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
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