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mitchco

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  1. Hello @cpdk welcome to Audiophile Style and thanks for your comment. Yes, the answer is in the 5th paragraph of the conclusion. Kind regards, Mitch
  2. @JR4321 Great question!! I can offer one perspective with some thoughts: 1) Room correction is a technically complicated subject area as there are several aspects to it. As alluded to in the article, some aspects are; audio digital signal processing, loudspeaker directivity, Schroeder frequency, standing waves, room resonances, early reflections, late reflections, the ears amplitude and frequency response non-linearity’s to sound, what to correct, what not to correct, etc., is what makes it a technically complicated subject area. I have yet to find a well written “layman’s” article on how room correction works and what the results are supposed to be. I go into gory detail in my book, but it isn’t a summary article. There is such a thing as an ideal loudspeaker in an ideal room that can be modelled and emulated. Unfortunately, most explanations are either too technical or are (wildly) incorrect, which leads me to a 2nd reason. 2) Because of the technical complexity, most folks don’t have the technical skills to understand what they are seeing on a chart and how it correlates to what we hear. So it makes it difficult to know what the end result should be or sound like, and most importantly, how to obtain it. There have been several scientific studies that correlates what makes for a good sounding loudspeaker in a room with objective measurements. Floyd Toole and most notably, Sean Olive have undertaken several scientific studies in this area correlating peoples listening preferences with objective measurements with loudspeakers in rooms and room correction systems. I have linked to a few studies in this article, plus other room correction articles I have written. In fact, there are literally dozens of repeatable scientific studies in this area, but again requires some level of technical acumen to spend the time reading and understanding. 3) This ain’t your granddaddies 31 band eq 🙂 Modern Digital Signal Processing (DSP) software is very sophisticated and coupled with powerful computers one can alter the frequency and timing response of a loudspeaker in a room to just about anything you want. Put another way, not all DSP room correction systems/techniques are the same and some are orders of magnitude better than others. Some work in the time domain, others do not. Some offer tailoring of every possible parameter, others do not. The list goes on. But perhaps the most important point is that most folks don't know how powerful software room correction has become. 4) And the elephant in the room, double pun, is that loudspeaker manufacturers (mostly) do not want folks to use room eq with their loudspeakers for a whole pile of reasons (e.g. don’t mess with my voicing, mics aren’t ears, yet those are used in the design of the loudspeaker in the first place, and a laundry list of other myths built up over the years - some justified with very poor eq systems from yesteryear). The biggest reason being that if one were to use state of the art DSP room correction system on Speaker A and then use the same target frequency response for a similar Speaker B and then A/B them (which is difficult to do) one may be hard pressed to tell the two speakers apart. Kudos to some speaker manufacturers, who have the confidence in their products to actually make a statement like that. For example, Martin Mensink, designer of the Dutch and Dutch 8c, “I've had the Kii's and the 8c's side by side in my living room for a while. The Kii's too are remarkably good speakers. With just some subtle EQ the two could be made to sound very similar on most program material - to the extent that I might not be able to distinguish them in a proper blind test. I'm still amazed sometimes by the extent to which differences in sound can be explained by frequency response.” Having performed the same experiment above myself with the speakers mentioned and others, there is considerable truth to this. The irony is that for around $500, which includes the calibrated measurement microphone and state of the art room correction software, will make the biggest sonic optimization/improvement that one can make to your existing sound system that is both audible and measurable for the $’s spent. The caveat is one must know what one is doing to achieve a successful result. Companies like Dirac are making it easier to obtain a good result by limiting the amount of variables the user can play with. But as I mentioned in the article, that in itself is also a trade-off. At this stage in the progression of software based room correction adoption, most folks that have experienced good room correction, even just for levelling out the bass frequencies, would not go without it. The audible difference in smooth sounding bass versus uneven bass response, virtually everyone can hear the difference as the difference is considerable. We are talking going from +20 dB peak to peak ripple response in the low end to +- 3 dB envelope. Everybody that has a pair of ears can hear the difference 😉 Again, part of this is the education to know that below the room’s transition frequency into standing waves/room modes, is to know that the room is in control of the bass response, not the loudspeaker. It takes a bit of understanding/time to wrap ones head around that. Hope some of that is useful. I have considered writing a layman’s article on the subject area to help folks understand what is being measured, what is being corrected and why. Maybe it is time... Have a great weekend! Mitch
  3. @bobfa check out my review that includes adjusting Active Room Matching and PEQ's. I hope you are enjoying them!
  4. Hi @GSWaul Sorry for the delay. Not sure about your details, I presume the laptop and desktop on the same network? If so, it should be able to see it. You might have to make a few adjustments as described here: https://forum.avast.com/index.php?topic=221321.0
  5. @bwhitejrThank you for purchasing my book. While it does cover Acourate, the majority of DSP concepts and even the procedures, albeit implemented differently, apply to most "room correction" software. Both Acourate and Audiolense are fully user adjustable, produce "textbook" measurement results and sound great. Good luck!
  6. Good question! There are a couple of popular diy techniques using donationware that can definitely improve the response of your loudspeakers in your room: One is the Moving Mic Measurement (MMM) technique: http://www.ohl.to/audio/downloads/MMM-moving-mic-measurement.pdf But is only frequency correction and does not provide excess phase correction. For some that is good enough. Another is using REW with rePhase: https://www.dropbox.com/s/10xdhh83jokzbxv/REW_rePhase_tuto.pdf?dl=0 While it has excess phase correction, it is not the same as Audiolense (or Acourate or Dirac for that matter). But again, for some, good enough. I have compared all of the above, including the commercial software listed and others, and there is a reason for paying for commercial applications which results in the best correction, especially low frequency correction. The commercial software packages employ additional analysis and filtering techniques that are proprietary and not present in either REW or rePhase. Note that both REW and rePhase are not marketed as "room correction" software but can be DIY adapted to provide some of the capability of the commercial offerings. But that's the difference and what makes the commercial offerings unique, especially if it does a better job than the freeware, which based on my measurements and listening tests does. For example, here is the phase response of my large double 15" "ported" cabs along with dual 18" sealed subs crossed at 46 Hz in an asymmetrical setup, measured at 9 ft at the listening position from 10 Hz to 200 Hz. The speakers were DSP'd using Audiolense with True Time Domain (TTD) correction. The left and right REW measurements were made at the listening position with 1/12 octave smoothing in REW using the default window of 500ms with no Frequency Dependant Windowing or any other manipulation in REW. Note the vertical scale. I have yet to see anyone show a similar measurement, using the same REW settings I have, with REW and rePhase. However, all of them will improve the low frequency response of your loudspeakers in your room, which is a good thing 🙂 Hope that helps. Kind regards, Mitch
  7. Hi @One and a half yes, once you take the measurements, and happy with them, you are good to go, and can put the measurement mic away. Of course if you change speakers, speaker placement, listening position, etc., then you will need to measure again. Now that you have the measurements, you can play with the target response and try a partial correction from 600 Hz on down, or full range correction, different target curves, etc., and easily A/B them in Dirac Live Processor while listening to music. Once you have settled on what sounds best to your ears, it is set and forget and enjoy the music!
  8. While Sonarworks is indeed another option, it is not in the same category as Dirac, Acourate or Audiolense, it is amplitude correction only. It does not have any timing correction capabilities.
  9. @One and a half you need digital loopback capability to route the output of Roon to the VST and then to the DAC. On the PC something like https://www.vb-audio.com/Cable/ or https://vac.muzychenko.net/en/ should work. I see this person got it to work in a basic setup: https://community.roonlabs.com/t/roon-vst-plugins-hifiberry-digi-pro-possible/73928 There is also a Dirac doc on how to do it as well: http://diracdocs.com/Windows-Using_Dirac_plugins_with_players_not_supporting_plugins.pdf But, I have no idea about DLNA... Unfortunately, most hardware devices are limited by the processing power required for low frequency control. The lower the frequency, the more "filter taps" required. So most hardware is limited to 8,000 FIR filter taps, whereas on the PC we can easily do 65,536 or even 131,072 taps. https://www.deqx.com/ is about the only hardware solution I would consider, but it is considerably more expensive than Dirac...
  10. It is not so much the direction and cues, which is indeed affected to a certain degree, it is more of the “comb filtering” that colours the frequency response is the main issue. I can’t show it because of copyright issues, but if you happen to have a copy of Bob Katz’s excellent book on Mastering, there are a couple of charts that show the frequency response comb filtering issues when, in his case, a mastering desk is placed between him and the speakers. Bob also adjusted the angle of the desk to minimize the comb filter to almost be the same as without the desk there. When I “critically listen” I move my coffee table out of the way. Aside from the comb filtering colourations, I want to hear the full radiation pattern off the speakers for the best possible imaging. When I place the coffee table back, I can hear the comb filter coloration and the image height is reduced (i.e. the bottom half), by a bit, but since our ears/brains quickly adapt, it is soon forgotten and I am back to background music listening. I also have a “half back” couch as when I had a full back couch, it drove me nuts as I moved further back into the couch, I could easily hear the comb filtering from both channels reflecting off the couch and into my ears, aside from cutting off the ambient sound from behind me. But the most important aspect is that if one corrects for the “comb filtering” off the coffee table for example, then that correction is now embedded in the frequency response of the loudspeaker including it's off axis response. We know from Sean Olive’s and Floyd Toole’s extensive work that a smooth off axis frequency response is just as important as smooth on axis response (think spinorama). So if we correct for the comb filter off the coffee table, we have just coloured the off axis response of the loudspeaker. So now the combination of the direct sound and early reflections in the listening window sounds “coloured” even sitting in the sweet spot and worsens when one moves around the room. This is one of the reasons why folks mistaken that it is the DSP that has coloured the sound when what they have done is coloured the off axis response, so the later reflections after the coffee table are coloured and to our ears does not sound right (and never will!). There are no hard and fast rules, and I encourage folks to experiment. Having been using DSP extensively for a decade, these are some of my experiences.
  11. Thanks Ted, appreciate it. That's awesome about your room build! Would love to hear more about that or perhaps it is an article? Wrt objects between the speakers and/or chair/couch in the way... If using a full range correction, we are correcting mainly for room interactions below 600 Hz and the loudspeakers direct sound (and some near reflections like the speaker baffle, stand), so we want to move any objects that are in the direct sound path. Not move them out of the room, but off to the side temporarily during measurements and then replace when finished. So the DSP is correcting for room and loudspeaker, not room/loudspeaker and coffee table, for example. Using this approach yields the best sounding correction as one is dealing with the room and the speaker, with it's natural dispersion pattern, not broken up by objects in the way, or reflections from the couch getting in the microphone if using UMIK-1's little stand to sit on the top of the couch. Myself and others have tried corrections with objects in the way of the direct sound field and/or chair couch inches from the measuring mic and then with the objects moved to the side. The consensus is that with the objects temporarily out of the way for measurement and then returned after, sounds better than the correction with objects in the path and/or mic sitting on the chair or couch. I have tried this many times, in many rooms, and in every case, the correction sounds better with an unobstructed path to the measurement mic and placing the objects back for listening. Leaving objects in the direct sound path and/or having reflections from the couch back getting into the mic produces an inferior sounding correction. Not so much at low frequencies, but at frequencies above the rooms transition frequency, i.e. 600 Hz and above. It can alter the tone and sound like comb filtering or just unnatural sounding and then folks blame the DSP 🙂 Of course, folks are free to do whatever they want, but myself and others have found this measurement approach works more effectively for achieving the best sounding correction. Best of luck with your room! Mitch
  12. Nice speaker, a little bright sounding to my ears if I recall... Yes, they are time aligned. No, Dirac would not compromise the time alignment, if anything optimise further if possible. Yes, Dirac would be a benefit as we still have room modes to deal with. The boundary controls on the speaker are helpful, but rudimentary compared to what Dirac can do. Here is the 600 XD frequency response in my room. Dirac would indeed smooth out the response below 600 Hz:
  13. @ZapuanSorry for misunderstanding. If you know what you are doing, yes, because of additional features and/or exposing functionality for the user to control: - create multi-way digital crossovers of varying types, slopes, you have complete control. - linearize individual drivers. - time align individual drivers. - user control over frequency dependent windowing. Both low and high frequency window widths can be independently adjusted for both magnitude and excess phase correction. - the amount of correction applied.
  14. There is no reason why this would not work with Dirac. When the free trial is available, you might want to give it a try. The only issue is that Roon does not support VST plugin's at this time. So you would need to temporally try it in a different music player that supports VST plugins. Then at least you could ascertain the sonic benefit, which I think would be significant in your setup.
  15. Cheers @Zapuan As mentioned in the review, I did not get a chance to try Dirac 1.x, but according to the manufacturer, there are several improvements upgrading to D2. As I understand it, you are entitled to a free upgrade and given the simplified measurement process, it is worth a shot to try first 🙂
  16. Yes, I have seen both Toole's, plus Todd Welti's presentation on multiple subs, and Earl's paper, plus Duke LeJeune's swarm system. All show positive benefit's but one still needs room correction to obtain the smoothest bass response based on experiments I have run with multiple subs and locations. Even if your sub(s) are part of your two channel system (i.e. not using digital XO), room correction is still a major benefit, which is what Dirac does. However, the new Dirac Bass Management will provide further control, but also requires more than 2 DAC channels. REW Room Simulator works very well and is quite accurate! Room mode calculators are mostly based on the physical dimensions of one's room. Room construction and rooms treatments (to a certain degree as most do little below 100 Hz) have an impact, but below the room's transition frequency, it is all about the room ratio. The reality is that no matter where the speakers/subs are placed in the room relative to the listening positions, there is no escaping room modes. All that happens is that the dips and peaks move in frequency based on these (super)positions. One can get lucky and minimize, so it is worth the effort. However, in the end, still need room eq...
  17. I did not use a UMIK-1 for this particular article, even though I do have one and have not come across any ringing artefacts with it yet. But I am going to test it out extensively in an upcoming article. As the speakers are also gone, I can't retest easily to confirm what the issue was. I could have goofed on my loopback test and resampled somewhere in the chain like 48 kHz signal through a 44.1 kHz filter. When I get a chance, I will confirm using another speaker/test, but given the amount of time I have spent on trying to identify (pre)ringing artefacts previously, it certainly inaudible to these ears. The main point I was trying to get across was the time alignment. Understanding room correction is complicated A separate article is required to get though it all and I am contemplating writing one. In a nutshell, good room correction software will take an acoustic measurement and extract the minimum phase response and since speakers are minimum phase devices, correcting the frequency response will also correct the phase response. However, that is only half the picture as there is the room to deal with, which is what Dirac calls mixed phase or others call it excess phase, which includes room resonances and reflections. And in the time domain, we know what the ideal speaker target is by looking at the timing (i.e. step) response of an "ideal" minimum phase speaker. So... the room correction software corrects the time domain to follow the ideal minimum phase target. You can see that if you have my book or some of the articles here at AS show that as well. But the trick is that this is wavelength dependant, so at low frequencies, the correction time window is long (like 600ms for example) and as frequency increases the correction time window becomes shorter so above the rooms transition frequency we are looking at more of the direct sound plus baffle of the speaker. Often called frequency dependant windowing. In other words, at low frequencies, the room correction software has a large time window that corrects both the direct sound and reflected sound (i.e. room resonance) towards a target response. At high frequencies, it is mostly the direct sound because if we tried to also correct for mid-range and high frequency comb filtering room reflections, the high frequency response of the loudspeaker sounds like a dentists drill. This is how one can hear "over correction" in the high frequencies. Hope that helps somewhat. Like I say, requires a separate article to fully explain.
  18. Thanks! From: https://www.dirac.com/faq-general it looks the free trial is 14 days. Perhaps @flak can confirm...
  19. Yes, I did not have the subs connected, but the the little Purifi's have solid output below 30 Hz. But my Rythmik dual F18's go down to 6 Hz in my room. How is your 2.2 system connected? Is it Y cabling or ?
  20. Thanks @sem115 One will require an application to "host" the Dirac Live Processor (DLP) plugin. An oldy but goody is VSTHost on Windows. I just tried loading the DLP VST plugin and was successful. There are several VST host's available, some free, some $$'s, some standalone and some integrated with other music player software. The trick is routing the audio. I ran out of time to try DLP in this configuration. But essentially, you will want to route Qobuz or Roon's output through the application hosting the DLP plugin and then out to the DAC and rest of the system. Perhaps other members here have tried this approach and can assist.
  21. Dirac 2 is the latest version of Dirac’s Digital Room Correction (DRC) technology. In this article, I walk through the steps of using Dirac to optimize the response of the Purifi SPK4 demo kit that I recently reviewed here on Audiophile Style. Can we make an already great sounding speaker sound better in my room? And by better I mean smoothing out the low frequency response below the room’s transition (Schroeder) frequency, making small, broad band tonal adjustments in the midrange and top end to be a bit smoother and finally a timing correction of the impulse response. The short answer is yes we can. We can see the improvement in the measured frequency and timing response. I can also hear an audible difference with a tighter, more clear sounding bass, smoother overall frequency response and a more coherent timing response (i.e. stereo imaging and depth of field) across a larger sweet spot. With Dirac 2, designing and generating a partial correction from about 600 Hz on down below the room’s transition frequency is as simple as it gets. The sonic benefits are instant, much smoother bass response with no huge peaks and dips that plague virtually every room below Schroeder frequency. Full range correction comes with improved phase response (i.e. imaging and depth of field) covering a wider sweet spot, more on that in the subjective listening section. Before the walkthrough, let’s talk a bit about Dirac’s room correction technology. Technology Discussion First, what audio problem is room correction technology designed to solve? The unfortunate reality is that below a room’s transition (a.k.a. Schroder) frequency, the room controls the bass response arriving at your ears, not the loudspeakers. Due to modal resonances we get wildly different bass responses depending on where we place speakers in rooms relative to the listening position: The above chart is from Floyd Toole’s Audio Science Article. As Floyd says, “In the investigation of many rooms over the years, I would estimate that something like 80% have serious bass coloration.” For those of us that take acoustical measurements of audio systems, this comes as no surprise. The peaks and dips in the low frequency response is typical and exists in virtually every room that was not specifically designed using proper room ratios, aside from construction and acoustic treatments. Room modes are calculated based on one’s room dimensions where room resonances take control of the low frequency response regardless of the speakers being used. I like this Room Mode calculator as you can move the cursor along the frequency scale and it will output a tone at that frequency. If your computer is hooked up to your speakers (careful with the volume!), you can hear the resonances in your room by hovering the cursor over the modes in the graph. It is an ear opening experience and great for training ones ears to know what to listen for. Room modes can sound like “one note bass” where the room mode is so bad, all you hear is the single droning bass note that drowns everything else out. Or the “where did the bass go?” when certain bass notes are played. Or simply just uneven bass, some notes are there, some are gone, and some are too loud. Dirac 2 is designed to smooth out the peaks of the low frequency room response and fill in some of the dips across the listening area. Our ears are more sensitive to peaks than dips. A good primer on why we hear what we hear in small room acoustics as it relates to room correction is James (JJ) Johnston’s, “Acoustic and Psychoacoustic Issues in Room Correction.” Download the PowerPoint if you can. The first 31 slides are worth the read. Dirac uses a “mixed phase” filter approach which is a combination of IIR filters and FIR filters for efficiency. The white paper linked goes into detail behind the design, and can be summarized as: “A first main aim is to control not only the frequency domain properties of the system but also the time domain properties: The impulse responses as measured at different listening positions. In particular, we strive to reduce the “pre-ringings” (pre-echoes) that would otherwise result in an un-natural sound experience. Secondly, we use dynamic models of the sound system that are based on measurements at multiple listening positions. This is important for obtaining a robust design that works over an extended region to provide a large spatial area with good sound quality. Third, we may jointly optimize multiple loudspeakers to better control the sound pressures at different listening positions. This is done by precise phase control of the individual loudspeaker transfer functions at low frequencies. Joint optimization of a set of loudspeakers results in more distinct bass performance, better robustness of the compensation and better control of the impulse responses at different listening positions.” Room correction is a complicated (and mostly misunderstood) subject area. If there is interest, I might write another article to discuss this in more detail as it is not just “eq” that is being applied here (i.e. the time domain properties are equally important). Dirac 2 consists of two pieces of software that work together. The Dirac Live Processer (DLP) is a VST plugin that “hosts” the correction filters in a convolution engine. The Dirac Live application is used to measure your loudspeakers in your room, design the filters, and upload the correction filters to the DLP. One interesting difference to other convolution engines is that the DLP accounts for any latency issues, so when switching the DLP on or off, the audio change is instant as opposed to a ¾ second silence gap with a 65K tap FIR filter. There is also no latency when switching between filters, which makes A/B comparisons easy so you can “bracket” in on your preferred frequency response in a shorter period of time. Since the Dirac Live Processor is a VST plugin, any software music player or Digital Audio Workstation (DAW) that supports the VST plugin architecture can simply load the DLP. In this article, I am using JRiver Media Center 64 Bit DSP audio architecture to host the DLP: I clicked on Manage Plug-ins in JRiver and navigated to where the DLP VST was installed and loaded it. Here we are seeing the DLP loaded into JRiver and active with a correction uploaded into the first of eight slots available. The 2nd piece of Dirac Live is an application that you run on your computer that is mostly “wizard” driven, and steps you through the process of: Connecting and configuring audio I/O Adjusting playback and record levels Select the type of measurements you want to take (i.e. chair, couch area, etc.) Taking the measurements Designing a correction filter whether partial or full range using a target of your preference Examining the results of the correction in a simulation Uploading the correction filter to Dirac Processor Managing measurements, targets, corrections, etc. So let’s perform step 1 of the Dirac Live application wizard in the objective measurements section Objective Measurements For the measurements, I moved the couch and coffee table out of the way and the sub woofers are not active. This is the main screen you see once you have downloaded Dirac Live, installed it and connected to the DLP. Note that you are required to login to your Dirac Live account in order to connect to the DLP. Also, in order for the Dirac Live application to communicate with the DLP, the latter needs to be enabled and actively receiving an audio signal. Meaning that the DLP is enabled in JRiver and music is playing through Dirac Live Processor so it is “active.” Without the DLP being active, the Dirac Live application cannot connect to the DLP. Each step of the wizard comes with a help screen of what to do. It is concise and well written so you can quickly get the hang of what to do. Here we are selecting which recording device to use once we clear the instructions: Here I am using my Lynx Hilo ADC with a microphone preamp and calibrated measurement microphone. I have clicked into the area with the red square and loaded my measurement mics calibration file. In my case, the mic calibration file uses comma separated text whereas Dirac is expecting periods. Easy enough to open in a file editor and replace commas with periods. In most case, like using a UMIK-1 mic, the microphone calibration file will load without any calibration file modifications. Next step is to adjust speaker level and microphone gain settings: I wholeheartedly agree with “keep the master volume at the lowest setting and gradually increase it.” I have pressed play on the left channel and adjusting the levels to be in the “green.” If you have a sound level meter, 75 dB SPL C weighting at the listening position is loud enough and I would not go beyond 85 dB SPL, which to our ears, sounds twice as loud as 75 dB SPL. If you don’t have a sound pressure level meter (even a smart phone SPL app will do), then adjust to a comfortable playback volume. If you feel like covering your ears, then it is too loud. Clicking on next brings up the type of measurement you would like to take. These involve taking multiple measurements, the number of which depending on what type of listening arrangement is selected. The wider the listening area to cover the more measurements one needs to take. So it could be as few as 5 measurements and as many as 17: In this walkthrough, I tried two listening arrangements, a single chair with 5 measurements and the sofa wide imaging with 17 measurements. This is so I could also subjectively compare the two corrections. First lets walkthrough the single chair selection. Single Chair Click on Proceed to Measure: I decided to use the minimum amount of measurements for this setup to test out Dirac’s algorithms. Above, I just completed the 5th measurement. The idea is that you move the microphone relative to the placement as indicated in the diagram. In general, the mic placement is 40 to 60cm apart per placement and you can choose whatever order you want take the measurements by clicking into the location on the diagram. Note that the precise placement of the mic to the location is not crucial. As a side note, it is good to temporarily remove any objects between the speakers and the microphone positions, including the chair and or sofa at the listening position. We measure and correct for the loudspeaker and room with an unobstructed path and nothing around the microphone so we are getting a clean as a sound field as possible. We don’t want to correct for any reflections between the speakers and mic due to a coffee table or around the mic itself as that will lead to a colored sound. It is best to measure and correct for the unobstructed sound field first and then return the furniture to its original locations. Finally, I recommend getting a real boom mic stand for taking measurements. You can grab one at your local music store or order online. The little mic stand that comes with the UMIK-1 while cute, is not helpful as you need to place it on something to get ear height, which means reflections (i.e. colorations) are getting into the mic. Each time you take a measurement, Dirac will output a sweep signal starting with the left channel, then the right channel, then back to the left channel. So 3 sweeps per location times 5 locations equals 15 sweeps: Here is the screen you see when taking a measurement, with the abort button in case you need to abort the measurement for whatever reason. Once you are happy with your measurements, click on proceed to filter design: Dirac suggests a target curve based on the measurements you took. Generally speaking, one is looking for a downward tilt of the frequency response at the listening position. This is because most forward firing drivers in loudspeakers start off as omnidirectional at low frequencies and become progressively more directional at higher frequencies. The rising bass energy yields a steady state room curve with a downward tilt. How much tilt? The scientific research that Sean Olive (and earlier Floyd Toole) conducted over years shows that a slope of 20 Hz to around -8 to -10 dB at 20 kHz is what most listeners prefer as neutral sound. A high level overview of that can be found in, The History of the Harman Target. While the overview also discusses headphone target curves, which have the same preferred target (transfer function) as loudspeakers in rooms, it also discusses loudspeaker target preferences. More importantly, there are a listing of AES papers that one can get the details on. One example is the Subjective and Objective Evaluation of Room Correction Products: Of course, you are free to implement any target curve you want, but a good starting place is to use Dirac’s suggestion and if it is a bit too bright or dark sounding, one can always adjust the curve. Note that a flat in-room response is not the ideal target response as it will be exceedingly bright sounding. If you read through Sean Olive’s presentation referenced above, and in these slides, it is actually the tilting frequency response that our ears perceive as flat or neutral sounding at the listening position (see slide 25). Well worth the read. Looking at the analysis chart below, we are seeing the processed 5 measurements down to left and right, plus the suggested target curve: So these wonderful little 15L bookshelf Purifi speakers have a frequency response with the -3dB limits at 25 Hz and 22 kHz. Outstanding for such a small 2 way bookshelf speaker. Ctrl-click both checkboxes, top right to get both curves to display. Note the room resonances showing up below 300 Hz in my room and if I were to choose a partial correction frequency, which I have done a bit later, I would choose 600 Hz and below to be corrected as we can see by 600 Hz the “up and down” peaks and dips are no longer an issue. In fact, having been using room correction since 2011, and having seen many other people’s measurements of loudspeakers in rooms, 600 Hz seems to be the most common point where the room starts to have an influence on the frequency response and gradually taking over completely by the time one reaches the rooms transition frequency, which in my room is at 200 Hz. As linked earlier, you can use this room mode calculator to calculate your room’s transition frequency. See the bottom right panel where it says on one side measured and corrected on the other side? Dirac has already calculated the correction in the background and by unchecking the measurement and checking on the correction, we get: As one can see, a considerably smoother response. So not only smoothing out below 600 Hz, but also bringing down the broadband rise from 2 kHz to 5 kHz when compared to the original measure. Still a little dip left at 100 Hz, but as folks may have read in JJ’s presentation, our ears are not as sensitive to narrow band dips as compared to peaks, so all good. Note the (very) slight channel level difference in the top end frequency response. As @flak has commented here, “Dirac Live 2.x algorithm might even slightly compromise on FR response of one of the speakers in favor of the best possible phase coherence between them that is important for imaging.” We will be exploring that in the subjective listening section. Next I clicked on “Proceed to filter export” and here I am exporting the correction filter to DLP. I can give it a name and description: Click on Export filter and Dirac takes a few seconds to upload the filter: Once uploaded, I can navigate to JRiver where I have the DLP VST loaded and enabled: It is recommended to leave the delay and gain compensation on. I also adjusted the gain slider to about -6 dB for headroom management. The meters will peak red if there is any clipping, which I experienced none. I could probably go to 3 dB of headroom management, but I have plenty of power on tap. OK, now listening to music! But before I comment on the sound, I also wanted to try a partial correction to 600 Hz as can be seen as a foreshadow in slot 2 ☺ Partial Correction I can go back to my saved measurements, don’t forget to save! And back to the filter design section. Creating a partial correction is dead easy. All I need to do is grab the right marker that was previously at 22.1 kHz and drag the marker to the left to about 600 Hz: So the frequencies between 25.7 Hz and 587 Hz will be corrected and the rest left alone. Note, if I were using Dirac and Digital Room Correction for the first time, this is where I would suggest starting with a partial correction. This way one can focus your attention on just listening for an improved bass response and not a change in overall frequency response. We proceed to filter export and load the filter into the 2nd slot: As one can see, I have several filter designs loaded, but here we are focused on the partial correction. Let’s move on to the sofa, wide imaging measurements. Sofa Wide Imaging After you have selected Sofa and Wide imaging, proceed to take the measurements: In this screen shot, I am on my last of 17 measurements. 3 sweeps per measurement times 17 measurements = 51 sweeps. Whew, that’s a lot and took me about 15 minutes to get through them all. I did not use a measuring tape. I just took a guess as to where the mic should go at each location, either raising or dropping the mic height and placing the mic at the front or back of my sofa, which I have temporarily moved out of the way. Note these 17 measurements covered a 6ft wide x 2ft deep x 2ft of height variability where my couch is. At any time you can re measure a position. Again we proceed to filter design. I just used the same default target as before and then switching to the corrected filter response: The response is so smooth I need to have a look without the target obscuring the correction: Wowser! That’s based on 17 measurements around my three seat couch area. To give an idea of all of the responses around the area, Dirac has a “spread” feature that allows you to see this if you check the box in the lower right hand: Those are 17 measurements made of left and right channels at the various locations around my 6’ x 2’ couch area. One can see the variability of the measurements based on position and again, by about 500 Hz, there is less variability as the loudspeaker is in control of the frequency response and not so much the room. Switching the view to impulse response: The corrected impulse is on the left and measured impulse on the right. As one can see on the left, the impulse or transient response is a cleaned up version of the measurement. Vertical scale is amplitude and horizontal scale in milliseconds. The big vertical spike is the direct sound from the loudspeaker arriving at the microphone without any reflections. The smaller spikes, after the direct sound are early reflections. The corrected impulse response shows a clean spike and very little spikes (i.e. early reflections) after. Sound travels roughly 1 foot per millisecond, so it becomes easier to understand that right after the direct sound, are the reflections (i.e. diffraction) off the speaker cabinet and then the speaker stand and anything else nearby like floor bounce or front/side wall bounce or ceiling bounce, all of those reflections are contributing to the sound, after the direct sound and our ears have a pattern to integrate those early reflections (i.e. the Haas effect) and understand amplitude and frequency response. It is a complicated topic as it involves so many aspects of why we hear what we hear in small room acoustics; standing waves, room resonances, early reflections, later reflections, room ratios, room construction and treatments, digital filtering, FIR, IIR, our ears non-linear response to both amplitude and frequency response, the Haas Effect, which reflections are good, which are bad, and on it goes. At this stage, I just wanted to show that Dirac works in the time domain. Given multiple measurement locations, one can understand that the sound and reflections arrive at different times and in different ways depending on the mic location relative to the loudspeakers. Using the concept of superposition and what we know about the physics of how standing waves respond in a rooms, we can work out the pattern and therefore correct for a wider sweet spot, which is what the 17 measurements are about. Dirac knows the phase relationships between the measurements plus taking the speakers as a pair into account, is used to calculate the correction filter. A note about target curves. While I have been using Dirac’s suggested target curve based on the measurements, you are free to create your own target curves. As mentioned above in the Subjective and Objective Evaluation of Room Correction products, the most preferred or accurate or neutral response is one that is 20 Hz to about -8 or -9 dB at 20 kHz. Here I have dialed that in the target curve by grabbing the control points of the target and dragging them down a bit to get the desired target response. Of course, one can add and delete control points along the target (right click on the target). One can also save and load targets from the menu at the top left, beside the help icon. I was able to catch Dirac’s old correction computation as it was calculating the new correction along the new target: You can see how much I pulled down the target. It is not a lot, but is an audible difference. Just my personal preference. Of course, export to DLP and start listening: I have 7 of the 8 slots filled with a variety of filters to listen to. Before we get to the subjective listening section, here are a couple of “spot” verification measurements I made with REW. REW Verification Measurements This is just a “spot” verification of the correction. It is a loopback measurement where I am feeding the digital output of REW sweep test tone into the JRiver’s ASIO digital line input and it passing through Dirac Live Processor with the correction engaged and fed out my Hilo DAC to the Purifi amp and speakers, picked up by the measurement mic, through a mic preamp, Hilo ADC and routed to REW’s digital input. Then the measurement can be displayed in a variety of graphs to look at different viewpoints of the acoustic measurement of the room and loudspeakers. This is not a “gated” measurement as we do want the room’s low frequencies in the picture and using REW’s default impulse window of 500 milliseconds. One should note that this is not fully representative as it is one measurement made at the middle listening position. Ideally, I would take 17 measurements at the same or similar locations as I did in Dirac and vector average them in REW. And even then it still will not be quite the same as Dirac is applying more than just a vector average. However, it does verify that Dirac is doing what it is supposed to do: On the top are the Purifi SPK4 speakers measured at the listening position from my previous article and the bottom with the Dirac correction enabled. The speakers are in a 9ft equilateral triangle. I just split the screens apart to make it an easier comparison. Let’s start with the low frequency response and work our way up to the high frequencies, calling out the differences. Dirac was able to extend the low end frequency response in my room to below 30 Hz. -3 dB down at 22 Hz for the left speaker (as it is more in the corner of the room) and 28 Hz for the right speaker, as it is almost center in the room and does not get the corner boundary reinforcement. Compare that to -3 dB down at 32 Hz on the left and 45 Hz for the right speaker without correction. Looking at the 30 Hz to 60 Hz region on the correction, both speakers are matched closer together in amplitude response for a more solid bass response. This is in addition to being smoother and less peaky at 36 Hz for the left uncorrected speaker and 55 Hz for the right uncorrected speaker. Remember, our ears follow the “envelope” of the frequency response with our ears more attuned to peaks rather than narrow dips in frequency response. The big correction comes at 80 and 90 Hz as one can see in the uncorrected response. The total swing from the lowest point at 90 Hz to the highest point at 110 Hz is 23 dB. To our ears, a bass note playing at 90 Hz will be ¼ as loud as compared to a bass note at 110 Hz and vice versa. I.e. a bass note at 110 Hz will sound 4 times as loud as compared to the bass note at 90 Hz. This is classic room modes and we all got em, albeit at different frequencies dependent on one’s room ratio. While the dip is still there, it has been reduced by over 50% to ±5 dB which is very close to studio tolerance of ±3 dB. I suspect it would be in this range if I vector averaged 17 measurements in REW. Good enough for a spot check. Note Dirac pulls downs the peaks at 234 Hz, 461 Hz and 1.2 kHz. We also see a smoother response from 2 kHz to 5 kHz which just pulls down the brightness of the AMT tweeter just a bit. Finally, we extend the response a bit past 10 kHz. From good to great goes the speaker’s in-room frequency response, not only smoother but extended on both ends of the frequency spectrum. Let’s look at the timing or step response. Dirac makes a claim that “misaligned drivers in multi-way loudspeakers can be corrected by automatically applying different delays to different frequency ranges.” So let’s verify this claim by first looking at the step response of the uncorrected Purifi SPK4: So the spike (doublet) at time 0ms on the horizontal scale is the AMT tweeter and the negative going portion is perfectly blended into the start of Purifi’s PTT6.5 woofer. Classic or “textbook” passive crossover design. However, not truly time aligned. Let’s look at the corrected response: Here we see the drivers are perfectly time aligned with the straight vertical “step” as both drivers response are arriving at your ears at the same time. Claim verified. There is just a hint of preringing at the beginning of the step response. I know it is not audible, as I have experimented in detail with greatly exaggerated preringing examples in my book, “Accurate Sound Reproduction using DSP.” Completely innocuous with the main goal of time alignment of drivers achieved. While phase and time alignment are not supposed to be audible, in my listening tests I have performed over the years, they are to my ears. Excess phase correction in the low frequency response is audible to my ears as is time alignment of drivers. Perhaps a future article on more discussion around this, but suffice to say, Dirac is doing its job in the time domain. Subjective Listening Note the dual Rythmik F18 subs were not used in the evaluation. I did have to check a couple of times as the amount of sub 30 Hz output from the Purifi’s PTT6.5” (sub)woofer is astonishing. However, I do intend to use the subs in Part 2 of using Dirac Live Bass Management in an upcoming review. The first thing I wanted to try was use my reference bass recording, Madonna’s, “Power of Goodbye” to test out Dirac’s partial correction up to 600 Hz in my room. The reason I like using this recording for testing bass frequencies is because it has long sustained bass notes. This makes it easier to hear the room’s resonances. As expected, there are some bass notes that practically disappear and other bass notes that dominate the sound in my room. Enabling Dirac brings the level of bass notes to the same amplitude and the bass response is solid and powerful on each sustained note. It is still amazing to me as when I was working in recording studio control rooms, the only way to control room modes was the geometric and acoustic design of the control room. Aside from the room controlling reflections and acoustic treatments, the bass traps required to deal with very low frequencies at long wavelengths meant for a special acoustic design: a room within a room. The bass would pass through the shell of the inner room and be “trapped” in the outer shell so as not to pass back into the inner room. You can imagine the construction costs. Now at the flick of a digital switch, for truly a fraction of the cost, one can have the same smooth solid bass response at every frequency. What a time to be alive! Another fav track I have for listening to bass evenness is Rebecca Pidgeon’s “Spanish Harlem”. A very nice Bob Katz recording and master of an acoustic bass guitar, in addition to Rebecca’s wonderful vocal rendition. Spanish Harlem, in the key of G, uses the classic 1, 4, 5 progression. Here are the frequencies of the fundamental notes of the bass: So without Dirac enabled, the 62 Hz to 82 Hz range of bass notes are really down in sound pressure level and the 110 Hz bass note literally sounds like a “coke bottle” resonance it is so loud and dominating in my room. Enabling Dirac, brings back the proper level balance in the low end where each bass note is heard succinctly and even in level. When your ears starting tuning in, it really becomes apparent the differences. Having nice solid, even sounding bass arriving at your ears is a real treat. The reason why I say our ears need to tune in is because we are so used to listening to uneven bass, we are simply used to it and don’t give it a second thought. Remember JJ’s presentation. But once you hear how smooth it can be with your existing loudspeakers in your room, it is impossible to go back. Just the ability to smooth out the bass response is well worth the price of the software, aside from the additional benefits we are about to discuss. Let’s try the full range correction, but for the single seat. I find it a great boon being able to switch on and off DLP while listening in real time. There is no delay or interruption of sound or glitches of any sort, just a smooth and transparent switch. This makes is really easy to compare uncorrected to corrected or comparing different correction filters. This makes it easy to hear the differences instantly and then appreciate them over a period of time. SRV and Double Trouble’s, “Tin Pan Alley” is one of my favorite blues recordings. It was the first song and first take of the recording for the album that captured the feeling so well. When I was recording, I was always trying to get bands to “practice” before we recorded, but I was secretly recording as invariably the first take is sometimes the best take. Case in point this tune, wow! Great dynamic range that pulses with the emotion of the song. Great bass line that sounds nice and smooth with Dirac enabled. But what really stood out for me is the Stevie’s strat sound with the Fender Leslie Vibrotone giving it that rotating stereo effect. It sounds so clear and precise with Dirac enabled, almost spooky hearing SRV fingers sliding up and down the strings with the Leslie effect enabled. Goosebumps territory. I switched over to the sofa, wide imaging corrections and re-listened to the above tracks, switching from single seat to wide imaging, at the same time moving around on my couch. I wanted to see if the 5 measurements versus 17 measurements were audible in some way to my ears. I found the changes to be subtle with the 17 measurements offering a bit more clarity in the overall sound, especially moving to edges of my couch area. I must say the phantom center image was perfect between the speakers, no wandering of the center image versus frequency. Excellent spatial imaging with a 3D depth of field I only hear from speakers that are time aligned with both channels closely matching in frequency response. Even with the 5 measurements, Dirac’s room correction algorithm appears to be quite robust and does not require an overabundance of measurements to get good even bass and smooth overall frequency response. While I enjoyed the benefits of partial correction, my preference is for the full range correction in two areas. One advantage was smoothing any midrange and high frequency “peaks” which with the Purifi SPK4 demo kit were small, but makes a difference not only in the smoothness of the sound, but also the vocals and instruments in the midrange “sat” in the overall mix where I thought they should be rather than sitting in front of the mix being a bit forward in sound from 2 kHz to 5 kHz region. Finally, I like the full range correction as I can dial in a preferred target of simply what I am used to listening to. Personally, I find most speakers sound too “bright” to my ears and confirmed by my measurements. While I have performed some speaker reviews here on Audiophile Style, I have had many pass through my listening room and all too bright for my tastes. The only one that had the preferred response out of the box is the Dutch and Dutch 8c. The KEF LS50 a very close 2nd, but just voiced a little too bright in the 2 kHz range, but Dirac would do a nice job of toning that back just a bit. Maybe we will see that in a future article. Conclusion Well folks, it is 2020 and if you are not using digital room correction, at least partial correction to smooth the bass frequencies in your listening room, you are missing out on a simple but audible optimization. At a retail price of US $349, coupled with a US $100 calibrated measurement mic like the UMIK-1, this is a “no brainer” upgrade to your existing system that is both measureable and audible in a positive way. For existing Dirac 1.x customers, it is a free upgrade. In fact, while I am biased towards advanced DSP room correction, I know of no other way to achieve this level of sonic improvement for dollars spent. Dirac 2 has made the process of applying room correction as simple as possible. Setup the mic, take measurements and click next, next, next, through the wizard and be listening to the smooth bass tones in the shortest time possible. They have made it difficult for anyone to mess up ☺ It is interesting that Dirac chose the VST plugin route, but I can see that their market is not only audiophiles but also the semi/Pro audio market as virtually every Digital Audio Workstation (DAW) uses Steinberg’s VST software plugin architecture. Dirac also provides AU and AAX plugins for Mac users and for DAW’s that use that plugin architecture like ProTools. You can find Dirac Live on the largest marketplace for plugins at KVRAudio. I have all but given up going to audio shows due to the fact that below the (hotel) room’s transition frequency, the room dominates the bass response regardless of the type of or how expensive the loudspeakers used are. As Floyd Toole’s research shows that bass subjectively accounts for 30% of how we judge speakers sound quality. And “ANY loudspeaker can sound better after room EQ, so long as it competently addresses the bass frequencies - this is not a guarantee, but really is not difficult for at least the prime listener” In the case of Dirac, the sofa, wide imaging selection extends the bass sweet spot across at least three seats. To answer the inevitable question, which sounds better, Acourate, Audiolense or Dirac, since I have evaluated all three. I will say they all improve the low frequency response of virtually any system in any room. One can model the “ideal” loudspeaker in an “ideal” room and then it is a matter of which room reflections and how many are positive for the sound quality and how many are not, are taken care of. There is significant scientific research on this topic and with quite a bit of agreement. Having said that, all three DSP room correction software employ different strategies to arrive at the “ideal” result. In the case of Dirac, if you are looking for the shortest path to the result, this is it. That’s also a limitation. Other than different target responses, everything else is pretty much closed off and taken care for you with no user adjustable controls. I did not try Dirac 1.x, so have no way of comparing other than reading through the Dirac 1.x 42 page manual. I must say Dirac has greatly simplified the use of their room correction software for Dirac 2. I did not have to open a manual, I just started using it. I found the improvements made to my loudspeakers in my room by Dirac were audible to my ears and measurable in a positive way. Recommended. Mitch “Mitchco” Barnett. I love music and audio. I grew up with music around me, as my mom was a piano player (swing) and my dad was an audiophile (jazz). My hobby is building speakers, amps, preamps, etc., and I still DIY today. I mixed live sound for a variety of bands, which led to working full-time in multiple 24-track recording studios. Over 10 years, I recorded, mixed, and sometimes produced over 30 albums. I wrote a book on, “Accurate Sound Reproduction using DSP” and run an Accurate Sound Calibration service.
  22. The good ol Lynx Hilo has an excellent ADC. I can share measurements, but one can also find them here: https://www.audiosciencereview.com/forum/index.php?threads/review-and-measurements-of-lynx-hilo-adc-part-2.3596/ Also: https://www.gearslutz.com/board/showpost.php?p=14477582&postcount=1850 The Hilo is still in the top spot for AD DA loop testing linearity. You should be able to find a used one for around the price you are suggesting or maybe a bit more. Good luck!
  23. Thanks Ted! Both sound quite good on the NAD's if I understand your question correctly. If I use the Windows 7.1 config and play each channel individually or listen to Redscapes 7.1 demo, the virtualized locations are exactly to be as expected. Redscapes 2 channel virtualization sounds great too, like being in front of ideal speakers in an ideal room. I enjoy the tone of the NAD's as well as I like the Harman target curve for headphones. Yes, I heard the Smyth Realiser older version some time ago. Does work really well. But as you say, quite the price point difference 🙂 If one can handle a bit of diy, there is the Impulcifer that does what Smyth Realiser does and maybe worth some investigation. Very cool about your music room! I hope you show us when it is done. Kind regards, Mitch
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