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martes, 17 de marzo de 2015

audio file recomendation

It's best to have files in the native format of the connecting device; that reduces transcoding & resampling. Downsampling a 16kHz file to 8kHz to get it to a G.711 law device, on-the-fly, doesn't produce better results in perceptible audio quality than beginning with an 8kHz file

If you store all of your sound files as 48kHz signed linear files, but your clients were 8kHz a-law clients or 16kHz G.722 clients, then you'd have to down-sample each of those sound files, and downsampling costs CPU cycles

Voice channels on a PRI operate using G.711 u-law or a-law coding; 8kHz.

I don't know why your sound files at something greater than 8kHz aren't playing back properly.

The sampling rate in the file has to match the default sampling rate for the corresponding codec in Asterisk. .wav files are treated as being in the SLIN codec, which is 16 bit samples at 8kHz rate. Asterisk doesn't adapt to the metadata in .wav files.

8kHz is the Nyquist rate for 4kHz bandwidth. That was the carrier spacing on analogue multiplexes, and is based on the fact that most of the frequencies necessary for speech lie between 300Hz an 3.4kHz, giving 4kHz when you allow for reasonably realisable channel filters. For digital systems, it is the anti-aliasing filters than need to be realised. A lot of work has gone into making telephony systems efficient, and any better sound quality just wastes bandwidth.

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