michael hurd Posted July 3, 2003 Share Posted July 3, 2003 I have to agree wholeheatedly with artto on this one. Btw, I have some ceramic coated depleted urainium speaker cables for sale... only good for very short straight runs. Hey if you guys like to experiment with esoteric cable designs, why not hook up a variable resistor in series with your cables. Keep twisting the dial until you get the sound you like. Or the smoke you like... Quote Link to comment Share on other sites More sharing options...
Marvel Posted July 4, 2003 Share Posted July 4, 2003 Okay, I have made serious blunders here before (Ray might remember a long time ago), but I have two different charts for calculating wavelengths and they both give the same results. It is a simple formula really, but I get a lot shorter values than what's posted above. Both give the same results. 100 Hz = 138 in 10kHz = 1.38 in So where do the miles come in? added: I may have just answered my own question. This is the wavelength of these frequencies in air at 70 degrees C (one chart says F) at sea level. Marvel Quote Link to comment Share on other sites More sharing options...
Griffinator Posted July 4, 2003 Share Posted July 4, 2003 ---------------- On 7/4/2003 9:49:24 AM Marvel wrote: Okay, I have made serious blunders here before (Ray might remember a long time ago), but I have two different charts for calculating wavelengths and they both give the same results. It is a simple formula really, but I get a lot shorter values than what's posted above. Both give the same results. 100 Hz = 138 in 10kHz = 1.38 in So where do the miles come in? Marvel ---------------- The chart you're using, I suspect, is actually for sonic wavelengths in the air, not their electrical equivalent along a wire path. Quote Link to comment Share on other sites More sharing options...
Marvel Posted July 4, 2003 Share Posted July 4, 2003 Griff, Yup, I figured that out as you were posting. However, how does one reconcile the fact that all of the wavelength calculators are (mostly) concerning radio waves, not being tranmitted through wire, but through air? Then using these calculations for the design/size of an antenna to match the wavelength of the freq. to be "captured" and demodulated into something we can hear, i.e., an FM stations on 88.1 Mhz has an 88.1 Mhz carrier, to modulate the audio being broadcast. Our receivers demodulate that, stripping out the 88.1 Mhz freq, leaving us with he audio we want. I guess this gives the answer to why speaker manufacturers move the individual components of a cabinet to time align them instead of hocus pocus wiring schemes. Audio freq. in air are easier to adjust time wise than while going through the copper. Still, you move your head two inches and it might not matter much. Marvel Quote Link to comment Share on other sites More sharing options...
Griffinator Posted July 4, 2003 Share Posted July 4, 2003 I'm trying to upload a little experiment. I'll let you know when I sort it out... Quote Link to comment Share on other sites More sharing options...
Griffinator Posted July 4, 2003 Share Posted July 4, 2003 ---------------- On 7/3/2003 4:59:08 PM artto wrote: People are still arguing (after 50+ years) about the merits of time alignment of a few milliseconds between speaker drivers and now there are people who think they can hear 1/10th to 1/100 of that? ---------------- I don't know about .5 or .05ms, but I can certainly hear 5ms time delays. It's called phase shift and phase cancellation, and it's extremely audible. I deal with it constantly when micing sources. You put one microphone directly in front of a source, then put a second (identical) one two feet away. Record source. Listen to playback of source, one mic in the left, one mic in the right. It sounds all kinds of screwed up. I can demonstrate the audibility of a 5ms delay all day long to you - even to your ears, on your equipment. I'm going to experiment a bit with smaller delays (the joys of software!) and see exactly how tight I can get before I can no longer discern between source and delayed source. I'll get back with you on this one - maybe I can do a little demonstrating here on the forum... Quote Link to comment Share on other sites More sharing options...
Ray Garrison Posted July 7, 2003 Share Posted July 7, 2003 toddc, I was trying to keep it even simpler than that. Looking at this without going into any analysis about whether they are measuring something that can affect the sound or not, or whether the differences are due to the actual digital signal processing itself or to something much less esoteric (maybe the TOSLINK receiver was just a better design that the coax receiver, or maybe the coax link allowed more noise contamination, or whatever), I see this: We are measuring the signal in the analog domain with a jitter analyzer. We only change the cable and leave everything else alone. The image on the scope looks different. We change the cable back to the original cable. The image looks the way it did before we swapped cables. Ergo, changing the cable changed something, somehow. Don't necessarily know what, or how, or whether it's going to be audible, but something changed. End of point I was trying to make. As to exactly what they are measuring, and why, that'd reguire a rather lengthy answer. If you go to the Stereophile website and poke around in their archives searching for jitter analysis or some such thing, you'll find quite a bit of ink spilled on the topic of whether jitter affects sound, if so how, how much and what kind is most offensive, how to measure it, what causes it, that kind of stuff. I suppose you could replace looking at the scope with capturing the two different digital data streams to recordable CD, but I think that introduces more variables into the setup, and you'd still have to use basically the same setup to see what it was that you had recorded onto the CD. Quote Link to comment Share on other sites More sharing options...
toddc Posted July 7, 2003 Share Posted July 7, 2003 Ray, I was being a bit sarcastic, although it was directed at the topic not at you. It's not clear from this thread if you're discussing the impact the cable has on a Digital signal or an Analog signal. I'll concede they are very different and my comment was related specifically to the digital signal. If I read you correctly it sounds like youre discussing using an analog measuring device to look at the transmission of a digital signal, yes? My point was that if youre sending a digital signal you can very simply compare it to the original. An exact copy means success. If any 1s and 0s are missing then you have failure. Completely objective and quantifiable. Whatever else is happening in the cable is irrelevant. To use an analogy, if I'm typing right now on a DSL connection I'm sending a digital signal to the internet over my phone line. If my wife starts talking on the phone at the same time then a meter will measure more "stuff" on the phone line. There are several things happening in the wire that can be measured by an analog meter, but the digital transmission of my typing isn't warmed up by her pleasant voice talking on the same wire. And, the typos are still all MY fault. Similarly, a digital signal sent over a high-end digital interconnect will sound exactly identical to one sent over a cheap digital interconnect -- IF all of the 1's and 0's are getting through both cables. Usually, even cheapest cables meet this bar. To take this to the strongest point, the quality of ALL components prior to the DAC are irrelevant if all of the 1s and 0s are making it from the disc to the DAC. This is trivial with even the cheapest components today. Take the cheapest CD drive and plug its *digital output* into your high-end DAC using a $1 cable and the sound will be identical to a $2000 CD player sending its *digital signal* into the same high-end DAC over a $500 interconnect. Talking about anything else that can be measured on the cable in this digital signal scenario is deceptive and irrelevant, isnt it? A digital signal is just data. Quote Link to comment Share on other sites More sharing options...
Marvel Posted July 7, 2003 Share Posted July 7, 2003 Mark, You asked how a two terminal device can have signals traveling at different speeds down the cable. My guess, without reading all the PR that Monster puts out is that it is well known that AC travels on the outside of the conductor. Thats why high voltage transmission lines can use hollow cable. If the voltage doesn't travel down the middle, make it hollow. It costs less and is lighter. (Of course that is still 60 Hz, so your bass frequencies are still on the outside of the cable.) Maybe the higher freq. goes faster on smaller gauge wire. I'll add to this discussion with an interesting read by Tom Daley, designer of the unity horn. He says that on a single direct radiator, a 50Hz signal leaves the speaker much later than a 100Hz signal (assume both recorded together) this would mean you would want the higher freq. delayed a LOT more than what this wire would do. He also says a horn doesn't exhibit this as much as a direct radiator (fini probably has a direct radiator in his '63 Volvo). Here is the quote, can't remember just where I got it: 99% of the people do not realize that when a point source has flat response, its mid band acoustic phase lags behind the input signal by about -90 degrees (once all fixed time delays are accounted for). This broad band lag is equal to a time delay who's amount increases with decreasing frequency. This -90 degree operation is how some of the simpler measuring systems "determine" acoustic phase, it is a Hilbert transform of the amplitude and at low frequencies for a simple piston, this is a safe assumption. Consider how a normal "perfect" speaker spreads out a signal in time. Make an imaginary signal that has equal amplitude content from 100 Hz to 25 Hz, a specific waveshape which has this property. Take an imaginary perfect flat response speaker who's upper and lower cutoffs are way past our needed bandwidth. This mass controlled "flat" response speaker has a -90 degree lag or delay, at 100 Hz the phase shift is equal to a source 2.83 feet behind the speaker cone, at 50 Hz, the delay is equal to 5.66 feet, at 25 Hz is equal to 11.32 feet and so on. This test signal's wave shape defines the input "time" of each frequency component. When reproduced, the highest frequency component at 100 Hz emerges from the radiator 2.5 ms AFTER the signal arrived at the driver terminals. At 50 Hz, this component emerges 5 ms AFTER the signal hit the terminals and at 25 Hz, the signal emerges after 10 ms and so on. With the driver spreading the signals frequency components out in time, it is simply not possible to retain the same waveshape as the input signal, lower frequencies arrive progressively later in time than the original signal.. Any signal reproduced is done so with the spectrum rearranged in time by the drivers acoustic phase response. If one had a driver which had a very strong motor or a normal motor but very low moving mass, one gets an "over damped" response. This term is from filter design meaning that it is not optimally flat, excessively damped, rolling off too soon and gradually Should the slope of the response reach 6 dB per octave, the driver is operating in the Velocity controlled mode, while the response is not flat, the acoustic phase DOES track the input signal (zero degrees) and the different frequency components are not spread out in time. The waveshape of the input signal is more closely replicated as the frequency components are in the original "time" although the amplitudes are off 6 dB/oct. Each 3 dB /oct change in the slope produces a 45 degree change in phase. An over damped response more closely retains the time information where a flat amplitude response cannot. A proper LF horn can have flat acoustic response AND roughly zero degree acoustic phase. For a person more sensitive to "time errors", they will likely find an over damped system more realistic. For a person more sensitive to "amplitude errors" the traditional "flat response" system will be more satisfying. For the person lucky enough to have heard a proper lf horn system, you have heard that one can have "lightning fast" sounding bass and still make your pant legs flap. A normal "flat" response point source speaker HAS this kind of delay built right in and it is unavoidable (currently). All conventionally driven point source speakers MUST have the phase shift / delay if they are to have flat frequency response (dictated by the falling velocity, acceleration controlled response needed to offset the changing radiation resistance with frequency). Hope you enjoyed it, Marvel Quote Link to comment Share on other sites More sharing options...
reel 2 reel Posted July 7, 2003 Share Posted July 7, 2003 I guess i'll halfta brush up on my trig ,and read that again..... Quote Link to comment Share on other sites More sharing options...
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