[asterisk-users] IVR menu sound processing for AMR and GSM + live test available

[Arkadi Shishlov]

Hi!
We are in process of setting up an audio guide that will cover notable places of
our capital Riga, Latvia.
The target audience are tourists that dials a free phone number from a mobile
handset to listen to a 3 minute introduction to historic place.
All audio, 10+ languages are recorder in studio at 44KHz. The audio is stored on
server in A-law 8KHz because we'll be pushing it through E1 line.

I need an advice on improving audio quality taking in account that callers will
use cell phones that are most likely in 3G network coverage. So it means, I
believe, either AMR NB or GSM EFR.
Any hints on pre-filtering and volume normalization techniques that could be
beneficiary in this case?
Currently, a Sony Sound Forge speech preset -10dB is applied to normalize the
volume (AFAIK) and then audio is re-sampled with SOX -t raw -e a-law -r 8000 -c 1
Free and commercial software recommendations are fine.

It would be essential to get your comments (in email or by leaving a voice
message) about sound quality if you could call the menu at
sip:123 at riga.beta.lv (actually, any number at riga.beta.lv)

I'm not a sound processing expert, nor I believe people performing the recording
are IVR professionals. So please keep that in mind.
It looks likes there are some problems with English and Norwegian languages.

We also need a person who speaks Dutch language to perform a ~1h recording.
This is price sensitive assignment because the project itself is close to
non-commercial.

01 through 11 will select the language.
Then 0000 through 0013 will select the place.
You can press # to go up one level, and * will take you to 30sec feedback
recording (only when you're listening to the guide or just right after that).
Available languages are:
01=LV
02=EN
03=RU
04=DE
05=LT
08=NO
09=SE
10=PL
11=SP

not yet ready
06=EE
07=FI
12=FR
13=IT
14=AR
15=CN
16=JP
17=IN