-----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Kevin P. Fleming Sent: Friday, October 15, 2010 10:25 AM To: [email protected] Subject: Re: [asterisk-users] drop dead fix
On 10/15/2010 08:59 AM, Danny Nicholas wrote: > Hello list, > > I am about to have to dump Asterisk in favor of some other > VOIP/PBX solution; the reason? I have 304 voice prompts recorded as > 22Khz wav format files that sound like crumpling paper whenever I > convert them to the 8Khz wav/gsm format required by Asterisk. I was > considering trying the G.729 codec, but reading through the specs, I see > that the 8Khz conversion is going to dump me into the same pile of > dung. Any body have any suggestions? In addition to all the other comments you've received (including the fact that Asterisk does not "require" GSM format files), keep in mind that *any* product that plays these files over the PSTN is going to have to downsample them to 8KHz and, at a minimum, use G.711 companding. That is what the PSTN uses, so it's not possible to have higher fidelity than that. There were some comments in other replies about your files being 'quiet' (low average volume level)... this won't help your situation at all, because it means that any artifacts caused by resampling and compression/decompression will end up at a relatively high amplitude compared to the original signal (resulting in a low signal-to-noise ratio), and when the listener increases the volume level on their listening device, the noise level will be increased along with it. For these sorts of tasks, you really do want the source material recorded at a fairly high volume level. This appears to be the resolution to my problem - #1. Get my "recording talent" in an isolated environment so I can get "clean, loud" recordings #2. Dump the Audacity and Audiologic steps and just use SOX with the highpass and lowpass filters. Don't know if this will make "acceptable" GSM files, but should help with the WAV ones. Thanks to all who offered suggestions. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
