John Hasler wrote:
Mark Allums wries:In the US, a 56k dialup running at 53k (max allowed by law, and rarely achieved in practice) gives you just about the necessary bandwidth for voice over IP, *and nothing else*.That's 56k downbound. Upbound is 33k max.
Been upgraded to 44k. dialup still sucks though.
Not a voice-over-IP expert, and yes, I *have* heard of Nyquist.Good. Now read Shannon. The number of bits per second you can push through a channel depends on the bandwidth _and_ the signal to noise ratio.
Read him. I am being specific about the approximate reproduction of a "waveform" from digital samples here, not about maximum channel capacity. Two samples per "wave" ensures that all the frequencies are there, but three bits makes a more "accurate" reproduction. At the highest frequencies, the human ear probably can't tell the difference, which is why the sample rate of a standard CD is "only" 48kHz. (44.1 after mastering)
What modulation does Skype use? Some type of delta modulation would be my guess for dialup, but I have no real idea.
None. Skype surely uses some sort of compression, but the modem handles the physical layer using complex multitone modulation schemes.
Again, I am being specific about the encoding of the voice signal into bits, not the type of signal modulation done on the physical channel. Which is a type of angle modulation, specifically a phase encoding whereby each transition encodes three bits.
Delta modulation is good for a DAC with a small sample size. Remember those DOS games where they managed to reproduce a digitized sound out of the PC "beep" speaker? That was a one-bit DAC, and they used delta modulation to make it work.
My problem is, I think, that I don't use the standard terminology very well. Mark Allums