Le lundi 01 mars 2010 à 22:47 -0500, Christian Jaeger a écrit : > Hello > > I'm using Ekiga and Twinkle with the Diamondcard.us service as my main > phone service. People are telling me every now and then that the sound > quality is pretty poor, and in fact when I listen to my voice on the > echo service it's always been quite much worse than when using a > "normal" phone I think. It's regardless of the codec used, all that > are offered by Diamondcard are using 8000 Hz sampling rate. And I'm > using a headset and have checked sound quality locally, which is just > fine, as I can verify for example with (the bfr command is to delay > playback, so you can hear the echo better): > > arecord -t raw -f dat -r 22050 -c 1 | bfr -m 33075 | aplay -t raw -f > dat -r 22050 -c 1 - > > Now 8000 Hz is of course lower, and in fact when using that rate: > > arecord -t raw -f dat -r 8000 -c 1 | bfr -m 12000 | aplay -t raw -f > dat -r 8000 -c 1 - > > it sounds exactly as in the echo test. Now one thing I know is that if > you sample a signal, you need a low pass before the quantization, at > half the sample rate or lower, to remove frequencies that are too high > to be represented in the digitalized data. If the low pass is missing, > the high frequencies are turned to noise instead. So what happens when > a program samples data at a low frequency, will it somehow make the > sound card tune a low pass to the correct cut-off frequency when the > sample rate is changed? Sounds rather unlikely, and in fact the echo > test at 8000 Hz sounds exactly like suffering from that problem: you > can't distinguish "s" from "f" anymore, which may just be unavoidable > with no frequency spectrum above 4000 Hz, but worse, they sound very > loud and noisy, and for example an combined s and vowel, like they > often appear in french (e.g. "bourgeoisie" I think, I'm not a native > french speaker), sound really awful. Now there's a way to remedie > this: low-pass filtering can also be done computationally, so we can > digitize at say DAT quality (48000 Hz) and resample down to 8000 > samples/second programmatically, which is what the following command > line is doing: > > arecord -t raw -f dat -c 1 | bfr -m 36000 | sox -t raw -r 48000 -s -2 > -c 1 - -t raw -r 8000 - resample -ql |bfr -m 6000 | aplay -t raw -f > dat -r 8000 -c 1 - > > It sounds more hollow/muffled now, but much cleaner. No loud noises > anymore for sharp consonants. > > Would doing this in VoIP programs improve perceived sound quality for > the listener? I would expect so, but haven't been able to test. (All I > know is that more often than not I'm using Skype to call people now > (Skype-out service), because people are telling me they hear me > better.) > > Note that you can very well say "you're wrong, we don't need a > low-pass, just use a higher sampling rate then it sounds just fine", > which is of course partially true, with higher sampling rates the > relevance of a low-pass decreases quickly as the spectrum of > frequencies that are higher than half the sample rate becomes narrower > (and an increasing part of the consonant frequency range is digitized > correctly instead of as noise). But given that at least Diamondcard > only offers 8000 Hz, and there may be valid reasons for this (like > maybe the phone system is still using that frequency), what's your > position on this? >
Isn't it waht the "plughw:0,0" special ALSA device is supposed to do by itself ? We rely on it. -- _ Damien Sandras (o- //\ Ekiga Softphone : http://www.ekiga.org/ v_/_ Be IP : http://www.beip.be/ FOSDEM : http://www.fosdem.org/ SIP Phone : sip:dsand...@ekiga.net
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