It seems that we have 2 parties here who both don't understand a thing of what they are talkign about.
1. When you do not have a right to use certain code, still using it and distributing it with a GPL license still doesn't bring this code under the GPL. The GPL is pretty clear about that.
This means that the Mamba team seems to be wrong, regardless of if the code was partially derived from their project or not, it includes code to which the person who put it under the GPL had no right to use.
That said, the trade secret story is utter bullshit.
First of all, using a trade secret to gain a competative advantage is not only legal, it is THE reason for them to exist.
Stealing a trade secret is first of all a breach of contract, second, it can be a crime.
This is an issue between tge supposed owners of the code and the person who leaked it. No other peopel are involved, they are not part of the contract, and had no reason to believe this code was supposed to be a trade secret.
So.. they have a case against their developer, not againstr any end-user, and it is already doubtfull they have any case with regards to Mamba. If a secret is out, it is no longer a secret, and there the story really ends.
If the original code was derived from the Mamba project then the owners of that code may have to release it as GPL if they distribute it. It is not clear to me from the article if the original code was indeed derived from the Mamba project, but if so, then that eliminates any possible case against the Mamba team as well (a case which would not be able to go beyond fixing the damage at any rate)
Nom I am saying they are convicted of having illegally obtained monopolies (in the USA and EU) and nothing is done about it. You don't seem to like facts and may not find them convincing, well, guess that I stay with my original advice to you but should add that you act like you are incredibly stupid.
> That may be true, but that still doesn't make Linux any more useful to me, since it can't do what I need it to do.
According to your previous post, you did not get ti to work. That however means you can't judge if it can or can't do what you need, and besides, you change your agrument.
Matter of fact is, you can't get it to do what you need.
When a company has 80% proffit margins on a product, uses illegal methods to monopilize other markets using their existing monopoly and pays off politicians so that nothign gets done about it, I also do see why people really don't think much of taking a bit away from them.
As for myself, I have legit copies of win95, 98 and 2k, none of which are being used (no WIndows here, I rather just use an alternative if OI don't feel like giving my money to a company)
THat said, you (black mariah) should fuck off untill you learned to argue instead of producing a stream of 'cool' sounding words and insults.
So tell me.. how are you going to do those things without net access? (don't give me either 'you should have done it beforehand' or 'with another computer', there are many situations in which neither is/was a possibility)
Complaining about JAVA... first of all, take a peek here for some not so technical but still rather real problem.
Then, it seems to me that hotspot on x86 machines mostly ends up generating 486, and in few ocations 586 optimized native code, and there are quite a few cases where that won't be that difficult to beat with modern compilers and good optimizing with any pre-compiled language (including JAVA of course). This is not a theoretical limitation on JAVA performance in any way, but it is one in practise on one of the most often used platforms.
That said, I am happy with the performance of it esp. for serverside use, and my workstation is more then powerfull enough to not really notice startup delay of applications. I do not like its resource use, esp. when it comes to memory tho.
> Tube amps (older designs, at least) can be damaged if they're connected to the wrong load, whereas solid-state amps will do whatever you want until you try to pull too much current (at which point they will probably overheat and shut down).
Quite true, but not entirely what I was talking about.
The issue is as follows:
When you go measure the impednace of a speaker system over the entire audio spectrum, you will notice it isn't exactly the same everywhere, and ahs various peaks, most notably around the resonanse frequency.
A tube amp doesn't care about that, while generally spoken, a solid state one does.
You will also find that such an impedance/frequency graph differs from speaker system to system, and is in part why certain speakers go better with a certain (solid state) amp then others while their specs are very similar at first glance.
From your entire bug report, it seems pretty obvious that something is rather wrong with the ata controller and/or bios on your hardware.
I happen to have had 3 of the same Maxtor drives as you have, and the last surviving one is currently primary master in my router. (hint, replace them, I have had 2 of them give up within short time of eachother, and the 3rd one seems to be getting close to giving up)
All 3 have always worked with FreeBSD, Linux and Windows, without any confusion with regards to its CHS settings.
I did however follow the rather strong suggestion that my BIOS gives me to run the bios auto detect and select 'normal' mode for a drive on which I am going to use a unix like system. This results in a user configured drive (for as far as the standard cmos settings go)
Oh, and I also followed the recomendations to have the disk as primary, and a cd drive as slave instead of having it as primary like you have.
That it is not recognized that way during sysinstall seems rather suspicious to me.
Maybe FreeBSD needs some patches for detectign this broken hardware configuration, just as it did get a patch for dealing with the rather broken bios of the asus p2b-ds that I happen to use (apic is broken, resulting in an interupt storm when doing things according to the official standard)
If you think this problem affects more then your very specific case, I suggest talking about it on the mailinglists for -current, and try to be helpfull in getting a workaround, ie, that means accepting that it is in fact a problem of your hardware.
All that said, your comment regarding fdisk seems to be correct, and it should accept the alternative geometry if those fit within the physical number of sectors that the drive has.
> Well, yes, that was my point. They're a different instrument, like a piano is a different instrument than a harpsichord, or a saxophone is a different instrument than a trumpet.
Well, a modern digital piano is a very faithfull simulation in sound of the real thing, if that is enough depends a lot on what you are going to use it for, but for example for your average pop/rock band it is more then good enough, while for some jazz and concert pianists it really won't do.
It is a bit different with regards to things like a virtual saxophone but not because it can't be used as a decent simulation of the real thing.
(partial) virtualisations of existing instruments may be incomplete, but also offer many possibilities that the real thing wont offer. Ever tried playing a guitar like it was a saxophone?
The combination of a good keyboard and mouthpiece open very cool possibilities for expression with synthesizers, but beyond that I have seldom seen such things being used on stage, when you want a saxophone, that big shiny instrument looks so much better... and when you are recording a studio track and have someone who can play such an instrument, you are quite likely to have the 'real' instrument at hand anyway.
This changes with hard to transport things like organs and pianos, which is why you will find those used as replacement of the real thing quite a bit I think.
Trying to replace those with easer to transport alternatives has been a quest for a few generations (human generations..) now.
So well, virtualisations of other instruments tend to turn into instruments in their own right often simply because that is about the only interesting aspect of them, they allow you to do things to the instrument that would not be possible in the physical world.
> Digital recording and playback doesn't magically remove the "noise" in the music. Whether the reproduction is analog or digital, if it's accurate at reconstructing the original music it will reproduce the original noise.
It will not automagically recreate the typical noise, distortion and coloring of the PA system that was used during a live performance, it won't reproduce the typical coloring of the studio monitors.
The recording you get is incomplete in the sense that you do not get a recording of what it sounded like while the recording was mastered, you get (a copy of) the source without the listenign equipment.
> What are you getting at?
I don't know if this is what the parent meant, ut in my original post I was suggesting there is a difference between the kidn of soudn equipment used for amplification of a live performance, and the monitor system in a studio for example, and of course the acoustics of the room/hall/whatever is an entirely seperate story still.
You want to 'recreate' the typical limitations and coloring of such a system if you want something to sound very close to how the original recording was intended to sound. (and yeah, you would want some comparable acoustics as well)
There is simply a lot more to reproducing soudn then seeing no or very little distortion on a scope connected to the output of an amp, or with nearby mics when lookign at speakers. You can say a lot about the theoretical performance of such components, but the interaction between speakers, cable, amplifier and acoustics make that it is not easy to predict how music will sound based on such measurements.
It becomes a lot clearer when you go try to 'measure' what a listener would hear, but even in that case there is the simpel problem that a microphone and the human ear register sound in soemwhat different ways (ear has a very nice physical highpass filter that prevents certain types of interference that would occur when registering the same sounds with a microphone for example)
I'm not trying to say that audio is voodoo magic, but that there are many more factors that play a role then most people assume, and while you can strive for a system that produces the most accurate sound from a given source, that still will not get you any guarantee that you are hearing things as they were intended.
> I could question how complete they could be, given how minor differences in the design of a piano can change the sound,
Oh, you are right there, the simulations are incomplete except for the most basic cases. They do consider shape and material tho, but it is always a model and not reality that is used for the simulation, so it will never be perfect.. how close you get? it is good enough for a Jazz pianist to make use of it for fun things as having strings resonate on notes being played by just keeping the key open for example, but it is far from good enough to simulate all subtle details of a good concert piano.
> but it's only a start down the road to being able to genuinely simulating real instruments with digital ones: the piano has been analysed and studied and simulated with synthesizers to a far greater degree, I doubt that there's a "digital trumpet" or "digital violin" with anything like the range of the real thing.
Digital trumpet and sax (or better said, digital wind and reed instruments) exist, including 'real' mouthpieces for playing..
The fun thing is that those are more instruments in their own right then good simulations of the real thing.
> The point is, whether the recording session was analog or digital is less important than whether the instruments are physical, analog synthesizers, or digital synthesizers... and whether the playback is analog or digital is even less relevant.
I disagree.
Whatever the instruments are is an artistic choice, and has nothign to do whatsoever with being digital or analog.
Being able to capture and later reproduce what is being played depends first of all on the hardware and technology used for recording.
Having digital instruments may matter when doing digital recording, resampling can introduce nasty artifacts.
Your playback equipment matters mostly in the sense of it havign to be representative for similar equipment used while mastering or during the live performance, while also having to fit within your requirements for price, space and in the end, taste.
> When you start talking about taking that, sampling or resampling it to AAC on an iPod, and then playing it back through a tube amplifier instead of a solid state amplifier so you can "think about the music, [not] just listen"... there's a breakdown in thinking that has nothing to do with the music.
Oh, there I agree, and so much happened to the music that faithfull reproduction is no longer really what it is about.
The issue is that quality and kidn of playback equipment has to be 'good enough' for the purpose, which isn't exactly the same as 'best there is'.
It makes no sense to have ultra high range audio equipment in your car, capable of 120db dynamic range etc, when the dynamic range in your car is severely limited due to for example noise fromn trafic and your tires and engine.
The same applies for portable music normally, but I do at times use a portable mp3 player while laying in bed, and since my bedroom is extremely quiet, all of the sudden it is going to matter a lot which headphones I connect to it, and actually, using a big amp and some real headphones will do wonders in that case... whereas they wont matter much at all out on the street.
> Yet many electronic musicians will employ a compressor on a drum beat and the result can be a more lively and interesting sound, depending on the characteristics and tweaking of the compressor.
Yeah, quite true. Same applies for a bass guitar and similar instruments. (oh, and try an electric guitar without compression for some real fun;)
THey are usually a piece of hardware... but I bet there is software aroudn that does this as well.
I use the parametric equilizers of my audio mixer, which do the job very well for me (4 bands, all with adhustable center and range), and never really looked for a software soklution.
That said, my Boss sx700 has a digital arametric equilizer, so I'm sure there are software implementations, just never looked for one on my computer.
A graphic equilizer is easier to setup maybe because it gives you a 'graphic' view of what you are doing.. A parametric equilizer however produces a lot less phase shifting and similar problems because it only works for the frequency ranges where you actually need to change the sound.
You are ignoring one very important difference between solid state and tube based amps.
Solid state amps rely on feedback from speakers/cable etc to work without distortion, and as a result, depend a lot on what speakers you connect to it to get anywhere near that desirable ultra low distortion.
A tube amp doesn't rely on this, and willd eliver a constant quality regardless of what you connect to it.
As a result you may find that a tube amp does a lot better at driving crappy speakers then your solid state amp.
> is the skilled architect of a concert hall to be seen as better than the engineer who masterfully uses compression, eq, and other techniques?
The dymanics of a hall actually make the sound more proounced, compression etc makes it fit within the dynamic range of some emdium. Yes, those 2 are entirely different things.
You are right that both the architect and the sound tech aim at giving an as good as possible listening experience, but do so within completelyd ifferent sets of limitations.
> the interference patterns between the strings and in the body of the piano are going to be different.
Which has been known for a long time, and there are digital simulators of this particular aspect (they have een around for at least half a decade also)
Nonetheless, you have a very good point, an instrument is an instrument and sounds as it sounds. Its upto the mucisian to decide if it is what (s)he wants, and that is totally subjective and irrelevant when lookign at quality of reproduction, it is only relevant for the instrument and recording.
Reproduction should be as accurate as possible, ut there is a whole set of problems with that.
Are you listening to a live recordign that has been made of the soundboard on soem live gig?
Then you betetr play it back through comparable equipment and with comparable acoustics to get anythign near what was there, and even then you lose a lot of dynamics in the process (mostly during recording)
Are you listening to some studio recording? In that case there is no 'realistic reproduction' unless you are going to try to recreate the sound as it was on the studio monitors.
For those 2 cases you would already end up with different requirements for reproduction, and requirements which might actually be conflicting.
It seems that this all doesn't matter as logn as you are not overloadign the amp.
A 'modern' tube amp can be as colorless as a transistor amp, mosfet amps can approach the distortion as given by an overloaded tube ampetc etc..
Where it really differs, and where tubes beat the crap out of any transistor/mosfet amp is in that they usually do not rely on the feedback from the speakers/cable to work well, whereas a transistor based amp does. This affects distortion of the amp quite a bit.
> which I'm pretty sure includes (for them) being able to distinguish sound differences from different "power supplies, power cords, interconnect cables, and speaker cables"
A power crd should not matter, but there are very good and scientific reasons why the power supply, inter connect cables and speaker cables do make quite a difference.
That is not to say that you should buy all the audiophile bs, but you should buy good cables for conencting your equipment in general and your speakers in particular because the cable directly affects loss and in some cases interference.
The power supply is simply important because it has to be able to provide the power needs of the amplifier, and the better it is at that, the better the amplifier will perform.
It makes no sense however to overdo things, and spendign a fortune on cables is not needed for example.
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Hmm.. mabe I should start using a spell checker... and of course it is Mambo, not Mamba.. ah well..
It seems that we have 2 parties here who both don't understand a thing of what they are talkign about.
1. When you do not have a right to use certain code, still using it and distributing it with a GPL license still doesn't bring this code under the GPL. The GPL is pretty clear about that.
This means that the Mamba team seems to be wrong, regardless of if the code was partially derived from their project or not, it includes code to which the person who put it under the GPL had no right to use.
That said, the trade secret story is utter bullshit.
First of all, using a trade secret to gain a competative advantage is not only legal, it is THE reason for them to exist.
Stealing a trade secret is first of all a breach of contract, second, it can be a crime.
This is an issue between tge supposed owners of the code and the person who leaked it. No other peopel are involved, they are not part of the contract, and had no reason to believe this code was supposed to be a trade secret.
So.. they have a case against their developer, not againstr any end-user, and it is already doubtfull they have any case with regards to Mamba. If a secret is out, it is no longer a secret, and there the story really ends.
If the original code was derived from the Mamba project then the owners of that code may have to release it as GPL if they distribute it. It is not clear to me from the article if the original code was indeed derived from the Mamba project, but if so, then that eliminates any possible case against the Mamba team as well (a case which would not be able to go beyond fixing the damage at any rate)
Nom I am saying they are convicted of having illegally obtained monopolies (in the USA and EU) and nothing is done about it. You don't seem to like facts and may not find them convincing, well, guess that I stay with my original advice to you but should add that you act like you are incredibly stupid.
> That may be true, but that still doesn't make Linux any more useful to me, since it can't do what I need it to do.
According to your previous post, you did not get ti to work. That however means you can't judge if it can or can't do what you need, and besides, you change your agrument.
Matter of fact is, you can't get it to do what you need.
I don't like software 'piracy' but..
When a company has 80% proffit margins on a product, uses illegal methods to monopilize other markets using their existing monopoly and pays off politicians so that nothign gets done about it, I also do see why people really don't think much of taking a bit away from them.
As for myself, I have legit copies of win95, 98 and 2k, none of which are being used (no WIndows here, I rather just use an alternative if OI don't feel like giving my money to a company)
THat said, you (black mariah) should fuck off untill you learned to argue instead of producing a stream of 'cool' sounding words and insults.
So tell me.. how are you going to do those things without net access? (don't give me either 'you should have done it beforehand' or 'with another computer', there are many situations in which neither is/was a possibility)
Hehehehe. yeah, I can't type ;)
A food software encoder may be more tiem consuming, but can produce superior quality.
Multipass encoding is whaat you can't do in hardware, and what makes a huge difference when you want to create the best quality/size.
MPEG editing in itself is trivial, and can be done without (much) loss of quality indeed, but the same is not true for MPEG encoding.
Complaining about JAVA... first of all, take a peek here for some not so technical but still rather real problem.
Then, it seems to me that hotspot on x86 machines mostly ends up generating 486, and in few ocations 586 optimized native code, and there are quite a few cases where that won't be that difficult to beat with modern compilers and good optimizing with any pre-compiled language (including JAVA of course). This is not a theoretical limitation on JAVA performance in any way, but it is one in practise on one of the most often used platforms.
That said, I am happy with the performance of it esp. for serverside use, and my workstation is more then powerfull enough to not really notice startup delay of applications. I do not like its resource use, esp. when it comes to memory tho.
> Tube amps (older designs, at least) can be damaged if they're connected to the wrong load, whereas solid-state amps will do whatever you want until you try to pull too much current (at which point they will probably overheat and shut down).
Quite true, but not entirely what I was talking about.
The issue is as follows:
When you go measure the impednace of a speaker system over the entire audio spectrum, you will notice it isn't exactly the same everywhere, and ahs various peaks, most notably around the resonanse frequency.
A tube amp doesn't care about that, while generally spoken, a solid state one does.
You will also find that such an impedance/frequency graph differs from speaker system to system, and is in part why certain speakers go better with a certain (solid state) amp then others while their specs are very similar at first glance.
> Do not use screen savers.
All too true.
And I have similar experiences with win2k and power management.. tho I have seen ti work as well.
As for my own machines, most run either FreeBSD or Linux, and power management seems to work in both cases.
hmm.. here are such things as acpi and apm.. and suspend and resume features.
From your entire bug report, it seems pretty obvious that something is rather wrong with the ata controller and/or bios on your hardware.
I happen to have had 3 of the same Maxtor drives as you have, and the last surviving one is currently primary master in my router. (hint, replace them, I have had 2 of them give up within short time of eachother, and the 3rd one seems to be getting close to giving up)
All 3 have always worked with FreeBSD, Linux and Windows, without any confusion with regards to its CHS settings.
I did however follow the rather strong suggestion that my BIOS gives me to run the bios auto detect and select 'normal' mode for a drive on which I am going to use a unix like system. This results in a user configured drive (for as far as the standard cmos settings go)
Oh, and I also followed the recomendations to have the disk as primary, and a cd drive as slave instead of having it as primary like you have.
That it is not recognized that way during sysinstall seems rather suspicious to me.
Maybe FreeBSD needs some patches for detectign this broken hardware configuration, just as it did get a patch for dealing with the rather broken bios of the asus p2b-ds that I happen to use (apic is broken, resulting in an interupt storm when doing things according to the official standard)
If you think this problem affects more then your very specific case, I suggest talking about it on the mailinglists for -current, and try to be helpfull in getting a workaround, ie, that means accepting that it is in fact a problem of your hardware.
All that said, your comment regarding fdisk seems to be correct, and it should accept the alternative geometry if those fit within the physical number of sectors that the drive has.
Heh, and now I wish I had mod points today.. you and the poster of the parent (if not the same) are just too funny.
> Well, yes, that was my point. They're a different instrument, like a piano is a different instrument than a harpsichord, or a saxophone is a different instrument than a trumpet.
Well, a modern digital piano is a very faithfull simulation in sound of the real thing, if that is enough depends a lot on what you are going to use it for, but for example for your average pop/rock band it is more then good enough, while for some jazz and concert pianists it really won't do.
It is a bit different with regards to things like a virtual saxophone but not because it can't be used as a decent simulation of the real thing.
(partial) virtualisations of existing instruments may be incomplete, but also offer many possibilities that the real thing wont offer. Ever tried playing a guitar like it was a saxophone?
The combination of a good keyboard and mouthpiece open very cool possibilities for expression with synthesizers, but beyond that I have seldom seen such things being used on stage, when you want a saxophone, that big shiny instrument looks so much better... and when you are recording a studio track and have someone who can play such an instrument, you are quite likely to have the 'real' instrument at hand anyway.
This changes with hard to transport things like organs and pianos, which is why you will find those used as replacement of the real thing quite a bit I think.
Trying to replace those with easer to transport alternatives has been a quest for a few generations (human generations..) now.
So well, virtualisations of other instruments tend to turn into instruments in their own right often simply because that is about the only interesting aspect of them, they allow you to do things to the instrument that would not be possible in the physical world.
> Digital recording and playback doesn't magically remove the "noise" in the music. Whether the reproduction is analog or digital, if it's accurate at reconstructing the original music it will reproduce the original noise.
It will not automagically recreate the typical noise, distortion and coloring of the PA system that was used during a live performance, it won't reproduce the typical coloring of the studio monitors.
The recording you get is incomplete in the sense that you do not get a recording of what it sounded like while the recording was mastered, you get (a copy of) the source without the listenign equipment.
> What are you getting at?
I don't know if this is what the parent meant, ut in my original post I was suggesting there is a difference between the kidn of soudn equipment used for amplification of a live performance, and the monitor system in a studio for example, and of course the acoustics of the room/hall/whatever is an entirely seperate story still.
You want to 'recreate' the typical limitations and coloring of such a system if you want something to sound very close to how the original recording was intended to sound. (and yeah, you would want some comparable acoustics as well)
There is simply a lot more to reproducing soudn then seeing no or very little distortion on a scope connected to the output of an amp, or with nearby mics when lookign at speakers. You can say a lot about the theoretical performance of such components, but the interaction between speakers, cable, amplifier and acoustics make that it is not easy to predict how music will sound based on such measurements.
It becomes a lot clearer when you go try to 'measure' what a listener would hear, but even in that case there is the simpel problem that a microphone and the human ear register sound in soemwhat different ways (ear has a very nice physical highpass filter that prevents certain types of interference that would occur when registering the same sounds with a microphone for example)
I'm not trying to say that audio is voodoo magic, but that there are many more factors that play a role then most people assume, and while you can strive for a system that produces the most accurate sound from a given source, that still will not get you any guarantee that you are hearing things as they were intended.
> I could question how complete they could be, given how minor differences in the design of a piano can change the sound,
Oh, you are right there, the simulations are incomplete except for the most basic cases.
They do consider shape and material tho, but it is always a model and not reality that is used for the simulation, so it will never be perfect.. how close you get? it is good enough for a Jazz pianist to make use of it for fun things as having strings resonate on notes being played by just keeping the key open for example, but it is far from good enough to simulate all subtle details of a good concert piano.
> but it's only a start down the road to being able to genuinely simulating real instruments with digital ones: the piano has been analysed and studied and simulated with synthesizers to a far greater degree, I doubt that there's a "digital trumpet" or "digital violin" with anything like the range of the real thing.
Digital trumpet and sax (or better said, digital wind and reed instruments) exist, including 'real' mouthpieces for playing..
The fun thing is that those are more instruments in their own right then good simulations of the real thing.
> The point is, whether the recording session was analog or digital is less important than whether the instruments are physical, analog synthesizers, or digital synthesizers... and whether the playback is analog or digital is even less relevant.
I disagree.
Whatever the instruments are is an artistic choice, and has nothign to do whatsoever with being digital or analog.
Being able to capture and later reproduce what is being played depends first of all on the hardware and technology used for recording.
Having digital instruments may matter when doing digital recording, resampling can introduce nasty artifacts.
Your playback equipment matters mostly in the sense of it havign to be representative for similar equipment used while mastering or during the live performance, while also having to fit within your requirements for price, space and in the end, taste.
> When you start talking about taking that, sampling or resampling it to AAC on an iPod, and then playing it back through a tube amplifier instead of a solid state amplifier so you can "think about the music, [not] just listen"... there's a breakdown in thinking that has nothing to do with the music.
Oh, there I agree, and so much happened to the music that faithfull reproduction is no longer really what it is about.
The issue is that quality and kidn of playback equipment has to be 'good enough' for the purpose, which isn't exactly the same as 'best there is'.
It makes no sense to have ultra high range audio equipment in your car, capable of 120db dynamic range etc, when the dynamic range in your car is severely limited due to for example noise fromn trafic and your tires and engine.
The same applies for portable music normally, but I do at times use a portable mp3 player while laying in bed, and since my bedroom is extremely quiet, all of the sudden it is going to matter a lot which headphones I connect to it, and actually, using a big amp and some real headphones will do wonders in that case... whereas they wont matter much at all out on the street.
> Yet many electronic musicians will employ a compressor on a drum beat and the result can be a more lively and interesting sound, depending on the characteristics and tweaking of the compressor.
;)
Yeah, quite true. Same applies for a bass guitar and similar instruments. (oh, and try an electric guitar without compression for some real fun
THey are usually a piece of hardware... but I bet there is software aroudn that does this as well.
I use the parametric equilizers of my audio mixer, which do the job very well for me (4 bands, all with adhustable center and range), and never really looked for a software soklution.
That said, my Boss sx700 has a digital arametric equilizer, so I'm sure there are software implementations, just never looked for one on my computer.
Try a parametric equilizer instead..
A graphic equilizer is easier to setup maybe because it gives you a 'graphic' view of what you are doing..
A parametric equilizer however produces a lot less phase shifting and similar problems because it only works for the frequency ranges where you actually need to change the sound.
You are ignoring one very important difference between solid state and tube based amps.
Solid state amps rely on feedback from speakers/cable etc to work without distortion, and as a result, depend a lot on what speakers you connect to it to get anywhere near that desirable ultra low distortion.
A tube amp doesn't rely on this, and willd eliver a constant quality regardless of what you connect to it.
As a result you may find that a tube amp does a lot better at driving crappy speakers then your solid state amp.
> is the skilled architect of a concert hall to be seen as better than the engineer who masterfully uses compression, eq, and other techniques?
The dymanics of a hall actually make the sound more proounced, compression etc makes it fit within the dynamic range of some emdium. Yes, those 2 are entirely different things.
You are right that both the architect and the sound tech aim at giving an as good as possible listening experience, but do so within completelyd ifferent sets of limitations.
> the interference patterns between the strings and in the body of the piano are going to be different.
Which has been known for a long time, and there are digital simulators of this particular aspect (they have een around for at least half a decade also)
Nonetheless, you have a very good point, an instrument is an instrument and sounds as it sounds. Its upto the mucisian to decide if it is what (s)he wants, and that is totally subjective and irrelevant when lookign at quality of reproduction, it is only relevant for the instrument and recording.
Reproduction should be as accurate as possible, ut there is a whole set of problems with that.
Are you listening to a live recordign that has been made of the soundboard on soem live gig?
Then you betetr play it back through comparable equipment and with comparable acoustics to get anythign near what was there, and even then you lose a lot of dynamics in the process (mostly during recording)
Are you listening to some studio recording? In that case there is no 'realistic reproduction' unless you are going to try to recreate the sound as it was on the studio monitors.
For those 2 cases you would already end up with different requirements for reproduction, and requirements which might actually be conflicting.
It seems that this all doesn't matter as logn as you are not overloadign the amp.
A 'modern' tube amp can be as colorless as a transistor amp, mosfet amps can approach the distortion as given by an overloaded tube ampetc etc..
Where it really differs, and where tubes beat the crap out of any transistor/mosfet amp is in that they usually do not rely on the feedback from the speakers/cable to work well, whereas a transistor based amp does. This affects distortion of the amp quite a bit.
> which I'm pretty sure includes (for them) being able to distinguish sound differences from different "power supplies, power cords, interconnect cables, and speaker cables"
A power crd should not matter, but there are very good and scientific reasons why the power supply, inter connect cables and speaker cables do make quite a difference.
That is not to say that you should buy all the audiophile bs, but you should buy good cables for conencting your equipment in general and your speakers in particular because the cable directly affects loss and in some cases interference.
The power supply is simply important because it has to be able to provide the power needs of the amplifier, and the better it is at that, the better the amplifier will perform.
It makes no sense however to overdo things, and spendign a fortune on cables is not needed for example.