similar to: added background noise problem?

2 things, first an idea... next a question. QUALITY MEASUREMENT IDEA: I find it difficult to hear 2 voice samples and tell which is nearer the original, especially if the background hiss is slightly different. So what if you actually subtract the post-compression sound from the original and then listen to the DIFFERENCE. If you can't hear any voice except background noise and some hiss from

I'm using 22khz wavs because 16khz doesn't provide the quality i need in the original wavs. I can't get 32khz wavs because that part of the project is finished and not redoable. The .spx's don't have to be as high quality, so i guess i could convert them to 16khz. I was trying 17kbps, ultra-wideband, and VBR.

I'm running a X100P connected to a POTS line and a TDMP400P w/ two FXS daughter cards. Both calling out from one of the FXS phones (internally) or calling my home number (externally) the FXO card starts to freak out. By freak out I mean I can still hear but it sounds like you are underwater, there is an annoying hiss or buzz on the line as well. If I hang up and pick up another house phone

I was doing some testing with RC2 and I noticed that RC2 doesn't encode past 19kHz with this clip (-b256 and -b350). There are no problems with this clip like it was before, but this clip contains signal past 19kHz which is audible as a faint high-frequency hiss - and that hiss is gone in the encoded file since RC2 cuts off at 19kHz. I think that -b256 and -b350 should encode at least up to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi to everybody. First of all I hope this is the right place to discuss such an (nitpicky) issue. I've just been testing the current Opus release and for mere curiosity compared its performance to WMAPro with CD quality music at low bitrates (48 kbit/s). While Opus generally does a very good job, I found one particular example (a high pitched

Hi all, I started testing the RC1 encoder at ftp://sjeng.sourceforge.net/pub/sjeng/oggdrop.exe (based on branch_monty_20010708) On the songs I have tested so far (not much :) I did not hear any stereo issues, but there are some very noticeable problems with the produced files. Songs without much high-end will suddenly have one when encoded. (you'd expect it the other way around) It

Hi, I have speex running as a part of a voice conferencing app. Well, one under development anyway. I'm running VBR at quality 3 and get a "hissy-squelchy" background noise. This is fine, kinda, because the internal microphone in the laptop picks up hiss, the sound of the (actually very quiet) hard drive and generally speaking is of less than exemplary quality. To help

I'm wondering if anyone has seen (heard!) this before. I have a site which has Grandstream Budgetone 100 phones (don't laugh, they weren't my choice and I was quite angry when I heard they'd been installed )-: They have an asterisk box with a TDM400 card in it with 4 FXO ports and 4 lines to the telco (BT, in the UK) It basically works and does what it says it's supposed

I think you misunderstood my quality measurement idea. I mean if you subtract the original and the one after, the LESS voice that is less over or the LESS you can tell when someone is speaking, the better the compression. This is still subjective but I think its easier to tell this way because its easier to tell how much voice is remaining than to tell how much the compressed voice is missing from

Heya all, I'm experiencing some problems with a TDM40P and was wondering if anyone else on this list has similar experiences, or maybe even a possible solution. My setup is a dual PIII-750 with 1 gig of RAM, with an X100P, connected to an analog line to my telco, and a TDM40P with analog phones connected. The TDM-card is not sharing any interrupts, but the X100P is, with the 2 Adaptec SCSI

I'm trying to use Asterisk with a TDM400P and 2 analog lines, but I'm having a hard time getting the kind of audio quality that I'd like. I'm hoping to be able to use SIP phones to make calls through Asterisk and have the same quality as a regular analog phone connected to the PSTN. Are my expectations too high? Is it that once I'm using Asterisk as a gateway to the PSTN, I

Hi: I have a Sangoma A200 card installed in a server with two FXO modules and one FXS module. Analog sets connected to the FXS module have a "squeaky static" -- it's like static mixed with the sound of someone vigorously cleaning a window a few doors down. In other words, it's not a classic static noise, but it is noise, and it's distracting. Remote callers can hear this

Hello folks, Although Jack graciously permitted me to keep tinkering on the Postfish alongside OggFile work ;-) allowing it to sit there in a partially tinkered state continued to be a distraction. So yesterday and today I cleaned up the source repository and put together a prerelease. With this prerelease, I'm putting the project out of mind until I put out a beta of OggFile. Actually,

Hello folks, Although Jack graciously permitted me to keep tinkering on the Postfish alongside OggFile work ;-) allowing it to sit there in a partially tinkered state continued to be a distraction. So yesterday and today I cleaned up the source repository and put together a prerelease. With this prerelease, I'm putting the project out of mind until I put out a beta of OggFile. Actually,

Hi, I have got a hissing/humming noise on my handset that is connected to the S100U USB device.. I was running a CVS version of * that was about 2 to 3 weeks old and had perfect quality.. But that HDD died on me so I reinstalled and used the latest CVS, now I have this sound that is best describes as a similar sound to the noise a cpu fan makes.. Like a high piched hissing noise.. Any one

Hi, I have just built my home Asterisk box into a better PC that became available (still only a P2 350 but it only has to manage 1 analog line).. Anyway I have built it on Fedora Core 1.. I have an X100P and a TDM400P (1 module installed).. These cards were working fine in my older PC that was running my Asterisk at home.. The inbound calls via the X100P to my sip phones are working great..

Hi, After trying to use the prebuilt Windows 1.1.6 binaries I've founded that the DLL version doesn't export the mode variables properly, this could probably easily be fixed. Anyways, as for now I've downloaded the 1.1.6 source and built it myself and I'm linking speex statically to my application. I've written an experimental software which just packs 20 ms frames and sends

I've been tinkering with vgplay (my fp gain function and replaygain support code) as well as looking at FLAC replaygain code, with an eye to combining them into a general gain effect plugin helper, however I noticed that the November 19, 2003 Ogg Traffic talks about postfish, being a sound postprocessor with a noise filter and 'a revolutionary declipping filter', that Monty is

Hi, I had ordered a TDM40B and developers kit a few months ago. I have everything installed and working, with one exception - sound quality. When placing a call it sounds like a very bad cordless phone - lots of hiss / static in the background. This even happens with the dialtone, though it is much worse one the call is connected. This does not occur when the phone is directly connected to the

Hi every one, I have an unfortunate need to release a closed source BeOS media codec for Vorbis, basically I'm using headers under an NDA so I can't release them. (Yeah I know closed source boo hiss). So I have a couple of question about what I need to do for all this to be above board. I've made no changes to the libraries so thats not a problem. As far as I can see as