# Psychoacoustically-Correct room correction



## bobkatz (Mar 28, 2009)

There is as much science as art in room correction and I hope to get a little closer to being able to be more productive and scientific in my work. I want to congratulate John on having developed a fantastic product. I'll be sure to donate to REW's development before Christmas. I come to REW after slowly discovering that it is a more powerful product (tool) than FuzzMeasure, which I have been using up to this point. The main improvements that REW offers me over Fuzzmeasure are:

1) the ability to preset the window to my wishes for the next measurement

2) Truly remembering all the zoom settings for each module, and ability to display and switch between impulse, ETC, waterfall, etc. in tabs rather than in popup menus, tabs that remember the zoom and other settings that I had preset for each of these modules

3) Automatic determination of the peak of the impulse, saves a lot of time.

4) And most important, the ability to use a curved window (e.g. Hann or Tukey) and start that window any preset amount in front of the impulse and end the window any preset amount after the impulse. 

The next thing I'd like to conquer and see if REW is up to the job, is the ability to use different size windows for different portions of the frequency response measurement. The principle is that psychoacoustically, the ear responds more to the room response for low frequencies, and to the impulse UP TO THE FIRST REFLECTION for high frequencies. In other words, I want to combine two or more response curves, preferably by taking only one measurement. It would be tedious to have to set start frequencies and end frequencies and window settings by hand every time I want to take a different portion of the measurement. The article (actually a powerpoint) that describes these issues is by the famed James Johnston, and can be found at (have to avoid coding this since I have not made the requisite number of posts to be allowed to put a URL in this). The Pacific AES Northwest Section. Go to the aes dot org site, sections, pnw section and the link ppt dot html. Find the powerpoint presentation given by JJ, A Low-Complexity, Fast-acquiring Perceptually Tuned Room Correction Algorithm. 
PNW Section meeting, January 2008. JJ has also co-written an AES paper on this subject, which I have to find. My object is to boil down his technical largesse into a formula for REW that will allow us to marry two or more different window shapes and splice their frequency responses together, level-matched, to yield a more accurate low and high frequency response than the compromise window settings have yielded up to now. 

No, it's not a perfect world, as JJ indicates in his Powerpoint. You still have to think, and you still have to be able to interpret what you see, but I think this will be a step forward in advancing the science and the repetitivity over the art of this whole room measurement thing. Selfishly I am asking the developer of REW to see if he can add a feature to ease this multiple window thing and see if he is inspired by the idea of improved psychoacoustic room response measurements (and maybe in the future, room correction). Onward and upward....

Bob Katz

JohnM edit: link to presentations http://www.aes.org/sections/pnw/ppt.htm


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## JohnM (Apr 11, 2006)

Hi Bob,

Great to see you here, thanks for the kind words - very interested to hear your thoughts and suggestions on REW. Variable windowing is one of the (many) things on my REW todo list, though if you settle on a scheme with some number of window settings I could look at adding support for it. In the meantime you can experiment with merging measurements that use different window settings using the All SPL graph, in case you haven't come across that already.


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## sdurani (Oct 28, 2010)

bobkatz said:


> JJ has also co-written an AES paper on this subject, which I have to find.


Bob, is that the DTS room equalization paper that JJ co-wrote with Zoran Fejzo?


> [JohnM edit: link to presentations http://www.aes.org/sections/pnw/ppt.htm]


John, that link doesn't work since it has a bracket ] included in the URL at the end. 
Here is the corrected link: http://www.aes.org/sections/pnw/ppt.htm


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## JohnM (Apr 11, 2006)

sdurani said:


> John, that link doesn't work since it has a bracket ] included in the URL at the end.


Oops. Thanks Sanjay, fixed it.


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## bobkatz (Mar 28, 2009)

JohnM said:


> Hi Bob,
> 
> Great to see you here, thanks for the kind words - very interested to hear your thoughts and suggestions on REW. Variable windowing is one of the (many) things on my REW todo list, though if you settle on a scheme with some number of window settings I could look at adding support for it. In the meantime you can experiment with merging measurements that use different window settings using the All SPL graph, in case you haven't come across that already.


John, I'm humbled by your expertise and barely cracking the abilities of REW. One more thing it has over Fuzzmeasure is the ability to do 1 octave averaging. I use 1 octave averaging to see the big picture, try to set the "0 line" of reference for the 0 dB point with flat around zero from, say, 200 Hz to about 1 kHz. Speaking of "flat around zero"---I would like to see a relative 0 dB line ability in REW as well as its current SPL ability. SPL is good, but I like to think, "how far above my reference zero, and how far below my reference zero does this point fall?" If my reference is 70 dB, for example, I don't want to have to do a mental subtraction of 4 dB when I see that 50 Hz is 66 dB. It's a minor point, but just a suggestion for you.

I'll try the "all SPL" option with two different window settings. Or I may just wait till you get around to multiple window options. I'm going to confer with Jim Johnston about some recommended multiple window settings and frequency ranges and give you his recommendation to see if you'll be willing to implement it. I don't think it will be that easy for you, there's all kinds of ergonomic issues to cover. Let me give you a for instance (which may or may not be the best choice): For example, Hann window starting 100 ms. before the impulse and ending 200 ms. after the impulse, running 20 Hz to 200 Hz. And then a 20 ms. Hann window centered about the impulse covering 200 Hz to 20 kHz. You wil also have to conquer the different apparent gains and splice the two gains or offsets together at the 200 Hz point. I can't imagine what other troubles this proposal may cause ;-). 

Take care and thanks,

Bob Katz


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## tesseract (Aug 9, 2010)

Glad you could join us, Bob! It's really good to see you here.


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## bobkatz (Mar 28, 2009)

sdurani said:


> Bob, is that the DTS room equalization paper that JJ co-wrote with Zoran Fejzo? John, that link doesn't work since it has a bracket ] included in the URL at the end.
> Here is the corrected link: http://www.aes.org/sections/pnw/ppt.htm



I have found three JJ papers that truly take us to the 21st century of room correction, all co-written by JJ, of course. These papers can be purchased from the AES E library and are well worth reading and digesting. I'm just beginning to digest them myself! 

The papers are preprint 8314, Beyond Coding Reproduction of Direct and Diffuse Sounds in Multiple Environments

Preprint 7263, A Low Complexity Perceptually Tuned Room Correction System

and 

8379, DTS Multi-Channel Audio Playback System: Characterization and Correction. 

My dream is that REW can start to take us to the next level by at the least allowing us to integrate two (or more?) different windows effectively for low and high frequencies. As we learn more from JJ's wisdom, perhaps we can do more, but I'd love to see that first step. I've always known that long windows are wrong for the high frequencies, but now I have an idea what to do about it. 

Crossing fingers,


Bob


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## jlohl (Jun 6, 2007)

Other interesting doc that relates to the frequency dependant windowing used by Denis Sbragion's DRC-FIR for room correction and can be found here : http://drc-fir.sourceforge.net/doc/drc.html#htoc29

I personnaly use something near this :


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## bobkatz (Mar 28, 2009)

Thanks for the link, Jlohol. Your windowing curve looks much like what JJ recommends. May I ask what room analysis application are you using or applying that variable windowing in? And the fact that you use different windowing for different frequency analysis smoothing is, to say the least, interesting. 

It seems that Denis Sbragion does not justify his variable windowing for psychoacoustic reasons, but rather for some obscure reason that I cannot fathom: "The frequency dependent windowing is one of the most common operations within DRC. This type of windowing follow up directly from the fact that within a room the sensitivity of the room transfer function to the listening position is roughly dependent on the wavelength involved. This of course implies that the listening position sensitivity increase quite quickly with frequency."

The psychoacoustics that JJ and his fellows advocate have to do with real* psychoacoustic issues regarding the lesser importance of reflected sound in the ear's sensitivity as the frequency goes up. Other issues regarding the Schroeder frequency of the room. And thirdly the directivity of the loudspeakers as frequency increases. 

*It's ironic that I use real- and psycho- in the same sentence, but, well, psychoacoustics is real . Merry Christmas, everyone.


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## Mitchco (Apr 12, 2011)

Hi Bob, I am a big fan of your book on Mastering and sound productions.

In addition to Denis DRC software, there are two commercial software apps that allow users to adjust frequency dependent windowing: Audiolense and Acourate

I have been using/experimenting with Audiolense for 18 months and have been getting excellent results. Some of those results are outlined here, but could use an update with my latest findings.

Attached is what the Correction Procedure Designer looks like in Audiolense.

The trial version of Audiolense allows for 90 seconds of correction and the associated help file goes into detail on psychoacoustics and frequency dependent windowing.

You may also want to post your q’s at the Audiolense User Forum and Acourate User Forum

What I really appreciate about REW is all the different views just from one measure - cheers John!

Happy Holidays,

Mitch


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## jlohl (Jun 6, 2007)

> May I ask what room analysis application are you using or applying that variable windowing in?


I'm using my own application base on octave scripts.



> The psychoacoustics that JJ and his fellows advocate have to do with real* psychoacoustic issues regarding the lesser importance of reflected sound in the ear's sensitivity as the frequency goes up


We can learn much from perceptual audio coding and JJ worked a lot on those ! You may obtain the curve I showed by mixing frequency resolution (ERB bandwidth, near 1/6th of an octave in mid frequencies) and shortest time resolution of our ears (coding time window of a few ms at higher frequencies).
If you read the DRC doc, you will also notice that D Sbragion also uses a spectral enveloppe detection in the latest versions. Another interesting idea.

Bob's question is really fundamental for optimised louspeaker EQ. And there is still work to fully match measurements to listening quality.


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## bobkatz (Mar 28, 2009)

Dear Mitch: Happy New Year and thanks for your nice comments and pointers. It's ironic, but I'm actually using 1/6 octave analog filters in a Meyer equalizer on the subwoofers only, to deal with low frequency anomalies because I simply do not accept the compromise (loss of transparency) that an additional A/D/A conversion adds to my system. But eventually I may have to go that route because the power of DSP-based correction is far more surgical, powerful, reliable and correctable than any analog equalizer. 

What with home theatre playback forcing me to use the analog outputs of my Marantz A/V preamp for HDMI sources (Bluray), I would always have to have an additional A/D/A conversion to do DSP-based room (and speaker) correction. But for CD playback, and all of my high-end masters at up to 192 kHz, I should be able to construct a reliable digital in/out DSP-based system. Unfortunately, every one I've encountered so far or tried to construct either suffers from extreme instability when changing sample rates OR, they have to use an ASRC (which is a loss of transparency) in order to lock the DSP at a consistent rate. 

I think the problem is we are the 1%, the small percentage of audiophiles who really care about every aspect and every detail of our system, and the vast majority of listeners are willing to accept pre-made solutions that are not as transparent-sounding as possible. If I had the time (that is, if I were not working and were retired), I would have time to "roll my own". Does any of the digital correction systems that you describe have the ability to process digital-in sources in real time, sense the sample rate change reliably, lock to the new rate without causing glitches? In that case I'd be willing to add a high end six channel (oy!) 192 kHz ADC for playing the home theatre stuff, OR hack into the HDMI and extract six channnels of PCM digital out.


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## bobkatz (Mar 28, 2009)

Dear Mitch: Just visited your site and read your blog on your room and its progress. I see that you were an active sound engineer and subscribed to the Don Davis/Richard Heyser approach which I thoroughly endorse. My treated room and listening position has a superb ETC and it is its saving grace. Still, there are anomalies I want to correct for in the most transparent way possible. Your approach to dealing with an untreated room initially struck me as bass-ackwards until I read your full blog and saw all your acoustical efforts to get the ETC below -15 dB for all reflections, so your heart is definitely in the right place! But you seem to have much too much time on your hands! I see that the room correction community that you follow is progressively developing powerful tools that don't need much fiddling to get where you want to get. But still, audio is a full-time profession for me and I have to work day in and day out with this reproduction system. I really can't afford any down time. 

In my Studio B, we struggle with a DSP-based correction system that sounds pretty good, but it is locked digitally (my insistence) to the source sample rate with no sample rate converters in the chain, and so my assistant has to reboot that processor nearly every time he switches between a double and a single SR or changes digital sources. It's not ideal, but the increased sonic transparency is worth it, and at least for now we can afford the five minute aggravation. There are no clients in the room, usually, so its instabilities are not overwhelming. 

However, In Studio A, where I work, I cannot afford that kind of down time. I have to get things done. So I look forward to the day when one of the correction tools you mention mature to the point where they can take in digital in signals, NOT sample rate convert internally to suit its fixed filters, and output digitally to my DACs. This strikes me as a serious hobby and I don't mean to denigrate the fabulous work that has been accomplished so far (I'm impressed!), but I cannot imagine myself with enough free time to experiment yet with these systems in their current state. Someday, though! "In my copious free time."

Looks like a dedicated DSP-based correction system in Studio A is in my future, though!


Again, Happy New Year, everyone.


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## Mitchco (Apr 12, 2011)

Hi Bob,

Thanks for looking at the software and blog post. Since it is now a hobby for me, I do have all of the time in the world  Most of the time was spent on the “target” response as there seems to be very little information in the way of what a preferred in-room frequency response at the listening position should be.
I found some correlation here Btw, check out Brad’s multi-channel home theater system that is linked in that post. 

Wrt to Audiolense, it takes about 15 minutes to set up the mic, take a measurement, filter the measurement, apply a target curve, generate the digital FIR filters, plug them into a Convolver, and be listening to the result.

Audiolense can generate 64-bit FIR filters at all the usual sample rates, at a variety of taps (all the way up to a million taps). I use JRiver’s Media Center 18 64-bit audio engine and built-in Convolver which automatically switches native sample rates (and the Convolver) in real-time with no glitches.

Btw, there is no additional AD/DA conversion anywhere in my signal path, nor is there any sample rate conversion, it is all at whatever the source material’s native sample rate is and JRiver (and the Convolver) switch in real time with no glitches (in my system). With JRiver, this also can be done with 5.1 or 7.1.

I am using a Lynx Hilo as the DAC and the on-board monitor controller that outputs directly into my DIY Nelson Pass Class A amps. The ADC I use for the output of the mic preamp and measurement mic for REW and Audiolense. As an aside, I also use the ADC for vinyl ripping and I still do some amateur recording using these fabulous binaural mics.

Using JRiver’s loopback or ASIO line in capability, I can loop or route the swept sine wave (both Audiolense and REW) through JRiver and with a check box on the Convolver, I can measure the before and after in minutes. Note that JRiver’s Convolver engine eliminates any delay that the digital FIR filter introduces in the signal path. Again, all glitch free in my system. With JRiver's ASIO line in capability, any DAW software should be able to feed the line in of JRiver's audio engine, with or without Convolution...

Hope that helps. 

Happy New Year!

Mitch


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## bobkatz (Mar 28, 2009)

Dear Mitch:

Your reply (quoted entirely below with my questions inserted) is VERY revealing and enticing. Now I'm really thinking about doing this!



Mitchco said:


> Hi Bob,
> 
> Thanks for looking at the software and blog post. Since it is now a hobby for me, I do have all of the time in the world  Most of the time was spent on the “target” response as there seems to be very little information in the way of what a preferred in-room frequency response at the listening position should be.
> I found some correlation here Btw, check out Brad’s multi-channel home theater system that is linked in that post.


Regardless of the method of measurement, my experience is that if it measures flat it's probably going to sound bright! For reasons cited, for example, by JJ. Especially for home theatre but also for music. I don't use an x-curve, but when I play movies I usually have an additional 1 or 2 dB rolloff beginning at some high frequency. But this is going to be different if you narrow the high frequency measurement window down to near anechoic. So play the chicken and the egg thing until the high end sounds right ;-)



> Wrt to Audiolense, it takes about 15 minutes to set up the mic, take a measurement, filter the measurement, apply a target curve, generate the digital FIR filters, plug them into a Convolver, and be listening to the result.
> 
> Audiolense can generate 64-bit FIR filters at all the usual sample rates, at a variety of taps (all the way up to a million taps). I use JRiver’s Media Center 18 64-bit audio engine and built-in Convolver which automatically switches native sample rates (and the Convolver) in real-time with no glitches.


Does it apply 24-bit dither on the way out to the DAC?



> Btw, there is no additional AD/DA conversion anywhere in my signal path, nor is there any sample rate conversion, it is all at whatever the source material’s native sample rate is and JRiver (and the Convolver) switch in real time with no glitches (in my system). With JRiver, this also can be done with 5.1 or 7.1.


So you are using JRiver as a "live", real time digital in/out device?



> I am using a Lynx Hilo as the DAC and the on-board monitor controller that outputs directly into my DIY Nelson Pass Class A amps. The ADC I use for the output of the mic preamp and measurement mic for REW and Audiolense. As an aside, I also use the ADC for vinyl ripping and I still do some amateur recording using these fabulous binaural mics.
> 
> Using JRiver’s loopback or ASIO line in capability, I can loop or route the swept sine wave (both Audiolense and REW) through JRiver and with a check box on the Convolver, I can measure the before and after in minutes. Note that JRiver’s Convolver engine eliminates any delay that the digital FIR filter introduces in the signal path. Again, all glitch free in my system. With JRiver's ASIO line in capability, any DAW software should be able to feed the line in of JRiver's audio engine, with or without Convolution...
> 
> ...


Yes, this helps and it's very tantalizing! The Convolver cannot completely eliminate delay, it has to have some finite latency.... which is not a big deal except for lip sync. Can you adjust delay for lip sync, by the way? I still have a couple of other questions if you don't mind. In order to change sample rates without any asrc in the path, Jriver would have to have a set of pre-calculated filters on hand for each sample rate (or calculate them in real time which would be crazy). And seamlessly switch them as it detects sample rate changes. It also needs to be smart enough to detect loss of input sync and reset its dsp when good sync is detected, so as not to crash the code. So, do you have the software configured to do that already? No glitches when switching rates (or maybe a little click is ok). I dropped into the JRiver discussion forum and found references to adding lines to the code (which I'm not afraid of doing, of course) but I would like to know how mature that functionality is at this point. I could easily insert it in and out of my digital signal path while experimenting, and I can insert or remove the analog filters, so I suppose I won't lose any heavy time. My weekends are about to become much fuller .

What sound card or interface are you using with JRiver? What operating system does it run on? You've tantalized me enough so that I'm thinking of getting one of those one-card standalone stripped-down computers and a sound card and trying this out. I did not think that there was anything out there that was true dig-in/dig out with no sample rate conversion in the circuitry---that's stable as well! 

How responsive is the developer to ideas? I imagine very. JJ's idea of cancelling the first reflection (at least partially) is very intriguing as well.


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## Mitchco (Apr 12, 2011)

Audiolense automatically generates the FIR filters at whatever sample rates you select in a check box. It can generate a full set (44.1 to 192KHz) in a matter of a few seconds. In JRiver, the Convolver open file dialog box points to the newly minted config file which then refs the filters. All in less than a minute from filter generation in Audiolense to listening in Jriver. There is no hand coding or manipulating the files whatsoever. It is all generated and automatic.

Depending on the FIR filter length, there can be anywhere from 350 to 750 milliseconds of delay. However, Matt (head s/w dev) at JRiver has found a way to eliminate that delay to 0ms or pretty close. I can literally A/B the FIR filter while clicking the checkbox to turn the Convolver on/off and there is brief pause (silence in milliseconds), all while the music is playing in real-time. The music sounds "in time" as I check and uncheck the Convolver while playing. It is quite amazing. For video, there is an option to adjust for lipsync for any additional delays.

Wrt to target curve. Yes, flat is too bright for sure at the listening position. In Sean Olive's studies, that is referenced in the previous link, it seems that a straight line tilt that is 0dB @ 20Hz and -10 db @ 20 KHz seems to sound the most natural. Myself and others tend to agree. Ultimately, the tilt down point at 20KHz, by how many dB, I think is dependent on many things, like how live/dead the room is, directivity index of the speakers, etc.

However, the point I am making is that I prefer the straight line tilt, versus flat out to 2Khz or whereever and then roll off. The latter does not sound natural to me, but sounds rather "rolled-off" just like the "curve" The straight line like flat, but tilted down at 20KHz does sound more natural. I can attach a couple of REW graphs if interested to see what I am talking about or look at the top curve of Slide 24 in Sean Olives presentaton. I should point out that Audiolense also allows for generation of linear phase FIR filters. The result is a time aligned response at the listening position, especially if you have separate subs, or in my case horns, but that's a whole nother story.

With respect to dither, you will have to ask Matt that at JRiver as I don't know.

I use JRiver as a music player (does DSD as well) and streaming music, watching DVD's Blu-ray, streaming Netflix, etc. It does have a digital loopback and an ASIO line input that you can run any digital source through. My HTPC is nothing special running Windows 7 64-bit and I connect my Lynx Hilo via USB (there is no sound card or any other interface) and using Lynx's multi-client ASIO driver for all software. 

The latter is something to really think about as with ASIO it's low latency, but the Lynx driver is also multi-client. Which means multiple software programs can share the same I/O at low very latency. Works very well for me, but other DAC's and non-multi-client ASIO drivers may not, so YMMV and may require experimenting.

The functionality in JRiver and Audiolense is quite mature as I have been using both for over 18 months and there have been many new features that make this real-time native sample rate switching and Convolution (hosting Audiolense generated FIR filters) a reality.

Matt at JRiver is very responsive as the Convolver portion was a community driven feature request. Btw, JRiver can host any number of VST's as well. Bernt at Audiolense is also extremely responsive and his DRC solution is probably one of the most advanced as much work has been done in the time domain as well - reference back the attached pic of the ability to change frequency domain windowing. That's another topic too ;-)

Hope that helps. Cheers!

Mitch


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## bobkatz (Mar 28, 2009)

Wow, Mitch, thank you for the detailed reply. What I want to do is buid a separate PC dedicated to Audiolense and (if necessary), JRiver. But I would not use JRiver as a media playback system. I have MANY different Digital (AES/EBU or SPDIF) sources, three different DAWs (both Mac and PC), each with its own interface, to iTunes on the Mac to a Logitech Transporter, CD transport, etc. etc. These are all switched digitally through a digital router into a single DAC. Monitor gain is a Cranesong Avocet. 

So, if I want to play my DAW, it gets to Audiolense (as you describe below, via some kind of ASIO input... I guess that would be a soundcard with a digital input and an ASIO driver) JRiver takes it in and plays it using the Convolver in audio lense. 350-750 ms. of delay is untenable for video playback and very hard to get used to for audio playback because as I work there should be some correlation between the visual moving waveform and what I hear. I already have about 250-300 ms. of delay with my own buffers and that's on the threshold of unacceptable. I might be able to tolerate another 100 ms. I assume this extra delay would have to exist when using an external source (like a CD transport) but Matt at J River has found a way to get rid of it when using JRiver as the playback module. But basically I need a processor (room EQ), not a new media playback system. However, if I can get six channel PCM digital out from a computer-based Blu-Ray into the system and to an external DAC, and it can play SACD, CD, DTS, Dolby True HD, Blu-Ray, etc. I might be willing to get rid of my Marantz A/V preamp! Possibly you can even convince me to use a card-based DAC for the Blu-Ray and feed analog from it to the Avocet if it proves the most convenient for video-based sources. 

Lots of food for thought there, you've awakened the Kraken!

Bob



Mitchco said:


> Audiolense automatically generates the FIR filters at whatever sample rates you select in a check box. It can generate a full set (44.1 to 192KHz) in a matter of a few seconds. In JRiver, the Convolver open file dialog box points to the newly minted config file which then refs the filters. All in less than a minute from filter generation in Audiolense to listening in Jriver. There is no hand coding or manipulating the files whatsoever. It is all generated and automatic.
> 
> Depending on the FIR filter length, there can be anywhere from 350 to 750 milliseconds of delay. However, Matt (head s/w dev) at JRiver has found a way to eliminate that delay to 0ms or pretty close. I can literally A/B the FIR filter while clicking the checkbox to turn the Convolver on/off and there is brief pause (silence in milliseconds), all while the music is playing in real-time. The music sounds "in time" as I check and uncheck the Convolver while playing. It is quite amazing. For video, there is an option to adjust for lipsync for any additional delays.
> 
> ...


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## Mitchco (Apr 12, 2011)

Hey Bob,

Maybe I did not write as clearly as I should have. Audiolense's only purpose in life is to measure the room acoustics, apply it's speaker/room correction filter magic, and generate the FIR filters.

Once those FIR filters are generated, they then plug into JRiver's Convolution engine for playback of program material. So if you want to play your DAW, it's output would be routed to JRiver's digital input so it now can pass through the 64-bit Convolution engine, which hosts those Audiolense generated speaker/room correction FIR filters.

Wrt to the 350ms to 700ms delay that belongs to the FIR filter, JRiver adjusts the latency to 0ms. I don't know how Matt does it but it does.

So Audiolense is the producer of the FIR filters, which could be a one time only thing, unless you are like me and always experimenting with speaker positioning/room acoustics. All of the playback goodness is only available via JRiver and with it's 64 bit audio and Convolution engine. You could use just that as an independent audio engine and digitally route all of your other audio feeds through it as the last piece of software in the playback chain.

Cheers!

Mitch


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## bobkatz (Mar 28, 2009)

Happy January 1st! Well, this clears up everything. It makes sense (at least to me) what the deal with the delay is. the calculation time (latency) for the CREATION of the filters is in the hundreds of milliseconds. But of course the filters themselves are only a certain number of samples long, depending on the tap length. Once the filters are created, they become a set of coefficients (multipliers) for each sample of the audio, and the latency there can be very small, only a couple of milliseconds at most with typical CPU speeds. The way JRiver makes it "0 ms" insertion loss is not by breaking the laws of physics or inventing a "digital advance line", but simply to start with a fixed buffer length that's longer than the longest delay with the correction engine added, and then when he removes the correction engine, he restores the missing part of the buffer so the resulting total delay is always the same.

I'll start a dialogue with JRiver about the playback engine to see if they have dithering. It would be a requirement to recover the most depth from the material. 


Thanks for all your help, Mitch,


Bob




Mitchco said:


> Hey Bob,
> 
> Maybe I did not write as clearly as I should have. Audiolense's only purpose in life is to measure the room acoustics, apply it's speaker/room correction filter magic, and generate the FIR filters.
> 
> ...


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## acoustat6 (Mar 7, 2008)

Hi Mitch, You said "it seems that a straight line tilt that is 0dB @ 20Hz and -10 db @ 20 KHz seems to sound the most natural. Myself and others tend to agree. Ultimately, the tilt down point at 20KHz, by how many dB, I think is dependent on many things, like how live/dead the room is, directivity index of the speakers, etc.

However, the point I am making is that I prefer the straight line tilt, versus flat out to 2Khz or whereever and then roll off. The latter does not sound natural to me, but sounds rather "rolled-off" just like the "curve" The straight line like flat, but tilted down at 20KHz does sound more natural. I can attach a couple of REW graphs if interested to see what I am talking about"

I would love to see some of your REW graphs showing this. 
Thanks,
Bob
PHP143


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## jlohl (Jun 6, 2007)

> Ultimately, the tilt down point at 20KHz, by how many dB, I think is dependent on many things, like how live/dead the room is, directivity index of the speakers, etc.


I did an online calculator to do a rough estimation of the tilt depending on DI, room, distance. It generally gives a good starting point.
http://www.ohl.to/calculators/targetcurve.php


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## Mitchco (Apr 12, 2011)

bobkatz said:


> Happy January 1st! Well, this clears up everything. It makes sense (at least to me) what the deal with the delay is. … to start with a fixed buffer length that's longer than the longest delay with the correction engine added, and then when he removes the correction engine, he restores the missing part of the buffer so the resulting total delay is always the same.
> Bob


Thanks Bob, makes sense to me too. I hope you give JRiver’s audio engine a try and also Audiolense as both are state of the art. I should have mentioned that Audiolense also creates multichannel correction filters and digital XO.



acoustat6 said:


> Hi Mitch, You said "it seems that a straight line tilt that is 0dB @ 20Hz and -10 db @ 20 KHz seems to sound the most natural. Myself and others tend to agree. Ultimately, the tilt down point at 20KHz, by how many dB, I think is dependent on many things, like how live/dead the room is, directivity index of the speakers, etc.
> 
> However, the point I am making is that I prefer the straight line tilt, versus flat out to 2Khz or whereever and then roll off. The latter does not sound natural to me, but sounds rather "rolled-off" just like the "curve" The straight line like flat, but tilted down at 20KHz does sound more natural. I can attach a couple of REW graphs if interested to see what I am talking about"
> 
> ...


Hi Bob, sure attached is one of my measures at the listening position some 9 ½’ feet away. The horn tweeter rolls off just before 20 KHz and QB3 vented box with a 15” woofer rolls off at around 34 Hz. I also refer you to Sieglander’s response here: http://www.hometheatershack.com/forums/rew-forum/63614-what-does-good-curve-look-like.html



jlohl said:


> I did an online calculator to do a rough estimation of the tilt depending on DI, room, distance. It generally gives a good starting point.
> http://www.ohl.to/calculators/targetcurve.php


Hi jlohl, I tried your calculator, very nice and modeled quite close to my measured response!

Cheers,

Mitch


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## bobkatz (Mar 28, 2009)

JohnM said:


> Hi Bob,
> 
> Great to see you here, thanks for the kind words - very interested to hear your thoughts and suggestions on REW. Variable windowing is one of the (many) things on my REW todo list, though if you settle on a scheme with some number of window settings I could look at adding support for it. In the meantime you can experiment with merging measurements that use different window settings using the All SPL graph, in case you haven't come across that already.


John, I've just finally learned enough about REW to have grasped and tested your approach to having two different windows in the measurement. Let me please confirm your current state of REW: 

1) Is it currently necessary to make two different measurements, each one covering (sweeping) a different frequency range?

2) When splicing the two measurements together, is it currently possible to morph them, that is, gradually shift from one window size to the other? 

Your development speed and dedication is remarkable. I have rarely (maybe never) seen any application developer as dedicated to his application as you are! Please do not rush this request on my account... you have your priorities and I'm sure I can get by with the current rich feature set.


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## acoustat6 (Mar 7, 2008)

Hi Mitch, Impressive freq response! What do you use for eq? And can I assume it is full range? You appear to be +/- 2db, wow :clap:, across the entire freq range!
Bob


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## bobkatz (Mar 28, 2009)

acoustat6 said:


> Hi Mitch, Impressive freq response! What do you use for eq? And can I assume it is full range? You appear to be +/- 2db, wow :clap:, across the entire freq range!
> Bob


I hope it sounds as good as it looks ;-)

BK


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## JohnM (Apr 11, 2006)

bobkatz said:


> 1) Is it currently necessary to make two different measurements, each one covering (sweeping) a different frequency range?


It depends a little what the exact intent is, but you could likely just load the same full range measurement twice and apply different window settings to each via the IR Windows dialog.



> 2) When splicing the two measurements together, is it currently possible to morph them, that is, gradually shift from one window size to the other?


Not currently, that is in essence a variable windowing feature, which is on the list of things to do. The splicing switches from one measurement to the other at the splice point, but does level and phase align them at that point to provide a fairly seamless joint.


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## Mitchco (Apr 12, 2011)

acoustat6 said:


> Hi Mitch, Impressive freq response! What do you use for eq? And can I assume it is full range? You appear to be +/- 2db, wow :clap:, across the entire freq range!
> Bob


Hi Bob, yes, a 3-way full range kit designed by Bob Crites, driven by Nelson Pass DIY Class A amps, and a Lynx Hilo.

Audiolense is the Digital Room Correction (DRC) software I use to correct frequency and time domain issues for speakers and rooms.

The measurement I supplied is from Audiolense ver 4.4 Beta. I would be remiss not to post an updated REW graph of the latest Audiolense ver 4.6 which has visible improvements. I also included the response before DRC.



bobkatz said:


> I hope it sounds as good as it looks ;-)
> BK


Hi Bob, yes it does ;-) Especially given the dimensions of my room has one ratio a multiple of another, the speaker setup is offset along the wall, and horn loudspeakers are not the smoothest response, or time aligned in my case. I would imagine folks with better room ratios and smoother response speakers should best my response.

Not saying frequency response is everything, but coupled with a room with decent ratios, symmetrical setup, good RT60, waterfall, and ETC, should produce really good results, and with Audiolense, excellent results. 

I see Audiolense as the icing on the cake to fine tune timbre, balance channels, time alignment, and as Bernt points out, cleans up some early reflections and acts like an low frequency active bass absorber. You could say I am enjoying the sound, and looks like you soon will be too :sn:

Cheers,

Mitch


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## acoustat6 (Mar 7, 2008)

Hi Mitch, Thanks for the info and the graph. 
How many filters did it take to achieve the results? 
How many were gain filters and how many were cut filters?. 
How come the response rolls off at 50hz?

Hope you dont mind the questions!

Thanks,
Bob


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## Mitchco (Apr 12, 2011)

acoustat6 said:


> Hi Mitch, Thanks for the info and the graph.
> How many filters did it take to achieve the results?
> How many were gain filters and how many were cut filters?.
> How come the response rolls off at 50hz?
> ...


Hi Bob,

Some answers to your questions can be found in one of my blog posts at Computer Audiophile. 

But to summarize:
How many filters? A linear phase FIR filter, with 65,536 taps. 

Gain/cut - it depends on parameters in Audiolense such as max boost, target frequency response, partial correction, etc. See my blog post, near the bottom, shows a graphic of the target frequency response and correction filter. Post 10 has a pic of the Correction Procedure Designer which has also has some of the parameters. Consider downloading Audiolense as the help file goes into detail, which would be too lengthy to describe here.

Roll-off at 50Hz. The -3dB point is at 34Hz, but starts to roll at 50Hz. Looking at my target frequency response in the blog post starts to roll-off at 50Hz, that's part of it, and user defined. The other part is speaker placement. Ideally my Cornscala's would be placed near corners, firing down the length of the room. 

Unfortunately, I can't make that happen, so the speakers are firing across the width of the room, and away from the corners, with the right speaker (green trace in above graph) almost at mid-point along the long wall - hardly ideal and I am paying for it. 

Audiolense is doing a great job in correction for less than ideal room ratio (love Bob Gold's Room Modes Calculator) and speaker placement in the low end. And likely the 15" woofers and large QB3 vented boxes. See before and after correction, level matched ~85 dB SPL, in the REW overlay pic above. 

Hope that helps answer your questions.

Cheers, Mitch


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