So if they bid $10000 per click, they'd end up paying that
Only if there's another bidder who offers $9999 per click. At least, that's what I understand of the adwords bidding system. If there are no other bidders, you just pay $0.05 per click. And if you bid $100, one competitor $99, and a third one $0.05, then you will pay $99.05, the second one $0.06, and the third one $0.05, or something like that.
Hmm, the knives are made of zirconium oxide. Since it is an electrical insulator, it won't trigger the metal detectors, but zirconium is a heavy element and therefore will show up on the X-ray.
Besides, the metal detectors are mainly sensitive to magnetic metals (iron, steel, nickel in coins). With nonmagnetic metals you need much more to trigger it. My brass belt buckle, metal frame for my glasses, and titanium watch have never set off the metal detector. I suppose you could make a knife out of bronze.
It wasn't all that long ago that the electricity needed just to melt the silicon was more energy than the cell would generate throughout it's entire lifetime (they do degrade over time).
I don't know about how long ago you are talking, but the Energy return on investment varies between a factor 4 and a factor 17 for current solar cells, rather than a number below 1 as you are suggesting.
considering that CDDA can sample any sound that your ears can, and that each level is represents is indistinguishable from other levels by your ears, it's probably pretty close to perfect.
The theoretical dynamic range of CDDA is 92 dB. Considering that people in clubs routinely expose themselves to 110 dB(A) levels, which is 110 dB above the human hearing threshold, you would need an additional 18 dB, or 19 bits rather than the 16 bits of CDDA. It's even more if you consider that thumping basses can have enormous amplitudes even though due to the lower human sensitivity for low frequencies it stays well below the 110 dB(A) level. Hence, it makes some theoretical sense to have 24 bits/44 kHz recordings for the end user. Although I personally would prefer to invest the extra bandwidth into extra channels above the standard 2 stereo channels.
A dog would hear it a bit muffled because of lack of high frequencies, like how we hear AM radio. Every audio DAC has circuitry to make sure that there is very little output above 20 kHz and virtually no output above 22 kHz. But lack of high frequencies is a very different kind of distortion compared to the metallic warbling and ringing you have in low-bitrate MP3s.
Finally, there is the fact that users with accounts have, as of now, a degree of anonymity in the sense that the only information presented to the world is what they choose to. Their IP addresses are exposed only to admins.
Actually, not even regular administrators can do this. There are only some 15 wikipedians on English Wikipedia with checkuser privileges, and they may only use that privilege for very specific circumstances (basically deal with serious abuse).
It does mask (cut off) but it's based on the psychoacoustic modelling
Yes, they are based on psychoacoustic modelling. But I believe that it is mostly a few curves that define the hearing threshold for certain frequencies in the presence of a loud masking tone. The rest is trial and error, with lots of fine-tuning of a zillion parameters in the algorithm while listening to compressed music and asking the golden ears at hydrogenaudio to compare different versions of a codec (at least for the OSS ones). There is no algorithm that will give you the degree of transparency of an encoding as a number that realiably matches the results of double-blind trials.
Regarding generative losses of enoding: masking can for example be done by using the fact that a listener doesn't hear pre-echoes before sharp attacks as long as they don't come earlier than X milliseconds before. The encoder uses this fact to get the bitrate of sharp attacks down. But on the second encoding, the pre-echo might become 2X milliseconds rather than X milliseconds, and be audible.
I've seen reports on hydrogenaudio that codecs such as LAME that use complex psya modelling extensively do a worse job as a source for transcoding than fast high-bitrate codecs that have much simpler algorithms for throwing away information.
So it's entirely possible to make a lossy codec which loses NO information through further encodings (though what would be the purpose?).
I suppose you are talking about transcoding to the same bitrate MP3 with the same psychoacoustic model. That could be useful if you want to get rid of DRM by burning to CD and then re-ripping. But the question is whether transcoding to a lower bitrate or even different codec will give audibly different results from encoding directly from the source.
Yes, there will be a quality diff between #4 and #1, but it'll be the same miniscule PSNR loss as from #1 to #2. So unless you transcode a dozen times or something it won't really hurt you.
Every encoder will generate ringing and other artifacts. Every good encoder tries to put those artifacts just a bit below the hearing threshold according to an algorithm that has been tested extensively with normal music. However, encoders are generally not fine-tuned to deal with the unnatural type of noise that results from another encoding process, resulting in the noise ending up above the hearing threshold after the second time.
You might wish to check some double-blind test results on HydrogenAudio. Short version: reencoding 256 kbps MP3 to 128 kbps MP3 sounds horrible compared to 128 kbps MP3 straight from the lossless source.
I've seen a different solution against theft in a big-box chain store here in Netherlands. The more expensive or theft-prone gadgets are stored in clear plastic boxes that can be opened at the checkout counter in a way similar to the anti-theft tags they use in clothes stores. The plastic boxes are usually pretty scratched, but there's always a demo model, firmly attached to the shelves, that you can touch with your hands.
The store is called Media Markt, and I don't like buying stuff there since they try to make there prices appear low by making the store look like a flee market and there are ghettoblasters everywhere, playing music from different radio stations simultaneously. But the packaging is usually no problem.
Special relativity goes on to say that you can exchange time and spatial coordinates using the Lorentz transform, which preserves the length of the 4D position vector.
But on the other hand, the Lorentz transform has very different properties compared to for example a rotation in 3 dimensions, just because of that minus sign for time in the equations. A pair of points in xyzt space that are outside each other's light cone can never be transformed such that they end up inside each other's light cone. Also, a non-euclidian space geometry can be thought as being a hypersurface embedded in a higher dimensional Euclidian space, but that's something that you can only do for spatial dimensions, not for time dimensions. So in these respects, the time dimension is fundamentally different from the space dimensions.
But you're saying that the wave emitted on the "outside" of the speaker, you get a reflected wave that now has all the energy of the incoming wave and the cancellation wave, right?
No, the outgoing wave carries the same energy as the incoming wave was carrying. As the speaker membrane doesn't move, it doesn't do any mechanical work (force times distance). Of course, there will be some losses because of the resistance of the wires, but as far as the interaction with the sound waves is concerned, there is no energy transfer.
In both cases, you haven't changed the total amount of energy reaching your ear, it's just that some portion of the kinetic energy (sound) that can damage your ear is now thermal energy that won't.
No, that's not what happens. The sound is simply reflected. The speaker membrane acts as a rock-solid wall to the incoming sound waves because the current through the voice coil is just right to prevent the membrane from moving. Or in a different way: behind the speaker, the incoming wave and emitted wave cancel each other out. But on the opposite side of the speaker, away from you ear, a wave is emitted.
The problem is that a headphone typically has a very complicated frequency response resulting from the resonances in de closed volume between the eardrum and the headphone loudspeaker, and the attempts of the headphone designer to compensate for these resonances. (see for example here). The net effect is that the impulse response of the headphone/ear system with respect to electrical signals going into the speaker is about 1.5 ms. That means that even if you have full knowledge of the interaction of the headphone with a particular ear, you need to know what sound wave to cancel 1.5 milliseconds in advance. In this time, the sound can travel about 50 cm, which is obviously more than the 1-2 cm between the headphone speaker and the microphone.
So to make an effective noise-cancellating headphone, you have to compromise on sound quality in order to give it a quicker impulse response. Then you will have to accept that you will never be able to effectively cancel out high frequencies (above 1 kHz or so). Finally, you will still need to build some kind of lowpass filter such that you won't substract the higher frequencies with the wrong phase and thus increase the noise rather than decrease it. With all these constraints, you can be happy if you achieve 10 dB reduction.
Maybe I was a bit too pessimistic about 10 dB reduction. For lower frequencies, like from engines, it can work quite well. But I wouldn't expect miracles when blocking the whitish noise from computer fans.
I totally agree that building your own is the way to go, but most people are (for no good reason) scared or unwilling to do so.
Why? To build your own system you have to know exactly which CPU will work together with which motherboard; you need a stock of those tiny screws for mounting drives and if it doesn't work because of hardware incompatibility, because you zapped a component with a static discharge, or because you just bent a pin the first time you inserted a CPU, you're on your own. There are plenty of shops which will let you configure your own system and take their responsibility for component prices plus $50 assembling costs. At least, here in Netherlands, we have these shops.
Page history does not go "all the way back to the beginning" nor does it "store all edits," at least publicly. This record-keeping only occurs until after a certain number of edits.
I think it's actually that the older versions of the Mediawiki software automatically erased the edit history after a (few) months. That's why the history doesn't go back all the way to the first version for older (pre-2004) articles.
At best, they could only do ECB. To anyone who doesn't realize why this is a serious problem, check out [...]
I'd say the problem with ECB you mention is trivial to circumvent: just xor the key or the data with the block number before encryption.
I wonder about something else: how fast can you encrypt or decrypt? If I run gpg in cipher mode (CAST5 algorithm) on a big gzipped file, it takes about 0.5 second of CPU per 1 MB of data on my P4. My harddisk (nothing special) has a data transfer rate of 30 MB/sec. What kind of encryption hardware would one need to handle such an amount of data while still fitting inside a harddisk case without a lot of power draw?
the sound never actually leaves the soundcard, I suspect that it just stays digital the entire time,
Correct. Unfortunately, most consumer-grade soundcards resample all channels to 48 kHz, which means that the 44.1 kHz data stream will be resampled two times: once from 44.1 to 48, and then from 48 back to 44.1. Although it is in theory possible to do that without change of the data (48 k should contain redundant data), in practice the re-sampling will introduce artifacts. Resampling well is especially computationally intensive if the difference in sample rates is so small.
Anyway, these artifacts will probably be neglegible compared to the compression artifacts caused by the encoder. Especially mp3 (with LAME) does a very bad job dealing with compression artifacts from the previous encoding, even if the original encoding was a high bitrate. See also the hydrogenaudio.org wiki
with HTML you've got all your tags, but many people don't write them correctly. How often do you write a closing P tag? Do you close your IMG tags like you should ()?
That is perfectly valid HTML 4.01 strict, as are unclosed <tr>, <td>, and <li> tags. I write my web pages by hand or generate them with short hand-written perl scripts (that are valid HTML 4.01 transitional) and it is much easier to deal with the raw HTML if all these redundant closing tags aren't there. What also helps is to put all the layout details into CSS. The main reason that I don't use HTML 4.01 strict is that it is much more picky about embedding everything in a div or p.
It seems that nobody in this thread understands the problems with noise cancellation. Of course it is trivial to build an analog circuit that substracts the signal from a small microphone from the signal that goes to the headphone speaker. Unfortunately, that won't work. The problem is that a headphone typically has a very complicated frequency response resulting from the resonances in de closed volume between the eardrum and the headphone loudspeaker, and the attempts of the headphone designer to compensate for these resonances. (see for example here). The net effect is that the impulse response of the headphone/ear system with respect to electrical signals going into the speaker is about 1.5 ms. That means that even if you have full knowledge of the interaction of the headphone with a particular ear, you need to know what sound wave to cancel 1.5 milliseconds in advance. In this time, the sound can travel about 50 cm, which is obviously more than the 1-2 cm between the headphone speaker and the microphone.
So to make an effective noise-cancellating headphone, you have to compromise on sound quality in order to give it a quicker impulse response. Then you will have to accept that you will never be able to effectively cancel out high frequencies (above 1 kHz or so). Finally, you will still need to build some kind of lowpass filter such that you won't substract the higher frequencies with the wrong phase and thus increase the noise rather than decrease it. With all these constraints, you can be happy if you achieve 10 dB reduction.
Gummint could help by switching out traffic light, street lighting, etc. to more efficient LED power.
AFAIK street lights typically use very high efficiency light bulbs, such as low-pressure sodium lamps (200 lumen/watt). Compare that to an incandescent (12 lm/W) or CFL (60 lm/W). LEDs aren't all that efficient. With 30 lm/W they're better than incandescent, but worse than the other alternatives. They are good in applications that need small directional light sources with fast switching times.
I learned that my Stereo system consumes 22W when on "standby" and only about 35W when in use
As someone else said, consumer-grade wattmeters can be unreliable for measuring inductive loads such as the transformer in a stereo set. If it really uses 22 W, it should clearly feel warm to the touch, especially if you cover the ventilation holes for a few minutes.
My question is, if I'm paying for kilo-watt hours and watts are volts * amps, am I paying more when the voltage is higher? If so, is the power co. ripping me off?
In many household appliances, especially those with transformers and motors, the used power is not simply the product of volts and amps, since the current and voltage are not always in phase. A common rotating-disc electricity meter measures the true used power. So, yes, you pay more per amp at a higher voltage. However, most likely, the UPS will draw fewer amps when the line voltage is higher.
When there's enough space, expect the dozens of commentary tracks to include video instead of just audio.
As a movie producer, you will still need to pay a human or a team of humans to do the additional recording and editing. I don't expect the producers to increase their investments in the future uxhd-dvd with a factor 1000 just to fill up the available space.
Thus CD-quality is, if not perfect, then good enough that further improvements are ignorable for most people. CD-quality losslessly-compressed music is around 300MB/hour.
I agree that more than 44 kHz/16 bits doesn't make much sense for music playback, but I could imagine an upgrade from stereo to 40-channel surround sound that enables a full immersion into a three-dimensional sound field. Of course, most people wouldn't want to mount 40 loudspeakers in their living room.:)
Outlook is doing exactly what it needs to do, blocking download of images.
And since what version did Microsoft realize that that is the sane thing to do with email from untrusted sources? I recall having searched in vain for such an option in Outlook just 2-3 years ago for a colleague who didn't want to inform the spammers when he accidentally opened a spam message.
Slashdot Mirror
Only if there's another bidder who offers $9999 per click. At least, that's what I understand of the adwords bidding system. If there are no other bidders, you just pay $0.05 per click. And if you bid $100, one competitor $99, and a third one $0.05, then you will pay $99.05, the second one $0.06, and the third one $0.05, or something like that.
Hmm, the knives are made of zirconium oxide. Since it is an electrical insulator, it won't trigger the metal detectors, but zirconium is a heavy element and therefore will show up on the X-ray.
Besides, the metal detectors are mainly sensitive to magnetic metals (iron, steel, nickel in coins). With nonmagnetic metals you need much more to trigger it. My brass belt buckle, metal frame for my glasses, and titanium watch have never set off the metal detector. I suppose you could make a knife out of bronze.
I don't know about how long ago you are talking, but the Energy return on investment varies between a factor 4 and a factor 17 for current solar cells, rather than a number below 1 as you are suggesting.
The theoretical dynamic range of CDDA is 92 dB. Considering that people in clubs routinely expose themselves to 110 dB(A) levels, which is 110 dB above the human hearing threshold, you would need an additional 18 dB, or 19 bits rather than the 16 bits of CDDA. It's even more if you consider that thumping basses can have enormous amplitudes even though due to the lower human sensitivity for low frequencies it stays well below the 110 dB(A) level. Hence, it makes some theoretical sense to have 24 bits/44 kHz recordings for the end user. Although I personally would prefer to invest the extra bandwidth into extra channels above the standard 2 stereo channels.
A dog would hear it a bit muffled because of lack of high frequencies, like how we hear AM radio. Every audio DAC has circuitry to make sure that there is very little output above 20 kHz and virtually no output above 22 kHz. But lack of high frequencies is a very different kind of distortion compared to the metallic warbling and ringing you have in low-bitrate MP3s.
Yes, they are based on psychoacoustic modelling. But I believe that it is mostly a few curves that define the hearing threshold for certain frequencies in the presence of a loud masking tone. The rest is trial and error, with lots of fine-tuning of a zillion parameters in the algorithm while listening to compressed music and asking the golden ears at hydrogenaudio to compare different versions of a codec (at least for the OSS ones). There is no algorithm that will give you the degree of transparency of an encoding as a number that realiably matches the results of double-blind trials.
Regarding generative losses of enoding: masking can for example be done by using the fact that a listener doesn't hear pre-echoes before sharp attacks as long as they don't come earlier than X milliseconds before. The encoder uses this fact to get the bitrate of sharp attacks down. But on the second encoding, the pre-echo might become 2X milliseconds rather than X milliseconds, and be audible.
I've seen reports on hydrogenaudio that codecs such as LAME that use complex psya modelling extensively do a worse job as a source for transcoding than fast high-bitrate codecs that have much simpler algorithms for throwing away information.
I suppose you are talking about transcoding to the same bitrate MP3 with the same psychoacoustic model. That could be useful if you want to get rid of DRM by burning to CD and then re-ripping. But the question is whether transcoding to a lower bitrate or even different codec will give audibly different results from encoding directly from the source.
Every encoder will generate ringing and other artifacts. Every good encoder tries to put those artifacts just a bit below the hearing threshold according to an algorithm that has been tested extensively with normal music. However, encoders are generally not fine-tuned to deal with the unnatural type of noise that results from another encoding process, resulting in the noise ending up above the hearing threshold after the second time.
You might wish to check some double-blind test results on HydrogenAudio. Short version: reencoding 256 kbps MP3 to 128 kbps MP3 sounds horrible compared to 128 kbps MP3 straight from the lossless source.
I've seen a different solution against theft in a big-box chain store here in Netherlands. The more expensive or theft-prone gadgets are stored in clear plastic boxes that can be opened at the checkout counter in a way similar to the anti-theft tags they use in clothes stores. The plastic boxes are usually pretty scratched, but there's always a demo model, firmly attached to the shelves, that you can touch with your hands.
The store is called Media Markt, and I don't like buying stuff there since they try to make there prices appear low by making the store look like a flee market and there are ghettoblasters everywhere, playing music from different radio stations simultaneously. But the packaging is usually no problem.
But on the other hand, the Lorentz transform has very different properties compared to for example a rotation in 3 dimensions, just because of that minus sign for time in the equations. A pair of points in xyzt space that are outside each other's light cone can never be transformed such that they end up inside each other's light cone. Also, a non-euclidian space geometry can be thought as being a hypersurface embedded in a higher dimensional Euclidian space, but that's something that you can only do for spatial dimensions, not for time dimensions. So in these respects, the time dimension is fundamentally different from the space dimensions.
No, the outgoing wave carries the same energy as the incoming wave was carrying. As the speaker membrane doesn't move, it doesn't do any mechanical work (force times distance). Of course, there will be some losses because of the resistance of the wires, but as far as the interaction with the sound waves is concerned, there is no energy transfer.
No, that's not what happens. The sound is simply reflected. The speaker membrane acts as a rock-solid wall to the incoming sound waves because the current through the voice coil is just right to prevent the membrane from moving. Or in a different way: behind the speaker, the incoming wave and emitted wave cancel each other out. But on the opposite side of the speaker, away from you ear, a wave is emitted.
I'd also like to repeat what I mentioned a few weeks ago:
Maybe I was a bit too pessimistic about 10 dB reduction. For lower frequencies, like from engines, it can work quite well. But I wouldn't expect miracles when blocking the whitish noise from computer fans.
Why? To build your own system you have to know exactly which CPU will work together with which motherboard; you need a stock of those tiny screws for mounting drives and if it doesn't work because of hardware incompatibility, because you zapped a component with a static discharge, or because you just bent a pin the first time you inserted a CPU, you're on your own. There are plenty of shops which will let you configure your own system and take their responsibility for component prices plus $50 assembling costs. At least, here in Netherlands, we have these shops.
I think it's actually that the older versions of the Mediawiki software automatically erased the edit history after a (few) months. That's why the history doesn't go back all the way to the first version for older (pre-2004) articles.
I'd say the problem with ECB you mention is trivial to circumvent: just xor the key or the data with the block number before encryption.
I wonder about something else: how fast can you encrypt or decrypt? If I run gpg in cipher mode (CAST5 algorithm) on a big gzipped file, it takes about 0.5 second of CPU per 1 MB of data on my P4. My harddisk (nothing special) has a data transfer rate of 30 MB/sec. What kind of encryption hardware would one need to handle such an amount of data while still fitting inside a harddisk case without a lot of power draw?
I suppose it is the production capacity of the 99.99999% purity grade silicon they're talking about.
Correct. Unfortunately, most consumer-grade soundcards resample all channels to 48 kHz, which means that the 44.1 kHz data stream will be resampled two times: once from 44.1 to 48, and then from 48 back to 44.1. Although it is in theory possible to do that without change of the data (48 k should contain redundant data), in practice the re-sampling will introduce artifacts. Resampling well is especially computationally intensive if the difference in sample rates is so small.
Anyway, these artifacts will probably be neglegible compared to the compression artifacts caused by the encoder. Especially mp3 (with LAME) does a very bad job dealing with compression artifacts from the previous encoding, even if the original encoding was a high bitrate. See also the hydrogenaudio.org wiki
That is perfectly valid HTML 4.01 strict, as are unclosed <tr>, <td>, and <li> tags. I write my web pages by hand or generate them with short hand-written perl scripts (that are valid HTML 4.01 transitional) and it is much easier to deal with the raw HTML if all these redundant closing tags aren't there. What also helps is to put all the layout details into CSS. The main reason that I don't use HTML 4.01 strict is that it is much more picky about embedding everything in a div or p.
It seems that nobody in this thread understands the problems with noise cancellation. Of course it is trivial to build an analog circuit that substracts the signal from a small microphone from the signal that goes to the headphone speaker. Unfortunately, that won't work. The problem is that a headphone typically has a very complicated frequency response resulting from the resonances in de closed volume between the eardrum and the headphone loudspeaker, and the attempts of the headphone designer to compensate for these resonances. (see for example here). The net effect is that the impulse response of the headphone/ear system with respect to electrical signals going into the speaker is about 1.5 ms. That means that even if you have full knowledge of the interaction of the headphone with a particular ear, you need to know what sound wave to cancel 1.5 milliseconds in advance. In this time, the sound can travel about 50 cm, which is obviously more than the 1-2 cm between the headphone speaker and the microphone.
So to make an effective noise-cancellating headphone, you have to compromise on sound quality in order to give it a quicker impulse response. Then you will have to accept that you will never be able to effectively cancel out high frequencies (above 1 kHz or so). Finally, you will still need to build some kind of lowpass filter such that you won't substract the higher frequencies with the wrong phase and thus increase the noise rather than decrease it. With all these constraints, you can be happy if you achieve 10 dB reduction.
AFAIK street lights typically use very high efficiency light bulbs, such as low-pressure sodium lamps (200 lumen/watt). Compare that to an incandescent (12 lm/W) or CFL (60 lm/W). LEDs aren't all that efficient. With 30 lm/W they're better than incandescent, but worse than the other alternatives. They are good in applications that need small directional light sources with fast switching times.
As someone else said, consumer-grade wattmeters can be unreliable for measuring inductive loads such as the transformer in a stereo set. If it really uses 22 W, it should clearly feel warm to the touch, especially if you cover the ventilation holes for a few minutes.
As a movie producer, you will still need to pay a human or a team of humans to do the additional recording and editing. I don't expect the producers to increase their investments in the future uxhd-dvd with a factor 1000 just to fill up the available space.
I agree that more than 44 kHz/16 bits doesn't make much sense for music playback, but I could imagine an upgrade from stereo to 40-channel surround sound that enables a full immersion into a three-dimensional sound field. Of course, most people wouldn't want to mount 40 loudspeakers in their living room. :)
And since what version did Microsoft realize that that is the sane thing to do with email from untrusted sources? I recall having searched in vain for such an option in Outlook just 2-3 years ago for a colleague who didn't want to inform the spammers when he accidentally opened a spam message.