How to break into the professional audio industry – Find and keep your first video game jobs and recording studio jobs.

How to break into the professional audio industry – Find and keep your first video game jobs and recording studio jobs.

If you are reading this then you likely aware of how difficult it can be to get your start as a professional audio engineer. Whether you want to work in recording studios or the games industry, breaking in takes more than just talent and dreams. In this post I’ll give you a lot of tips on how to get your big break and stay employed once you do.

Free Impulse Responses from the iPad, iPhone and Behritone speakers

Free Impulse Responses from the iPad, iPhone and Behritone speakers

Impulse Responses for iPad, iPhone and Behritone speakers
Impulse Response for iPad, iPhone and Behritone speakers

I have worked on a few iOS games lately and realized how useful it would be to simulate the iPhone 4S and iPad 2 speakers. It can be a real pain in the neck to transfer sounds to your iOS device when testing or waiting for a new build to see how your sounds will work on each devices speakers. Well, I decided to sample the impulse response from each device to do just that!

These impulse responses will work in most convolution reverbs as long as they can import .wav files. Make sure you change it to mono to really simulate what will happen! I’m using Izotope Trash 2 to do this on my computer, but you can use any software that allows importing impulse responses and summing stereo to mono signals.

So, what can you use these for? I’m glad you asked!

Any sound designer or developer working on game audio can use this on a master audio bus in their DAW to simulate how an iPhone or iPad will alter their sounds. Anyone making music for portable devices like smartphones can do the same and get an idea of what frequencies will jump out on these small speakers! If you work in post you could use them to simulate a conversation on a smartphone or music through one of these devices.

Pretty useful eh?

By the way, if you want to make your own impulse responses for your convolution reverbs using a free tool check out one of my other posts for a clear tutorial.


BONUS: I created an impulse response for my Behringer Behritone speaker. This is a grot box that I use for mono testing and small speaker compatibility checks. It’s kind of a cheap Auratone speaker, which is funny considering Auratone was designed as a cheap speaker for contrast… but I digress…


CLICK HERE TO DOWNLOAD THE iPad iPhone and Behritone Impulse Responses


Please let me know if you found these impulse responses useful and I will make some more 🙂 If you want some other impulse response I’d love to hear about it in the comments below!
All product names used in this webpage and download are trademarks of their respective owners, which are in no way associated or affiliated with Aaron Brown. These trademarks of other manufacturers are used solely to identify the products of those manufacturers whose tones and sounds were sampled during impulse response capture.


Thanks for visiting the Aaron Brown Sound blog! Come back soon for more posts about video game audio, audio engineering, sound design, composing and all other things relating to being an audio professional 🙂

Studio Monitor A/B test

I recently had to opportunity to work with the very talented sound designers at Naughty Dog on Uncharted 3! During this project I was using Dynaudio Air Series speakers exclusively. These speakers sounded amazing. They were accurate, natural, and non-fatiguing (which is critical when you work long hours!). I LOVED these speakers.

This made me realize that I hadn’t heard quite a few brands of speakers out there. I began to wonder what I could be missing. Don’t get me wrong, in the past I’ve used monitors that range from Behringer Truths and Alesis MKIIs to Yamaha NS10s and Genelecs. However, many brands like Adam, Dynaudio and JBL have come out with some amazing speakers lately.

I talked with my good friend, a fellow audio professional named Preston Smits, and we decided to set up a listening session. I called around and eventually found a Guitar Center in Arvada that had many of the speakers I am looking to buy. Warren in pro audio took great care of us.

We decided to compare speakers ranging from $1000 to about $2500 that were available. We also threw in the Behringer Truth as a low range comparison speaker. The results were ear opening to say the least!

Here are the details of the shootout:


  1. Behringer Truth (ribbon tweater version)
    1. $500/pair
  2. Dynaudio BM5a
    1. $1000/pair
  3. Adam A7x
    1. $1400/pair
  4. JBL LSR4328P
    1. $1550/pair
  5. Genelec 8040a
    1. $2400/pair

Note: We wanted to listen to the new Focal Solo 6 speakers, but no one keeps them in stock 🙁

I compiled a CD made of .flac files from a variety of sources and genres. This included songs from John Williams, ACDC, The Wallflowers, Peter Gabriel, Tom Petty, Daft Punk and Skrillex. These songs all demonstrated different qualities of the speakers which was essential to REALLY understand how each speaker sounds.

While listening to the CD we flipped between each speaker taking notes on things like the speakers’ depth, frequency response, detail, transient response, and if we thought the speakers would be fatiguing. We didn’t compare notes or tell each other what we were thinking while listening so we wouldn’t influence each others decisions.

After listening to 6 songs over and over we compared our notes. After 2 hours of listening to these speakers our notes were about 95% the same. We both heard the same qualities of the speakers.


Enough with the details. You are probably wondering which speakers won the shootout…. Unfortunately, professional audio gear eventually comes down to a matter of personal taste rather than having one clear victor. That being said we ended up picking two of the speakers as our definite favorites. Read on to find out which speakers we loved the most!


Smooth, detailed, clean, 3d sound with a high end clarity better than any other speaker I’ve ever head. When ADAM extended their response up to 50kHz. it was NOT a gimmick. I’m absolutely sold on these speakers. They may need a sub if you do heavy music genres that extends below 50 Hz.

JBL LSR4328:

Punchy, accurate, wide imaging, high transient response with excellent low end response. These had the best low end response. These speakers were the most accurate on the transients of the hi-hat/shaker/sibilance region of the frequency spectrum.

Genelec 8040a:

Crisp (in a bad way when compared), forward, scooped sound (lacking mid range detail), big low end detail. Sounded very inaccurate when compared to the JBL or ADAM speakers.

Dynaudio BM5a:

Boxy low end, collapsed image, and dull when compared to the other speakers. These were a disappointment to me. Their low end sounded boxy with poor transient response. It almost sounded like their speakers weren’t getting enough power to be driven clean in the low frequencies. These did have better detail than the Behringer speakers, but they weren’t the big step up to pro speakers I had hoped them to be.

Behringer Truth (Ribbon):

Collapsed image, least detail of the bunch, ok transient response. Cheapest sound of the bunch as well it should be at only $500 for the pair 😉



You really couldn’t go wrong with either of these speakers. The ADAM A7X had a smoother sound whereas the JBL LSR4328P had a more transient sound. The ADAMs did extend higher and give more detail on the highest frequencies (which is VERY pleasing to the ear I might add) while the JBLs give you more detail in transient things like high hats, snare transients and the like. The ADAMs could turn out to be less fatiguing. The JBLs have a wider dispersion of sound. The JBLs also offer calibration utilities to deal with different rooms. The ADAMs have no such option.

It all comes down to preference when you hit this level, but I can’t imagine anyone disliking either of these excellent sounding speakers. Personally, I liked the ADAM A7X the most. It would depend on your own tastes and applications.

In my opinion, both of these put every Genelec I’ve heard to shame in terms of detail, accuracy, imaging, and more.

2nd place: Genelec 8040a

The Genelec’s sounded a bit harsh, crisp, and scooped in the mids in comparison. They did offer a nice low end response and good transient detail. I wasn’t shocked at this. I’ve used many varieties of Genelec speakers in the past and never loved them.

This test was PROOF that there are better speakers out there for less money. Don’t believe me???? Go do your own test and tell me what YOU hear 🙂 Both the ADAM and JBL speakers offered a more detailed sound that filled the frequency spectrum. YMMV of course!

3rd place: Dynaudio BM5a

This was a disappointment to me. I have read many things online saying these speakers compare to ADAM speakers. I’m hear to tell you that the ADAM A7X is FAR superior to the Dynaudio BM5a. The Dynaudio sounds boxy in comparison and the transients in the low end just don’t come through clean at all. I’d avoid these speakers at this price. To be fair, I did really enjoy my Dynaudio Air Series speakers in the past so perhaps this is just their attempt at a low end speaker.

4th place: Behringer Truth (Ribbon)

Yeah, no big surprise here. This low end speaker was more of a control and certainly proved the difference between a $500 pair of speakers and a $1400 pair of speakers. These are a decent option for beginners with small budgets, but do yourself a favor and get a great pair of speakers if you can find any way at all to afford it.


Some people say that as long as you “know” your speakers you can mix with them. I believe there are limits with this type of thinking. Low end speakers lack TONS of detail when compared with high end speakers. Bottom line: If you can’t hear what’s missing or going wrong then you can’t fix it. The detail provided by a good pair of speakers affects EVERY PART OF YOUR SOUND! Without them you can’t trust yourself when mixing, tracking, or editing!

Speaker technologies have come a long way. There is a large market of studio monitors out there and never enough time to compare them. However, now that I have taken the time to A/B these speakers I have truly found what I’ve been missing in my home studio. I bought a pair of Adam A7X speakers and I’m 100% satisfied!

Digital Microphones

Digital Microphones


Digital microphones are far from a new thing.  Neumann has been developing the technology for over 9 years.  Now many more companies are showing off digital microphones.  In fact, it seemed the trend at AES 2010 was digital microphone technology.   I used to think of digital microphones as either a fad or unnecessary.  Now I believe they absolutely have their use and will eventually find their way into my normal workflow.  I invite you to read on and you may find yourself a believer in this emerging technology as well.

If you aren’t familiar with digital microphones they do have a few complications that don’t exist with analog microphones.  Digital microphones use a format called AES 42.  This isn’t a typical AES/EBU signal and requires a device to interpret the signal to normal AES connections.  If you want to invest in digital microphones it will cost you extra to accept this type of signal and there isn’t a large variety of devices capable of handling their signals yet.  Prior to this years AES I thought all that hassle wasn’t worth what they might offer an audio professional.  That being said, I now think that in many situations the pros far outweigh the cons.

This year AES had lots of digital microphone technology from Neumann, Schoepps, Line 6 and other companies.  I also got a special presentation from Neumann about their digital technology as well, but I will talk more about that later.

Schoepps introduced a shotgun microphone that uses a microphone on the back to further reduce off axis rejection by a whopping 16dB!  Obviously this is technology that traditional analog microphones can’t achieve.  By reducing off axis sounds the shotgun mic is even more directional without any coloration at all that can alter the most directional of microphones.

Schoepps digital shotgun microphone

Line 6 introduced digital wireless technology using digital microphones.  By plugging in a digital microphone to digital wireless you can send full frequency content along a carrier instead of a squashed lossy analog signal.  The result is, in theory, a lossless wireless microphone at a distance of up to ~300 ft.  Digital wireless also send 4 of the same signal and checks them all later to be sure the signal hasn’t been altered by drops or interference!




As a sound designer this really caught my attention.  By using digital lav microphones or condensors I could record sounds on location in full quality.  Imagine putting a lav microphone on a skate board as a professional rides around a skate park or a remote condenser that allows you to control it’s polar pattern and pre-amplification remotely.  The Line 6 tech is limited to a lav and a handheld mic at the moment, but the lavalier can also use a Schoepps capsule.  I look forward to seeing where this particular tech goes.

A valuable part of digital microphones is that their signal is converted immediately and is therefore unnaffected by long cable runs.  This may not make a bit difference in a perfectly wired studio, but imagine a cable run of 200 feet in a live venue or even longer in a broadcast situation.  Now audio professionals can achieve full quality sound at great distances.

I was fortunate enough to attend an event sponsored by Neumann Berlin and Sennheiser USA on digital microphones at SkyWalker Sound on Sunday during AES 2010.  It was here that I really got to hear how digital microphones are used by todays professionals.  We also got to witness Leslie Ann Jones do a mic shootout of analog Neumann microphones and their equivalent digital microphones.  It all started with an informative presentation by Wolfgang Fraissinet, the President of Sales and Marketing for Neumann Berlin, and Mike Pappas from Jazz 89 KUVO.

The first presentation was all about the Canadian olympics.  All of their audio was in 5.1 including sound effects and music.  Their choirs and orchestras were all recorded using Neumann digital microphones.  They used the digital microphones primarily to prevent signal loss over long cable runs, but also for their sonic qualities.  Their work all sounded amazing and everything they presented used only the digital microphones.

Another reason they loved the digital microphones was the extremly low noise floor they can provide.  They worked with lots of stacked tracks and they said even with all faders up over multiple transitions the noise never became an issue.  This was important to them because they made all of their 5.1 cues overlap without the need for crossfading.  I was very impressed with their results.

The second part of the presentation involved tracking a live quartet called Quartet San Francisco.  They played a rendition of Eleanor Rigby on the orchestral sound stage that sounded ABSOLUTELY AMAZING!

We then went into the SkySound control room to listen to the results.  Overall I felt that the digital microphones were a bit more focused and provided a bit more clarity and high end than their analog counterparts.  Most of the crowd heard things the same way.  Both recordings turned out stellar with Leslie Anne Jones running the session through Millenia preamps and the Neve board.

The general consensus was that these microphones, while not a replacement for the current analog microphones, are more of a new flavor available to engineers.  Digital microphones will probably not replace analog microphones, but rather give engineers a way to problem solve long cable runs, improve wireless technology, and simply give us new ways to record sound not possible with regular analog microphones.

Overall, I now understand that digital microphones absolutely have their place in professional audio.  Hopefully this post has helped change your opinion about the usefulness of digital microphones.  I can’t thank the Neumann and Sennheiser people enough for inviting me to such a great event.  It has opened my eyes and my ears to the potential that digital technology has to offer microphones.

Sampling Spaces (Make your own impulses for convolution reverbs) – Part 2


How To Sample Reverb Units In Your DAW
Tutorial: How to sample your Reverb Units to use them in Convolution Reverbs.

Before you haul all of your gear out into the world and attempt to sample real spaces, it is worthwhile to make a few Impulses of reverbs you have in your studio.  I’m aware that convolution reverbs take WAY more CPU than your average reverb unit, but I do think it’s important to follow this guide to really get the process down.  By doing this you can demystify convolution reverb sampling and have a better understanding of how it all works before wasting a lot of time sampling real spaces.  To do this you only need a few things.  Really, it is a pretty simple thing to do.  Let’s get down to business!

List of gear you need to sample a Reverb unit.

  1. DAW and Convolution Reverb unit (Obviously)
    1. I”ll be using REVerence which comes free with Cubase 5.
  2. Hardware or Software Reverb unit.
  3. DeConvolver to decode the recorded signal.
    1. Voxengo DeConvolver – This program is free to try and awesome.  It can generate your sine sweep impulses and DeConvolve sampled spaces so you can put the IRs into your Convolution Reverb Unit.

Once you have the necessary components you can follow these steps to sample a reverb unit.

  • Run Voxengo DeConvolver.  Click the button at the bottom that says Test Tone Gen.
DeConvolver Test Tone Generator
DeConvolver Test Tone Generator
  • Now load the SineSweep into your DAW of choice.  Put the reverb unit you want to sample into the first Plug-in Insert.  I used a Hall Church setting that is in RoomVerb from Cubase 5.  Any reverb unit will work.
    • Set the reverb unit to 100% wet so it will be entirely outputting a reverb signal.  I also recommend turning the volume down to about -8 to allow headroom.
  • Bounce the signal going through your reverb unit.  Be sure to set the bounce region to be long enough to keep the reverb tail.  I left about 2 seconds for the church hall.
  • Take the bounced signal and import it into Voxengo DeConvolver to DeConvolve the signal.  First load your original Sine Sweep test tone file at the top, then load the bounced signal into the program.  See the screenshot below for my settings.
    • Click process to generate your very own Impulse Response file!

CONGRATULATIONS!  You have just taken the steps to make your very own Impulse Responses for convolution reverbs.  That wasn’t so bad was it??

  • Now make a new track in your DAW with a convolution reverb.  Import the IR into your Convolution Reverb and you are ready to hear your sampled room at work.  After comparison you can hear that they are almost exactly alike.
Sampling Reverb Units in Cubase 5
Screenshot of Sampling Reverb Units in Cubase 5

This process isn’t limited to reverb units!

That’s right!  You can sample EQs, your favorite filter settings, or really anything you can put as an insert.  This is where you can start to get creative.  I hope this has demystified the process of sampling reverbs for you all.  I also hope you all start to get creative with this process.  If you do please take the time to comment and let me know what you’ve come up with.

Now that you know how to sample gear, you can use similar steps to sample real world spaces.  Stay tuned to the blog for Part 3 where I’ll show details on how to sample an acoustic space using a variety of gear.  This is where it will start to get REALLY useful.

-Aaron B


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Sampling Spaces (Make your own impulses for convolution reverbs) – Part 1

iPhone microphone - FFT Analysis
iPhone microphone – FFT Analysis

Places with interesting acoustic properties have always been interesting to people around the globe: Cavernous canyons have mystical qualities, old churches can produce heavenly reverbs, and large man made halls can all create larger than life sounds. All of these spaces can be useful to audio engineers, but until fairly recently it was extremely hard to reproduce the exact properties of a particular acoustic space. Can you imagine how useful it would be to “sample” a room’s properties and recreate it anywhere and at anytime? Now, with so many convolution reverbs available to audio engineers, it has never been easier to do! You only need a few things to start creating your own impulse responses! Ahhh, what a great time to be an audio engineer.

I’ve recently decided to start sampling spaces. As a sound designer and musician these will be useful in many ways. For example, I can sample a world class recording studio to give samples a more realistic room sound. Outdoor canyons can give an extra touch of realism to a space. I can also sample a trash can to see what texture that can add to my mixes.  Another use is to sample locations you have recorded in so you can simulate them later for overdubbing or post.  There are a multitude of creative applications for convolution reverbs.  So why is it that none of the audio engineers I know take the time to record their own impulse responses for convolution reverbs?

Over the years i have come to realize that if things aren’t simple and easily accessible they don’t get done. Now re-read that last sentence and really internalize it. To demonstrate this try putting your instrument out in the open for a week and see how often it gets played. Now take the same instrument, put it in it’s case and keep track of how often you play it. I guarantee you play it more when it is readily available.

It is for this reason that I’m making a simple tutorial on how to sample your own acoustic spaces. I will attempt to sample a variety of spaces using a variety of gear; iPhone, portable PA (crate taxi), Microtrack, MBox, Rode NT4 and Studio Projects C4 microphones. These results should tell you what results you can get from these devices. They should also give you all the information you need to make your very own impulse responses!

I will create the steps over a series of posts since otherwise this would be one giant post.  Stay tuned for more soon!



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Banshee Talk Box – Part 1

Ever since I heard Daft Punk’s “Around The World” I have wanted to recreate that sound.   Now I can finally make this sound and you can too!  Last week I broke down and bought a used Rocktron Banshee talk box!   Since it was used I had to buy a replacement tube at Home Depot. I just received the power supply i bought online (It takes a 2000 ma 9V adapter).  Now it is as good as new and I’m ready to rock!!!  Behold, the Rocktron Banshee in all is glory!


Rocktron Banshee Talk Box
Rocktron Banshee Talk Box

For those of you who don’t know the difference between Talkboxes, Autotuning and Vocoders I’ll quickly explain what kind of sounds they make.   I will attempt to simplify this as much as possible.  There are plenty of resources online to get more detailed info about their history and how they actually work.  Videos of the Talk Box in use coming soon!

  • Vocoders: These are processors that make voices sound robotic and synthetic.  To oversimplify a very complex effect, vocoders essentially take a voice, modulate it with another sound source that has been divided into frequency bands,  add some noise and let you control the levels of each sound.   Most of today’s vocoders allow you to control the pitch of the signal and even play combinations of notes.
    • Example: Imogen Heap – Hide and Seek
    • Vocoders available to buy: Waves Morphoder (plug in), Native Instuments Vokator (plug in), Prosoniq Orange Vocoder (plug in), Nord Modular (hardware), Electro Harmonix Voice Box (Stomp Box).
  • Autotune: Autotune is an effect that has become used in most recordings these days.  These work by taking a signal, defining a set of notes that the sound can use, and only allowing the signal to use those notes.  Many autotune devices can also be used with a keyboard to retune the vocal live, or in a sequencer, based on the notes being triggered while the signal is playing.  Their sound ranges from subtle pitch correction to inhuman-like  pitch accuracy and vibrato.   It sounds less robotic than vocoders has a high intelligibility.
  • Talk Boxes: This is the sound I’ve wanted for so long.  Talk boxes take an amplified signal, put it through a tube that goes to your mouth, and let you use your mouth as a filter.   Microphones are only used to record the signal or send it to a PA.  The sounds you can make with them range from simple formant filter (A, E, O, etc) to forming words while you play an instrument.  It is most typically used with distorted guitars or synthesizers.  It is important to note that TalkBoxes can only exist in hardware unlike Vocoders and Autotune devices.  Since your mouth is the filter it has to be a signal sent to your mouth to form that signal into words or formants.  It is for this reason that Talk Boxes tend to sound like talking instruments more than processed voice.  I guess it could be a virtual plug in if someone with years of free time could synthesize the human voice, create a digital representation of the mouth cavity, create words in real time, and side chain in a signal….  That’s a lot of work for something so simple.  I recommend we quit trying to make everything a VST and accept that some things make more sense to exist as hardware.
    • Examples: 2 Pac “California Love”
    • Talk Box devices available to buy: Rocktron Banshee, Heil Talk Box, Dan Electro Free Speech, Custom Built.
      • Personally, I like the banshee option the most for it’s blend of simplicity and tone.  Heil seems to have better tone, but there is more set up hassle involved.  Dan Electro is commonly reviewed as the cheaper lesser option.  There are a few tutorials online to build your own for cheap.  If you do this I’d love to hear your results.

Come back soon to hear a demo of the Rocktron Banshee in a video demo!  Thanks for stopping by 🙂


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Intonation – Fix your pitch

Intonation Matters!
Intonation Matters!  Do you want the notes on your killer solo to be this out of tune?

Have you ever picked up an stringed instrument that seems in tune, but as you play some chords up the neck it begins to sound out of tune?  Perhaps you started to play a great lick only to find it just sounds kind of flat.  I know I have and I used to blame cheap instruments and poor craftsmanship. As it turns out this could just mean the instrument isn’t set up properly.  If you try to record an instrument that isn’t set up you will quickly find that tuning becomes a big issue.  Playing with other people is also difficult because your tuning changes while theirs may be constantly in tune.  The good news is that this can be very easy to fix!

Intonation is one of the most important things to understand with many stringed instruments. Electric guitars, bass guitars, and even instruments like banjos all have adjustable intonation. Instruments like acoustic guitars make it much more difficult to adjust the intonation and will require help from a luthier. This is because the bridge on acoustics isn’t easily adjustable like those on electric guitars.

How to adjust intonation:

You can test your instruments intonation by tuning the first open string correctly then playing the 12th fret on that string to see if it is still in tune. If the 12th fret note is sharp then your string length needs to be longer. If the 12th fret note is flat then the string needs to be shorter.  Adjust the bridge of your instrument appropriately. Now re-tune the string and test again. Once it is in tune on the open fret and 12th fret you can move on to the next one. Do this for all of the strings on your instrument and **BAM** you’ll have a properly tuned instrument ready to rock!

Note that none of this covers the action of the guitar .  That is a different issue that is also very important.  I may make a post on this later.

A smart man once said a picture is worth a thousand words.  Below you will find a few pictures to show you exactly what I mean 🙂


Here is a diagram of how to adjust the intonation on a banjo.

Bridge of a banjo
Bridge of a banjo

Electric instruments usually just need a regular or small screwdriver to adjust their intonation.  Follow the same steps as above and do each string one at a time.  You’ll have a properly tuned  instrument in no time!  One thing worth mentioning is that you don’t want to press too hard on the 12th fret while making these adjustments.  If you press too hard it will bend the string and produce a sharper pitch than you normally do while playing the guitar.  Just try to press the 12th fret as hard as you would playing normally.

Fender Fat Strat - Bridge properly intonated
Fender Fat Strat Bridge – This is what mine looks like with good intonation. 
Bass Guitar Bridge - Ibanez GSR200
Bass Guitar Bridge – Properly intonated Ibanez GSR200.

It should be just as in tune on an open string as it is on the same strings 12th fret of your instrument.  I know it’s not perfect in these screenshots, but it’s really hard to get a screenshot while it’s perfectly in tune.
Here are snapshots I took of what the tuning should look like on a tuner using GuitarToolkit on my iPhone.   It’s a great tool!  Buy it if you have an iPhone.  You won’t regret it 🙂

6th string in tune on the guitar
Tuning 6th String
6th string 12th fret in tune on the guitar
12th fret of the 6th string also in tune on the guitar.  This proves that the 6th string is properly intonated.

Video demonstration coming soon!

I’d like to thank the good people at Exploring Music for showing me how to properly set up my banjo. They have very helpful employees and cheap lessons. I took a few violin lessons there and I plan on returning!

-Aaron B.


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