KeithL
Administrator 
Posts: 10,261
|
Post by KeithL on Nov 19, 2024 11:02:05 GMT -5
I think I need to clarify a few things here... First of all, on our processors, and pretty much any others that I know of, the bass management is done by the same DSP as the other processing. The DSP "does the digital processing"; which is basically a single big multi-function program; it wouldn't make sense to run parts of it on different DSPs. (Some processors might actually have multiple DSP engines... but usually they're sharing the same functions... like a multi-core or multi-threaded CPU...) (In the past there have been processors that had entirely separate analog bass management circuitry - but I don't know of any today.) The other thing is that, for the most part, there is no validity to the idea of "cheaper DSPs not sounding as good", at least not in the context of bass management. It's not nearly as simple as that. How a DSP "sounds" is going to depend on the "quality" of the "math" it performs... because the DSP itself is purely digital. There is no "separate DSP for the bass management". And since the bass management inside a processor is being done by the same DSP as everything else, and is "part of the same software", there is no reason for the math to be "done at a different quality". And, incidentally, there is no "high-pass section" or "low-pass section" per-se... there are just "channels" to which various audio streams are routed to. And, in a given single DSP, they're almost certainly going to all be of the same "quality". Now, when you use an entirely separate processor, like a miniDSP, this is not the case. The miniDSP has its own processor, which is totally independent of the processor in your AVR or pre/pro, so the audio quality they deliver is also independent of each other. THAT PROCESSOR may work at a different bit depth, or a different sample rate, or in other significantly different ways. HOWEVER, when we consider the pros and cons of using something like a miniDSP, that is usually not the major issue. The biggest issue is usually that, in order to move the signal between "boxes", we are required to add multiple extra conversions... and those are where quality is often lost. For example, if you add a miniDSP to one of our processors "for bass management".... You're taking our analog output, whose sound quality will be determined by our processing, and our DACs... Converting it back into digital using the A/D converters in the miniDSP... Doing digital processing inside the miniDSP... Then converting that signal back into analog using the DACs in the miniDSP... So, instead of one conversion between digital and analog after the digital surround sound signal has been decoded, you now have THREE... And, in addition to those conversions, you have more analog circuitry, at multiple different places in the signal path... And, of course, more interconnects, and more connectors, and more ground connections, and more power supplies, and so on... Now, to be fair, many of those things are almost certainly less critical when it comes to the range of frequencies handled by a subwoofer... But you absolutely have made the signal path a lot more complicated... And added several more opportunities for things to be less than ideal... It's also worth mentioning that you still need to understand the purposes (purposes plural) of bass management... One part of bass management is to route the low frequency content to the subwoofer... And, if you have multiple subs, to handle dividing that content between them... (And all of the fun things you can do with adjusting the relative phase and timing between them.) But the other part of bass management is to route those low frequencies AWAY FROM your main speakers. This is what the high-pass part of the bass management crossover does. One reason to do this is that even large full range speakers may struggle with very low bass... Another is that, even if your main speakers don't noticeably struggle, it's still better to avoid sending bass to a speaker below the frequencies it can produce. The additional cone movement potentially causes more distortion at higher frequencies... (To put it crudely... you're wasting amplifier power, heating the speaker up more, and making the cone wave around more, for no real benefit.) And, finally, if your main speakers are in fact full range, and you aren't using a high-pass filter, then your main speakers are also subs. They are making output at frequencies that overlap with those handled by your subs... so they interact with your subs.And you're only going to be able to achieve all of the capabilities of bass management if your bass management includes that high-pass filter... (Which means that not just the content destined for the subwoofer, but the full frequency range, must go through the bass management.) What most people who use something like a miniDSP are actually doing is NOT "using the miniDSP for bass management". What they're doing is using the miniDSP for additional detailed control of multiple subs in additional to the bass management in the processor itself. Hi marcl - I almost sent a prompt response to your previous post, but then stopped to think a bit more before replying. After further consideration, no, I don’t think that my objections concerning “bass management” pertain only to analog sources. Why? Because my last post on the topic mixed up two different things, digital conversion (A/D-D/A) and Digital Signal Processing (DSP). The former applies only to analog signals that have to be converted to digital for whatever reason and then be converted back to analog. Cheap converters cause readily audible sonic degradation (a widely-measured and agreed-upon artifact). Although this degradation is inaudible at subwoofer frequencies, it is VERY audible at higher frequencies. But digital bass management also creates audible artifacts in purely digital bitstreams! Even if you’re feeding your AVR a purely digital signal (streaming, digital output from a disc player, etc.) the “lowest-possible-cost” DSP chip in your processor that’s being used for bass management is still creating audible distortion in it’s high-pass section. I’ve found, aegain and again that I get cleaner sound (particularly in the treble end) when I don’t use DSP bass management. By adjusting the low-pass frequency via the sub’s plate amp, adjusting phase at the sub, and carefully matching gains, I am more likely to achieve an “invisible blend” between the satellites and the sub(s) than when I’m using digital bass management. Glenn Okay I think I understood the distinction (my deleted post went into that more  ) and to the first point of A/D-D/A yes I had no issue doing it for subs but I didn't want to run my main L/R through the miniDSP 2x4 Balanced or 2x4 HD. But what about the miniDSP Flex? Seems like that would do the job very well ... and a scenario might be analog 2-channel, but you want to have Bass Management and Dirac. In your last paragraph are you saying you run all speakers full range without a high pass, and just set your crossover in the sub plate amp high enough to fill in the bottom? Do you do this with speakers that have to be crossed at 80-120Hz or do you mean just 2-channel with mains that go down to 40-50? And regarding the use of DSP bass management in the processor ... maybe keithl can chime in on whether the DSP Bass Management is done in its own “lowest-possible-cost” DSP chip, or if it's done in the same DSP chip that handles everything else ... i.e. the signal goes through it whether you do Bass Management or not. |
|
|
|
Post by Boomzilla on Nov 19, 2024 11:06:15 GMT -5
Hi marcl - Excellent comments! ...and to the first point of A/D-D/A yes I had no issue doing it for subs but I didn't want to run my main L/R through the miniDSP 2x4 Balanced or 2x4 HD. But what about the miniDSP Flex? Seems like that would do the job very well ... and a scenario might be analog 2-channel, but you want to have Bass Management and Dirac. I haven't heard the miniDSP products, so I couldn't say. But I will say that DACs and DSPs have improved a LOT over the past decades. Most oversample the signal to minimize loss, phase shift, and distortion. But that said, the best sub crossover I've ever heard was all-analog, designed by Jon Dahlquist. To use it, you had to know the input impedance of your power amp to calculate what capacitor you needed to insert for the crossover point you wanted. The high pass was 6dB / octave (first-order) and completely passive. The low-pass was active at 18dB / octave (third order). This made for a "phase coherent" crossover and it sounded wonderful! To date, I've never heard ANY digital bass-management system that came close. EDIT - OK, I take that last sentence back! I've owned ONE crossover that was equal to the Dahlquist - It was made by JL-Audio (and cost ~$3,000). In your last paragraph are you saying you run all speakers full range without a high pass, and just set your crossover in the sub plate amp high enough to fill in the bottom? Do you do this with speakers that have to be crossed at 80-120Hz or do you mean just 2-channel with mains that go down to 40-50? Whether to use a sub at all depends (to me) on the main speakers' -3dB point. If I'm using speakers with the roll-off in the low to mid-30s, then I generally don't think a sub is needed at all. But almost ALL speakers sold these days roll off at 40 Hz. or higher - even extremely expensive full-tower models. I think this is criminal! So, regardless of size and cost, if the speaker can't get down to the low 30s, it needs a sub. For those (many, many) speakers that need a sub, I've heard better sound quality by running the speakers without a high-pass filter upstream. I then blend in the sub below. And regarding the use of DSP bass management in the processor ... maybe keithl can chime in on whether the DSP Bass Management is done in its own “lowest-possible-cost” DSP chip, or if it's done in the same DSP chip that handles everything else ... i.e. the signal goes through it whether you do Bass Management or not. As you say, Mr. KeithL would need to answer that, but my suspicion is that processors & AVRs use a single DSP chip for both effects AND bass management. But that STILL doesn't mean that the DSP chip isn't a really cheap one! Glenn |
|
KeithL
Administrator
Posts: 10,261
|
Post by KeithL on Nov 19, 2024 11:30:00 GMT -5
Our big processors actually have two DSP "engines"... The actual Decoding and Bass Management is done on the first DSP... And this first DSP runs at a rate that depends on the source... The Dolby and DTS decoder software modules themselves run at 48k... (This is actually determined by the software in the licensed decoder software modules.) Other types of sources can run at different rates... and the DSP will match the sample rate of the incoming data... The sixteen output channels are then passed to the second DSP engine which does all "post processing"... Post processing includes things like EQ, Dirac Live filters, trims, distance and lip synch delays, and such... The second DSP engine basically always runs at 48k... Note that the Dirac Live filters are basically special filter modules that replace the standard PEQ filters when they're in use. These filter modules are downloaded to the processor by the Dirac Live software... The calculations that create these filters from the measurements are all done in the Dirac Live software on the computer... [ I do apologize if this seems to be at odds with my previous post. From a user perspective I tend to think of "the DSP engine" as a single thing. Much as, from a user perspective, it doesn't matter which processing my computer does in the CPU or the GPU. (And, in many modern computers and programs that detail may actually be somewhat negotiable.) Suffice to say that the DSP engines don't "sound different"; they simply produce "the specified output". For example, at the level of the decoders, any Dolby decoder that passes certification should be delivering "the proper output". Differences in sound quality generally shouldn't occur that that level. ] I've had that happen... more than once... Or had the draft simply disappear after I walked away for a while... I've also had a post disappear when I clicked the Create Post key... The answer is to back up your post... In this case, if you type something long and profound, highlight it and clip it to the clipboard (Ctrl-C) in Windows.... If it "mysteriously vanishes" you can get it back by pasting it (Ctrl-V)... And, if not, you can just forget about the clipboard... (If you do Ctrl-C again it will overwrite whatever was there before...) Yes I very often do that. In this case in my reply to Boom maybe I was more concise in the rewrite. Over the years I swear Powerpoint used to let me close files without saving every once in a while. But while we're talking .... can you tell us, does all the Dirac processing, surround and Atmos decoding, PEQ, bass and treble trims ... and Bass Magagement ... all happen in the same DSP chip? |
|
|
|
Post by marcl on Nov 19, 2024 13:05:14 GMT -5
From a user perspective I tend to think of "the DSP engine" as a single thing. Much as, from a user perspective, it doesn't matter which processing my computer does in the CPU or the GPU. (And, in many modern computers and programs that detail may actually be somewhat negotiable.) Suffice to say that the DSP engines don't "sound different"; they simply produce "the specified output". For example, at the level of the decoders, any Dolby decoder that passes certification should be delivering "the proper output". Differences in sound quality generally shouldn't occur that that level. ] I think the key concept is that calculations don't degrade a signal (by downsampling or whatever) and are just math that has no effect on sound quality unless there is some flaw in the calculations themselves. Cost of the DSP doesn't affect how well calculations are done. Maybe it determines the speed, which may limit WHAT is done or not done .... i.e. You do 1000 FIR taps in Dirac and not 4000 ... you don't do Dirac at 96KHz ... or you can't do ART. Now it gets back to stereo vs multichannel. Because with multichannel you will have to do delays and Bass Management in the digital domain. Comparing a stereo-only pure analog signal path with only analog filters, if any .... well that's a big apple:orange leap from a processor that does what our processors do. Consider one of our amigos who sees the significant value of using Dirac in his room, but wants to have a 2-channel system with a specific DAC ... is there any value to having that DAC if he gives up Dirac? |
|
|
|
Post by leonski on Nov 19, 2024 15:34:46 GMT -5
This bass treatment goes a long way back. To the early 60s....at least....when sealed box speakers such as the original line form AR (AR3a, an object of lust) came out.
My NAD1700 Tuner Preamp had an 'INFRASONIC' filter, back panel switchable. Turn Table RUMBLE can cause all sorts of Havoc. Including inaudible woofer cone motion.
Ever seen your woofer pump back and forth, maybe even to its excursion limits? I don't remember the exact spec, but I think 20hz was IT. Of course, modern digital media
has no problem with 20 hz. And that filter is not needed.
My 2-way Maggies are now 3-way, with the addition of a sub. Sub gets a full range signal and I use ITS crossover at maybe 45hz high pass. My preamp has a low-pass
adjustible for the main speakers which I set to about 50 to 55hz.
I must ask Marcl a Question, if I may? Are all 'programs' for the manipulation of sound done to the same precision or accuracy (2 different things) level? I suspect not.
Bass may be easier and HF may have other problems which an inferior PROGRAM (which any of a dozen processors can run) may not handle as well.....
|
|
KeithL
Administrator
Posts: 10,261
|
Post by KeithL on Nov 19, 2024 15:46:27 GMT -5
I would say that's ENTIRELY up to your friend... To be quite honest, IN MY PERSONAL OPINION, I don't see a compelling argument either way. I personally find that Dirac Live offers significant benefits in a surround sound system... But, to be quite honest, I don't find that to always be the case in a good stereo system... It seems to often do an excellent job of cleaning up "vague imaging" and "cleaning up the presentation". And, since these are things that I often find problematic with surround sound systems, I find this to be a major benefit. But, to be quite blunt, the sorts of problems that I run into in stereo systems are not the sorts of things that Dirac Live seems to be good at fixing. And, in a stereo system without active room correction, I would be more likely to notice subtle differences between DACs. ......................................... I think the key concept is that calculations don't degrade a signal (by downsampling or whatever) and are just math that has no effect on sound quality unless there is some flaw in the calculations themselves. Cost of the DSP doesn't affect how well calculations are done. Maybe it determines the speed, which may limit WHAT is done or not done .... i.e. You do 1000 FIR taps in Dirac and not 4000 ... you don't do Dirac at 96KHz ... or you can't do ART. Now it gets back to stereo vs multichannel. Because with multichannel you will have to do delays and Bass Management in the digital domain. Comparing a stereo-only pure analog signal path with only analog filters, if any .... well that's a big apple:orange leap from a processor that does what our processors do. Consider one of our amigos who sees the significant value of using Dirac in his room, but wants to have a 2-channel system with a specific DAC ... is there any value to having that DAC if he gives up Dirac? |
|
|
|
Post by marcl on Nov 19, 2024 17:04:25 GMT -5
This bass treatment goes a long way back. To the early 60s....at least....when sealed box speakers such as the original line form AR (AR3a, an object of lust) came out. My NAD1700 Tuner Preamp had an 'INFRASONIC' filter, back panel switchable. Turn Table RUMBLE can cause all sorts of Havoc. Including inaudible woofer cone motion. Ever seen your woofer pump back and forth, maybe even to its excursion limits? I don't remember the exact spec, but I think 20hz was IT. Of course, modern digital media has no problem with 20 hz. And that filter is not needed. My 2-way Maggies are now 3-way, with the addition of a sub. Sub gets a full range signal and I use ITS crossover at maybe 45hz high pass. My preamp has a low-pass adjustible for the main speakers which I set to about 50 to 55hz. I must ask Marcl a Question, if I may? Are all 'programs' for the manipulation of sound done to the same precision or accuracy (2 different things) level? I suspect not. Bass may be easier and HF may have other problems which an inferior PROGRAM (which any of a dozen processors can run) may not handle as well..... I don't know about how digital filters are implemented ... and my EE degree is way too old for me to remember how to do analog filters, though I remember something about Mrs Butterworth and Chevy Chase. But today I think crossovers are often done with a fourth order Linkwitz-Riley filter, 24db/octave with flat summed response and linear phase at the crossover frequency. I use a pair of XKitz L-R24 analog filters at 60Hz between my Maggies and subs. When I did crossovers in the miniDSP I also used L-R filters either 24 or 48 db/octave. I think there's really good reason to high pass the mains. Having woofers wasting energy with low frequencies they can barely reproduce adds distortion and wastes amplifier power, seems to me. |
|
|
|
Post by marcl on Nov 19, 2024 17:14:17 GMT -5
I would say that's ENTIRELY up to your friend... To be quite honest, IN MY PERSONAL OPINION, I don't see a compelling argument either way. I personally find that Dirac Live offers significant benefits in a surround sound system... But, to be quite honest, I don't find that to always be the case in a good stereo system... It seems to often do an excellent job of cleaning up "vague imaging" and "cleaning up the presentation". And, since these are things that I often find problematic with surround sound systems, I find this to be a major benefit. But, to be quite blunt, the sorts of problems that I run into in stereo systems are not the sorts of things that Dirac Live seems to be good at fixing. And, in a stereo system without active room correction, I would be more likely to notice subtle differences between DACs. I think the key concept is that calculations don't degrade a signal (by downsampling or whatever) and are just math that has no effect on sound quality unless there is some flaw in the calculations themselves. Cost of the DSP doesn't affect how well calculations are done. Maybe it determines the speed, which may limit WHAT is done or not done .... i.e. You do 1000 FIR taps in Dirac and not 4000 ... you don't do Dirac at 96KHz ... or you can't do ART. Now it gets back to stereo vs multichannel. Because with multichannel you will have to do delays and Bass Management in the digital domain. Comparing a stereo-only pure analog signal path with only analog filters, if any .... well that's a big apple:orange leap from a processor that does what our processors do. Consider one of our amigos who sees the significant value of using Dirac in his room, but wants to have a 2-channel system with a specific DAC ... is there any value to having that DAC if he gives up Dirac? What makes my brain hurt is the idea that resonance modes don't matter with 2-channel but they do with HT. Play 1, 2, 6, 16 channels in any room where humans live and there will be a peak typically +10db somewhere between 30 and 70Hz followed by a cancellation -5db ... at least, relative to the nominal response above 200Hz. You can't fix it with bass traps, you can only make it a little better. Often the peak to null is worse ... even as much as 20-25db, as I see in the Dirac FB Group when people post their Dirac measurements and ask how to set their Target Curves. If you don't use Dirac to fix it, you can use PEQ. But if you don't fix it ... it's there. I have seen it in every 2-channel room I've been in the past 6 months between my audio club and CAF. |
|
|
|
Post by Boomzilla on Nov 19, 2024 17:21:14 GMT -5
MOST anything that can be done to room acoustics via DIRAC becomes moot if the room is symmetrical and has proper acoustic treatment. That isn’t always the case, though, which is why DIRAC (and YPAO, etc.) exist.
In my experience, electronic room correction of any flavor is still a poor substitute for room treatment & symmetry, but if you need it, it’s better than nothing…
Since 90% plus of my listening is simple stereo, I generally find that the less electronics I have in the signal path, the better the sound.
My listening choices erase most of the negatives associated with running my satellites full range:
*. I prefer high-sensitivity speakers
*. I listen at VERY low volumes
*. I sit fairly close to the speakers to minimize room echo
Even with notes way below my (ported) satellites’ cutoff frequency, there is no audible distress from the woofer cones due to the very low volumes.
Although my “analog crossover or none at all” comments are true for my own situation, I can easily understand why they don’t apply universally!
|
|
|
|
Post by fbczar on Nov 19, 2024 18:26:52 GMT -5
MOST anything that can be done to room acoustics via DIRAC becomes moot if the room is symmetrical and has proper acoustic treatment. That isn’t always the case, though, which is why DIRAC (and YPAO, etc.) exist. In my experience, electronic room correction of any flavor is still a poor substitute for room treatment & symmetry, but if you need it, it’s better than nothing… Since 90% plus of my listening is simple stereo, I generally find that the less electronics I have in the signal path, the better the sound. My listening choices erase most of the negatives associated with running my satellites full range: *. I prefer high-sensitivity speakers *. I listen at VERY low volumes *. I sit fairly close to the speakers to minimize room echo Even with notes way below my (ported) satellites’ cutoff frequency, there is no audible distress from the woofer cones due to the very low volumes. Although my “analog crossover or none at all” comments are true for my own situation, I can easily understand why they don’t apply universally! Boom, Dirac corrects both frequency and impulse response. I have used Dirac in several shapes of extremely well treated rooms and in each case Dirac has improved the sound in the room. The only caveat I would make relative to Dirac is that you actually must know how to use it for it to be effective. No doubt, far more folks do not know how to use Dirac properly than do. In my experience, Dirac has been an asset with everything from Transmission Line speakers to Magnepans, but has been most effective with Magnepans. I assume the impulse correction is a major plus for dipoles. I understand that the way you listen to music, especially in the near field, may make Dirac less effective, but I would put my money on a properly treated room with Dirac than just a properly treated room, in most cases. Obviously, your comment was limited to stereo music. I assume you feel differently about home theater systems and Dirac, but I would be interested in your opinion. |
|
|
|
Post by Boomzilla on Nov 19, 2024 19:15:08 GMT -5
Hi fbczar - Since my wife just doesn’t watch movies at all anymore, then rather than tie up the living room when I want a movie, I fire up my iPad, plug the AirPods in, and enjoy. Apple’s “Spatial Audio” can’t compete with a real surround-sound system, but I enjoy it just fine. I’d think that, in general, DSP would be a HUGE enhancement to a multi-channel system. But when I tried it with stereo, I concluded that in my room, to my ears, and the way that I listen, I liked the music better without DSP. Of course, I was hearing YPAO, not DIRAC. Thanks for the feedback - I appreciate it! Glenn |
|
|
|
Post by leonski on Nov 20, 2024 4:55:52 GMT -5
MOST anything that can be done to room acoustics via DIRAC becomes moot if the room is symmetrical and has proper acoustic treatment. That isn’t always the case, though, which is why DIRAC (and YPAO, etc.) exist. In my experience, electronic room correction of any flavor is still a poor substitute for room treatment & symmetry, but if you need it, it’s better than nothing… Since 90% plus of my listening is simple stereo, I generally find that the less electronics I have in the signal path, the better the sound. My listening choices erase most of the negatives associated with running my satellites full range: *. I prefer high-sensitivity speakers *. I listen at VERY low volumes *. I sit fairly close to the speakers to minimize room echo Even with notes way below my (ported) satellites’ cutoff frequency, there is no audible distress from the woofer cones due to the very low volumes. Although my “analog crossover or none at all” comments are true for my own situation, I can easily understand why they don’t apply universally! Boom, Dirac corrects both frequency and impulse response. I have used Dirac in several shapes of extremely well treated rooms and in each case Dirac has improved the sound in the room. The only caveat I would make relative to Dirac is that you actually must know how to use it for it to be effective. No doubt, far more folks do not know how to use Dirac properly than do. In my experience, Dirac has been an asset with everything from Transmission Line speakers to Magnepans, but has been most effective with Magnepans. I assume the impulse correction is a major plus for dipoles. I understand that the way you listen to music, especially in the near field, may make Dirac less effective, but I would put my money on a properly treated room with Dirac than just a properly treated room, in most cases. Obviously, your comment was limited to stereo music. I assume you feel differently about home theater systems and Dirac, but I would be interested in your opinion. Your point about most not knowing how to use Dirac echos what I said many moons ago in another thread. Sure, Let's just stipulate it is good and correct software. But than it comes down to measurement technique and IF you can get repeatable system in place. All measurment systems have inherent errors and your job is to minimize the operators (that's YOU!) contribution.... |
|
|
|
Post by marcl on Nov 20, 2024 6:53:39 GMT -5
I would say that's ENTIRELY up to your friend... To be quite honest, IN MY PERSONAL OPINION, I don't see a compelling argument either way. I personally find that Dirac Live offers significant benefits in a surround sound system... But, to be quite honest, I don't find that to always be the case in a good stereo system... It seems to often do an excellent job of cleaning up "vague imaging" and "cleaning up the presentation". And, since these are things that I often find problematic with surround sound systems, I find this to be a major benefit. But, to be quite blunt, the sorts of problems that I run into in stereo systems are not the sorts of things that Dirac Live seems to be good at fixing. And, in a stereo system without active room correction, I would be more likely to notice subtle differences between DACs. To your last sentence, I think a case could be made that it actually demonstrates the significant value that Dirac provides. Beyond correcting modal resonances of the room (which other filter methods can do well, if not quite AS well) the phase and impulse response correction that is unique to Dirac, and unachievable any other way, results in such clarity and improved transients and imaging ... that any nuance of a difference between one DAC and another is rendered imperceptible. |
|
|
|
Post by leonski on Nov 20, 2024 15:24:05 GMT -5
I would say that to KNOW how your measurement system interacts with the equipment, speakers AND room, I'd say to do as good a setup as you can manage..
Follow the directions and any helps from known good / expert users.
Record all measurements.....Speaker spacing? Distance from walls? I'd even keep track of Line Voltage. Even record placement of microphone with orientation.
This will be a baseline number / value and can change later as changes are made to the setup or room......
THAN? Run maybe 5 scans in a row without touching anything. Don't even move that pillow and let the cat BE.......
What you'll get is an idea about how repeatable s system you have. And by system I mean 'measurement'......
I would suspect variations from scan to scan. these indicate the measurement limits and a certain 'confidence' that any given
amplitude at any given frequency is 'real'.....and potential margin of error....
|
|
|
|
Post by leonski on Nov 21, 2024 1:18:30 GMT -5
Any home bakers out there? Being winter I baked my first bread of the season this morning using my maybe 10 year old, Costco bought OYSTER machine.
When I bought the last 2 (one a gift) the intent was for wife to have it. WRONG! It's mine! And makes a wonderful loaf of French which is my standby.
The only thing I wish to do is get a good SOURDOUGH starter and experiment with sourdough Pizza Crust.....
Any other machine choices out there? I see a couple in the 200$ to 300$ range but that's just nutty. ALL use a variation of the same system with a
motor driven stirring paddle and a twistlock baking pan.
I MAY end up building a rising box out of a cooler, electric blanket and temp controller, but THAN I'd really be in for it but could make double loaves
or even some other shapes.
Within the next couple days? Stand Mixer comes out and after an ingredient inventory, I'll start on a couple rounds of COOKIES. I make the standard
3, which are Chocolate Chip, Peanut Butter and Oatmeal. I made Scones once and need to revisit that. Fresh is 100x better than what you get at the coffee shop.
|
|
|
|
Post by Boomzilla on Nov 22, 2024 16:21:22 GMT -5
Subwoofer phasing...
For the moment, I'm using a sub that lacks a phase switch or dial. I want to get the sub to blend with the satellites. Since I can't vary phase or polarity at the sub, I'm thinking that I can achieve the same by using the "distance" setting in the AVR? In theory, by varying the relative distances between the satellites and the sub, I should be able to adjust phase exactly as if I had a phase dial on the sub's plate amp?
Now since the C/O frequency is 40 Hz, and the wavelength of a 40 Hz wave is 28 feet, there'll have to be some serious changes to the relative distances. But is there anything I've overlooked that would prevent this plan from working?
Another "trick" that might help would to put a SPL meter halfway between the sub and the satellite, play a 40 Hz tone and then increment the distance setting in the AVR by 5 feet at a time, noting the SPL at each setting. The distance setting that results in the highest SPL would be the one where the sub and satellite are in phase.
|
|
|
|
Post by marcl on Nov 22, 2024 16:35:10 GMT -5
Subwoofer phasing... For the moment, I'm using a sub that lacks a phase switch or dial. I want to get the sub to blend with the satellites. Since I can't vary phase or polarity at the sub, I'm thinking that I can achieve the same by using the "distance" setting in the AVR? In theory, by varying the relative distances between the satellites and the sub, I should be able to adjust phase exactly as if I had a phase dial on the sub's plate amp? Now since the C/O frequency is 40 Hz, and the wavelength of a 40 Hz wave is 28 feet, there'll have to be some serious changes to the relative distances. But is there anything I've overlooked that would prevent this plan from working? Another "trick" that might help would to put a SPL meter halfway between the sub and the satellite, play a 40 Hz tone and then increment the distance setting in the AVR by 5 feet at a time, noting the SPL at each setting. The distance setting that results in the highest SPL would be the one where the sub and satellite are in phase. You're on the right track. If you use REW to measure one of your main speakers and your sub separately from around 30 to 100Hz you can look at the impulse response of each and see the timing difference. You can change the distance of the main speaker on the AVR farther away than it actually is to let the sub "catch up". Measure again until they're aligned. Check the phase on the overlays screen to see if they are in phase at 40Hz. |
|
|
|
Post by Boomzilla on Nov 22, 2024 20:09:53 GMT -5
When I had a laptop, I used to use REW, but I no longer have it, and my closest machine is a desktop at the far end of the house. Oh well...
|
|
|
|
Post by marcl on Nov 23, 2024 7:27:01 GMT -5
When I had a laptop, I used to use REW, but I no longer have it, and my closest machine is a desktop at the far end of the house. Oh well... What worries me about the idea of using an SPL meter and sine wave and adjusting distance for the loudest bass, is that you could possibly be adjusting to maximize excitation of a 40Hz resonance mode. It could also result in a cancelation at some other frequency. That would give you a big peak at 40Hz, but then what? Do you have PEQ to flatten it out? And how would you know if you can't measure? |
|
|
|
Post by Boomzilla on Nov 23, 2024 17:37:14 GMT -5
I can run a 20-20Khz with a cell phone app. Since all iPhones use the same microphone, the app's built-in EQ compensates. Is it as accurate as my UMIK-1 with REW? Not! But it's good enough to show peaks and dips.
Roon has its own built in DSP that can create notch filters, etc. So, between the two, I should be able to get close.
|
|
Hi marcl - 
) and to the first point of A/D-D/A yes I had no issue doing it for subs but I didn't want to run my main L/R through the miniDSP 2x4 Balanced or 2x4 HD. But what about the miniDSP Flex? Seems like that would do the job very well ... and a scenario might be analog 2-channel, but you want to have Bass Management and Dirac.
