Arkadi Shishlov
2010-Apr-08 23:48 UTC
[asterisk-users] IVR menu sound processing for AMR and GSM + live test available
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
Steve Edwards
2010-Apr-09 02:08 UTC
[asterisk-users] IVR menu sound processing for AMR and GSM + live test available
On Fri, 9 Apr 2010, Arkadi Shishlov wrote:> 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 get: -- Executing Dial("SIP/501-0961b3a8", "sip/123 at riga.beta.lv") in new stack -- Called 123 at riga.beta.lv -- Got SIP response 488 "Not acceptable here" back from 213.21.217.130 -- SIP/riga.beta.lv-09623508 is circuit-busy -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000
Arkadi Shishlov
2010-Apr-09 08:42 UTC
[asterisk-users] IVR menu sound processing for AMR and GSM + live test available
On 04/09/10 05:08, Steve Edwards wrote:> On Fri, 9 Apr 2010, Arkadi Shishlov wrote: > >> 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 get: > > -- Executing Dial("SIP/501-0961b3a8", "sip/123 at riga.beta.lv") in new stack > -- Called 123 at riga.beta.lv > -- Got SIP response 488 "Not acceptable here" back from 213.21.217.130 > -- SIP/riga.beta.lv-09623508 is circuit-busyIt was A-law only, on purpose. Just added U-law to the list.