There has been some discussion on this forum regarding what role, if any PC's will play in audio systems in the future. I believe that PCs will likely play a huge role in DIY systems in the future. This will have little to do with library convenience and music archival and everything to do with sound quality. I believe that adaptive signal processing will change the face of home audio systems forever; it just has not caught on yet.
Consider the following vision/prognostication:
As a DIYer, you rip the crossovers out of a few old speakers, which collectively have about 7 mismatched speaker cones. You build seven cheap gainclones, one for each transducer. You plug each gainclone into a board that has a IEEE1394 receiver chip and as many DAC's as you decided to solder onto your PCB (which someone else designed and can hold may DACs though you don't have to populate it all the way) and one ADC.
You buy a cheap microphone of a popular model and plug it into ADC. You plug the CODEC board into your iMac G5 or whatever PC you already have laying around and hang the mic from the ceiling about where your head would be.
You install open source software that has been hacked together collaboratively by DIYers all over the world, grad students at engineering universities as course projects, etc.
You tell this software a few things, like which speakers correspond to which signals in your digital music source and which model microphone you are using (other DIYers have posted models for the frequency response of several common mics).
If you're on a mac, you hit a big pulsing blue button labeled "Calibrate" and if you're on Linux you run a shell script with a name like "sp34kr_c4llbr8_2349x293.ruby" with 13 mostly documented parameters and walk away
The system automatically determines the frequency response (Amplitude AND phase) of each speaker/amp/dac, how much of each frequency each speaker/amp/dac can handle without non-linear distortion.
The system then plays a recording of Robin Williams "Good Morning Vietnam!!!" or something and you know it is save to come back into the room.
The software has determined an appropriate FIR (this will surely all be linear processing at least at first) filter for the output from each source signal to the listener, and you now have a fully calibrated and equalized "reference" system.
Since the overhead of having to tune each transducer and each amplifier carefully by hand has been removed, each transducer and each amp can be designed to work excellently and whatever portion of the frequency range it is best for.
Even the frequency response of your room has been compensated for, reducing the need to futz with speaker and furniture placement.
Of course, everyone has their own tastes. Once the computer has found its own baseline for your system, you can tinker with the filters all you want, re-equalizing certain recordings differently, etc. Since other DIYers out there have already built similar "reference" systems, you can share the filters you have come up with others.
The hardest part of this building this system will be designing and tuning the auto-calibration software, and fortunately that is the part of the system that can be created by specialists and shared with everyone for free.
Although I had the idea of closing the audio feedback loop independently a few years ago, I would be very surprised if I was the first, and several large american corporations have probably already patented it at the wonderful USPTO. I found one product implementing these ideas that uses custom hardware and is intended for a different market, but it should provide some indication of the technical feasibility of these ideas.
See the sabine REAL-Q2 Real-Time Adaptive Equalizer
http://www.sabine.com/newsite/realq.html
P.S. Active Room Equalization has also been discussed in another thread; this idea is different primarily in the 1 amp per cone and the use of PC's as the digital signal processor and the source rather than proprietary outboard hardware and algorithms. i.e. the Behringer Ultracurve Pro 8024 that was discussed in the other thread.
Anyone else out there share this vision?
Consider the following vision/prognostication:
As a DIYer, you rip the crossovers out of a few old speakers, which collectively have about 7 mismatched speaker cones. You build seven cheap gainclones, one for each transducer. You plug each gainclone into a board that has a IEEE1394 receiver chip and as many DAC's as you decided to solder onto your PCB (which someone else designed and can hold may DACs though you don't have to populate it all the way) and one ADC.
You buy a cheap microphone of a popular model and plug it into ADC. You plug the CODEC board into your iMac G5 or whatever PC you already have laying around and hang the mic from the ceiling about where your head would be.
You install open source software that has been hacked together collaboratively by DIYers all over the world, grad students at engineering universities as course projects, etc.
You tell this software a few things, like which speakers correspond to which signals in your digital music source and which model microphone you are using (other DIYers have posted models for the frequency response of several common mics).
If you're on a mac, you hit a big pulsing blue button labeled "Calibrate" and if you're on Linux you run a shell script with a name like "sp34kr_c4llbr8_2349x293.ruby" with 13 mostly documented parameters and walk away
The system automatically determines the frequency response (Amplitude AND phase) of each speaker/amp/dac, how much of each frequency each speaker/amp/dac can handle without non-linear distortion.
The system then plays a recording of Robin Williams "Good Morning Vietnam!!!" or something and you know it is save to come back into the room.
The software has determined an appropriate FIR (this will surely all be linear processing at least at first) filter for the output from each source signal to the listener, and you now have a fully calibrated and equalized "reference" system.
Since the overhead of having to tune each transducer and each amplifier carefully by hand has been removed, each transducer and each amp can be designed to work excellently and whatever portion of the frequency range it is best for.
Even the frequency response of your room has been compensated for, reducing the need to futz with speaker and furniture placement.
Of course, everyone has their own tastes. Once the computer has found its own baseline for your system, you can tinker with the filters all you want, re-equalizing certain recordings differently, etc. Since other DIYers out there have already built similar "reference" systems, you can share the filters you have come up with others.
The hardest part of this building this system will be designing and tuning the auto-calibration software, and fortunately that is the part of the system that can be created by specialists and shared with everyone for free.
Although I had the idea of closing the audio feedback loop independently a few years ago, I would be very surprised if I was the first, and several large american corporations have probably already patented it at the wonderful USPTO. I found one product implementing these ideas that uses custom hardware and is intended for a different market, but it should provide some indication of the technical feasibility of these ideas.
See the sabine REAL-Q2 Real-Time Adaptive Equalizer
http://www.sabine.com/newsite/realq.html
P.S. Active Room Equalization has also been discussed in another thread; this idea is different primarily in the 1 amp per cone and the use of PC's as the digital signal processor and the source rather than proprietary outboard hardware and algorithms. i.e. the Behringer Ultracurve Pro 8024 that was discussed in the other thread.
Anyone else out there share this vision?
> I certainly don't share that vision.
>
> Why would anyone want to build a system around 7 junk speakers?
Thanks for the links wimms, and sorry Bob, I'll be more clear on what I meant. You wouldn't use junk speakers... When I said old I had in my head "old but high quality", and the cones are mismatched because they came from different sources, not a matched set. i.e. maybe a cone was damaged beyond repair but its sister is still good. The odd number was to point out the extra flexibility auto-calibration could buy you in flexibility of parts usage.
The idea I was trying to get across is that the overhead, primarily in design and tuning, for adding extra speaker cones would be greatly reduced. The idea is to get the best sound for the money, but using more, less expensive cones, and using your PC in place of commercial DSP based hardware.
The cones don't have to have a wide or uniform frequency response; you just have to have at least one cone per source channel that is good at reproducing any given frequency range. Thus, less expensive cones. Likewise, the amp doesn't have to have a uniform response either. Perhaps it would be possible to even do non-linear processing to cancel out the non-linear effect of a transistor that swings far from its designed bias point. The use of more, cheaper redundant elements could pay off in sound quality per dollar. (or maybe it never will, I don't see why not)
One of the reasons we naturally find simple designs to be more effective and reliable is that the engineer (you don't have to have a degree. if you can successfully build a stereo from parts you are an engineer in my eyes)
> I disagree with another point as well. You start messing around with EQ > to correct for the room and you've destroyed the first arrival signal - > the important one.
I haven't studied frequency response (if it is even reasonably linear) of echoing cavities yet. (I've only seen waveguides and resonant cavities in an E+M context.)
Why is the first arrival signal is (necessarily) destroyed? That is certainly a possibility with a poor implementation of the algorithm, but I don't see why it is a necessity for all such algorithms. Assuming an inevitable and noticeable effect on the first arrival, it may still be worth it:
If your playing space has an awful echo, the effect of the echo and problems with equalization will be worth correcting even if this does have a detrimental effect on the first arrival. If you have a great listening space, the equalization might not be worth it at all. One of the sound systems I want to improve the most is the PA we use at our weekly swing dances. We dance in the top floor of a church with a stone floor and a vaulted ceiling.
http://www.cu-swing.org/gallery/weekly/DSC04234_b
Our fender PA is on the table to the right. There is a fairly large powered subwoofer on the floor outside the frame. This is obviously an extreme case. Again, perhaps the average DIYer's listening space is good enough that equalization would never catch on.
Heck, I would think that even if you never actually use the equalization for playback, I would still expect it to be an excellent tool for designing and tuning your traditional passive crossovers and amps. i.e. if you assume your amps to be perfect, or you have a good model for its imperfections, you can use it with each cone separately, and the software will suggest an optimal location for the crossover point(s) for each speaker. There are a bunch of calibration schemes you could use... eventually the community would probably settle on standard routines/guidelines.
Having all of your crossover / equalization being done in a central, easily re-programmable place also gives you a lot of freedom to devise creative new methods of minimizing distortion.
Suppose you have two transducers that are both efficient in one frequency range, but complimentary distortion characteristics with respect to volume. However, one has low distortion only at high volumes while the other has low distortion only at lower volumes. Perhaps you could intelligently, perhaps even, distribute the power between the two transducers.
Maybe one cone has really high fidelity, but distorts at high volumes. Cones with neighboring frequency responses could help pick up the slack when the volumes are high even though they are less efficient power-wise in the mid frequencies.
>
> Why would anyone want to build a system around 7 junk speakers?
Thanks for the links wimms, and sorry Bob, I'll be more clear on what I meant. You wouldn't use junk speakers... When I said old I had in my head "old but high quality", and the cones are mismatched because they came from different sources, not a matched set. i.e. maybe a cone was damaged beyond repair but its sister is still good. The odd number was to point out the extra flexibility auto-calibration could buy you in flexibility of parts usage.
The idea I was trying to get across is that the overhead, primarily in design and tuning, for adding extra speaker cones would be greatly reduced. The idea is to get the best sound for the money, but using more, less expensive cones, and using your PC in place of commercial DSP based hardware.
The cones don't have to have a wide or uniform frequency response; you just have to have at least one cone per source channel that is good at reproducing any given frequency range. Thus, less expensive cones. Likewise, the amp doesn't have to have a uniform response either. Perhaps it would be possible to even do non-linear processing to cancel out the non-linear effect of a transistor that swings far from its designed bias point. The use of more, cheaper redundant elements could pay off in sound quality per dollar. (or maybe it never will, I don't see why not)
One of the reasons we naturally find simple designs to be more effective and reliable is that the engineer (you don't have to have a degree. if you can successfully build a stereo from parts you are an engineer in my eyes)
> I disagree with another point as well. You start messing around with EQ > to correct for the room and you've destroyed the first arrival signal - > the important one.
I haven't studied frequency response (if it is even reasonably linear) of echoing cavities yet. (I've only seen waveguides and resonant cavities in an E+M context.)
Why is the first arrival signal is (necessarily) destroyed? That is certainly a possibility with a poor implementation of the algorithm, but I don't see why it is a necessity for all such algorithms. Assuming an inevitable and noticeable effect on the first arrival, it may still be worth it:
If your playing space has an awful echo, the effect of the echo and problems with equalization will be worth correcting even if this does have a detrimental effect on the first arrival. If you have a great listening space, the equalization might not be worth it at all. One of the sound systems I want to improve the most is the PA we use at our weekly swing dances. We dance in the top floor of a church with a stone floor and a vaulted ceiling.
http://www.cu-swing.org/gallery/weekly/DSC04234_b
Our fender PA is on the table to the right. There is a fairly large powered subwoofer on the floor outside the frame. This is obviously an extreme case. Again, perhaps the average DIYer's listening space is good enough that equalization would never catch on.
Heck, I would think that even if you never actually use the equalization for playback, I would still expect it to be an excellent tool for designing and tuning your traditional passive crossovers and amps. i.e. if you assume your amps to be perfect, or you have a good model for its imperfections, you can use it with each cone separately, and the software will suggest an optimal location for the crossover point(s) for each speaker. There are a bunch of calibration schemes you could use... eventually the community would probably settle on standard routines/guidelines.
Having all of your crossover / equalization being done in a central, easily re-programmable place also gives you a lot of freedom to devise creative new methods of minimizing distortion.
Suppose you have two transducers that are both efficient in one frequency range, but complimentary distortion characteristics with respect to volume. However, one has low distortion only at high volumes while the other has low distortion only at lower volumes. Perhaps you could intelligently, perhaps even, distribute the power between the two transducers.
Maybe one cone has really high fidelity, but distorts at high volumes. Cones with neighboring frequency responses could help pick up the slack when the volumes are high even though they are less efficient power-wise in the mid frequencies.
Ignore the Luddites. I think your vision is worth pursuing. Mercator and I have been discussing a similar idea, although not as ambitious as yours. I like the idea of multi-amping and active crossovers. I like the idea of crossovers in software, particularly given that sources will all become digital (or will be ripped into being so). DSP (even if done on a PC) produces much better filters. I like the idea of PCs because they're cheap. Add in the autoequalization and it sounds pretty good.
Of course, it begs the question, how do you know what you like? I think your preferred settings would vary with source material. Maybe we need controls like soundstage and brightness and transparency instead of bass and treble.
I don't understand Bill's concern with first arrival signal - why would it be any different?
Of course, it begs the question, how do you know what you like? I think your preferred settings would vary with source material. Maybe we need controls like soundstage and brightness and transparency instead of bass and treble.
I don't understand Bill's concern with first arrival signal - why would it be any different?
Bill Fitzpatrick said:
Judging from your bio, probably a lot more than me about practical matters and the current state of technology. I'm still being steeped in academia, which means I'm full of wacky ideas about technology that would take years to be accepted (or become economical), in the unlikely chance that they get accepted at all.
That said, I think this is a pretty solid idea; the requisite technology is already available, though I don't know if it is economical (in time and money) for DIYers to implement something like this yet.
Two cents from an audiophile.
I believe crossovers is really hard to tame. Adding the difference in units (tweeter, woofer.. etc), the sound from multi way speakers almost always feel altered to my ears. No matter what you do, it just doesn't sound smooth near/at the crossover point; at least not like full range units.
I think once the signal is altered you can't change it back, no matter how you do it. Even if the graph shows up correct, my ears would tell me otherwise.
I think the technology is not advance enough to fool the brain yet.
I believe crossovers is really hard to tame. Adding the difference in units (tweeter, woofer.. etc), the sound from multi way speakers almost always feel altered to my ears. No matter what you do, it just doesn't sound smooth near/at the crossover point; at least not like full range units.
I think once the signal is altered you can't change it back, no matter how you do it. Even if the graph shows up correct, my ears would tell me otherwise.
I think the technology is not advance enough to fool the brain yet.
paulb said:
With a PC as both your source and an active element in the audio loop, it is easy to do things like having settings that are persistent for the recording. EQ settings can get stored in ID3 tags and shared between users a-la CDDB. Only thing that prevents that now is that so few people have callibrated systems, eq settings have no meaning. Using a PC to close the feedback loop allows everyone to have a callibrated system.
I haven't spent a lot of time having deep discussions about subjective/artistic qualities as soundstage, brightness, transparency, so I don't ~really understand what people mean by them. I have vague notions, but have not lived enough to be comfortable discussing them, let alone comment on and if/how you could improve them independently through DSP. I know what I like when I hear it, but I don't have a language for it yet.
We always start from engineering critera of good, and then add the human / artistic element. Using a PC for closed loop DSP only provides a new framework that seems to me to make a lot engineering sense, and allows a whole new playground for people to express themselves artistically through creative design. This time the design just happens to be mostly on a computer where most of what you can think up is possible.
Have to go design differential amplifiers... (this time it's homework)
Here's my imagination:
1: Cover *all* the walls with "Speaxels" - a transducer that takes as input a low-level signal, and outputs linear excursion to a 5 cm^2 "cone". Think small amp & speaker driver all in one. Arrayed like pixels. Individually addressable via RF, powered by power connectors embedded in the wall, and cooled via the wall as well.
2. The listener then puts on the "Audiophool Hat". This is a hat, or some other artifice which has spatial indicators on it, like those silly suits with reflectors that they use for motion capture. We need to locate the listener's ears in space. Or maybe we do it with microwaves, or lasers, or sharks with lasers on thier heads or whatever. Spatial location accomplished somehow.
3. Press play.
4. Let the computer figure out how to put the sweet spot at the listener's head by varying the volume, phase, delay, etc.. from each Speaxel in real time, and following the listener anywhere in the room. Need to do hi frequencies? One speaxel active. Need to do low frequencies? Many speaxels at once. Need to kill a reflection? Modulate the speaxel in anti-phase when the wave form impinges on it. etc. Heck, with 3 actuators behind a planar speaxel surface, we could change the pitch of the surface for more/less diffusion.
5. Perfect Sound Forever. Send me a postcard if you make any money at it.
1: Cover *all* the walls with "Speaxels" - a transducer that takes as input a low-level signal, and outputs linear excursion to a 5 cm^2 "cone". Think small amp & speaker driver all in one. Arrayed like pixels. Individually addressable via RF, powered by power connectors embedded in the wall, and cooled via the wall as well.
2. The listener then puts on the "Audiophool Hat". This is a hat, or some other artifice which has spatial indicators on it, like those silly suits with reflectors that they use for motion capture. We need to locate the listener's ears in space. Or maybe we do it with microwaves, or lasers, or sharks with lasers on thier heads or whatever. Spatial location accomplished somehow.
3. Press play.
4. Let the computer figure out how to put the sweet spot at the listener's head by varying the volume, phase, delay, etc.. from each Speaxel in real time, and following the listener anywhere in the room. Need to do hi frequencies? One speaxel active. Need to do low frequencies? Many speaxels at once. Need to kill a reflection? Modulate the speaxel in anti-phase when the wave form impinges on it. etc. Heck, with 3 actuators behind a planar speaxel surface, we could change the pitch of the surface for more/less diffusion.
5. Perfect Sound Forever. Send me a postcard if you make any money at it.
David@NY said:
Thanks for the reply David. Is the tradition that the crossover is as sharp a transition as possible so that you don't have two cones outputing the same signal at slightly different phase, and the resulting interference patterns in space? Given that you prefer a full range system over a multi way system either way, how do digital crossovers compare to their analog counterparts in the same price range?
Do you think closed loop DSP could have value in a full range system? It still adds possibilities. If you already use a digital source, your current setup is already a special case of this system. Modifications could be arbitrarily subtle.
drewm1980 said:
I'm not familiar with the technical stuff so bear with me if my comments seem silly.
Traditional (passive) crossovers suffers from mechanical parts inconsistencies. Digital crossover in theory should alleviate such problem. However our experiences with digital may still not advanced enough to tackle the issue. For some reason, the timing issue in analog domain is more pleasurable to the ear than in digital domain. So in digital domain timing is critical. We're having enough trouble keeping two devices in complete sync (transport + DAC), let along multiple devices (please correct me if I'm wrong).
If you're into digital crossover route, you may also want to check out the traditional route -- create separate amps for each driver unit and put the (analog) crossover right on the amp stage. However one potential problem with that is you may have inconsistent response at different volume (each driver respond differently as volume changes). We may fix that in digital domain using DSP.
I have no idea what closed loop DSP does so I can't comment on it.
I'd very much to see the project suceed. As I see it digital may resolve some issues but not all. As far as "perfect sound forever", digital may not be the direction because there's still a lot of stuff missing between the 0s and 1s.
Good thread!
Drew,
Nice ideas. Maybe not so much the "junk cones" but a lot of it makes sense. Make the best out of what you've got for system and room.
Bill,
What is the point of your sniping? If you have any direct knowledge of what you're criticizing, it would be incumbent on you to SHARE IT.
Denis Sbragion explained to me that it is wrong to think of DRC as "equalization". The direct signal (zero'th reflection) is NOT modified by the DRC filter at all unless it needs correction in the first place due to system response issues. DRC does cancel wall reflections and will precisely time/phase align the drivers and by improving the ideality of impulse response will provide much flatter frequency response. This is NOT equalization.
I am still experimenting with DRC and am far from expert with this technology but it's obvious that it has a benefit and is economical even now. Many in diyAudio.com have verified this.
-Robert
Drew,
Nice ideas. Maybe not so much the "junk cones" but a lot of it makes sense. Make the best out of what you've got for system and room.
Bill,
What is the point of your sniping? If you have any direct knowledge of what you're criticizing, it would be incumbent on you to SHARE IT.
Denis Sbragion explained to me that it is wrong to think of DRC as "equalization". The direct signal (zero'th reflection) is NOT modified by the DRC filter at all unless it needs correction in the first place due to system response issues. DRC does cancel wall reflections and will precisely time/phase align the drivers and by improving the ideality of impulse response will provide much flatter frequency response. This is NOT equalization.
I am still experimenting with DRC and am far from expert with this technology but it's obvious that it has a benefit and is economical even now. Many in diyAudio.com have verified this.
-Robert
Good point, I asked the wrong question.Bill Fitzpatrick said:
Why is that a problem? Of course you're altering it. You are correcting it (whatever that turns out to mean). I think what you're saying is, if you fix some things, you'll screw up others. This is the "you'll never get it working right" POV. Could be true. Fun to find out. You can learn much more from failure than from success.
Re: Good thread!
Hey, I'm not the one who stuck the word "equalization" into this discussion. All I said was EQ is a bad idea. No need to be all in my face.
I still want to know what the remark about the project meant. I didn't know there was a project. This is just a discussion which will ultimately die. It's way to advanced for the likes of those who are participating in it.
RFScheer said:
Hey, I'm not the one who stuck the word "equalization" into this discussion. All I said was EQ is a bad idea. No need to be all in my face.
I still want to know what the remark about the project meant. I didn't know there was a project. This is just a discussion which will ultimately die. It's way to advanced for the likes of those who are participating in it.
Re: Re: Good thread!
AndyN - I suspect your "speaxels" idea as proposed would suffer from air rushing around each individual cone, thus eliminating any bass response. A whole lot of little cones does not result in one large cone. However, perhaps your could attach a lot of actuators to one large planar membrane. I've seen something very similar before... for telescopes.
I forget what it was called, but the idea is that you have one huge deformable mirror, and somehow you can adaptively deform it to cancel out atmospheric disturbances. Supposedly this technology has completely obsoleted space telescopes for certain kinds of observation. Maybe the elements could repond intelligently but passively to an external source. (i.e. they don't try to drive the wall, just change its local impedance)
Still another dream technology- maybe you could just always output a large amplitude, sub-sonic signal that the wall modulates up into higher frequencies... since this is just a fun daydream, the speaxels could even be powered by energy taken from the incoming signal the same way wireless keycards are.
David@NY -
I thought more about crossovers, and I'm certain this must have been discussed in other threads. The two cones in either will necessarily have some overlap in their frequency responses, and I believe there are limitations on how well the phase from one can agree with the other.
If you super-impose to sinusoids at the same frequency, you get a third sinusoid of the same frequency, but of a magnitude that will depend on the phase difference between the two original signals. Thus, differences in the phase responses of the two cones will affect the magnitude response of the entire speaker, mostly in the cross-over region. My guess is that this is the "unsmoothness" you say you can hear with any multi-way system.
paulb-
I thought more about your idea... I think it would be very feasible to have user-tunable parameters for things like the relative importance of smooth magnitude response, coherence of first arrival signal, non-linear distortion, power consumption of the system, etc.
The way it would work is that during the identification phase, the system models the system and the room (very little user interaction). After that, the user gets to help tune the filters (not necessarily all linear ) used for playback to his/her tastes.
AndyN - I suspect your "speaxels" idea as proposed would suffer from air rushing around each individual cone, thus eliminating any bass response. A whole lot of little cones does not result in one large cone. However, perhaps your could attach a lot of actuators to one large planar membrane. I've seen something very similar before... for telescopes.
I forget what it was called, but the idea is that you have one huge deformable mirror, and somehow you can adaptively deform it to cancel out atmospheric disturbances. Supposedly this technology has completely obsoleted space telescopes for certain kinds of observation. Maybe the elements could repond intelligently but passively to an external source. (i.e. they don't try to drive the wall, just change its local impedance)
Still another dream technology- maybe you could just always output a large amplitude, sub-sonic signal that the wall modulates up into higher frequencies... since this is just a fun daydream, the speaxels could even be powered by energy taken from the incoming signal the same way wireless keycards are.
David@NY -
I thought more about crossovers, and I'm certain this must have been discussed in other threads. The two cones in either will necessarily have some overlap in their frequency responses, and I believe there are limitations on how well the phase from one can agree with the other.
If you super-impose to sinusoids at the same frequency, you get a third sinusoid of the same frequency, but of a magnitude that will depend on the phase difference between the two original signals. Thus, differences in the phase responses of the two cones will affect the magnitude response of the entire speaker, mostly in the cross-over region. My guess is that this is the "unsmoothness" you say you can hear with any multi-way system.
paulb-
I thought more about your idea... I think it would be very feasible to have user-tunable parameters for things like the relative importance of smooth magnitude response, coherence of first arrival signal, non-linear distortion, power consumption of the system, etc.
The way it would work is that during the identification phase, the system models the system and the room (very little user interaction). After that, the user gets to help tune the filters (not necessarily all linear ) used for playback to his/her tastes.
way to advanced for the likes of those who are participating in it
Oh man! I love a challenge like that.
This sounds like a good idea to me. As has been pointed out, most of the pieces of the puzzle are already in place, with the software glue remaining to be built.
The main objection from the audiophile perspective would seem to relate to the deficiencies in sampling frequency and resolution of the digital hardware. I believe it is reasonable to expect this technology to improve, therefore getting started on the software aspect of the 'project' would be a good idea.
Count me in, it fits right in with my current plans anyway.
/Dave
Oh man! I love a challenge like that.
This sounds like a good idea to me. As has been pointed out, most of the pieces of the puzzle are already in place, with the software glue remaining to be built.
The main objection from the audiophile perspective would seem to relate to the deficiencies in sampling frequency and resolution of the digital hardware. I believe it is reasonable to expect this technology to improve, therefore getting started on the software aspect of the 'project' would be a good idea.
Count me in, it fits right in with my current plans anyway.
/Dave
- Status
- This old topic is closed. If you want to reopen this topic, contact a moderator using the "Report Post" button.