Archive for the ‘Sound Systems Book’ category

Sound Design Live eBook

July 25, 2013

I recently wrote the forward for an eBook that Nathan Lively has published called Sound Design Live: Build Your Career As A Sound Engineer. It’s a great read and I’ll include the forward below.

* * *

sound-design-live-ebook-coverAudio engineering technology has changed, but the work is still all about connections. Not through CobraNet®, Dante or AVB but rather the personal connections of telephones, email, social media and old-fashioned face to face. Connecting to creative artists, crew, managers, producers and audiences. If you are already in this field you are somewhere in this interconnected network. If you are wanting to get involved, welcome to the ultimate work in progress. This book is all about connections and why they are the most important, valuable and motivating forces in the industry. Nathan Lively plays the role of network hub and monitors traffic in this book to give you a glimpse into the absolutely real experience of our peers and mentors in this trade. The voices in this book have vastly different viewpoints, passions, and experience. Artists who use technology for self-expression, technologists who thrive on being a conduit for artists to reach their audience, and folks who have worn many different hats. If Nathan had brought them all together in one room to discuss audio, there would be at least as many passionate disagreements as points of concordance. Two points they would all agree on are the importance of having a passion for this field of work and attention to networking and relationships. So much of this field is serial monogamy, and therefore we must be careful to maintain good relationships, not burn bridges and keep that little black book up to date with all the folks you might want to see in the future. This book examines the relationship issues that are so important for getting into, and staying in, this business.

Nathan’s choice of speakers and topics provides a mix of information and experiences that I have not seen collected in one place. The viewpoints are refreshingly honest and free of the laundry lists of gear that characterize 90% of words written about this field. These are not all superstar designers with mega-million dollar projects. Dive bars, home studios, educators, starving artists and manufacturers are here as well. The contributors are very real people in whom it is impossible not to see a part of yourself, just as much as it impossible to not find a viewpoint that you had never considered or understood before.

Nathan’s “Sound Design Live podcasts” have brought these people’s voices to the internet and will continue to do so in the future. Here and now is a collection of assembled wisdom and experience that I believe will open your mind to the many ways you can expand your role in the audio professional network.

Sound System: Design & Optimization hits 10,000

September 13, 2010

Thanks for your support

I am grateful to announce that Sound Systems: Design & Optimization has reached the milestone of 10,000 copies sold. Around half of these are the English language editions and the remaining half split equally between the Spanish and Chinese language editions. Work is currently underway to translate the 2nd edition into Chinese. I want to extend my thanks to everyone that has supported this project, either by buying the book, promoting it to others, or helping me write it. Thanks especially to my editor Catharine Steers at Elsevier, all of my peer reviewers and those that contributed photographs and perspective pieces for the book. Thanks also to John and Helen Meyer, Gavin Canaan, Mac Johnson and all the staff there that continue to support my educational efforts through their sponsorship of my seminars and to all who have taken their valuable time to attend them. Thanks also to everyone at LiveDesign, Rational Acoustics, TC Furlong, and others that have helped promote this book.  Also to Ana Lorentz for translation of the Spanish Edition and to Magu for his help in that effort. Finally my highest gratitude goes to my wife Merridith who negotiated the deal and was one of just two people (along with Thorny) to read the whole book during its development.

When this was written I felt that less than 1250 books would be a failure, and anything more than 2500 would be a success. Reaching 10,000 in less than 4 years is far beyond my wildest dreams.

So thanks a million, I mean a ten thousand.



 In case you are interested: Here is what I did with the proceeds from the book: I crossed back over the art/science line to a 1978 Gibson Johnny Smith, and a 2009 Breedlove Bossa Nova.

1978 Gibson Johnny Smith

Breedlove Bossa Nova

New York Trainings – Updated with Pics

May 19, 2010

I am in NYC this week for two rounds of classes: SIM3 training in Brooklyn and the Broadway Sound Master class on the NYU campus in the East Village. The SIM class is at City Tech in Brooklyn – which is where John Huntington teaches. We have several of his students joining the class which is really nice. This trip marked a first for me – even though I have been coming to NYC quite regularly since 1984 this was the 1st time I ever set fot in Brooklyn. We arelocated over by the legendary brooklyn bridge and I can see it from the school – so hey – I can add another borough to my list……Manhattan, Queens – been 2 places there – La Guardia AND JFK – wow, and now two places in Brooklyn – City Tech AND Peter Luger – the famous steakhouse.

We are halfway through the class and pretty much right on schedule. Looking forward to the BSMC this weekend – always a great learning experience for me

****** John Huntington was kind enough to take some pics of the seminar.


Off to Mexico for AES and SIM3 Training – Updated

April 25, 2010

I have been busy putting together new material for SIM 3 training, and AES seminar and the upcoming Broadway Sound Master Class. You have seen some of the work in progress below, but I have had to push to get things ready for showtime.  Sorry for the delay in getting more things posted and for my lateness in response to Goran. Just pushing it right now – LOTS of really good stuff coming- phase circles galore but priorities………..

AES Expo

I was invited to give a talk for the AES at the Sound Check Expo in Mexico City

This is a big audio trade show in Mexico City. Lots of  levels of gear mixed together: Pro Audio, Music Industry Audio, Guitars, Pianos, Disco lighting , and the most popular event was getting the autograph of a hot young girl singer. I am sorry but I never made it to the front of the line. :-(.

I did a talk for about 1.5 or  hours and it was like giving a speech at the United Nations. There was a faint spanish language echo in the room when I spoke, about 500 ms delayed. It was a translator in a booth at the rear and everybody in the audience was listening on headphones. Wow, this guy was fast – and good, because I even got a few laughs at my jokes. I remember doing a translated seminar once in China. 4 days without a single laugh – until I tripped and fell down on the stage – the crowd loved THAT!   OK back to Mexico. The lecture was very well attended and it was a great honor to have so many people there. We covered alot of interesting topics including subwoofers steering and fun stuff like that. I was told that this was the best attended training session of the convention (about 120 people) and that felt really good. If only I could have gotten the singer girl to join me on stage we would have REALLY filled up the place!

Here are two pics from the seminar. The first one shows me at the podium. I don’t remember the bubbles floating around the room, but you can see them in the picture. The second one shows the view in the room.

AES Sound Check Expo

SIM3 Seminar

 Next on the agenda was Meyer Sound Mexico where we held a SIM3 Training. It was the usual 4 day session, but in Mexico City the sessions are marathons. Typically we go from 10 am to 7pm, but two of the days we went past 9 pm. The Mexico city schools are some of the most interactive of all the schools I do. The students are sendiing up a constant stream of interesting and challenging questions and we cover SO MUCH material. Sometimes the order in which we cover them is a bit crazy, but we cover TONS of topics.

Working with me on this seminar was Oscar Barrientos and Mauricio Ramire(el Magu). These are expert teachers in their own right so it is great to have them to translate and enhance. My typical style, when doing a translated seminar, is to (try) keep my talking short, to make quick switches to the translators. With these guys, because they know the subject so well, they can follow the concepts and even expand on what I say when they move it into Spanish. This helps speed things along alot – because typically a translator has no audio knowledge, does not know the terms, and certainly does not know FFT analysis. (In Korea once we had a translator QUIT at lunch time the 1st day – because she was too humiliated by all the students telling her she was translating all the audio words wrong. A student, Sean Cho, took over the job and saved the seminar).

Most of the students had been to Magu’s and/or Oscar’s training courses before and a few (Eduardo Brewer from Venezuela and Jorge and Juan Carlos Yeppes) had even been to my course before. It is the ultimate honor for me to have engineers return to my course.  The advanced level of the students helped us to move along at a very fast pace.  I am always very grateful about the way i am treated in Mexico. They are SO GOOD to me.

I also had the honor of meeting Luis Pinzon. He is the only person I know with 3 copies of my book – 1st & 2nd edition ingles, and 1st edition espanol. I happily signed all 3 for him. I wish I had brought a Chinese version to give to him. That would have completed the set!  . Luis also gave me his cable checker – which is quite amazing.

I was taken to some really nice restaurants by Antonio Zacarias and also  we went for Tacos to El Charco – which I highly recommend.  Also went to a Chinese restaurant in a shopping mall near Meyer Sound Mexico – I DON’T recommend this place, unless you want to die.  Funny though ….A week later I had Mexican food in China – The Mexican food in China was better than the Chinese food in Mexico, but Mexican in Mexico and Chinese in China worked out the best.

So here are some photos from the Mexico seminar, taken by Eduardo and Hermes. I think I have the names right on the class photo – if not help me out please. Also if you have some others – please send them to me.

Thanks for inviting me to Mexico, and I hope I can come back soon —actually I WILL be back in Mexico this Novemeber – but it  a cruise vacation – so I won’t be working 🙂

Until next time,

Hasta luego y Buena Suerte


Mexico SIM3 Class 2010

6o6 is getting some hands on experienceNow we know who was reading email during class

Yes this is supposed to make sense..........

Magu, Oscar, Paco and 6o6

Eduardo, Magu, Hermes, 6o6, Oscar and the bald guy


Uncoupled Array Design: Beginnings and Endings (Updated)

March 28, 2010

** Update:  A downloadable version of the calculator to do this work is available (courtesy of Daniel Lundberg). Go to the bottom of this post for preview and instructions. 


When a coupled array is assembled, its operating range is limited primarily by its power capability. Even very large arrays will congeal fairly quickly and once they have joined together let no phase tear them asunder. Wow! Not often that we can work hard-core religion language into speaker array theory (not to say there is not a lot of mysticism out there in line array theory land).  So coupled arrays, once joined, once fully formed will maintain there shape over distance, finally either running (literally) out of air, or into the wall.  

Uncoupled arrays are quite the opposite. They can’t wait to destroy themselves. The battle begins with each speaker owning  its piece or real estate close by, in front of it. As we continue forward we have a happy meeting with the neighboring speaker’s response. They greet with an in-phase handshake and we have a crossover, known as the unity line.  At this point the speakers are working together and the line that runs from speaker center to center (through the crossover) is approximately unity gain. This is exactly what we want to happen – an extended line of unity gain, wider than a single speaker. Ideal for frontfills, underbalconies, parade routes, racetracks and more. This is both a happy beginning AND a happy ending.  How so? The beginning part is obvious, but the ending part…………well what I mean here is that this beginning is the best response we will get. It is all downhill from here as the more distant areas directly in front of each speaker no longer have sole ownership of the coverage. The others speakers are spilling in and they are arriving late. VERY late in acoustic terms. The displacement between the speakers (a factor that is large in an uncoupled array) now creates a very rapidly changing variation of time offsets between the elements. The result is combing that moves rapidly down in frequency and becomes stronger with each step we go deeper into coverage.  

How far can we go before we throw up the white flag and surrender? One could evoke a variety of subjective answers such as: until it sucks, or until I can afford another set of speakers to take over etc., but these are not very satisfying to me. There is a verifiable milestone: three’s company. When we reach the point where the entire length of the coverage line is within the pattern of three sources we have reached full immersion into the combing. Three is a magic number. With three sources arrayed along a line, or an arc it is impossible to find a location that is equidistant to all three. This guarantees two or three arrival times from speakers operating within their coverage  angle. That is the fight I was talking about before. The only way to stop the fight is to drown it out with another much louder speaker – like a mains to take over for your frontfill, or stop it – like a back wall for your underbalconies.  

In my book I go through a set of design calculations for uncoupled line source and point source arrays. The variables are the coverage shape of the speaker (The Forward Aspect Ratio/FAR), the spacing, and the splay angle. From these we can determine where the coverage will start (D unity) and where the coverage should end (D limit). If you know the speaker and where your audience starts, you can determine the spacing, and where you will need to connect to the mains. If you have fixed positions you can get the right speaker model etc.  

An example reference chart using a 80 degree speaker in an uncoupled line source is in the book.  This shows nicely how to solve for this particular model and then one can refer back to the FAR chart to get the angle/FAR conversion for other speakers.  

Uncoupled line source design reference for an 80 degree speaker

Design procedure for the same speakers as above

Another example reference chart uses a 90 degree speaker in an uncoupled point source source in the book.  In this case the splay anglwe variable is added to the equation.  

Design reference for 90 degree speaker in an uncoupled point source array

Design Procedure example for a 90 degree speaker

It is not possible to put an XL file into the book and not practical to give a separate chart for each speaker angle/spacing etc.  but folks that bought the book don’t have a working calculator/spreadsheet that they can go to on their computer so I was in the process of making one for the blog and then Daniel Lundberg contacted me with his calculator based on this same concept. Whereas mine was derived from observing the trends and behiavior of many, many, array interactions, Daniel’s goes to the heart of the trigonometry involved.  

So over the past few days I ran through some different models of speakers athdifferent angles and spacing to check for consistency between a) my published values derived through observation of other speakers at other angles  

b) Daniel Lundberg’s values derived through trigonometry and geometry  

c) what we can see on the MAPP plots now  

The good news is they are all in very close agreement.  The largest discrepancy is in the limit values for the longest range, and even these are relatively close.  

Comparison of observed and mathematically derived values 45 deg speaker with 4m spacing and 0 deg splay 45 deg speaker with 4m spacing and 4.5 deg splay

45 deg speaker with 4m spacing and 9 deg splay 45 deg speaker with 4m spacing and 13.5 deg splay

45 deg speaker with 4m spacing and 18 deg splay

45 deg speaker with 4m spacing and 22.5 deg splay


Here is what the downloadable version of the calculator to do this work looks like (courtesy of Daniel Lundberg). You can have a copy of it. Free. 

HOWEVER, the security rules of this blog host prohibit me from posting an XL file. 

Therefore, if you want a working copy of this calculator, you will need to send me an email request to If you think this is just a trick to get you on my mailing list…………


Phase alignment of spectral crossovers

March 11, 2010

This is a continuation of the impulse response subwoofer thread. Here are some screens from Sounds Systems: Optimaztion and Design that deal with phase alignment of crossovers.

The figure above shows the phase responses brought together between sub and main.  The crossover is not steep and therefore the phase response overlap over a wide range.

The LF and HF components of a 2-way xover

The combined response of a 2-way xover

Line Array Gain Taper (Breaking the Line II)

March 10, 2010

After Bob Gardam’s comment on Breaking the Line I decided to give a quick go of the hypothetical scenario I had proposed in my reply:

12 boxes,  6 @ 2 degrees @ 0 dB, 3 @ 4 degrees @ -2 dB, 3 @ 8 degrees @ -4 dB.

I made screen dumps of 3 scenarios:

1) 0,-2, -4 dB taper – the system as it would be if operated below limiting

2) 0, 0, -2 dB taper – the system as it would be if the top section only was limiting

3) 0, 0, 0 dB taper – the system as it would be if the whole system was limiting – or if there was no gain tapering

As expected the compression reshapes the HF range most noticeably. The honed agular shape – longer throw for the uppers – becomes rounded so that the relative level in the near areas goes up.  This is most easily seen in the 4 kHz response because there is minimal fingering, but is also present in the 8 k Hz response.

The 1 kHz response carries on the trend in a similar – yet reduced fashion. Notice that the main frontal lobe is barely affected. Waht you see is an increasing bulge in the underside heading toward the near seats.

The 250 Hz response requires a very careful look to spot the changes. Two things are happening. As the taper is reduces by compression the additional coupling of the lower boxes steers the main lobe downward by a whopping 1 degree.  Not exactly a game changer. People who can hear that in their system should check out the products of Acoustic Revive.  The other change is that the beam has narrowed very slightly. This can be seen by the markings I made of the original shape. The mechanism causing the narrowing is the same as the downwaqrd beam steer – increased summation causes both.

So here is your choice – loud HF in the front ALL the time, or only when we drive it limiting. Bear in mind also that the limiting simulated here is a brick wall. Actual limiters will be more forgiving so results in the field would be somewhat LESS than that pictured.


A great honor – and great company

March 5, 2010

I am humbled and pleased to announce that I have been included in a list of “50Powerful People” in the field of entertainment technology by LiveDesign Magazine. It is a HUGE honor for me to be included in this list with the likes of Abe Jacob, Jonathan Deans, Mark Holden, John Meyer, Scott Lehrer, Tom Clark, Mark Menard, Nevin Steinberg  in the audio filed as well as the big names of lighting, staging and projection. My thanks go out to the folks at LiveDesign and I hope I can live up to this award.

Here is an an explanation of the list:

Here is the link to the full list:

Phase Alignment of Subs – Why I don’t use the impulse response

February 8, 2010

OK. The saga continues down below……………………….

What is the best way to phase align our subwoofers to the mains? There is a hint in the way the question was phrased. I didn’t say time align (and it is not because I am afraid of copyright police). I say phase align because that is precisely what we will do. Simply put, you can’t time align a subwoofer to the mains. Why? because your subwoofers are stretched over time – the highest frequencies in your subwoofer can easily be 10-20 ms ahead of the lowest frequencies. Whatever delay time you choose leaves you with a pair of unsettling realities: (a) you are only aligning the timing for a limited ( I repeat LIMITED) frequency range, and (b) you are only aligning the timing for a limited ( I repeat LIMITED) geographical range of the room. So the first thing we need to come to grips is with is the fact that our solution is by no means a global one. There are two decisions to make: what frequency range do we want to optimize for this limited partnership and at what location.

Let’s begin with the frequency range. What makes the most sense to you? 30 Hz (where the subs are soloists) , 100 Hz (where the mains and subs share the load) of 300 Hz (where the mains are soloists)?  This should be obvious.  It should be just as obvious that since we have a moving target in time, that there is not one timing that can fit for all.

Analogy: a 100 car freight crosses the road in front of you. What time did the train cross the road? The answer spans 5 minutes, depending on whether you count the engine, the middle of the train, or the end. Such it is with the question: when does the subwoofer arrive? (and is also true for when does the main arrive?) How do we couple two time-stretched systems together? In this case it is pretty simple. We will couple the subwoofer train right behind the mains. The rear of the mains is 100 Hz and the front of the subs is the same. We will run the systems in series. The critical element is to link them at 100 Hz. (I am using 100 Hz as an example – this can, and will vary depending upon your particular system).

The procedure is simple. measure them both individually, view the phase and adjust the delay until they match. You have to figure out who is first and then delay the leader to meet the late speaker. This will depend upon your speaker and mic placement. I say this is simple – but in reality , it is quite difficult to see the phase response down here. Reflections corrupt the data – it is a real challenge.  Nonetheless, it can be done. It’s just a pain.

When I get a moment I will post up some pics to show a sub phase-align in the field. 

Wouldn’t it be nice if there was a simpler method? Like using the impulse response to get a nice simple answer directly in milleseconds, instead having to watch the fuzzy phase trace.  It is absolutely true that the impulse response method is easier.  In my next post I will explain why the easy way lacks sufficient accuracy for me to ever use with a client.

******************  Part II *****************************************

FFT measurement questions and answers

The first thing to understand about an impulse response is that it is a hypothetical construct. This could, to some extent, also be said about our phase and amplitude measurements, but it is much more apparent – and relevant with an impulse response.

The response on our analyzer is always an answer to a question. The amplitude response answers the question: What would be the level over frequency if we put in a signal that was flat over frequency. This is not hard to get our heads around. If we actually put in a flat signal (pink noise) we would see the response directly in a single channel. If not, we can use two channels and see the same thing as a transfer function. This makes it a hypothetical question- what would the response be with a flat signal – even if we use something like music.

Same story with phase but this gets more complex. Seen any excitation signals with a flat amplitude AND phase response?  You won’t find that in your pink noise. Pink noise achieves its flat amplitude response only by averaging over time. Random works for amplitude – but random phase – yikes – this will not get us any firm answers. In the case of phase we need to go to the transfer function hypothetical to get an answer – the phase response AS IF we sent a signal with flat phase in it. Still the answer is clear: this is what the system under test will do to the phase response over frequency.

Impulse response

The impulse response display on our FFT analyzer answers this question: what would be the amplitude vs. time response of the system under test IF the input signal was a “perfect” impulse.  Ok……….. so what is a perfect impulse?  A waveform with flat amplitude AND phase. That can’t be the pink noise described earlier, because pink noise has random phase. So what is it?  A single cycle of every frequency, all beginning at the same time. Ready set, GO, and all frequencies make a single round tripand stop. They all start together, the highest freq finishes first, and the lowest finishes last. If you looked at this on an oscilloscope (amp vs time) you would see the waveform rise vertically from a flat horizontal line, go to its peak and then return back to where is started.

IF the “perfect” impulse is perfectly reproduced it will rise and fall as a single straight line. The width of the line (in time) will relate to the HF limits of the system. The greater the HF extension, the thinner the impulse. As the HF range diminishes, the shortest round trip takes more time, and as a result the width of the impulse response thickens as the rise and fall reflect the longer timing. A system with a flat phase response has a single perfect rise and fall in its impulse response and a VERY important thing can be said about it: a single value of time can be attributed to it. The train arrives a 12:00 pm. All of it. 

The impulse response on the FFT analyzer is not an oscilloscope. We do not have to put in a perfect impulse. We will use a second generation transfer function, the inverse Fourier transform (IFT) , which is derived from the xfr frequency and phase responses. This is the answer to the hypothetical question: what would the amplitude vs time response be IF the system were excited by a perfect impulse. 

If the system under test does not reproduce the signal in time at all frequencies, then the impulse response shape will be modified. Any system that does NOT have a flat and amplitude and phase response will see its impulse response begin to be misshapen. Stretching and ringing, undershoot and overshoot will appear around the vertical peak. Once we are resigned to a non-flat phase response we must come to grips with the fact that a single time value can NOT describe the system. The system is stretched. The time is stretched. The impulse is stretched.

This is where the FFT impulse response can be misleading. We can easily see a high-point on the impulse response, even one that is highly stretched. Our eyes are naturally drawn to the peak – and most FFT analyzers automatically will have their cursors track the peak – and lead us to a simple answer like 22.4 ms, for something that is stretched 10ms either side of that. And here is where we can really get into trouble: we can nudge the analyzer around to get a variety of answers to the same question (e.g. the same speaker) by deciding how we want to filter time and frequency: ALL OF WHICH ARE POTENTIALLY MISLEADING BECAUSE NO SINGLE TIME VALUE CAN DESCRIBE A STRETCHED FUNCTION.

Did I mention that all speakers (as currently known to me) are time stretched?  So this means something pretty important. The simplistic single number derived from an impulse response can not be used to describe ANY speaker known (to me) especially a subwoofer.

Does a stretched impulse response tell you what frequencies are leading, and by how much? Good luck.  You would have a better chance decoding a German Enigma machine than divining the timing response over frequency out of the impulse. This brings us back to the heart of the problem with our original mission: we are trying to link the low frequencies of the main speaker (100 Hz) to the high frequencies of the subwoofer (100 Hz). The peaks of these two respective impulse responses are in totally different worlds. They are both strongly prejudiced toward the HF ranges of their particular devices which means the readings are likely to be the timings of 10 kHz and 100 Hz respectively. 

Simple answers for complex functions. Not so good.  That’s it for the moment. Next I will describe some of the different ways that impulse responses can be manipulated to give different answers and when and where the impulse response can provide an accurate means of setting delays.

********************** Part III ***********************************************

The linear basis  of the impulse response

 Those of us using the modern FFT analyzers that are purpose-built for pro (and amateur) audio have been spoiled. We have grown so accustomed to looking at a 24 or 48 point/octave frequency response display that we forget that this is NOT derived from logarithmic math. The FFT analyzer can only compute the frequency resp0nse in linear form. The quasi-log display we see is a patchwork of 8 or so linear computations put together into one (almost) seamless picture.  Underlying this is the fact that the composite picture is made up of a sequence of DIFFERENT time record lengths. Bear in mind that the editing room floor of our FFT analyzer is littered with unused portions of frequency data. We have clipped and saved only about half the freq response data from any of the individual time records.

How does this apply to the impulse response? Very big. The impulse response is derived from the transfer function frequency response (amp and phase). It is a 2nd generation product of the linear math. The IR is computed from a single frequency response – from a single time record – which means it comes from LINEAR frequency response data.  The inverse fourier transform (IFT) cannot be derived from the disected and combined slices  we use for the freq response. The IR cannot contain equal amounts of data taken from a 640 ms, 320 ms, 160 ms…. and so on down to 5ms  to derive it response. Think it through……… there is a time axis on the graph. It has to come from a single time event.

The IR we see comes from a single LINEAR transform. The importance is this: linear data favors the HF response. If you have 1000 data points, 500 of them are the top octave, 250 the next one down and so on. This means that our IR peak – where the “official” time will be found, is weighted in favor of the highest octave. If you have a leading tweeter, The IR will find it ahead of the pack (in time and level). The mids and lows will appear as lumpy foothills behind (to the right) of the Matterhorn peak. If you have a lagging tweeter, the IR will show the lumpy foothills ahead of the peak (to the left), but the peak will still be the highest point.  Our peak-finding function will still be drawn to the same point – the peak.

Now consider a comparison of arrival between two speakers – if they both extend out to 16 kHz (mains and delays) then the prejudice of the IR in favor of the HF response evens out. If we find the arrival time for both we can lock them together. Their response will be in phase at 16 kHz and remain in phase as we go down – (TO THE EXTENT THAT THE TWO SPEAKER MODELS ARE PHASE MATCHED).  This is a PARALLEL operation. 10kHz is linked to 10 kHz and 1k to 1k and 100 to 100 for as long as they share their range. If the speakers are compatible, one size fits all and the limitations of the IR are even on both sides of the equation. If they are not compatible over frquency, we will need to see the PHASE response to see where they diverge, and solutions enacted within this viewpoint. – later on that.

Now back to the subs…………

It should be clear now that the linear favoritism over frequency will NOT play out evenly in joining a sub to a main speaker. This is also true of aligning a woofer and tweeter in a two-way box. This problem holds for ANY sprectral crossover tuning. Linear frequency math does not have a and fair and balanced perspective over frequency. If you are looking at devices with different ranges they are subject to this distortion.  The location of the peak found in our IR is subject to the linear focus. If the main speaker is flat the peak will be found where there are more data points: the top end – 4 to 16 kHz. All other freq ranges with appear RELATIVE (leading or lagging) to this range. If you have a speaker that is similar to 100% of the speakers I have measured in the last 26 years, then one thing is certain: the response at 100 Hz is SUBSTANTIALLY behind the response we just found at 8 kHz.

The sub is NOT flat (duh!!) so there is a tradeoff game that goes on in the analyzer. As we lose energy (frequency rising) we gain data points (liner acquisition)  so the most likely place the peak will be found is in the upper areas of the subwoofer range and/or somewhat beyond,  before it has been too steeply attenuated.  If you have a subwoofer that is similar to 100% of the speakers I have measured in the last 26 years, then one thing is certain: the response at 30 Hz is SUBSTANTIALLY behind the response we just found at its upper region.

One of the reasons I have heard given as the reason to use the IR values alone to tune sprectral crossovers (subs+mains, or woofer+tweeter) is that the IR gives us “the bulk of the energy” for each driver and aligning “bulk of the energy1+bulk of energy2 = maximum bulk of energy.”  Sounds good in text. But it does NOT work that way. You are making a series connection at a specific freq range, not a parellel connection (where bulk might apply). Futhermore, the bulk formula is flawed anyway – because the linear freq nature if the IR means that the two “bulks” are weighed differently.

********************** Part  IV ******************

There are a variety of ways to compute an impulse response on an FFT analyzer. All of them haqve an effect on the shape of the response, how high the peak goes, and where (in time) the peak is found. Without going hard into the math we can look at the most decisive parameters.

VERY SIMPLIFIED IR Computation Features

1) The length of time included after time zero (the direct sound), in seconds, milliseconds etc.:  This differs from the the actual time record captured, since there is positive and negative time around time zero – but the math there is not important . In the end we have a span of time included in the computation.  This puts and end-stop on our display – we can’t see a 200ms reflection if we have only 100ms of data after the direct sound. We could, however choose to display less than the full amount of data we have. The visual may be a cropped version of the computation, or it could be the full length.  The capture time also limits how low we can go in frequency. We can’t see 30 Hz if we only have 10ms of data. Most IR response have the option of large amounts of time, so getting low frequencies included will not be a big issue. The fact that the frequency response is LINEAR means that frequency weighting favors the HF – no matter how long – or short our capture is.

2) Time increments/FFT resolution/sample freq:  How fine do we slice the response in time. The finer the slices, the  more detail we will see. More slices = higher frequencies. If we have slice it into .02 ms increments (50 kHz sample rate) we can see up to 25 kHz. If we slice at lower sample rates, the frequency range goes down. The same speaker, measured over the same amount of time, with different sample rates/time increments will include different frequeny ranges – and therefore MOST LIKELY will have its impulse peak centered at a different time. This is important. The speaker did not change, but our conclusions about it did. This is a non-issue if we are comparing two speakers that each cover the same range – they would both have the same shift applied to them. But if we have one speaker with a full HF range and one without the playing field just got tilted. If one speaker really has no HF, and the other one does – but it is filtered by the anaylzer, then we can assume that synchronizing the two peaks will put the speaker in phase.

Vertical scale: Linear/Log:  The uncultured version of the IR is linear in time, freq and level. This means that things that go negative will peak downward while positive movement goes upward. Polarity (and its inversion) can be seen. The down side of this is that the linear vertical scaling translating vewry poorly visually toward seeing the details of the IR such as late arrivals, reflections, etc. Worse yet is trying to discern level differences in linear. The Y axis does not read in dB. It reads in a ratio and this has to be converted. Upward peaks have a positve value and downward have a negative value. The strength of an echo can be computed by the ratio of the direct to the echo levels – and converted by the log 20 formula into dB. Where it strange is when you try to compute positive direct sound to a negative going reflection.

The log version is obtained by the Hilbert transform and shows the vertical scale in dB. But the downside is that there isn’t a downside. Pun intended. What I mean is that the negative side of the impulse is folded over with the positive and these are combined into a single log value. This can now be displayed in dB since everything is going one way. This has various names: Energy-time-curve (ETC) amoung others. The visual display is blind to the polarity but I am told by sam Berkow that the cursor in SMAART shows whether or not the energy is positive or negative – even though it all displays positive.


So once again we are back to the same place. If you are going to use the impulse response alone (I say you because it will not be me) to align speakers in different freq ranges you are prone to computational items that will affect the HF and LF sides of the equation differrently. One technique I have seen advocated is the push down the sample freq so low that the upper regions of the HF speaker are filtered out. The idea is this: if the Xover is 100 Hz, then drop the resolution of the analyzer down to filter out the region above 100 in the HF speaker. Then we will see the impulse response at 100 Hz of BOTH speakers – and VOILA we have aour simple answer.  BUT – one impulse response (the HF) has filtered the device by computation – the other (the LF) is filtered by a filter. We have a merger of  the VIRTUAL – a computationally created phase shift and freq response filtering (which we don’t hear) with an actual – the filter response of the Xover.  It is possible that the value for the impulse will give the correct reading so that the Xover is actually in phase – possible – not probable – but we won’t know until we measure the phase – which is the whole point of this exercize.

Simply put: why bother with a step-saving solution ( Xover alignment by IR)  if it is so prone to error that you have to do the second step (Xover alignment by phase) any way? If a step is to be skipped it is the IR – not the phase.


Cardioid Subwoofers

January 27, 2010

A hot topic of discussion is cardioid subwoofers, so this will be the place to get that topic going. At the moment I will use this as a test to see if I can upload a few graphics from the book here.  The first will be some pics that describe the behaviour of end-fire arrays.

The above figure shows the timing chain for a set of four speakers in the end-fire configuration. The basic principle is a game of acoustic “leap frog” where the rearmost speakers jump sequentially over the front unit. The timing is set up so that all four speakers are synchronised at the front of the 1st (the rightmost in the figure) speaker. This “in phase” situation causes the signal to sum additively in this forward direction. The phase angles shown at the right for 3 different freq ranges are color coded to reflect their position on the phase cycle (green = +/- 90 deg, yellow = 90-120 deg and red = >120 degrees. The key item here is NOT the exact phase angle, but rather the amount of AGREEMENT in phase between the 4 speakers. In this case 31 Hz shows perfect agreement at 98 degrees so the addition wiull be strong. 63 Hz shows 4 speakers all synch’d at 198 deg and will achieve the same effect.  

Meanwhile in the rearward direction (shown on the left ) the timing chain reveals four speakers out of time as they move over the rearmost speaker. The 4 elements are all at different times and range over 17.4 ms apart. The phase responses also fall apart – ranging from a 1/6 cycle (65 deg at 31 Hz) to 2.16 cycles (125 Hz) and all sorts of values in between. These disparities in phase values cause the amplitude response to sum very poorly in the rearward direction, the intended result of the design.  In sync at the front – scrambled at the rear. The side directions fall somewhere between and the end result is shown in the 3 polar plots at the bottom of the chart. 

This figure shows an alternate spacing/timing configuration. Instead of a constant spacing (as the upper version shows), with a consistent delay timing, this config has a log spacing – and log adjusted timing. The leap-frog game is still played the same at the front – everybody in sync at the front cabinet, but thing play out somewhat differently at the rear. The difference is small but illustrates the options we have available to us.

This 3rd figure has a different twist to it. In this case the intent is NOT to have perfect synchronicity at the front of the array. Instead the timing sequence is set up so that they are slightly off – such that there is about a 90 degree spread at 125 Hz, 45 degree spread at 63 Hz etc.  This creates a less than perfect addition at front/center – but causes a better addition at the front corners. The result  is a flattened front and an overall triangular shape.  This configuration was shown to me by Mitchell Hart way down in Australia.