil1 said:


> I think in a car it is more important to get an average of several readings. I've heard this over and over again and I've found that it works very well.


Very true. Several readings at several angles/several mic position.
We really need to average those measurement to have a better look at high frequency content. Low frequency shouldn't change much.


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## ErinH (Feb 14, 2007)

masswork said:


> Very true. Several readings at several angles/several mic position.
> We really need to average those measurement to have a better look at high frequency content. Low frequency shouldn't change much.


I definitely agree. I actually made a thread discussing it.
http://www.diymobileaudio.com/forum...-rta-tuning-importance-spacial-averaging.html


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## masswork (Feb 23, 2009)

Hi Erin,
i wonder why you still need RTA when you have ARTA or maybe some other impulse-based measurement softwares?

RTA/pink noise is subject to external noise, while using sweep sine we can have it clearer.


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## masswork (Feb 23, 2009)

Umm, i just notice trueRTA has this quick sweep as the signal generator.
So it very looks like an impulse-based measurement but with RTA packaging


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## 14642 (May 19, 2008)

Hey, what midbasses should I buy?

I'm amazed by how many other posters' questions are actually answered in this thread in a discussion of measurement tools. I suggest that everyone reads this thread a thousand times and thinks carefully about all of their questions about, why doesn't this have a digital input, do I need a laser pointer to aim my midbass, should i use a tape measure to set time alignment, how do I make it sound like the bass is in the front, should I use 6dB per octave slopes instead of 24dB to avoid phase shift...and the list goes on and on. 

All of the answers are in this thread. It's like Where's Waldo?


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## ErinH (Feb 14, 2007)

masswork said:


> Hi Erin,
> i wonder why you still need RTA when you have ARTA or maybe some other impulse-based measurement softwares?
> 
> RTA/pink noise is subject to external noise, while using sweep sine we can have it clearer.


Honestly? It's just easier to me. I don't bother with impulse measurements since my days of smaart made me realize how much of a chore it is to do that in a car. 
True is pretty much plug and play RTA software and that's why I use it. 
The gui is much more to my liking especially in regards to averaging and overlaying multiple plots.


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## ErinH (Feb 14, 2007)

masswork said:


> Umm, i just notice trueRTA has this quick sweep as the signal generator.
> So it very looks like an impulse-based measurement but with RTA packaging





bikinpunk said:


> trueRTA, as it seems more like an FFT program but in RTA clothing ...


Agreed.


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## ErinH (Feb 14, 2007)

Andy Wehmeyer said:


> Hey, what midbasses should I buy?


ahhh. That's more like it.


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## masswork (Feb 23, 2009)

bikinpunk said:


> Honestly? It's just easier to me. I don't bother with impulse measurements since my days of smaart made me realize how much of a chore it is to do that in a car.
> True is pretty much plug and play RTA software and that's why I use it.
> The gui is much more to my liking especially in regards to averaging and overlaying multiple plots.


Yeah, RTA software is easier and it's very near real-time 

Smaart? A thing i don't like with smaart is the impulse chart is dBFs, not %dBFS. This means we can't see the polarity of the speaker we want to check.


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## masswork (Feb 23, 2009)

Andy Wehmeyer said:


> Hey, what midbasses should I buy?
> 
> I'm amazed by how many other posters' questions are actually answered in this thread in a discussion of measurement tools. I suggest that everyone reads this thread a thousand times and thinks carefully about all of their questions about, why doesn't this have a digital input, do I need a laser pointer to aim my midbass, should i use a tape measure to set time alignment, how do I make it sound like the bass is in the front, should I use 6dB per octave slopes instead of 24dB to avoid phase shift...and the list goes on and on.
> 
> All of the answers are in this thread. It's like Where's Waldo?


This is why i love DIYMA 

"Hey, what midbasses should I buy?" 
is 90% question i got in my local forum  Luckily many audio enthusiasts in my country are rich man (me excluded). They do buy Accuton, Micro Precision drivers that costs $10,000 !!! Some of them sound like s**t because they don't know what to do with them - they never measure.


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## SSSnake (Mar 8, 2007)

> So, how does the sample window fall in line here?
> aRTA allows the user to set both sampling rate and window time, IIRC and I never quite understood (for lack of effort) why/how it mattered. It seemed to me like you'd want a longer window for low frequency measurements, and smaller window for high frequency content. Is that even valid?
> Furthermore, how exactly does the FFT sampling rate play into this? Is it just another way of communicating the samples per second the system is collecting and you need more samples/second for higher frequency content than you do for lower?
> personally, this has never really "clicked" for me. You may have to give me a walk through at the GTG.
> ...


I haven't ignored the questions but most seem to centered on a particular products application of FFT/RTA. Give me some time and I will dig thru (removed TrueRTA after I lost my license and was going to have to re-purchase the program and I am not very familiar with the PE Omni setup). Besides, right now most of my free time is dedicated to de-bugging the MS-8 issues.


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## highly (Jan 30, 2007)

SSSnake said:


> I haven't ignored the questions but most seem to centered on a particular products application of FFT/RTA. Give me some time and I will dig thru (removed TrueRTA after I lost my license and was going to have to re-purchase the program and I am not very familiar with the PE Omni setup). Besides, right now most of my free time is dedicated to de-bugging the MS-8 issues.


Feel free to give me a call on the MS-8 stuff. Lord knows I've spend more than my fair share of time with it and have a pretty good feel for how to get it to do what you think you want it to!

If you don't still have my number handy just drop me an email and we'll go from there. Looks like Andy is watching as well, so you can probably bounce things off of him too...

-Todd


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## SSSnake (Mar 8, 2007)

Thanks Todd!

I am in discussions with Andy currently. The MS-8 is doing things I don't think most people would have seen. I will generate a thread to document the issues later but I don't want to put it out there until we get it fixed (I don't want anyone jumping to conclusions and bashing the MS-8 for no reason).

Andy,

If you are tracking this thread, check your inbox 

Thanks

Charles


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## 14642 (May 19, 2008)

Hey Charles,
I got your email. It's pretty easy for me to pop in here and whip out an answer. Your questions require more serious thought and a bit of reading, so it takes me a little longer. For me, the last week has been more like Whack-A-Mole than Where's Waldo, but I'll sit down tomorrow morning with a cup of coffee and try to figure this out.


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## ErinH (Feb 14, 2007)

Not necessarily pertinent to this particular thread, but since it does involve RTA/FFT software I mentioned earlier, here's a link to the Dayton Omnimic thread with some plots I managed to get quickly (while the baby was napping).

http://www.diymobileaudio.com/forum/diyma-sq-forum-technical-advanced/96060-dayton-omnimic.html


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## masswork (Feb 23, 2009)

Quick question for RTA users:
Do you subtract background noise? Is it required?

For example:
1. Quiet room. No sound detected by ear. Run RTA without pink noise. Speaker is quiet. RTA shows 100Hz +20dB and 200Hz 15dB.
2. In the same room, now we run RTA with pink noise. Speaker plays pink noise. Now RTA shows 100Hz +80dB and 200Hz 75dB.

Which one is correct from speaker/system POV:
1. 100Hz is +5dB louder than 200Hz
2. 100Hz is actually as loud as 200Hz


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