On Tue, Apr 05, 2005 at 08:26:45AM -0700, Ralph Giles wrote:> AM radio is lower quality (mono) but I don't know > what the digital equivalent would be.Just a minor nit-pick: AM radio can be stereo. However its use is almost nonexistent. See <http://users.hfx.eastlink.ca/~amstereo/amstereo.htm> for more information.> Telephone is nominally 8 kHz mono > (i.e. really bad) though I think the use of digital voice codecs in the > last 20 years may have improved on this a bit.Telephone lines (POTS) have a frequency range of 300-3400Hz. That means 7kHz mono should be enough, although 8kHz is generous towards the transition bandwidth/roll-off. - Andrew
[ > Telephone is nominally 8 kHz mono (i.e. really bad) though I [ > think the use of digital voice codecs in the last 20 years may [ > have improved on this a bit. [ [ Telephone lines (POTS) have a frequency range of 300-3400Hz. That [ means 7kHz mono should be enough, although 8kHz is generous towards [ the transition bandwidth/roll-off. 7 kHz would require a tighter brick-wall filter than even CD, are you even sure it's possible to go from flat to silent in 100 Hz (between 3400 Hz and 3500 Hz)? Almost all telephone connections are digital. Certainly long distance, and probably local as well. Since that is all sampled at 8 kHz, you'd be reducing the bandwidth even further by sampling at only 7 kHz. In other words, the standard sampling rate for telephone voice is certainly 8 kHz. ... and for grins: [ > AM radio is lower quality (mono) but I don't know what the digital [ > equivalent would be. [ [ Just a minor nit-pick: AM radio can be stereo. However its use is [ almost nonexistent. Sure, but is AM stereo more or less widely used than dual-FM stereo (where the side channel difference signal is FM modulated rather than AM modulated)? Brian Willoughby Sound Consulting
Brian Willoughby wrote:> [ > Telephone is nominally 8 kHz mono (i.e. really bad) though I > [ > think the use of digital voice codecs in the last 20 years may > [ > have improved on this a bit. > [ > [ Telephone lines (POTS) have a frequency range of 300-3400Hz. That > [ means 7kHz mono should be enough, although 8kHz is generous towards > [ the transition bandwidth/roll-off. > > 7 kHz would require a tighter brick-wall filter than even CD, are you even > sure it's possible to go from flat to silent in 100 Hz (between 3400 Hz and > 3500 Hz)? > > Almost all telephone connections are digital. Certainly long distance, and > probably local as well. Since that is all sampled at 8 kHz, you'd be reducing > the bandwidth even further by sampling at only 7 kHz. In other words, the > standard sampling rate for telephone voice is certainly 8 kHz.Yup, it's 8kHz. The channel is theoretically from 0-4kHz (with only 300-3000 being available to the user). Also, telephone quantization is only 8 bits (rather than the typical 16) and the quantization levels are not linearly mapped. 8,000 samples per second, each 8 bits = 64kbps = one "B" (Bearer) Channel in the telephone world. MPEG Audio Layer 3 is optimized for 64kbps because of its ubiquity.
On Wed, 6 Apr 2005, Brian Willoughby wrote:> 7 kHz would require a tighter brick-wall filter than even CD, are you even > sure it's possible to go from flat to silent in 100 Hz (between 3400 Hz and > 3500 Hz)?It's not how many Hz, but how many octaves. As well, "flat" to "silent" is relative, you can probably afford to start rolling off a bit earlier (say, put the filter at 3400 Hz, since filters are spec'd at their -3db points if I recall correctly) and it just has to get "silent enough"--if you're using 8-bit audio the spectrum at 3500 Hz needs be reduced only half as much as it would for 16-bit audio. Still, that is a filter about twice as steep as one used for a CD. On the other hand, oversampling and digital filters can help greatly with this, and with the slower sample rates and having fewer bits to process you can oversample further for the same price.> Almost all telephone connections are digital. Certainly long distance, > and probably local as well.Oh, yes, definitely. You would have to have a very, very old switch to have analogue past your local loop. Last one I saw served only a few thousand people spread across a few villages on the west side of Vancouver Island, and even that was retired in the mid-90s.> In other words, the > standard sampling rate for telephone voice is certainly 8 kHz.That's certainly what it's sampled at when converted (at your ISDN TA, at a multiplexer, or at the switch end of your local loop); 64 Kbps is the standard for a digital channel in telephone equipment. (The reason you sometimes get only 56 Kbps for data is due to encoding and synchronisation issues; with some encoding systems sending long streams of certain values, say a constant stream of zeros, will cause the other end to lose sync.) So have to resample to change that sampling rate. Data compression is probably your better bet. cjs -- Curt Sampson <cjs@cynic.net> +81 90 7737 2974 http://www.NetBSD.org Make up enjoying your city life...produced by BIC CAMERA