Then again, my USR Pilot 100 (which I upgraded from 128k to 512k memory) is less than half full, stores all my phone + addresses, and plays chess. It doesn't even have a backlit display, but I use it all the time.
Amazing how the first iteration of a device (before it was renamed Palm Pilot even), which is almost 3 years old, is still very useful. Not many companies make hardware with that kind of useful longevity!
Let me get this straight: Windows2000 doesn't work with a bunch of games. Let's say they don't release a patch. People complain "shitty product because these games don't run and there's no patch yet."
OR...
Windows2000 doesn't work with a bunch of games. The day after its release, a patch is released, allowing those games to work in Windows2000. People complain "shitty product because there's already a patch for it."
Golly.
At least I can look forward to never needing a kernel patch ever again after 2.4.0 is released!
Actually it used to, in an earlier iteration of the site (6-12 months ago, if I remember). At the time, that message said something to the effect of Thanks for checking out our source code! We believe in community participation in making our HTML code better -- it's OPEN SOURCE -- yadda yadda...
That's very paraphrased, but I did find it funny that the statement really did have "OPEN SOURCE" in all caps, like it was specifically targeted to get the attention of the geek community. Eventually, somebody probably told Gore or his webmaster that the concept of "open source HTML" doesn't make much sense.
Still trying to figure out the concept of that site (everything2.com).
Looks like one of those "new way of thinking" sites that I really can't figure out in 3 minutes, at 7:45am, before the coffee has taken effect... guess I better go through the University!
I did read your write-up, and I gotta wonder, what movies got a good review from CAPalert?? I clicked on several and didn't find a single film that's not pure evil. Even Toy Story 2 had its problems... one of the characters told a lie, there was arguing between characters, Barbie danced in swimwear (GASP!!) Even worse, the movie depicted a video game killing, theft, threat with physical violence... how Hollywood can release this rubbish is beyond me. To be fair, Toy Story 2 was only given a yellow light, and actually praised some by the reviewer.
It took a lot more than seven to kill off Saddam Hussein. At one time I had the whole thing memorized (From watching the DVD subtitles, of course) and it went something like:
FUCK SHIT COCK ASS PUSSY TITTIES... uh, cunt, something, something, another synonym for feces, blood-drenched frozen tampon popsicle, something something, BARBARA STRIESAND!!!!
You're missing the entire point of the movie, blatantly. I'd sum it up in one line from the movie's script:
"Remember what the Motion Picture Association of America says: horrific, deplorable violence is OK as long as nobody uses naughty language." (slightly paraphrased)
The movie uses foul language to make a point, one that most conservatives would actually agree with. Calling this movie trash based on the fact that it has foul language would be like calling Schindler's List trash because it has graphic violence, the word "Fuck", and full frontal nudity. All on NBC no less. Schlindler's List uses these elements to display the utter horror that was the holocaust. South Park uses the same elements, but in a bitingly humorous way (it actually has some of the most clever satire you'll ever see, and some of the best musical scoring of any movie in recent memory) but then again, you haven't seen South Park, so how can you possibly pass judgment on it? Based on what others who haven't seen the movie have said?
Well yeah, that's assuming your DAC is doing a nice job of reconstructing the sine wave. If you sample any waveform at 120 samples/sec with a fundamental of 60Hz and that only has harmonic overtones (120,180, etc) the output will always be the same. As far as I know, most DAC's don't do this very well at their frequency limit, in which case the output would be more "connect-the-dots" and you'd have a triangle waveform with some slightly rounded-off tips.
Most audio pros work with the digital audio at 96k and then downsample right before the CD master. On the other hand, my SBLive insists on outputting 48khz, which makes that card useless for digital transfers to/from DAT that wind up on CD. I just do everything at 44.1k (on another soundcard) and I'm happy.
morons who know nothing about signal processing beyond Nyquist-Shannon theory but here goes.
I know what you mean-- those retarded idiots that don't know anything about nanophase solid-state physics beyond the Hall-Petch relationship annoy the crap out of me too. [/sarcasm]
However, you did give a reason why 96kHz audio is preferable (especially during the mixing stage) compared to 44kHz 16-bit.
True... for a speaker to produce a true square wave, it can't have any intertia. For that matter, the cone has to tunnel from A to B without going the distance between for each cycle, requiring a negative energy field, which would probably have a side effect of destroying the planet. However, as soon as the Klipsch Promedia's come off of backorder, I'll have the next best thing.
There are actually upper limits on the SNR of an analog system, too, that result from the effects of thermal noise, but I don't know what they are, off-hand. Also, you never truly have infinite bandwidth.
That's a shame, because that will be a requirement for Windows 2010. The problem with keeping a system thermally noise free (absolute zero) is that as soon as you have a signal, you have heat, and thus noise.
The thing is, I find that comparing devices that can only exist in theory to be rather pointless.
And thus is the difference between theoretical science and engineering! Anybody have a massless, frictionless pulley that I can borrow?
Thank you! I was about to spend a couple hours doing research to back up my claims which I simply and "intuitively" knew to be true. The engineers are correct in that you don't lose any INFORMATION in a signal if you sample at at least twice its frequency. However, that doesn't account for the, uh, "extra" information that's added. When transmitting data across a wire, you don't really need to worry about subharmonic distortion. But as I said in another post... there's more to reproducing the original sound than signal theory suggests. I take it you're a musician?
Let's compare an ideal, perfect AD converter with an ideal, perfect analog recorder. With 16-bit audio, the best S/N ratio you can hope for is 96db, am I correct? I haven't seen the math behind this, I've just read this in several places as being the theoretical S/N ratio for 16-bit digital audio. With an ideal, perfect, doesn't-exist analog system, it's much higher. Like I said, the closest thing is the human ear. If I had my way, all my digital equipment would work @ 24 bit 96kHz. But that's expensive... When I said "analog is better than digital" I was referring to the ideal conditions. Of course, I master digitally on a DAT, because my analog setup would introduce more noise.
Here's what I know about DSP: Korg, Proteus, Tascam. And that the SBLive does sample rate conversions when you don't want it to. At least in the end, there's always beer.
Okay, I understand what you're saying. I haven't had much education in signal theory either. But after some thought, I do know that if you're given a set of sample points, there's one and only one solution for a sine wave that fits those points. I wasn't thinking of this earlier though... I was pretty much assuming that the sample points were just going to be read only as voltage levels on the output. There were some other issues I didn't address that cause loss of sound quality with AD/DA conversion. Some people will do a master on a DAT at 48kHz, and then do a sample rate conversion to 44.1kHz which kinda screws things up. But that's not what this discussion was about.
It says any waveform can be reconstructed perfectly if the sampling rate is twice the highest freq. component in the signal.
It seems to me that if you had (for example) a 100Hz sawtooth wave, you couldn't ever reproduce it digitally. Well, with filters you could, and in essence a speaker cone is a filter, but let's just stick to theory. A sawtooth wave is the sum of an infinite series of harmonics, ie sine wave components. Therefore, you'd have to sample at infinite frequency to reproduce it perfectly. Now, most people can't hear above 15kHz anyway and the realistic upper limit is 20kHz, so generally we're OK with 44.1kHz sampling.
BUT.... let's say you have a low, 60hz sine way. You sample it at 120Hz. If you play it back digitally, at the 120Hz sample rate, do you get a 60hz sine wave? Absolutely not. You get a 60hz base with a bunch of higher harmonics thrown in, because you're playing back a 60hz triangle wave. A triangle wave contains higher frequency components that weren't there in the original recording. This is noise. So, you have to apply a filter on the output that blocks out all frequencies that are over half your sampling rate.
So, *my* interpretation of Nyquist's theorem is that if you sample at twice the frequency of the highest component you care about, you won't lose any information. But my point I was trying to make, before a bunch of engineers jumped on my case, was that the playback waveform has all the original sounds plus some additional unwanted artifacts, which has to be taken care of with filtering. In my mind, that's not a perfect reproduction. Fortunately, in a CD, most of the unwanted noise is well above the human range of perception, although there are other factors at play that can cause reduction in sound quality when recording to CD.
On another note (bad pun), there is a noticeable difference in sound quality between 24bit 96kHz audio vs. 16-bit 44.1kHz. According to Nyquist's theorem and the frequency response range of the human ear, that shouldn't be the case. I suppose signal theory alone doesn't completely account for sound quality.
What universe do you live in? In mine, you can't do that. The best you can do is store a signal that is an analog of the source
Thanks for rephrasing what I already said.
And this quantization noise can easily be less than the noise introduced in going to an analog medium. Easily. What consumer-available analog medium has better SNR than you get with 16-bit quantization?
Did I say consumer-level? I was speaking on principle. Let's see... a near perfect analog system would consist of 1) somebody playing an acoustic instrument, and 2) a human ear. Better SNR than 16-bit quantization, and the only noise in the system is going to be due to bloodflow in the ear (ok, maybe dogs and traffic if you're not in a sound room). Anything else processing the sound in between, digital or analog, is just going to add noise.
Please, learn Nyquist's theorem. Audio is bandlimited. You can reconstruct it perfectly from discrete samples. No "sawtooth wave" at all.
...as long as the highest frequency is less than half the sampling rate. Yes, I know. In actuality, you can't (without filters) recreate a perfect SINE WAVE (that was my point), but you'll get something that's the same frequency. However, anything that's not a sine wave would have harmonics added which you won't hear anyway, and don't need. Now I'm sure some more DSP engineers are going to want to jump in again and smack me around, but I'm speaking purely from a musician's standpoint.
Nyquist's theorem is useful for data transmission. If it is applied to audio, then there's really no reason that a classical music CD should sound worse than actually being there, even when played on the best hardware. But then we'd be getting into psychoacoustics.
Correct me on this if I'm wrong (but do it nicely). If the original sound source has two signal components of 60kHz and 65kHz, there will be a tertiary tone of 5kHz as a result of the other two being superimposed. Is that 5kHz tone sampled successfully with a 44.1kHz sampling rate?
There are some artifacts and noise that even the best DSP's will add, solely due to the nature of converting to digital. Imagine you have a recording with lots of high's, like around 5kHz. What happens when you try to convert a 5kHz waveform to digital? Well if you're sampling at 44.1kHz, it'll take (on average) 8.82 samples to record one cycle of the 5kHz wave. Try to draw a complete crest and trough with only 8 points-- it's pretty jagged! Plus, you really can't have that.82 of a sample on the end... what actually happens is that the actual analog waveform falls out of phase with the sample rate, and then your sample points don't line up exactly on the peaks and valleys most of the time, because 44100 samples per second is not divisible by 5000 cycles per second. When the come back into phase, the tone sounds louder. What you hear off the CD then is not a true 5kHz tone, you hear a 5kHz jagged waveform that's being amplitute modulated, causing new frequencies to appear that weren't in the original waveform.
I agree -- analog is better quality than digital, in theory, because a digital signal is only an approximation of the original analog source. Think about what "analog" means. You store a signal on one medium that is a direct analog of the recorded signal. However, with digital, you're taking repeated samples, and approximating each sample to the nearest quantized level determined by the bit depth. So you lose some quality converting to digital. And then you lose some more when going from digital back to analog, which you HAVE to do with sound or you can't hear it. You can build an purely analog sound system that introduces less noise than digital. The big advantage digital has is the ability to make exact reproductions, with no loss from generation to generation. That doesn't change the fact that you can't make a true 10khz sine wave on a CD (roughly 4 sample points per cycle, and you actually have a sawtooth wave that phases in and out w/ the sample rate). To get the best of both worlds for audio, you need to go digital with a very high sample rate (96kHz) at 24-bit depth. That way you have a much better digital approximation of an analog signal.
Read the article! They're NOT making a black hole, it's just an optics experiment. They want to study the effect of light in a moving medium, where the speed of light in that medium is very slow. They won't even have to "stir" it that fast. The worst that could happen is that, as more photons pile up at the "event horizon", where they will eventually just be absorbed by the medium and converted to heat, it will raise the temperature of the bose-einstein condensate enough to undergo a phase change (back to regular old cold rubidium atoms, I suppose). There may be a dim flash of light during the phase change... just a guess. The whole experiment is about as dangerous as playing with a laser pointer.
Those won't work if your company uses a proxy server for web access. You have to set your browser's proxy setting to anonymizer or proxymate, which means you'll no longer be able to go through your company's proxy server, which kind of defeats the purpose. Right now I have to have my modem dialed in to another ISP, with the addresses I need to go to (for some reason the Surf Nazis are blocking The Onion!) listed in the "do not use proxy for addresses beginning with..." section. They load more slowly but I can get to them.
In my opinion, anything that's sending/recieving the full local pathnames of files is BAD. Why doesn't napster use logical shares like an http server? It's obviously much safer to send something like "//clientname/soundtrack/something.mp3" as opposed to "C:\private\pr0n\kiddiepr0n\other\mp3\soundtrack \something.mp3"
Now that this is being talked about, it shouldn't be long before somebody comes up with a hacked client that peruses through unsuspecting Napster PC's.
Slashdot Mirror
Gee, I wonder if it's time to upgrade yet.
Then again, my USR Pilot 100 (which I upgraded from 128k to 512k memory) is less than half full, stores all my phone + addresses, and plays chess. It doesn't even have a backlit display, but I use it all the time.
Amazing how the first iteration of a device (before it was renamed Palm Pilot even), which is almost 3 years old, is still very useful.
Not many companies make hardware with that kind of useful longevity!
(Devil's Advocate)
Let me get this straight: Windows2000 doesn't work with a bunch of games. Let's say they don't release a patch. People complain "shitty product because these games don't run and there's no patch yet."
OR...
Windows2000 doesn't work with a bunch of games. The day after its release, a patch is released, allowing those games to work in Windows2000. People complain "shitty product because there's already a patch for it."
Golly.
At least I can look forward to never needing a kernel patch ever again after 2.4.0 is released!
Actually it used to, in an earlier iteration of the site (6-12 months ago, if I remember).
At the time, that message said something to the effect of
Thanks for checking out our source code! We believe in community participation in making our HTML code better -- it's OPEN SOURCE -- yadda yadda...
That's very paraphrased, but I did find it funny that the statement really did have "OPEN SOURCE" in all caps, like it was specifically targeted to get the attention of the geek community. Eventually, somebody probably told Gore or his webmaster that the concept of "open source HTML" doesn't make much sense.
Still trying to figure out the concept of that site (everything2.com).
Looks like one of those "new way of thinking" sites that I really can't figure out in 3 minutes, at 7:45am, before the coffee has taken effect... guess I better go through the University!
I did read your write-up, and I gotta wonder, what movies got a good review from CAPalert??
I clicked on several and didn't find a single film that's not pure evil. Even Toy Story 2 had its problems... one of the characters told a lie, there was arguing between characters, Barbie danced in swimwear (GASP!!) Even worse, the movie depicted a video game killing, theft, threat with physical violence... how Hollywood can release this rubbish is beyond me. To be fair, Toy Story 2 was only given a yellow light, and actually praised some by the reviewer.
It took a lot more than seven to kill off Saddam Hussein.
At one time I had the whole thing memorized (From watching the DVD subtitles, of course) and it went something like:
FUCK SHIT COCK ASS PUSSY TITTIES... uh, cunt, something, something, another synonym for feces, blood-drenched frozen tampon popsicle, something something, BARBARA STRIESAND!!!!
You're missing the entire point of the movie, blatantly.
I'd sum it up in one line from the movie's script:
"Remember what the Motion Picture Association of America says: horrific, deplorable violence is OK as long as nobody uses naughty language." (slightly paraphrased)
The movie uses foul language to make a point, one that most conservatives would actually agree with. Calling this movie trash based on the fact that it has foul language would be like calling Schindler's List trash because it has graphic violence, the word "Fuck", and full frontal nudity. All on NBC no less. Schlindler's List uses these elements to display the utter horror that was the holocaust. South Park uses the same elements, but in a bitingly humorous way (it actually has some of the most clever satire you'll ever see, and some of the best musical scoring of any movie in recent memory) but then again, you haven't seen South Park, so how can you possibly pass judgment on it? Based on what others who haven't seen the movie have said?
Actually, it was "donkey raping shit eater" and "shit faced cockmaster."
These are to be used separately from "testicle shitting rectal wart" and "eat penguin shit, you ass spelunker."
Well yeah, that's assuming your DAC is doing a nice job of reconstructing the sine wave. If you sample any waveform at 120 samples/sec with a fundamental of 60Hz and that only has harmonic overtones (120,180, etc) the output will always be the same. As far as I know, most DAC's don't do this very well at their frequency limit, in which case the output would be more "connect-the-dots" and you'd have a triangle waveform with some slightly rounded-off tips.
Most audio pros work with the digital audio at 96k and then downsample right before the CD master. On the other hand, my SBLive insists on outputting 48khz, which makes that card useless for digital transfers to/from DAT that wind up on CD. I just do everything at 44.1k (on another soundcard) and I'm happy.
morons who know nothing about signal processing beyond Nyquist-Shannon theory but here goes.
I know what you mean-- those retarded idiots that don't know anything about nanophase solid-state physics beyond the Hall-Petch relationship annoy the crap out of me too. [/sarcasm]
However, you did give a reason why 96kHz audio is preferable (especially during the mixing stage) compared to 44kHz 16-bit.
True... for a speaker to produce a true square wave, it can't have any intertia. For that matter, the cone has to tunnel from A to B without going the distance between for each cycle, requiring a negative energy field, which would probably have a side effect of destroying the planet.
However, as soon as the Klipsch Promedia's come off of backorder, I'll have the next best thing.
There are actually upper limits on the SNR of an analog system, too, that result from the effects of thermal noise, but I don't know what they are, off-hand. Also, you never truly have infinite bandwidth.
That's a shame, because that will be a requirement for Windows 2010. The problem with keeping a system thermally noise free (absolute zero) is that as soon as you have a signal, you have heat, and thus noise.
The thing is, I find that comparing devices that can only exist in theory to be rather pointless.
And thus is the difference between theoretical science and engineering! Anybody have a massless, frictionless pulley that I can borrow?
Thank you!
I was about to spend a couple hours doing research to back up my claims which I simply and "intuitively" knew to be true.
The engineers are correct in that you don't lose any INFORMATION in a signal if you sample at at least twice its frequency. However, that doesn't account for the, uh, "extra" information that's added. When transmitting data across a wire, you don't really need to worry about subharmonic distortion. But as I said in another post... there's more to reproducing the original sound than signal theory suggests.
I take it you're a musician?
Let's compare an ideal, perfect AD converter with an ideal, perfect analog recorder.
With 16-bit audio, the best S/N ratio you can hope for is 96db, am I correct? I haven't seen the math behind this, I've just read this in several places as being the theoretical S/N ratio for 16-bit digital audio.
With an ideal, perfect, doesn't-exist analog system, it's much higher. Like I said, the closest thing is the human ear.
If I had my way, all my digital equipment would work @ 24 bit 96kHz. But that's expensive...
When I said "analog is better than digital" I was referring to the ideal conditions. Of course, I master digitally on a DAT, because my analog setup would introduce more noise.
man am I bored.
Here's what I know about DSP: Korg, Proteus, Tascam. And that the SBLive does sample rate conversions when you don't want it to.
At least in the end, there's always beer.
I'll stick to quantum mechanics from now on...
Okay, I understand what you're saying. I haven't had much education in signal theory either. But after some thought, I do know that if you're given a set of sample points, there's one and only one solution for a sine wave that fits those points. I wasn't thinking of this earlier though... I was pretty much assuming that the sample points were just going to be read only as voltage levels on the output. There were some other issues I didn't address that cause loss of sound quality with AD/DA conversion. Some people will do a master on a DAT at 48kHz, and then do a sample rate conversion to 44.1kHz which kinda screws things up. But that's not what this discussion was about.
It says any waveform can be reconstructed perfectly if the sampling rate is twice the highest freq. component in the signal.
It seems to me that if you had (for example) a 100Hz sawtooth wave, you couldn't ever reproduce it digitally. Well, with filters you could, and in essence a speaker cone is a filter, but let's just stick to theory.
A sawtooth wave is the sum of an infinite series of harmonics, ie sine wave components. Therefore, you'd have to sample at infinite frequency to reproduce it perfectly. Now, most people can't hear above 15kHz anyway and the realistic upper limit is 20kHz, so generally we're OK with 44.1kHz sampling.
BUT.... let's say you have a low, 60hz sine way. You sample it at 120Hz.
If you play it back digitally, at the 120Hz sample rate, do you get a 60hz sine wave? Absolutely not. You get a 60hz base with a bunch of higher harmonics thrown in, because you're playing back a 60hz triangle wave. A triangle wave contains higher frequency components that weren't there in the original recording. This is noise. So, you have to apply a filter on the output that blocks out all frequencies that are over half your sampling rate.
So, *my* interpretation of Nyquist's theorem is that if you sample at twice the frequency of the highest component you care about, you won't lose any information. But my point I was trying to make, before a bunch of engineers jumped on my case, was that the playback waveform has all the original sounds plus some additional unwanted artifacts, which has to be taken care of with filtering. In my mind, that's not a perfect reproduction. Fortunately, in a CD, most of the unwanted noise is well above the human range of perception, although there are other factors at play that can cause reduction in sound quality when recording to CD.
On another note (bad pun), there is a noticeable difference in sound quality between 24bit 96kHz audio vs. 16-bit 44.1kHz. According to Nyquist's theorem and the frequency response range of the human ear, that shouldn't be the case.
I suppose signal theory alone doesn't completely account for sound quality.
man harmonics
man fourier
And then read my reply to billybob jr.
What universe do you live in? In mine, you can't do that. The best you can do is store a signal that is an analog of the source
Thanks for rephrasing what I already said.
And this quantization noise can easily be less than the noise introduced in going to an analog medium. Easily. What consumer-available analog medium has better SNR than you get with 16-bit quantization?
Did I say consumer-level? I was speaking on principle. Let's see... a near perfect analog system would consist of 1) somebody playing an acoustic instrument, and 2) a human ear. Better SNR than 16-bit quantization, and the only noise in the system is going to be due to bloodflow in the ear (ok, maybe dogs and traffic if you're not in a sound room). Anything else processing the sound in between, digital or analog, is just going to add noise.
Please, learn Nyquist's theorem. Audio is bandlimited. You can reconstruct it perfectly from discrete samples. No "sawtooth wave" at all.
...as long as the highest frequency is less than half the sampling rate. Yes, I know. In actuality, you can't (without filters) recreate a perfect SINE WAVE (that was my point), but you'll get something that's the same frequency. However, anything that's not a sine wave would have harmonics added which you won't hear anyway, and don't need. Now I'm sure some more DSP engineers are going to want to jump in again and smack me around, but I'm speaking purely from a musician's standpoint.
Nyquist's theorem is useful for data transmission. If it is applied to audio, then there's really no reason that a classical music CD should sound worse than actually being there, even when played on the best hardware. But then we'd be getting into psychoacoustics.
Correct me on this if I'm wrong (but do it nicely). If the original sound source has two signal components of 60kHz and 65kHz, there will be a tertiary tone of 5kHz as a result of the other two being superimposed. Is that 5kHz tone sampled successfully with a 44.1kHz sampling rate?
Since a large portion of the slashdot audience knows very little about signal theory... maybe a little "intuitive" analysis would help.
There are some artifacts and noise that even the best DSP's will add, solely due to the nature of converting to digital. .82 of a sample on the end... what actually happens is that the actual analog waveform falls out of phase with the sample rate, and then your sample points don't line up exactly on the peaks and valleys most of the time, because 44100 samples per second is not divisible by 5000 cycles per second. When the come back into phase, the tone sounds louder. What you hear off the CD then is not a true 5kHz tone, you hear a 5kHz jagged waveform that's being amplitute modulated, causing new frequencies to appear that weren't in the original waveform.
Imagine you have a recording with lots of high's, like around 5kHz. What happens when you try to convert a 5kHz waveform to digital?
Well if you're sampling at 44.1kHz, it'll take (on average) 8.82 samples to record one cycle of the 5kHz wave. Try to draw a complete crest and trough with only 8 points-- it's pretty jagged! Plus, you really can't have that
I agree -- analog is better quality than digital, in theory, because a digital signal is only an approximation of the original analog source. Think about what "analog" means. You store a signal on one medium that is a direct analog of the recorded signal.
However, with digital, you're taking repeated samples, and approximating each sample to the nearest quantized level determined by the bit depth. So you lose some quality converting to digital.
And then you lose some more when going from digital back to analog, which you HAVE to do with sound or you can't hear it.
You can build an purely analog sound system that introduces less noise than digital. The big advantage digital has is the ability to make exact reproductions, with no loss from generation to generation.
That doesn't change the fact that you can't make a true 10khz sine wave on a CD (roughly 4 sample points per cycle, and you actually have a sawtooth wave that phases in and out w/ the sample rate).
To get the best of both worlds for audio, you need to go digital with a very high sample rate (96kHz) at 24-bit depth. That way you have a much better digital approximation of an analog signal.
Read the article! They're NOT making a black hole, it's just an optics experiment. They want to study the effect of light in a moving medium, where the speed of light in that medium is very slow. They won't even have to "stir" it that fast. The worst that could happen is that, as more photons pile up at the "event horizon", where they will eventually just be absorbed by the medium and converted to heat, it will raise the temperature of the bose-einstein condensate enough to undergo a phase change (back to regular old cold rubidium atoms, I suppose). There may be a dim flash of light during the phase change... just a guess.
The whole experiment is about as dangerous as playing with a laser pointer.
Those won't work if your company uses a proxy server for web access. You have to set your browser's proxy setting to anonymizer or proxymate, which means you'll no longer be able to go through your company's proxy server, which kind of defeats the purpose.
Right now I have to have my modem dialed in to another ISP, with the addresses I need to go to (for some reason the Surf Nazis are blocking The Onion!) listed in the "do not use proxy for addresses beginning with..." section. They load more slowly but I can get to them.
In my opinion, anything that's sending/recieving the full local pathnames of files is BAD.k \something.mp3"
Why doesn't napster use logical shares like an http server? It's obviously much safer to send something like
"//clientname/soundtrack/something.mp3"
as opposed to
"C:\private\pr0n\kiddiepr0n\other\mp3\soundtrac
Now that this is being talked about, it shouldn't be long before somebody comes up with a hacked client that peruses through unsuspecting Napster PC's.
Obviously! Any time a MS product performs better than the competition in a benchmark, it can ONLY be because MS paid somebody off. Excellent analysis!