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Speaker to Room Calibration Walkthrough



While visiting a friend in The Big Smoke recently, I had an opportunity to assist in tuning a set of DIY speakers to his critical listening environment. This is a walkthrough of what we did, how it sounded, and lessons learned. I tried to present this in a step by step format so if desired, by following the same steps, you could obtain similar results. Here is pic of my friend’s rig:




Let’s inventory the gear. The DIY satellite speakers are enclosed in a designed (using Thiel Small parameters) sealed box using a Seas 6 1/2” woofer that has a characteristically smooth frequency response. The tweeter is an interesting Philips ribbon design.




The passive sub-woofer is a sensitivity matched 10” Pioneer polypropylene driver with a dual voice coil that replaced the original Paradigm driver in which the surround rotted away.




In the above picture, the bass cabinet is upside down showing the two vented ports on the bottom. The speaker is mounted such that if the ports are facing the rear wall, the speaker is facing the listening position (i.e. firing forward).


In my friends listening tests, plus mine, we found the bass response was smoother with the driver firing towards the rear wall, with the ports facing the listening position. The bass frequency response was smoother, both from a listening, and as it turns out, a measurement perspective.


The satellites and passive sub are powered by 100 watt per channel vintage NAD power amplifier, using the preamp section of the NAD integrated amplifier to drive the amp.




Source music is iTunes on an iMac G5.


Here is a check list I assembled for this calibration task. As mentioned at the top of the article, I tried to write it in easy to perform steps.


Step 1 Check to see if the speakers and listening position can be set up in an equilateral triangle.


It is not the end of the world if it can’t be done for whatever reason. In my friend’s case, given the limited space, this is how it turned out. But once the speakers were tuned to the room, it was impressive to hear the rock solid 3D soundstage that these speakers reproduce. There are a couple of reasons for this that will be brought forward later.


Here is an example "blueprint" of the type of speaker to room interface setup required for playback if the goal is to hear what the mixing engineer intended to be heard as accurately as possible. This means trying to achieve symmetry as close to a tight tolerance as possible. Measuring the symmetry (i.e. physical distance) between speakers, walls, and listening position is the key to accurately reproducing (i.e. decoding) the soundstage.




Here is the specification.




Both Dolby Labs, “5.1 Channel Music Production Guidelines”, http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/4_Multichannel_Music_Mixing.pdf and “ITU-R BS.775-1* MULTICHANNEL STEREOPHONIC SOUND SYSTEM”, http://www.gearslutz.com/board/attachments/bass-traps-acoustic-panels-foam-etc/234800d1305311642-sticky-links-itu-r-bs775-1.pdf (and ANNEX 1 for stereo) have similar “specifications” when it comes to interfacing the speakers to the room.


Full disclosure, in a previous career, I worked as a recording/mixing engineer for 8 years. I was fortunate enough to have worked in very nice control rooms, with state of the art acoustics and monitoring. I was lucky to have been involved in the ground up build of this facility with Chips Davis: http://chips-davis.com/ I learned a lot. By far the best sounding critical listening environment I have ever heard.




Those Urei/JBL 813B "time aligns" were being driven by 2.4 kilowatts of Crown power. The bottom end was super tight and still went to 20 Hz. Using the mid and high frequency trim pots, we would adjust the tonal balance to typically that of the B&K target frequency response (discussed later). This was the only eq required as the room itself was designed and built to be as acoustically transparent as physically possible (LEDE certified). Almost all pro facilities since then are based off of this deign. It was a revolution in control room design.



In order to certify a mixing, mastering, broadcast, video post-production, critical listening, etc., facility, it means implementing this specification. So if the goal is to accurately reproduce what the mixing engineer hears in their chair, it makes sense to use the same specification in our own critical listening environments.


Note this is how stereo and 5.1 music is mixed using this "speaker to room" specification. If we want to properly “decode” what was mixed, then our speakers to room interface should conform to this specification. The Dolby document, “5.1 Channel Music Production Guidelines” is a great read. Even though a bit dated gear wise, almost every piece of music we listen to has been recorded, mixed, and mastered using this speaker to room interface specification.


Step 2 “Voice” the speaker to room interface by using the following two methods. One is by ear the other using REW acoustic measurement software. Let’s go by ear first.


By Ear. I voice (i.e. play music on) one speaker at a time. I play music that has a solid bass line, preferably with a wide range of notes. Ideally a bass line that produces significant output at 20 Hertz and moves up and down the musical scale.


What we want to do is listen to the music while moving the speaker either closer or further away from the rear wall. Here is the method I use. Looking at my friend’s setup, I would stand parallel, facing the side of one of the speaker cabinets. One ear is focused towards the listening position, with the other ear focused on the wall behind the speakers. So I am standing 90 degrees perpendicular (i.e. sideways) to how I would normally sit and listen to the stereo.


While the music is playing, on one speaker at a time, I am slowly moving the speaker either forward out into the room or backward towards the wall. What am I listening for? I am listening for the evenness of the bass notes. As the bass notes are playing up and down the musical scale, each note should sound equal in level (i.e. loudness) to the ear. Technically, I am trying to “hear” the room modes.


If one or more different bass notes drops quite a bit in level that means I am in a position in the room where those bass notes (i.e. range of frequencies) cancel each other out or in a “null” spot. Conversely, if some bass notes sound loud or boomy, relative to other notes, then I am standing in a position in the room where those bass notes add together. However, there is likely one or more spots in the room where the speaker’s bass notes sound the most even. That’s where the speakers should be placed, if the geometry works out. In other words start with a rough equilateral triangle and fine tune from there.


Here is an illustration that animates this concept of waves adding or canceling depending on room position.




Further reading: http://www.acs.psu.edu/drussell/Demos/superposition/superposition.html


Note this is relative to the listening position as well, which further complicates the mixing of waveforms. Ideally, the listening position is as far away from all reflecting surfaces as possible. This means towards the middle of the room. Called, a reflection free zone. Here is a picture of my critical listening environment, when I remove the table that is. The listening position is away from the rear and side walls as much as possible. The floor and ceiling and wall/glass behind the speakers are the early reflectors, but are dampened by the thick carpet I have between the speakers and listening position.




Back to my friends room. Another way to voice the speakers to room interface is by using a computer software measurement system. Fortunately, high resolution software measurement systems, too expensive ten years ago, are now state of the art for a commodity price or even free.


One such fine piece of measurement software is REW http://www.hometheatershack.com/roomeq/ , which unbelievably is free. In conjunction with a calibrated measurement microphone http://www.content.ibf-acoustic.com/catalog/product_info.php?cPath=30&products_id=35 this state of the art measurement system can assist in optimizing any speaker to room interface. In fact, given the computing power, sophisticated measurement software, and calibrated mic, the resolution of the measurements made by this system exceed what our ears are capable of hearing.


First we need to calibrate the sound card. What this means is that the REW software is going to measure the frequency response of the sound card and if there are any deviations from “flat” from 20 Hz to 20 KHz, the “calibration file” will compensate. This takes any of the sound card’s frequency response variations out of the measurement equation. That way we are measuring the true frequency response of the device under test, which in our case are the speakers in the room.


This is the measurement of my Lynx L22 pro sound card, frequency response is flat out past 50 KHz. Since this measurement is taken in external loopback mode, the frequency response is the combined response of both the input/output analog amplifiers and ADC/DAC components. You achieve external loopback by connecting the output of your sound card back to the input of same sound card. You can find good instructions here: http://www.hometheatershack.com/roomeq/wizardhelpv5/help_en-GB/html/calsoundcard.html#top




As an aside, it is amazing to me that the onboard chip in my Dell laptop has an ADC and DAC that supports up to 24/192. This is the onboard chip! Soon we will see this level of DAC in smart phones.


The same calibration approach applies for the measurement microphone as well. We want to plug in its calibration file, supplied by the microphone manufacturer, into the REW software. One of the reasons I like this measurement kit http://www.content.ibf-acoustic.com/catalog/product_info.php?cPath=30&products_id=35 is because the mic preamp is calibrated to the microphones sensitivity. This means I can calibrate SPL’s as well if I want to meet a proposed monitoring level specification like Bob Katz’s K System: “An Integrated Approach to Metering, Monitoring, and Leveling Practices” http://www.digido.com/level-practices-part-2-includes-the-k-system.html Excellent read and proposal in my opinion.


My mic comes with two calibration files, one for the mics on axis frequency response for when the mic is pointing down the center line towards the speakers, and 90 degree off axis response for when the mic is pointing up. I have used both and for room measurements, I prefer the 90 degree diffuse field correction. Also relevant if you are tuning a 5.1 system.


Here is my mic’s calibration file. This is the frequency response graph and partially what the 2 calibration files contain from 20 Hz to 20 KHz.








This is a pic of the microphone position used for the following measurements in my friend’s listening room. Note that the listening position is at the back wall, as opposed to the middle of the room, as space is limited. There is a sneak peek of the frequency response graph on REW’s software.




Now we are in a position to measure the frequency response, but first we need to set some levels and calibrate the SPL. That is covered in REW’s guide: http://www.hometheatershack.com/roomeq/wizardhelpv5/help_en-GB/html/measurementlevel.html#top and http://www.hometheatershack.com/roomeq/wizardhelpv5/help_en-GB/html/inputcal.html#top


But before we take any full range frequency measurements, we want to “voice” the speaker’s low frequency response to the room, like we did with our ears. In some respects, this is a validation that we have found the right spot based on our ear’s voicing of the room.


Aside from swept sine waves, we can use REW’s Real Time Analysis (RTA) function to pressurize the room (like blowing on a coke bottle to form the resonances). We output pink noise for low frequencies, and while watching the real–time display, moved the speaker/sub combo to and from the rear wall to achieve the most even response possible. As it turns out, where I voiced the speakers by ear turns out to be the same spot that measures the flattest frequency response in the low end. Here is what that looks like in REW:




Step 3 Now measure up and ensure we have the best possible symmetry to within a 1/4” tolerance.

Based on voicing the speaker to room interface for low frequency response, we need to ensure that each speaker is exactly the same distance from the rear wall. While I have used tape measures in the past, it is truly worth the investment in a digital laser measurer like http://www.amazon.com/Bosch-DLR130K-Digital-Distance-Measurer/dp/B001U89QBU


Why? Here is a bit of math to better understand why this is important: http://www.sengpielaudio.com/calculator-wavelength.htm This is a wavelength calculator. If I type in 20,000 Hz, the wavelength is 0.675 of an inch. That’s a little more than a 1/2". So if my equilateral triangle is out, or off center, or the distance from the speakers to the rear wall are different, then “comb filtering” occurs. All it takes is for the tolerance to be out slightly more than a 1/2". It not only affects the frequency response, but the soundstage as well. Ethan Winer does an awesome job explaining this: http://ethanwiner.com/believe.html


Bottom line. Try and ensure that the equilateral triangle between the speakers and listening position is within a 1/4” tolerance. Same goes for the distance between the speakers and the rear and side walls. Try and make it as symmetrical as possible. I shoot for a 1/16” – 1/8” to a 1/4" max. tolerance. I learned this from Chips Davis. When that studio was built in 1986, he used a laser survey standard to layout out the room from the blueprints. He viewed it as the most critical part of the build and spent a great deal of time getting it right on the button for every measure.


It may take a couple hours of work, measuring distances, readjusting, re-measuring, rinse and repeat, down to below 1/4" tolerance. The reward? Most people have not heard this level of precision towards neutral tonal balance and decoding of the soundstage.


Here is a pic using the laser distance measurer to ensure both speakers are exactly the same distance (I got it down to a 1/16” tolerance) from the rear wall.





Step 4 measure the frequency response of each speaker (and sub). Now that everything is configured to a tight tolerance, sound card and mic calibrations loaded into REW, we can take a full range measurement.




A couple of points to keep in mind while looking at this frequency response plot. One is that the REW software has considerably more measurement resolution than our ears do. “The ear tends to combine the sound within critical bandwidths, which are about 1/6 octave wide (historically thought to be 1/3 octave). This has led to the practice of averaging frequency response over 1/3 octave bands to produce beautiful-looking frequency response curves. In my opinion this is misleading.” From “Music and The Human Ear” Another excellent read in my opinion: http://www.silcom.com/~aludwig/EARS.htm


I agree. I use 1/6 octave smoothing on the measured frequency response, which more closely represents how our ears hear. Here is the same frequency response as above, except with 1/6 octave smoothing applied.




Major difference and provides a clue as to how much (displayed)resolution the software and components are capable of measuring.


I know it seems far away from a flat frequency response, as compare to the frequency response of the Lynx L22 sound card above. To quote Ethan Winer, "The room you listen in has far more influence on what you hear than any device in the signal path, including even the loudspeakers in most cases." I agree.


For example, here is the frequency response of one of my speakers in my listening room:




Similar, but different than my friends room. However, unfortunately, typical of small room acoustics. All of our listening environments suffer from this. Some more than others. To look into this more, I highly recommend Bob Gold’s Room Mode Calculator: http://bobgolds.com/Mode/RoomModes.htm to plot the room modes, cut off frequency, and other important acoustical parameters to take into consideration.


To better understand the output of the calculator, Ethan Winer, again, does an excellent job of explaining this and what can be done in his article, “Acoustic Treatment and Design for Recording Studios and Listening Rooms: http://www.ethanwiner.com/acoustics.html


As an aside, when we were running REW full range sine wave sweeps, there was an audible “chirp” detected in the right speaker. It was for just the briefest moment of time, but still audible during the sweep. We ran the test a number of times while feeling around the cone of the Seas driver. As it turns out, while we were soldering a connection earlier, we forgot to tighten up one of the bolts that mounts the driver. So we were hearing the vibration of the speaker frame against wood at a certain resonant frequency due to one bolt not being tightened up. After tightening the bolt, and re running the sweeps, the chirp was gone.


Btw, other folks are obtaining similar results. CA’s own Nyal Mellor and Dallas Justice worked together to use acoustic measurements for speaker placement: http://www.whatsbestforum.com/showthread.php?6050-A-good-example-of-how-to-use-acoustic-measurements-to-place-speakers


Great job!


CA’s wgb113 also wrote a blog post: http://www.computeraudiophile.com/blogs/Room-EQ-Wizard Nice frequency response!


Another interesting speaker room calibration discussion is Bruce Brown, The Pro Audiophile: http://www.whatsbestforum.com/showthread.php?5893-Speaker-Room-calibration


Step 5 Calibrate the speakers to a known target frequency response reference.


So now that I have measured the frequency response, now what? Considerable research and listening tests show that we do not want or prefer a “flat” 20 Hz to 20 KHz frequency response at the listening position. In fact, most of the research shows that we prefer a “target” frequency response at the listening position to be sloped downward from 20 Hz to 20 KHz, typically down -6dB at 20 Khz.


Reading this excellent paper from B&K, http://www.bksv.com/doc/17-197.pdf and based on the listening/measurement tests, the “optimum” target frequency response at the listening position is:




Translating the target frequency response into numbers: 0 dB at 20Hz, -0.5 dB at 200 Hz, -3.0 dB at 2 KHz, and -6 dB at 20 KHz. In addition, looking at the work Dr. Sean Olive http://seanolive.blogspot.ca/2009/11/subjective-and-objective-evaluation-of.html has done, both in measuring and listening tests, a very similar frequency response "slope" emerges as the preferred frequency response curve at the top of this diagram: https://docs.google.com/file/d/0B97zTRsdcJTfY2U4ODhiZmUtNDEyNC00ZDcyLWEzZTAtMGJiODQ1ZTUxMGQ4/edit?hl=en_US&pli=1




It is also stated that listeners do not prefer a “flat” frequency response, but rather the downward sloped response. “Flat in-room response is not the preferred target” from Sean’s slides.


I concur, here is the frequency response of one of my speakers at the listening position, corrected for perfectly flat. I am using Digital Room Correction (DRC) to achieve this flat response of 20 Hz to 20 Khz +- 3db at the listening position. It is definitely too bright and has another consequence of moving the soundstage too far forward. Note these graphs are 1/12 octave smoothing.




Because I use DRC software (i.e in the digital domain), I can easily change the “target” frequency response in less than a few minutes and be listening to a different “calibration”.


Here I am following the “optimum” B&K and similar Harman target frequency response at the listening position.




To my ears, and obviously to the folks that took the tests in both articles, this frequency response “slope” at the listening position sounds the most natural. It’s not too bright, not too dull, but just right. It also has the best sound stage in my opinion, not too far back, not too far forward, but just right. To my ears, best resembles the tonal balance that I took for granted while working in those state of the art acoustic and monitoring facilities.


Because of the ease in which I can change target frequency response, I have listened to dozens of different target frequency responses. By doing this, I have learned that a few dB up or down deviation from the B&K and Harman target's, makes a big difference in tone quality (i.e. timbre) and sound stage. That is why it is critical to calibrate the sound card and have a calibrated microphone, as a few dB off calibration makes a big difference and can produce incorrect results.


So how does this help my friends room tuning? Well, he has tweeter level controls so I can “calibrate” the frequency response slope to match my target frequency response that I know is already calibrated to a known target. Here I have overlaid 3 frequency responses. The one in blue is my friends, the one in green is mine, and the one in red is mine also, but with DRC (Audiolense) enabled.




It is not a fair comparison as I am using DRC (which makes a significant difference). But the point being is that I can match the high frequency response “slope” of my friends speakers to match the target for the best tonal balance and soundstage. Our listening tests confirmed that this is the right tonal shape at the listening position, even if the bottom end is a bit more variant in frequency response.


Step 6. Repeat as required. It may take a few iterations to fine tune the system as there are a lot of variables involved. For example, while the mic was pointing straight up, I used the wrong calibration file which accounts for a 4 dB difference at 20 KHz. Noticeable. You may want to space apart the iterations over days so that you can settle in and form an opinion as to the sound quality.


It has been my experience that there is a direct correlation to what we measure and see with graphs with what we hear. It just takes practice and patience to tune into both and correlate the two. If you take some DRC software for a spin, make sure you can quickly switch between targets. It is very interesting to see that small measured changes in frequency response slope, makes for large audible tonal and soundstage changes.




“The room you listen in has far more influence on what you hear than any device in the signal path, including even the loudspeakers in most cases”. As quoted from Ethan Winer. I would say by orders of magnitude difference based on my experience. This is where significant improvement to any existing sound system can be had for a relatively small investment.


Where to go from here?


If you are interested in Digital Room Correction (DRC), and advanced topics like True Time Domain correction (i.e. time alignment), I wrote a series of articles called, “Hear music the way it was intended to be reproduced” starting with: http://www.computeraudiophile.com/entries/6-Hear-music-the-way-it-was-intended-to-be-reproduced-part-1


Note with Audiolense DRC, your head is not “locked” into one mic position. Audiolense has the capability for multi-seat DRC so that you can tune the sweet spot to cover a couch area, for example:




If DRC is not for you, then try some simple passive acoustic treatments. This is a great starting point: http://www.gearslutz.com/board/studio-building-acoustics/610173-acoustics-treatment-reference-guide-look-here.html


Digital Audio has been around since the early 80’s, that’s 30 years ago. According to Moore’s law, http://en.wikipedia.org/wiki/Moore%27s_law we are using computing power that contains 100,000 times more transistors than in 1980. Today we have very advanced software working at 64 bit data paths, http://wiki.jriver.com/index.php/Audiophile_Info where noise and distortions are below our ability to hear them. That means Digital Room Correction, using digital FIR filters, has no audible effect on the sound quality when inserted into the playback chain.


REW is powerful acoustic measurement software as it can also measure Energy Time Curves (ETC’s) which will help you achieve a reflection free zone in your listening area. Also using REW’s waterfall measurement capability will show how sound decays in your room, typically over a 300 millisecond window. This is useful to see if bass traps are required as this is the most common problem, along with early reflections...


My goal in writing this was to walkthrough a typical speaker to room calibration and demonstrate the benefits of such an effort in hopefully easy to perform steps.


After a month of listening, my friend is very happy with the end result (both audible and visual).


Happy Listening!


Recommended Comments

Really nice job mitchco. I'm impressed by the fact that you took the time to do this, and do it so well. To come by all this info laid out so nicely in one place is, well, so fine. This would have come in very handy for me about a year ago.




Now if folks who complain about their system's sound would at least look into some of this, instead of fretting about what to buy next.





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Thanks Mitch, What can I say... you da man! ;-) I know you like to use Audiolense DRC, but if I remember right, it is not Mac compatible. Darn! I would like to be able to correct for multiple listening positions. To my knowledge REW doesn't do that. Do you know if REW will EQ in the time domain... if that would help? Speaker placement is VERY limited for me, as is listening position,... a few inches in all directions.




This novice is still reading everything I can find; trying to get my head around the various measurements, parameters for those measurements, and what each measurement can tell you. Have you come across a simple primer for this info? The hardware is all ordered. The calibrated mic is on back order... patience, patience! Now it looks like I'll need a digital laser measurer also. Better to do it right the first time as I see it!




I appreciate your time and effort to do these posts for our benefit.


Thanks and Cheers, Rod

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Wow excellent write up, lots of information about this is out there. But a real world, step-by-step process like you wrote is what people like me who are new to this hobby need!




Thank you for your time and effort involved in this!

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Ha! Familiar ribbon tweeters, I've had speakers with same/similar tweeters since early 90's, now in our bedroom:






Mine has quite different mounting to further improve directivity pattern and timing properties.




I like the sound of those tweeters.

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Mitchco. Well done!!!!!!!!!




I have yet to read this, but did a scan and want to echo the others in thanking you for this comprehensive step-by-step!!




On a side note, while the blog area is great, its often overlooked vs forums. Chris's site isn't well organized to enable one to keep up to date on blogs as with the rest of the site. I'd recommend you post on the general or speaker forum area a notification of this entry, so others can find it.




I look forward to reading it all carefully...

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For the shout out Mitchco! Wait until you see the results after fully treated room and a dialed in PARC only on the powered subs. I should be done with my installation this weekend. Since i listen to a lot of DSD, my goal is to get as flat a frequency response without DSP. Room acoustics is my final frontier. Looking back, it should have been the first thing I did.

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Of course, what caught my eye in the picture of the studio rig you helped design was the ancient IBM monitor sitting there. Dates it to the early to mid 1980s technology wise.




I would love to hear you expound a bit more on the equipment in that pic, and what you would choose today. :)





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Another great blog post. You walked us through the steps you took with your friend well and the pictures were a definite help. My progress of tuning my room and setup is taking much longer than I wish but I keep stumbling upon info and tips like you've provided here and I'm kind of glad I'm not done yet!





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"Two commercial software for measure and create a filter that allows for acting on a wider listening area"...








Having spend a few months (trying to) following DRC proceedures and products, I'm not familiar with these offerings. I looked quickly at the links, but rather than be seduced by manufacturers' hyporbola, would you briefly describe your knowledge of these and how they differ from more well known solutions like REW and Audiolense.





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It's simple, they can be used natively with the Mac




IK ARC includes the microphone, performs measurements at several points, it's based on Audyssey www.audyssey.com/, generates the filter that is managed as a plug-in Audio Unit format.




Dirac takes measurements at several points, manages up to 8 output channels, it is a sw that processes any audio signal from every sw (eg: any browser you use for streaming music)




REW & DRC are powerfull and complete but are not easy to use even more if you own an Apple. In addition they forcing to one listening position.





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Do these support exporting the correction filter impulse response for use in other processors?

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ARC http://www.ikmultimedia.com/arc/features/ generates a series of filters that can be uploaded and managed by its plug-in (VST or AU) in any software that can manage the plug-in (there is also a version of ARC for use with DAW and ProTools).




Dirac must be considered a "virtual card" that operates independently from the sw playing music (Mac/PC ---> every player or sw supporting audio ---> Dirac ---> sound card/DAC)

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...can you use any convolution engine of your choice with these, rather than the plugins associated with the software?




Dirac must be considered a "virtual card" that operates independently from the sw playing music (Mac/PC ---> every player or sw supporting audio ---> Dirac ---> sound card/DAC)




I guess this doesn't work with players using ASIO or WASAPI exclusive mode. Only with WASAPI shared mode, or one of the old audio APIs.




Not all players support VST plugins either, but lot of player on the other hand have built-in convolution engine for the purpose of running DRC filters. This is how REW, Audiolense, DRC-FIR & co are used most of the time.




ARC didn't seem to support 176.4/192 rates or above, how about Dirac?

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First of all I am an Apple user so I'm interested in products that work with Apple and I know nothing about PC versions, sorry :-)




Both sw measure the environment, create the filter, apply the filter.


No need to use other sw. Both meet all the necessary functions.




I do not see why you want to use other convolution engine [*].




Dirac curves are manually adjustable (ARC doesn't)




AudioUnit is a format created by Apple and many players support it.




Dirac does not need plug-ins nor uses plug-in




[*] If someone has enough time, desire and ability, can use REW [**] and DRC[***]




[**] For example, I can not use REW because it does not work with Firewire audio cards.




[***] DRC is powerful, precise, perfect but it is complicated and forces the user to a single listening position.




I apologize but as English is not my language, I suggest you to read the link of the two products especially for the PC version.

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I do not see why you want to use other convolution engine




Because it's already in the player and this would make applying the filter more flexible and platform independent from creating the filter. For example running the filter on a Linux-based music server, or a separate processor hardware.




I have access to a Mac I could use to make measurements and create filters, but I'm not using it for performing the actual playback.




Neither one seems to have a trial that would be useful for trying out if this is possible. So essentially answer the question if there's a similar software to Audiolense for Mac?




I apologize but as English is not my language




Mine neither..




I suggest you to read the link of the two products especially for the PC version




I did already, but it didn't answer the question about possibility of decoupling the filter creation and running the filter...

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(I hope Sean Olive can add his perspective to this topic.)




DEQX seems to be the only popular room correction product that uses parametric EQ. The others use convolution.




The convolution-based systems attempt to correct numerous narrow-band frequency response aberrations and comb filter effects, and to correct for the spatial offset between loudspeaker drivers and between the loudspeakers themselves, but it's unclear to me whether this actually is beneficial.




(I'm speaking only of room correction, not of active crossovers where these features clearly are useful.)




In convolution-based systems, the corrections can be highly dependent on small changes in listening position. The designers have different approaches to ameliorating this problem. However, I don't know of any convolution-based system that allows you override its attempts to correct for narrow-band frequency response aberrations. All you can do is adjust the slope and extension of the target curve.




Parametric EQ corrects only lower Q frequency response aberrations, but I believe studies have found these are the aberrations that are most audible. Advantages of parametric EQ are that you can readily tweak the EQ and listen for the results, and it would seem to have less potential for doing harm.




Room EQ Wizard (donation-ware) can automatically calculate suggested parametric EQ settings that you can use as a starting point.




My comments are merely hypothetical; I have not actually tried any of these yet, but I hope to take the plunge soon.

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The convolution-based systems attempt to correct numerous narrow-band frequency response aberrations and comb filter effects




Not necessarily, usually the design software takes care not to try to do unnecessary micro-optimizations. Good side with convolution methods is that frequency and phase responses can be corrected independently.




Room EQ Wizard (donation-ware) can automatically calculate suggested parametric EQ settings that you can use as a starting point.




REW can also export impulse response of the parametric correction which in turn can be run in a convolution engine. I've used this method quite a lot.




Naturally, you can run any kind of parametric EQ with convolution engine. But not vice versa.

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Nicely done.


I have one comment regarding this statement:




" Those Urei/JBL 813B "time aligns" were being driven by 2.4 kilowatts of Crown power. The bottom end was super tight and still went to 20 Hz. Using the mid and high frequency trim pots, we would adjust the tonal balance to typically that of the B&K target frequency response (discussed later). This was the only eq required as the room itself was designed and built to be as acoustically transparent as physically possible (LEDE certified). Almost all pro facilities since then are based off of this deign. It was a revolution in control room design."




Those speakers are/were known to have quite poor off-axis response meaning the reflected sounds arriving at the listening position, if not absorbed, were very colored. There is some speculation that the LEDE room, which is very absorptive at the front end was designed as a band-aid to fix bad speakers.




It is more environmentally responsible to design and use loudspeakers that sound and measure good both on and off axis. That way, you don't need to fill the room with 2 tons of fiberglass to turn the bad sound into heat. Start with decent speakers and save on amplifier power and fiberglass.




As far as LEDE rooms go, many people found them acoustically unnatural and difficult to work in. The psychoacoustics on which they are based are also flawed. Davis argued that a strong early reflection after 10 msec (he called them "Haas kickers") would further extend the precedence effect (i.e. mask later arrival reflections), which we've shown not to be true (see http://www.aes.org/e-lib/browse.cfm?elib=6079). The precedence effect perceptually takes care of most early reflections to minimize their effect on image shift-localization artifacts. Early reflections produce positive effects like increased intelligibility, image broadening, spaciousness and envelopment. So why the need to get rid of them other than to sell a bunch of acoustical treatment and impress clients?




Control room design in the recording industry was and still is largely based on fads and trends, with very little design based on solid science and psychoacoustics.

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Hey Paul, you would laugh as the IBM monitor was thrown into the picture by one of our techs at the time cause he thought it was cool, but was not used at all in the in the studio :-0




We were the first studio in Canada to be all digital. We had 48 tracks of digital using 2 x Sony PCM 3324a (with the Apogee filters) and 3 or 4 (I can’t remember) Sony PCM 3202 2 channel machines. The picture here only shows part of the tape room:








There is a story around the Neve console that it came from England and a list of bands that used it. I can’t remember the details, but I reached out to another bud (the tech!) that worked with me and see if he remembers.






Just to set the record straight, I did not design the studio with Chips. I was one of the resident house engineers at the time and we had 3 studios. Myself and a few others got to hang with Chips throughout the build, from the ground up. What was really cool was to see the “designed” frequency response and ETC’s before the studio was built and then when the frequency response (and ETC and all of the other stuff that makes up a certified LEDE room) was measured, it came out to being almost identical. Pretty amazing given it was almost 30 years ago.




Today? It’s all Digital Audio Workstations (DAW’s) for about 1/1000 of the cost. What’s going to be in another 10 years?




Thanks tonmeister86 for your comments. I hear you. Remember this was almost 30 years ago and at that time, it was state of the art. I think this comment sums it up: http://www.gearslutz.com/board/5575228-post7.html An LEDE room had several design criteria to be met, not just frequency response, in order to be certified. Since then most people agree that a reflection free zone at the listening position is what most people strive for, with diffusion on the rear wall and some bas traps to tame room modes, but still keep the room live sounding. I don’t like the “dead end” sound of too much absorption as can be evidenced in the picture of my room acustics. But I do appreciate Chip’s contribution to advancing the science (and sound!) of control room acoustics.




With respect to the 813B’s, what I liked most was the “time alignment” by Ed Long as Mix Magazine states, “…the UREI 813 was the most successful large format studio monitor ever made.” http://tecfoundation.com/hof/06techof.html#10 Surely they must have done something right… Btw, I do like the http://www.audioheritage.org/html/profiles/jbl/4430-35.htm as well with the Bi-Radial horns.








These did not measure so well off axis either: http://www.jblpro.com/pub/obsolete/443035.pdf It is too bad that large format, 3 –way speakers are not as popular anymore as the analogy goes, there is no substitute for cubic inches ;-) I would be curious to know what speakers you think are the shizzle.




Hey Miska, yes, those ribbon tweets do sound pretty amazing, especially the detail and soundstage they throw. I am always amazed when I hear my friends system on how good they sound.




Aloha Bob! Not sure if you are aware of this, but using digital FIR filters, you have an incredible degree of control over the filter “taps” which means, “…the amount of "filtering" the filter can do; in effect, more taps means more stopband attenuation, less ripple, narrower filters, etc.” http://www.dspguru.com/dsp/faqs/fir/basics In Audiolense, you can set the number of taps from 0 to 65,535 and anywhere in-between. So you can have almost infinite control over filter width from 20 Hz to 20 KHz. Of course you can also limit the correction to whatever frequency range you want to cover. So if you only want to work on the rooms modes below the room’s calculated cutoff frequency, you can. In some respects it is like an infinitely variable parametric eq, but in the digital domain.




With Audiolense DRC, using “just” frequency correction, I have found no need to engage the multi-seat correction as it sounds good anywhere in the room and sounds excellent seated anywhere on the couch. Using True Time Domain (TTD), does require the use of multi-seat correction as this mode of operation does involve correcting the time domain, in addition to the frequency domain. For me, this is the best sound possible and reminds of the “time aligned” sound that I used to hear through the Urei time align speakers. It is a real treat to hear crystal clear 3D sound while minimizing the impact of the dreaded “small room acoustics.”




With respect to the other DRC solutions, I have no experience with these and encourage readers to try them out, presuming there is a trial version to evaluate. I would love to hear feedback of other DRC software and how it works for folks.




As a 20 year professional software engineer, I can say with experience that most people underestimate the power of software and computers. As mentioned in my post, digital audio has been around over 30 years. Over that time, any music that has been recorded using an ADC or played back through a DAC has passed through a digital signal processing (DSP) chain. Every modern day ADC/DAC employs a digital and analog anti-aliasing filter. http://en.wikipedia.org/wiki/Anti-aliasing_filter Without DSP, there would be no computer audio or Computer Audiophile :-)




Today given the processing power of computers and that that the mass majority of consumer and professional digital audio software programs are under a grand, it will only get better, cheaper and faster. That multi-million dollar facility I worked in years ago is easily replaced with under $10,000 in computer, software, and a few good mics and can sound better! I wonder what it will be like in 10 years…

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Hi @mitchco, why do you recommend pointing the microphone to the ceiling instead of pointing forward, each position corrected with its file? Considering 2 channel stereo system in a listening room. What are pros and cons of each method?



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@Matias, that was from 9 years ago :-) As long as you have mic cal files for both 0 and 90 degrees, does not matter too much. Today the recommendation for 2 channel is pointing the mic between the two speakers and for surround pointing the mic up.



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