[mythtvnz] [OT] audio recording issue

[Stephen Worthington]

On Thu, 04 Jun 2009 19:34:27 +1200, you wrote:

>On Thu, 2009-06-04 at 19:08 +1200, Jason Haar wrote:
>> No - I tried all that. In the end I brought up an Asterisk server,
>> forced only G729 and G726 and still it's sounds awful - in fact I
>> basically don't hear any difference between the different codecs.
>> Fundamental issues like hearing static when I say "s" words is common.
>> Running "file" on the voicemails show they are all 8000Hz, whereas the
>> "asound -f cd" which sounds great is 44KHz - which is what I think the
>> problem is. However, I am not an audio-guru so it's complete guesswork
>> for me.
>
>I'm not surprised, a lot of people find G729 and friends sound less than
>good. Asterisk is narrowband (8 kHz) too, like the traditional phone
>network.

Yes, 8 kHz sampling is too little - sound will be distorted no matter
how good the codec is.  The human voice does normally have frequencies
in it > 4 kHz that matter to voice quality.

>If you're doing end to end VoIP and what is in between supports wideband
>codecs (G722, CELT, some Speex etc.) for example FreeSWITCH then you can
>get great sound. You'll need decent audio hardware too.
>
>If you're using a VoIP provider or going out to the PSTN at all then it
>will be narrowband.

PSTN should be 64 kbit/s, which gives quite good quality with the
codecs normally used for that.  I forget which ones they are.  But
they are now used for all PSTN calls - the old landline phone is
connected to a card at the exchange that digitises the call.  None of
the phone network is analogue now except the local loop.

>Typically people find a hardware SIP phone gives much better call.
>quality

I would have thought that using the same codec as is used in a
hardware SIP phone would give the same sound quality from a PC.

>hads