Right so first off - sorry, I don't mean to start clan wars with this question.
I've built a few kits, and for reading material I've bought LDC, the "Sound Reproduction..." third edition by Floyd E.Toole, and "Testing Loudspeakers" by D'Appolito... and read bits and pieces of them.
Also, I happen to have a DATS V3, and a Umik1 - I will perform all necessary measurements.
Usually when reading about crossover design it's all about passive / analog crossover design, and most questions I've asked regarding crossover design with DSP ended up in arguments between people discussing things that in my opinion were more related to analog crossovers - probably because that's what these people have been doing for the past few decades.. :-/
So to make a short story very long - I'll be using class D amplifiers to individually power my drivers and DSP to do the crossovers and PEQ when necessary.
Is there any reading material available on how to design crossovers when using DSP, instead of "learn how to do it in analog, including a lot of stuff you won't need when doing it using DSP, then convert that crossover to DSP" ?
I've built a few kits, and for reading material I've bought LDC, the "Sound Reproduction..." third edition by Floyd E.Toole, and "Testing Loudspeakers" by D'Appolito... and read bits and pieces of them.
Also, I happen to have a DATS V3, and a Umik1 - I will perform all necessary measurements.
Usually when reading about crossover design it's all about passive / analog crossover design, and most questions I've asked regarding crossover design with DSP ended up in arguments between people discussing things that in my opinion were more related to analog crossovers - probably because that's what these people have been doing for the past few decades.. :-/
So to make a short story very long - I'll be using class D amplifiers to individually power my drivers and DSP to do the crossovers and PEQ when necessary.
Is there any reading material available on how to design crossovers when using DSP, instead of "learn how to do it in analog, including a lot of stuff you won't need when doing it using DSP, then convert that crossover to DSP" ?
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Use your ears and don't be afraid of using most of your available PEQs on each channel to flatten the SPL response of each driver. Toole talks about the difference between ±3 dB and ±1 dB (suitable smoothing applied) on the sound quality.
I can attest to the difference in sound quality that Toole talks about. This is advice which is in direct contrast to others that I've seen talk about "minimizing" PEQ use. I find this advice to be not helpful, and ultimately, not correct. Toole talks about the JBL M4 having very flat SPL response, and others have stated that the total number of PEQs used in the M4 design is in excess of 20 biquads (probably in addition to FIR filtering to flatten phase) --those trying to DIY using their own DSP crossovers and drivers/horns/boxes, etc. That last 5% of sound quality that you get from further flattening the SPL and phase response disproportionately increases the subjective sound quality, in my experience.
A couple of threads on the subject of using REW and a DSP crossover to achieve flat SPL and phase response:
https://community.klipsch.com/index...ew-to-find-parametric-equalizer-peq-settings/
https://community.klipsch.com/index...rew-to-determine-time-delays-between-drivers/
A PDF tutorial on using DSP crossovers can be found here (you will need to login to download the PDF):
https://community.klipsch.com/index...drivers/page/11/&tab=comments#comment-2586621
Here's a discussion of the subconscious effects of flattening phase through the crossover interference bands using loudspeakers having full-range directivity control and controlled early reflections in-room that may be of interest.
Chris
I can attest to the difference in sound quality that Toole talks about. This is advice which is in direct contrast to others that I've seen talk about "minimizing" PEQ use. I find this advice to be not helpful, and ultimately, not correct. Toole talks about the JBL M4 having very flat SPL response, and others have stated that the total number of PEQs used in the M4 design is in excess of 20 biquads (probably in addition to FIR filtering to flatten phase) --those trying to DIY using their own DSP crossovers and drivers/horns/boxes, etc. That last 5% of sound quality that you get from further flattening the SPL and phase response disproportionately increases the subjective sound quality, in my experience.
A couple of threads on the subject of using REW and a DSP crossover to achieve flat SPL and phase response:
https://community.klipsch.com/index...ew-to-find-parametric-equalizer-peq-settings/
https://community.klipsch.com/index...rew-to-determine-time-delays-between-drivers/
A PDF tutorial on using DSP crossovers can be found here (you will need to login to download the PDF):
https://community.klipsch.com/index...drivers/page/11/&tab=comments#comment-2586621
Here's a discussion of the subconscious effects of flattening phase through the crossover interference bands using loudspeakers having full-range directivity control and controlled early reflections in-room that may be of interest.
Chris
Thanks!
I haven't fully processed what you're saying - english isn't my native language so I might need to read it a few more times and ponder it over - but to me it sounds like you're more talking about using DSP PEQ to apply a house curve..(?)
I'm more interested in how to design a proper crossover for f.e. a 2-way speaker - should I use Linkwitz-Riley, Butterworth, or something else.. How do I properly determine how to pick my crossover frequency, the filter / order to be used, how does phase shift / time alignment apply when using DSP, etc..
I haven't fully processed what you're saying - english isn't my native language so I might need to read it a few more times and ponder it over - but to me it sounds like you're more talking about using DSP PEQ to apply a house curve..(?)
I'm more interested in how to design a proper crossover for f.e. a 2-way speaker - should I use Linkwitz-Riley, Butterworth, or something else.. How do I properly determine how to pick my crossover frequency, the filter / order to be used, how does phase shift / time alignment apply when using DSP, etc..
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I don't use those "named" crossovers--like Tom Danley also has talked about--in order to avoid the 90-360 degrees of filter-induced phase shift. That last link that I posted talks about this. You can use out-of-band stopping filters to curtail lower frequency energy to the higher frequency drivers.
Chris
Chris
Chris,
thanks for replying!
I now am completely befuddled with what you wrote so I will now take some time and googling to figure out what it's supposed to mean :-D
If you could dumb it down that would also be appreciated.. 🙂
thanks for replying!
I now am completely befuddled with what you wrote so I will now take some time and googling to figure out what it's supposed to mean :-D
If you could dumb it down that would also be appreciated.. 🙂
Hi,
There is different aproach to implement treatments using a dsp.
I'll describe mine, others are possible (and maybe better).
First i linearise each way for their intended pass band plus/minus one octave. For this i need peq. Then i (try) to time align them at intended xover freq.
From there you should be able to use textbook kind of filters like LR24, LR12,.... ( the 'named ones' Cask is talking about, the presets you find into your dsp)
From there i apply baffle step compensation, my target curve for hi and then 'voice' if i feel the need for it. This voicing i take weeks to evaluate.
Cask answer to you is at the core of all this because dsp are usually powerful and offer a lot of options.
In my view the real issue is not in the use of many treatments but to be sure it is needed. In other words the measurements and how to interpret them as they 'll tell you what to correct and how.
The advice to not overprocess is valid for me. Not for Cask or other hardcore knowledgable users but for beginners. Once you have played a bit and did comparison between profile you can soon make your own way through this and decide by yourself what you like or not.
About the kind of complementary filters, each solution has it's pro and cons.
I suggest you to read pdf about Linkwitz riley and Bessel filters from Rane's white papers.
https://www.ranecommercial.com/kb_article.php?article=2134
https://www.ranecommercial.com/kb_article.php?article=2123
There is pdf if you scroll all down the page.
How to choose xover freq? Well there is many ways... i usually base mine on acoustic properties ( i try to keep an uniform directivity if possible). A bit of psycho acoustic and trial and errors...
There is different aproach to implement treatments using a dsp.
I'll describe mine, others are possible (and maybe better).
First i linearise each way for their intended pass band plus/minus one octave. For this i need peq. Then i (try) to time align them at intended xover freq.
From there you should be able to use textbook kind of filters like LR24, LR12,.... ( the 'named ones' Cask is talking about, the presets you find into your dsp)
From there i apply baffle step compensation, my target curve for hi and then 'voice' if i feel the need for it. This voicing i take weeks to evaluate.
Cask answer to you is at the core of all this because dsp are usually powerful and offer a lot of options.
In my view the real issue is not in the use of many treatments but to be sure it is needed. In other words the measurements and how to interpret them as they 'll tell you what to correct and how.
The advice to not overprocess is valid for me. Not for Cask or other hardcore knowledgable users but for beginners. Once you have played a bit and did comparison between profile you can soon make your own way through this and decide by yourself what you like or not.
About the kind of complementary filters, each solution has it's pro and cons.
I suggest you to read pdf about Linkwitz riley and Bessel filters from Rane's white papers.
https://www.ranecommercial.com/kb_article.php?article=2134
https://www.ranecommercial.com/kb_article.php?article=2123
There is pdf if you scroll all down the page.
How to choose xover freq? Well there is many ways... i usually base mine on acoustic properties ( i try to keep an uniform directivity if possible). A bit of psycho acoustic and trial and errors...
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Dear Pygmy,
I guess you're interested in using IIR filters, as you talk about Butterworth & Linkwitz-Riley crossovers and parametric EQ. Now, since IIR filters are supposed to mimic their analogue counterparts, you will need to learn every aspect of analogue filter design before attempting digital ones. Unfortunately, there seems to be no shortcut to this.
Now, the step-by-step procedure for making an IIR filter:
1) Design the analogue filter and obtain its poles and zeroes. This requires knowledge in complex arithmetic, as it is EE material.
2) Apply pre-warping to compensate for the frequency warping related to digital filters.
3) Use Bi-linear transformation to map every pole and zero of the analogue filter to the digital domain.
4) Quantise the results to the appropriate number of bits (24/32/64 etc.), and generate coefficients for the digital filter.
5) Check the digital filter for stability and coefficient scaling (if stable).
6) Compare the responses of analogue and digital filters to see if they're more or less the same.
7) If everything is OK, then use the filter.
Now, the shortcut is to purchase a professional audio processor that will do all the above jobs (and sometimes even more) for you. Popular brands are Behringer, dbx etc.
All the best.
I guess you're interested in using IIR filters, as you talk about Butterworth & Linkwitz-Riley crossovers and parametric EQ. Now, since IIR filters are supposed to mimic their analogue counterparts, you will need to learn every aspect of analogue filter design before attempting digital ones. Unfortunately, there seems to be no shortcut to this.
Now, the step-by-step procedure for making an IIR filter:
1) Design the analogue filter and obtain its poles and zeroes. This requires knowledge in complex arithmetic, as it is EE material.
2) Apply pre-warping to compensate for the frequency warping related to digital filters.
3) Use Bi-linear transformation to map every pole and zero of the analogue filter to the digital domain.
4) Quantise the results to the appropriate number of bits (24/32/64 etc.), and generate coefficients for the digital filter.
5) Check the digital filter for stability and coefficient scaling (if stable).
6) Compare the responses of analogue and digital filters to see if they're more or less the same.
7) If everything is OK, then use the filter.
Now, the shortcut is to purchase a professional audio processor that will do all the above jobs (and sometimes even more) for you. Popular brands are Behringer, dbx etc.
All the best.
To the OP:
If you're using high efficiency horn loaded drivers, I actualy don't recommend these two brands--if "hi-fi" is your intended goal. If you're looking for rock-bottom prices without compromising sound quality, I'd recommend miniDSP "2x4 HD" (2-In, 4-Out) or "4x10 HD" (4-In, 8-Out). The miniDSP "2x4" (without the "HD") has pretty severe gain issues.
If you're planning to stick with one for a number of years--and loudspeaker projects, I'd recommend saving your pennies and investing in a Xilica XP series--which is quieter and a bit more hi-fi for higher efficiency projects.
The Behringer DCX2496 has analog section issues as described in a long thread on this forum, as apparently does the lowest priced dbx Driverack PA, etc. I'd also avoid the extreme low-cost DSP crossovers that are aimed at the automotive marketplace--unless you don't care so much about sound quality.
Chris
Another option (that hasn't been mentioned so far) is to implement DSP filters using software under Linux on a low-powered, fanless computer plus a high quality audio interface. I have been doing this for years. This can be IIR or FIR filters. Compared to hardware DSP you are much less restricted regarding how many filters you can use and how, and if you have a very complicated crossover with many corrections this is important. The downside is some latency due to internal buffering and I/O delays, so you cannot really hope to use the end product along with video. For audio only use it is no problem. One advantage of this is that the DSP processing hardware is decoupled from the audio I/O hardware (ADC and DAC) and you can upgrade either one at any time without having to reinvent the wheel or learn a new platform/interface.
Also, when using IIR filter via hardware (e.g. miniDSP, etc.) or the software route I described above, you do not need to "design" the filters - only implement them properly. Filters are parameterized, by e.g. frequency, Q, polarity, gain, etc. and you only need to plug in the correct numbers for these parameters after figuring out what they should be via a crossover design package, which in turn will require good measurements as input.
Also, when using IIR filter via hardware (e.g. miniDSP, etc.) or the software route I described above, you do not need to "design" the filters - only implement them properly. Filters are parameterized, by e.g. frequency, Q, polarity, gain, etc. and you only need to plug in the correct numbers for these parameters after figuring out what they should be via a crossover design package, which in turn will require good measurements as input.
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Many ways to do it, and I find that it’s important to work in broad strokes and avoid very sharp or boost PEQs. Try to cut peaks but don’t boost dips (more than 1-2dB) to flatten the raw response.
I find the Harsch XO the best sounding to my ears. It has nice transient snap and is tine aligned with minimal phase wrapping. It’s flat with a 55deg bump near the XO frequency.
https://www.diyaudio.com/community/threads/s-harsch-xo.277691/
I used to use DSP and active XO’s but have converted to purely passive crossovers and I like the sound of passives better (no noise added) and it’s flexibility. I have a passive XO dev kit and find that I can build and test any XO in a few hours and require fewer iterations vs DSP which requires days or weeks and hundred sweeps. Once done, I have a unitized speaker that I can swap amps in and out as I have a lot of amps I like to listen to and don’t what to have to redo the XO every time. If you have a set system you won’t touch once you get it to play the way you like, active DSP is fine.
Good luck!
X
I find the Harsch XO the best sounding to my ears. It has nice transient snap and is tine aligned with minimal phase wrapping. It’s flat with a 55deg bump near the XO frequency.
https://www.diyaudio.com/community/threads/s-harsch-xo.277691/
I used to use DSP and active XO’s but have converted to purely passive crossovers and I like the sound of passives better (no noise added) and it’s flexibility. I have a passive XO dev kit and find that I can build and test any XO in a few hours and require fewer iterations vs DSP which requires days or weeks and hundred sweeps. Once done, I have a unitized speaker that I can swap amps in and out as I have a lot of amps I like to listen to and don’t what to have to redo the XO every time. If you have a set system you won’t touch once you get it to play the way you like, active DSP is fine.
Good luck!
X
Wow...my experience is just the opposite (dramatically so, in fact), in terms of the number of measurements required (which I find is the same or greater for passive networks--but first using a DSP crossover to decide what needs to happen in in terms of balancing and notch filtering of the final network.
Additionally, changing out amplifiers becomes more problematic with passive networks, which are sensitive to input impedance changes of the passive network+drivers.
Noise is a function of the quality of the DP crossovers employed. The recommended brands/models that I highlighted above were specifically made to avoid these problems. I find that many people make the mistake of using DSP crossovers of too low quality (even though they think that they "sound fine" when they really don't), then assign blame to the whole DSP crossover approach. I find that a DSP crossover+multiamping always results in better sounding loudspeakers. YMMV.
This is where I started with the last linked thread above (post #2, above), but moved on to filters that don't induce phase growth (higher frequencies-->lower frequencies). When I finally figured it out, the difference was a bit breathtaking (i.e., spectacular, but subjectively difficult to describe) using fully-horn loaded loudspeakers having full-range directivity in a room treated for early reflections just around the loudspeakers. Hence the name of the thread: "Subconscious Auditory Effects of Quasi-Linear-Phase Loudspeakers".
Chris
Hi Chris,
There is some aspect of making a passive crossover that is very deliberative. It makes you take careful baseline raw data and work with that in the XO simulator. I suppose one could do the same with a DSP XO simulator (in REW for example) and then upload the settings to the DSP and remeasure. Part of my “training” on DSPs led me to be more careful and deliberate with my passive XO dev process. So I literally measure one set of raw sweeps. I simulate the XO and the predicted response and measurement matches almost spot on. I design the passive XO to have as smooth impedance sweeps with smooth phase plots and nominal impedance that all my amps can handle. So it’s very useable with many amps. I don’t purposely design a “difficult to drive” XO.🙂
There is some aspect of making a passive crossover that is very deliberative. It makes you take careful baseline raw data and work with that in the XO simulator. I suppose one could do the same with a DSP XO simulator (in REW for example) and then upload the settings to the DSP and remeasure. Part of my “training” on DSPs led me to be more careful and deliberate with my passive XO dev process. So I literally measure one set of raw sweeps. I simulate the XO and the predicted response and measurement matches almost spot on. I design the passive XO to have as smooth impedance sweeps with smooth phase plots and nominal impedance that all my amps can handle. So it’s very useable with many amps. I don’t purposely design a “difficult to drive” XO.🙂
I use a process that starts with a miniDSP 2x4 HD, and two amps, to design a 2-way crossover. I listen to this for weeks/months, and when I have settled on the sound, I measure the speaker response with the digital crossover, and each driver with the crossover; and each driver "raw". Also I use a DATS V3 to measure the impedance of the drivers in the cabinet.
Then I use XSim to get an analog crossover designed to get the response as close as possible, to the digital version. Because I am aiming for the analog crossover, I only use a volume level on the tweeter, in addition to the crossover in the miniDSP 2x4HD. Initially, on the first speaker I did this with, I used PEQ on each driver, and on the speaker - but I figured out that this would be next to impossible to design into an analog crossover. On that first speaker, I ended up adding a notch filter on the tweeter, to flatten a particular hump in the response.
I find that the fact that I can do this, and end up with what I think, are pretty damn good speakers - with the use of several computer programs: Hornresp (to design the cabinets), REW, miniDSP software, and XSim - is pretty damn amazing.
Then I use XSim to get an analog crossover designed to get the response as close as possible, to the digital version. Because I am aiming for the analog crossover, I only use a volume level on the tweeter, in addition to the crossover in the miniDSP 2x4HD. Initially, on the first speaker I did this with, I used PEQ on each driver, and on the speaker - but I figured out that this would be next to impossible to design into an analog crossover. On that first speaker, I ended up adding a notch filter on the tweeter, to flatten a particular hump in the response.
I find that the fact that I can do this, and end up with what I think, are pretty damn good speakers - with the use of several computer programs: Hornresp (to design the cabinets), REW, miniDSP software, and XSim - is pretty damn amazing.
It seems to me that what you are introducing is precisely what the OP was referring to as argumentative?...
Perhaps it would be better to introduce discussions of "passive crossovers are [somehow] superior" elsewhere to keep this thread from diverging? 🙂
"Crossovers may be implemented either as passive RLC networks, as active filters with operational amplifier circuits or with DSP engines and software. The only excuse for passive crossovers is their low cost. Their behavior changes with the signal level dependent dynamics of the drivers. They block the power amplifier from taking maximum control over the voice coil motion. They are a waste of time, if accuracy of reproduction is the goal." S. Linkwitz
Chris
I did say that if you only have one setup and don’t need to change amps that DSP active is a good way to go. I don’t believe passive XO are cheaper. They can be less expensive but generally, DSP active is more cost effective.
I don’t have strong opinions one way or another on what is better. I have heard some of the best sound I have ever heard come from an active system with an AMT tweeter synergy horn and front loaded bass horn so no argument from me.
I don’t have strong opinions one way or another on what is better. I have heard some of the best sound I have ever heard come from an active system with an AMT tweeter synergy horn and front loaded bass horn so no argument from me.
I have played with both passive and digital xo in vituixcad, with real impedance curve.
To me the biggest advantage of digital xo is that your digital signal processing doesn’t care about the impedance, whereas the passive components will add non linearities to your filters that you have to attenuate with more passive components. Am I right ?
To me the biggest advantage of digital xo is that your digital signal processing doesn’t care about the impedance, whereas the passive components will add non linearities to your filters that you have to attenuate with more passive components. Am I right ?
Back to the original question:
A more advanced technique, which I learned later, is to use VituixCad2 to design the DSP filter. First we measure the on/off axis responses of our individual drivers (installed in the cabinet) in 10 degree increments and import them into VituixCad. Then we optimize the DSP filters until the desired combination of on-axis, listening window, and sound power curves are optimized.
All of my experience so far is with IIR (infinite impulse response) filters, which basically means the DSP process mimics an analog filter. Changing magnitude has a direct and predictable effect on phase.
Another technology is the FIR (finite impulse response) filters. With this technology, a designer can change magnitude without changing phase, or the other way around. There are pros and cons to each method, but I have no experience with FIR so I won't talk about the pros and cons...
You may find something useful in my two threads. I learned a lot along the way, so my errors and victories are all documented...
https://www.diyaudio.com/community/threads/new-active-satori-textreme.366347/
https://www.diyaudio.com/community/threads/new-active-3-way-hypex-and-sb.352767/
I do not know of any book, article, or website. My first DSP active crossover was made by following the Hypex Filter Design (HFD) software documentation. In post #6, Krivium basically describes a short, condensed version of this process. A good designer can use the on-axis response of the various drivers, and iteratively measure-adjust-listen, measure-adjust-listen until a good result is achieved. This process can work well, but it needs to be combined with all the standard good design practices.
A more advanced technique, which I learned later, is to use VituixCad2 to design the DSP filter. First we measure the on/off axis responses of our individual drivers (installed in the cabinet) in 10 degree increments and import them into VituixCad. Then we optimize the DSP filters until the desired combination of on-axis, listening window, and sound power curves are optimized.
All of my experience so far is with IIR (infinite impulse response) filters, which basically means the DSP process mimics an analog filter. Changing magnitude has a direct and predictable effect on phase.
Another technology is the FIR (finite impulse response) filters. With this technology, a designer can change magnitude without changing phase, or the other way around. There are pros and cons to each method, but I have no experience with FIR so I won't talk about the pros and cons...
You may find something useful in my two threads. I learned a lot along the way, so my errors and victories are all documented...
https://www.diyaudio.com/community/threads/new-active-satori-textreme.366347/
https://www.diyaudio.com/community/threads/new-active-3-way-hypex-and-sb.352767/
I can recommend an ebook about FIR ( Michba's ( Mitch Barnett): 'Accurate sound reproduction using DSP' but it goes way beyond the OP question: it dive into 'room correction' and FIR very deeply ( including mic measurement techniques).
In my view it is one of the most up to date and interesting approach but i think too one must first learn using IIR. Both approach have their pro and cons and IIR ( whatever the technology it is implemented with) still have a place.
Fir complementary filters ( xovers) are great. The latency is the real issue but they won't help a lacking design ( layout of drivers, the basic acoustic 'concept' of the loudspeaker).
They can help pushing the last 1/4octave ( of usable bandwith just before break up) of a loudspeaker without real nasty though.
Funny you mention your more advanced technique hifijim, i lately used more and more MMM measurement technique into my 'routine'.
In my view it is one of the most up to date and interesting approach but i think too one must first learn using IIR. Both approach have their pro and cons and IIR ( whatever the technology it is implemented with) still have a place.
Fir complementary filters ( xovers) are great. The latency is the real issue but they won't help a lacking design ( layout of drivers, the basic acoustic 'concept' of the loudspeaker).
They can help pushing the last 1/4octave ( of usable bandwith just before break up) of a loudspeaker without real nasty though.
Funny you mention your more advanced technique hifijim, i lately used more and more MMM measurement technique into my 'routine'.
I agree that "overprocessing" can be an issue, but you also do not learn the limitations of DSP if you "worship minimalism". My advice is to try it, and if not successful, clear it and try again. The only loss is your time (which isn't loss because you're also learning). My advice is to "try it and listen".
Without knowing how to do it wrong, there is probably a lot of improvement available that's still on the table--with DSP crossovers. The guys that I know that believe that someone's else's DSP settings are "gospel"--basically never learn how to use their crossovers.
After I had started on the basic FAQs and training approaches using DSP that I posted (post #2, above), I came across Mitch's book about three years ago. Mitch will remotely dial-in your system for ~$500-1K (USD), but it takes about $4-5K (USD) in software and hardware to get to the point where he can dial it in for you remotely. One of the guys on the K-forum enlisted Mitch's services and turned out very pleased, but for the typical DIYer, this is way beyond the typical budget for this sort of activity.
That's why I continue to assemble a series of tutorials using REW, a UMIK-1, and typical lower- and medium-budget DSP crossovers (miniDSP, Xilica, EV, Yamaha, etc.) to do the job (There are two more pdf tutorials in rough draft format--waiting for the demand to appear to complete and proof them.) The first tutorial (pdf) is very basic and will not really enable a novice learner the ability to fully get what they want out of their systems--but just enough to get them going. The initial tutorial really is setting the stage by introducing the concepts and the dimensions of the problem space to novice users so that they can develop their own concepts of the problem space.
I've been helping many guys on the K-forum dial-in their systems using REW measurements and the PC-based applications that come with the DSP crossovers remotely (via emailing measurements and DSP crossover settings), ~70-80 guys now over the past 5-6 years. I've also tried to answer questions as they arise from the guys that want to learn how to use all of REW's resources to dial-in and to help identify the source of in-room issues. That's where some of the text has come from in the very short introductory tutorial threads I created. There are other Q&A threads on the K-forum with guys asking "dumb questions"--which aren't so dumb after all.
I think there is a very large potential user base to make use of DSP crossovers--sort of like the present group of people that use passive networks--only a much larger group since it's DSP crossovers are much more powerful tool in achieving better in-room results. I see this as a potential for an addition to the existing body of "audiophile expertise"--only in dialing in their systems manually (i.e., not using "room correction software"...of which none of the packages has worked that well for me).
Most of what I see from other sources is far too pedantic/formulaic, or do not use the tools available to the typical DIYer, and none describe the processes I use to glean usable information from the measurements and to make updates to the DSP setting...iteratively. The real issues are not really DSP mathematics, but physical understanding of small room acoustics and loudspeakers, and how to come up to speed quickly on those subjects via use of freeware tools like REW.
Chris
Yes Mitch use Acourate which isn't cheap and the ebook is definitely oriented with this software in mind ( this is a step by step along the soft) but this is his routine/approach i find the more useful. Not that i did many different after all but he introduce some example of treatments i was doubtful ( correction of low end early reflection) but been convinced by his results ( everything is planned, executed then measured to check).
He did more or less the same with other software package like Audiolense. It's easy to find on internet and worth a read too ( it's at audiolifestyle iirc).
Thank you doing this kind of documents Chris.
Maybe you could create a thread and make them availlable here too? That would be worthwile for some members i'm sure.
He did more or less the same with other software package like Audiolense. It's easy to find on internet and worth a read too ( it's at audiolifestyle iirc).
Thank you doing this kind of documents Chris.
Maybe you could create a thread and make them availlable here too? That would be worthwile for some members i'm sure.