Wind isn't reliable, but paired with a thermal or hydro station that can do reasonably quick changes to load
Unfortunately that means that you you cannot save a single kW on conventional production capacity for every MW of installed wind power. Mind you, a significant part of the price you pay for electricity is because of the investments in infrastructure. A better way to deal with it is making special arrangments with energy-intensive industry to let them adjust their power consumption from minute to minute to keep the total current draw constant. Of course in exchange for a highly reduced price per kWh.
Algae are an extreme pain in the arse to grow. They require loads of sun, loads of CO2
Tell this to anyone who has an aquarium, artifical pond in the garden, or swimming pool. I'd say it is an extreme pain in the arse to prevent algae from growing in any water that is exposed to light. I have seen them grow in cooling-water hoses that were only exposed to fluorescent light tubes. Mind you, the cooling-water circuit was filled with deionized water, so all the nutrients must have been leaking out of the various stainless steel, brass and plastic parts.
This post is without karma bonus and let's see what happens to a link to the slashdot article on Wikipedia. I would expect that there is a nofollow tag, even if it this post is moderated up.
As far as I know nofollow is already related to karma. This post is posted with karma-bonus and let's see what happens to a relevant link to an explanation of the religious type of karma. For comparison I will reply to myself without a karma bonus.
One thing I keep meaning to try is a pair of -0.5 or -1.0 Dioptre eyeglasses to move the focal plane of my vision back a few feet when using a screen or reading. I reckon this could reduce the effects of short distance fixation for long periods, reduce eyestrain, and maybe even help restore my once-perfect distance vision.
I think you mean +0.5 or +1.0 diopter. Those are called reading glasses and youc an buy them for almost nothing.
What this also allows is for use of the plugs as perfect-fitting in ear headphones- they are used on stage by pro musicians as a substitute for stage monitors.
Creative idea, but I suspect that the frequency response of the result will be less than optimal. A good headphone has actually a very nonflat frequency response, since the air space from eardrum to speaker membrane acts as a resonator the effects of which the headphone should be compensated for. Putting a large object in there with a narrow air channel would likely affect the sound that you hear.
Windows is still installed, and paid for, and Firefox is no threat to Windows. Firefox drives 0 users away from Windows.
The reason MS would like to control the browser market is that browsers could be used to run platform-independent applications, i.e. applications for which you don't need Windows.
I now carry earplugs with me everywhere and end up using them several times a day. That sucks, because it's impossible to find earplugs with a nice flat frequency response, so you end up with badly colored audio.
Visit an audiologist and ask him to make you a pair of musician's earplugs. They will be molded to the shape of your ear and they contain a filter with a flat frequency response. Be prepared to shell out $200 or so.
What you want is a sinewave sweep from 10Hz to about 18KHz over a period of about 60 seconds.
In addition to the problem of the frequency response of your playback device as the sibling said, you have to normalize it for the 'healthy' frequency response curve of a human, i.e. the dB(A) sensitivity curve. At 0 dB(A) (the hearing treshold), the correction factors are:
50 Hz: +37 dB
100 Hz: +20 dB
500 Hz: +2 dB
1 kHz: 0 dB
3 kHz: -8 dB
8 kHz: 0 dB
10 kHz: +10 dB
18 kHz: +14 dB
Negative numbers mean less sensitivity, positive numbers mean increased sensitivity. Most lower-end mass-market loudspeakers give way too much sound at 100-200 Hz as a cheap way to give the impression of 'strong bases' without addressing the difficult range of 20-100 Hz, so that may compensate a bit for the ear sensitivity.:-)
the areas I work in here in southcentral Alaska were covered by an ice sheet 1,000 feet thick just 9,000 years ago, but 65 million or so years ago it was hot and humid,
And where was current Alaska back then? Continental drift is a few cm per year, so it could have been several 1000 km away from where it is now.
what does "same" really mean? Since they describe this entity with different characteristics, what is the "sameness" referring to?
It means that the believers acknowledge this fact. It's why Christians have a bible with the Old (jewish) and New (christian) Testament. Muslims will point out that there are references in the Bible to a prophet that turns out to be Mohammed. Because of that, muslims are allowed to marry a christian or jewish believer.
Why is not Zeus also the same god who just happens to be endowed with other characteristics by his prophets?
Why is not English actually Chinese spoken with a strong accent? There is no historical connection between them. The Greek/Roman gods were supposed to be human-like beings who happened to be immortal.
Faith in Christianity, Judaism and Islam stems from [...] All claim there's is the one true god.If you believe in one, that means two are wrong - and those two have exactly as much to support them as yours.
It IS the same god they are talking about, silly. Jesus and Mohammed were just prophets that had different views about how to live one's life in order to please that god.
that Goooooogle users actually know what they are looking for when they search on the Internet, and so use Goooogle, and people that don't know what they are looking for are happy with whatever Yahoo! and others dish up?
This is probably an important factor (although Yahoo doesn't seem to be too bad, it is just not being updated with fresh content as aggressively as Google). However, what's more interesting is how this shows how reluctant or ignorant people are to change things. It can be argued that OSS often is less polished than the commercial equivalents and that the time to get used to it is a barrier against change. But Google is better, both in search results and in user-friendliness, than for example MS search. Even with zero investment and a large potential improvement, people don't change. Why?
The nice thing is that you can put the amplifier very close to the signal source such that the weak signal doesn't have to travel any long distance over which it can pick up RF interference.
I'm not sure whether 24-bit ADCs really exist. Sure, professional audio recordings are done with 24 bits, but I doubt that you'll have more than 18 bits effective resolution in the ADC, and even then there aren't any microphones that can do much more than 85 dB SNR (14 bits) if you don't put them inside a trumpet or so. But then, I never really dived into this since all lab applications that I've seen have much more significant noise sources than the quantization of the ADC. Also, with audio signals you can do a few tricks based on the fact that you only need to sample signals that are guaranteed to contain only frequency components below 48 kHz or so.
you're still going to have to fight the SNR which, if anything, will have decreased (noise added by the amping, which in theory could be largely removed by amping the baseline signal and doing the differential stuff after the amping)
It boils down to the fact that it is much easier to design analog electronics with a good SNR than an ADC with a lot of bits.
The poster is a bit short on the details regarding what s/he considers to be 'good' quality data.
With a 16 bits ADC you might get 14 bits effective resolution at the 10 V input range. The last 2 bits are often mostly noise even if you shortcut the input. Always use differential mode instead of single-ended such that noise from ground loops is eliminated. (It means it does an analog substraction of the signals on two inputs, rather than compare them to the common ground that may be noisy)
If your noise requirements are much higher, then the best thing to do is to amplify the signal you're looking for before it goes into the ADC board. Use lock-in techniques if you can. For example if you're trying to see variations of 0.001 V on top of a 5 V signal, find a way to modulate the 0.001 V signal (e.g.chop the light source at 1000 Hz) and use a lock-in amplifier to measure the oscillating 0.001 V component and amplify it to some value that is easier to send to an ADC.
No, it's not. The first harmonic of 23 kHz is at 46 kHz. Harmonics increase in frequency.
OTOH, many sine wave sources create distortions that are audible at lower frequencies. E.g. a cheap sound card that internally resamples from 44.1 to 48 kHz may create all kinds of new frequency components; even without resampling, cheap DACs could have an inferior reconstruction filter (if you sample a 20 kHz sine wave at 44 kHz, you only have 2.2 points per oscillation period. It requires some heavy signal processing to reconstruct the original sine wave). Even in analog electronics, the amplifiers may make a bit more noise when they are dealing with high amplitude signals.
However in the process of converting air vibrations to and from an electrical signal, something other than what can be measured with any instruments is lost.[...] Anyone listening through an electronic channel will not be able to make out you conversation at all, whereas the person you are talking with will have no problem [...] through binaural recording...sounded fantastic when listened to with earphones
I'm not sure that we actually have a disagreement here. You are basically stating that you lose information when you try to collapse the three dimensional sound field into a monaural channel and much less if you do it with a good binaural recording since then the sound field at the listener's position is exactly reproduced during playback through headphones. My statement is that this physics behind this is well-understood rather than that the ear is a mysterious device that can pick up differences that are impossible to quantify with a measurement. It's just that you have to pick the right type of measurement. (The only thing that is hard is what the brain does with the binaural signal, but it doesn't matter much whether that binaural signal is live or comes from headphones and a binaural recording.)
By the way, the proper way to do a binaural recording is not with condensor mikes inside the ears, at least not if you play back through a normal headphone. The frequency response of a good headphone is corrected for the acoustic response of the pinnae and the ear channel. Normally you make a binaural recording by putting omnidirectional mikes roughly at the place where the headphone will be sitting during playback.
Chances are, both amps use the same type of devices (BJT vs MOSFET) and the same exact amplifier circuit.
In the amplifier challenge, the user is allowed to pick any two commercially available amplifiers to his liking (a $200 versus a $1000 is acceptable). An equalizer will be inserted into the signal chain of one of them (user can pick which one) if they have measurable different frequency responses. The whole point of this contest is to prove that simple electrical measurements are sufficient as a sound-quality indicator.
The electronic parts of the audio chain have always been and still are much superior to the transducers at each end, notably the speakers. [...] same pianist play the piano itself, even the most tin eared person would easily be able to tell the difference.
It indeed has to do with the transducers, but I don't think this is a fair comparison. The main reason the original sounds different from the recording is the fact that it is fundamentally impossible to reproduce the three-dimensional sound field with only two loudspeakers, even if the loudspeakers themselves had been transducing their input voltage into air pressure.
If you record a piano in stereo, you will measure the sound pressure at two points, which includes contributions coming directly from the piano and the acoustics of the environment. If you play back the recording in the same environment, you get the room acoustics for a second time. Apart from that, the way the sound from two loudspeakers reflect from the walls is different from the reflections of a piano standing between those loudspeakers.
What I mean is: this has nothing to do with the human ear being more sophisticated than electronics can measure. If you put two (test) microphones at the listener's position, you would see a different signal depending on whether the recording was playing or the original musician. In theory, it is possible to use DSP techniques to pre-compensate the loudspeaker signal such that the playback signal is identical to the original at the test microphone's position, but shifting the microphones by just 1 cm would completely undo this precompensation.
If you can't tell a 192kBps MP3 from the original, you are either deaf or have horrible speakers.
I did blind experiments with LAME abr with a high-end sound card and Sennheiser headphones and I could not hear the difference from 160 kbps ABR. I must be deaf from having had 12 years of classical music education. 192 kbps is roughly --alt-preset standard which is generally agreed by the lossy sound compression community to be indistinguishable from the original for most (i.e. more than 50%) people.
Sure, two mediocre amplifiers will sound... mediocre.
Why would two mediocre amplifiers sound mediocre in exactly the same way? There is only one way for an amplifier to perform perfectly (output voltage same as input voltage apart from a constant scaling factor). There are infinitely many possibilities to deviate from this.
Quality is very difficult to measure, simply because the ear is a hell of a lot more sophisticated and sensitive to nonlinearities than any man-made instrument.
I would say that the fact that most people can't hear the difference between a 192 kbps mp3 and the original 1.4 Mbps (700 kbps with lossless FLAC compression) proves that the human ear is quite unsophisticated.
Search the web (or Google Groups) for Richard Clark's 10000 dollar car amplifier challenge. If you can reliably hear the difference between two amplifiers that are matched in properties that are trivial to measure, you earn 10000 dollars. (trivial properties are: frequency response, output level, clipping, white noise)
I agree with you that THD is not a very good quantifier. True harmonics (i.e. x Hz sine wave in, 1% at N*x Hz out) are very hard to hear. But if the distortion were a low-amplitude frequency sweep with the same amplitude you can bet that you would hear it.
If the holeys in a mesh are half the size of the average wavelength of the radiation, practically none will get through,
Try wrapping a mobile phone (1800 MHz = 17 cm wavelength) in aluminum foil and just leave a small hole that allows you to look at the signal strength indicator. You will be surprised.
Your argument is only valid (and then only to a certain extent) if both of the following conditions are met:
The incident radiation is (approximately) a plane wave (i.e. the source is many wavelengths away and there are no antennas and such in the neighborhood).
The receiver is at least a few times the wavelength away from the aperture.
Close to the aperture you will still have a significant electric field (it's called the near field). In addition, in the near field of a radiation sources you have a magnetic field component that may penetrate thin layers of aluminum.
With a fully enclosing piece of aluminum foil, without any holes, you would do a better job.
Did you read the disclaimer at the end? Google excludes many results if they're too similar. You can click that link at the end to show all of the 28k+ results if you like, Dit you actually try clicking that link? You won't get any more results.
My pet peeve is the almost unlimited combination of wall wart connectors, polarity, output voltage, output current, etc.
For the connectors, they are actually standardized to some extent: different voltages are supposed to have different connectors such that you can't plug a 12 V power supply into a device that wants 3 V. Unfortunately, the polarity is not standardized.
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Unfortunately that means that you you cannot save a single kW on conventional production capacity for every MW of installed wind power. Mind you, a significant part of the price you pay for electricity is because of the investments in infrastructure. A better way to deal with it is making special arrangments with energy-intensive industry to let them adjust their power consumption from minute to minute to keep the total current draw constant. Of course in exchange for a highly reduced price per kWh.
Tell this to anyone who has an aquarium, artifical pond in the garden, or swimming pool. I'd say it is an extreme pain in the arse to prevent algae from growing in any water that is exposed to light. I have seen them grow in cooling-water hoses that were only exposed to fluorescent light tubes. Mind you, the cooling-water circuit was filled with deionized water, so all the nutrients must have been leaking out of the various stainless steel, brass and plastic parts.
This post is without karma bonus and let's see what happens to a link to the slashdot article on Wikipedia. I would expect that there is a nofollow tag, even if it this post is moderated up.
As far as I know nofollow is already related to karma. This post is posted with karma-bonus and let's see what happens to a relevant link to an explanation of the religious type of karma. For comparison I will reply to myself without a karma bonus.
I think you mean +0.5 or +1.0 diopter. Those are called reading glasses and youc an buy them for almost nothing.
Creative idea, but I suspect that the frequency response of the result will be less than optimal. A good headphone has actually a very nonflat frequency response, since the air space from eardrum to speaker membrane acts as a resonator the effects of which the headphone should be compensated for. Putting a large object in there with a narrow air channel would likely affect the sound that you hear.
The reason MS would like to control the browser market is that browsers could be used to run platform-independent applications, i.e. applications for which you don't need Windows.
Visit an audiologist and ask him to make you a pair of musician's earplugs. They will be molded to the shape of your ear and they contain a filter with a flat frequency response. Be prepared to shell out $200 or so.
In addition to the problem of the frequency response of your playback device as the sibling said, you have to normalize it for the 'healthy' frequency response curve of a human, i.e. the dB(A) sensitivity curve. At 0 dB(A) (the hearing treshold), the correction factors are:
50 Hz: +37 dB
100 Hz: +20 dB
500 Hz: +2 dB
1 kHz: 0 dB
3 kHz: -8 dB
8 kHz: 0 dB
10 kHz: +10 dB
18 kHz: +14 dB
Negative numbers mean less sensitivity, positive numbers mean increased sensitivity. Most lower-end mass-market loudspeakers give way too much sound at 100-200 Hz as a cheap way to give the impression of 'strong bases' without addressing the difficult range of 20-100 Hz, so that may compensate a bit for the ear sensitivity. :-)
And where was current Alaska back then? Continental drift is a few cm per year, so it could have been several 1000 km away from where it is now.
It means that the believers acknowledge this fact. It's why Christians have a bible with the Old (jewish) and New (christian) Testament. Muslims will point out that there are references in the Bible to a prophet that turns out to be Mohammed. Because of that, muslims are allowed to marry a christian or jewish believer. Why is not Zeus also the same god who just happens to be endowed with other characteristics by his prophets?
Why is not English actually Chinese spoken with a strong accent? There is no historical connection between them. The Greek/Roman gods were supposed to be human-like beings who happened to be immortal.
It IS the same god they are talking about, silly. Jesus and Mohammed were just prophets that had different views about how to live one's life in order to please that god.
This is probably an important factor (although Yahoo doesn't seem to be too bad, it is just not being updated with fresh content as aggressively as Google). However, what's more interesting is how this shows how reluctant or ignorant people are to change things. It can be argued that OSS often is less polished than the commercial equivalents and that the time to get used to it is a barrier against change. But Google is better, both in search results and in user-friendliness, than for example MS search. Even with zero investment and a large potential improvement, people don't change. Why?
GSM is basically a stripped-down ISDN (D as in 'digital'). All handshaking is done digitally.
The nice thing is that you can put the amplifier very close to the signal source such that the weak signal doesn't have to travel any long distance over which it can pick up RF interference.
I'm not sure whether 24-bit ADCs really exist. Sure, professional audio recordings are done with 24 bits, but I doubt that you'll have more than 18 bits effective resolution in the ADC, and even then there aren't any microphones that can do much more than 85 dB SNR (14 bits) if you don't put them inside a trumpet or so. But then, I never really dived into this since all lab applications that I've seen have much more significant noise sources than the quantization of the ADC. Also, with audio signals you can do a few tricks based on the fact that you only need to sample signals that are guaranteed to contain only frequency components below 48 kHz or so.
you're still going to have to fight the SNR which, if anything, will have decreased (noise added by the amping, which in theory could be largely removed by amping the baseline signal and doing the differential stuff after the amping)
It boils down to the fact that it is much easier to design analog electronics with a good SNR than an ADC with a lot of bits.
With a 16 bits ADC you might get 14 bits effective resolution at the 10 V input range. The last 2 bits are often mostly noise even if you shortcut the input. Always use differential mode instead of single-ended such that noise from ground loops is eliminated. (It means it does an analog substraction of the signals on two inputs, rather than compare them to the common ground that may be noisy)
If your noise requirements are much higher, then the best thing to do is to amplify the signal you're looking for before it goes into the ADC board. Use lock-in techniques if you can. For example if you're trying to see variations of 0.001 V on top of a 5 V signal, find a way to modulate the 0.001 V signal (e.g.chop the light source at 1000 Hz) and use a lock-in amplifier to measure the oscillating 0.001 V component and amplify it to some value that is easier to send to an ADC.
OTOH, many sine wave sources create distortions that are audible at lower frequencies. E.g. a cheap sound card that internally resamples from 44.1 to 48 kHz may create all kinds of new frequency components; even without resampling, cheap DACs could have an inferior reconstruction filter (if you sample a 20 kHz sine wave at 44 kHz, you only have 2.2 points per oscillation period. It requires some heavy signal processing to reconstruct the original sine wave). Even in analog electronics, the amplifiers may make a bit more noise when they are dealing with high amplitude signals.
I'm not sure that we actually have a disagreement here. You are basically stating that you lose information when you try to collapse the three dimensional sound field into a monaural channel and much less if you do it with a good binaural recording since then the sound field at the listener's position is exactly reproduced during playback through headphones. My statement is that this physics behind this is well-understood rather than that the ear is a mysterious device that can pick up differences that are impossible to quantify with a measurement. It's just that you have to pick the right type of measurement. (The only thing that is hard is what the brain does with the binaural signal, but it doesn't matter much whether that binaural signal is live or comes from headphones and a binaural recording.)
By the way, the proper way to do a binaural recording is not with condensor mikes inside the ears, at least not if you play back through a normal headphone. The frequency response of a good headphone is corrected for the acoustic response of the pinnae and the ear channel. Normally you make a binaural recording by putting omnidirectional mikes roughly at the place where the headphone will be sitting during playback.
In the amplifier challenge, the user is allowed to pick any two commercially available amplifiers to his liking (a $200 versus a $1000 is acceptable). An equalizer will be inserted into the signal chain of one of them (user can pick which one) if they have measurable different frequency responses. The whole point of this contest is to prove that simple electrical measurements are sufficient as a sound-quality indicator.
It indeed has to do with the transducers, but I don't think this is a fair comparison. The main reason the original sounds different from the recording is the fact that it is fundamentally impossible to reproduce the three-dimensional sound field with only two loudspeakers, even if the loudspeakers themselves had been transducing their input voltage into air pressure.
If you record a piano in stereo, you will measure the sound pressure at two points, which includes contributions coming directly from the piano and the acoustics of the environment. If you play back the recording in the same environment, you get the room acoustics for a second time. Apart from that, the way the sound from two loudspeakers reflect from the walls is different from the reflections of a piano standing between those loudspeakers.
What I mean is: this has nothing to do with the human ear being more sophisticated than electronics can measure. If you put two (test) microphones at the listener's position, you would see a different signal depending on whether the recording was playing or the original musician. In theory, it is possible to use DSP techniques to pre-compensate the loudspeaker signal such that the playback signal is identical to the original at the test microphone's position, but shifting the microphones by just 1 cm would completely undo this precompensation.
I did blind experiments with LAME abr with a high-end sound card and Sennheiser headphones and I could not hear the difference from 160 kbps ABR. I must be deaf from having had 12 years of classical music education. 192 kbps is roughly --alt-preset standard which is generally agreed by the lossy sound compression community to be indistinguishable from the original for most (i.e. more than 50%) people.
Sure, two mediocre amplifiers will sound... mediocre.
Why would two mediocre amplifiers sound mediocre in exactly the same way? There is only one way for an amplifier to perform perfectly (output voltage same as input voltage apart from a constant scaling factor). There are infinitely many possibilities to deviate from this.
I would say that the fact that most people can't hear the difference between a 192 kbps mp3 and the original 1.4 Mbps (700 kbps with lossless FLAC compression) proves that the human ear is quite unsophisticated.
Search the web (or Google Groups) for Richard Clark's 10000 dollar car amplifier challenge. If you can reliably hear the difference between two amplifiers that are matched in properties that are trivial to measure, you earn 10000 dollars. (trivial properties are: frequency response, output level, clipping, white noise)
I agree with you that THD is not a very good quantifier. True harmonics (i.e. x Hz sine wave in, 1% at N*x Hz out) are very hard to hear. But if the distortion were a low-amplitude frequency sweep with the same amplitude you can bet that you would hear it.
Try wrapping a mobile phone (1800 MHz = 17 cm wavelength) in aluminum foil and just leave a small hole that allows you to look at the signal strength indicator. You will be surprised.
Your argument is only valid (and then only to a certain extent) if both of the following conditions are met:
- The incident radiation is (approximately) a plane wave (i.e. the source is many wavelengths away and there are no antennas and such in the neighborhood).
- The receiver is at least a few times the wavelength away from the aperture.
Close to the aperture you will still have a significant electric field (it's called the near field). In addition, in the near field of a radiation sources you have a magnetic field component that may penetrate thin layers of aluminum. With a fully enclosing piece of aluminum foil, without any holes, you would do a better job.Did you read the disclaimer at the end? Google excludes many results if they're too similar. You can click that link at the end to show all of the 28k+ results if you like,
Dit you actually try clicking that link? You won't get any more results.
For the connectors, they are actually standardized to some extent: different voltages are supposed to have different connectors such that you can't plug a 12 V power supply into a device that wants 3 V. Unfortunately, the polarity is not standardized.