search for: quailti

Ok this is my first post and I will try to keep it short. I have searched everywhere and haven't found an answer to my question I have two Trixbox servers that are connected over the Internet via an IAX2 connection. We are experiencing very poor sound quality. I have tried many different codecs gsm, ilbc, g729, g711 and all seem to have the same problem. (All though g729 seems to work the

Hi everybody, I need to know about sound quailty issues from those who have experience with Tormenta2 PRI Interface. Also how to make it work with new versions of Asterisk and Zaptel. And also suggestion if it is a good idea to switch to some newer card from Sangoma or Digium, or Tormenta should work fine. I have sound breaking many times a day over the trunks. Server is AMD Athlon 2.4 GHz with

hi all, Here is the situation. We are currently upgrading our infrastructure to use LDAP as our authentication mechanism and I intend to also have Samba look at LDAP. I had this working in a test case using 2.2 as a PDC awhile ago. Should I be using 3.0 or 2.2 for this implementation? I realize that 3.0 is still quite alpha-quailty, but I've also read one or two interview with the

Hello, I have a large collection of audio files contains music in mp3 format, due to need to free space of hard disk, I need to reduce their size. It seems opus is the best format for this purpose, in order to have the quality of original mp3 files, currently I use ffmpeg command to convert them to FLAC and then use opusenc, the official opus encoder, to convert FLAC files to opus. By using one

I compiled speex1.2 on my linux box and they used speexenc to transcode an 16khz 16bit PCM mono wav file into speex. I used quailty 4. When I try and decode the file with speexdec back into the wav I get an error unknown nb_cbl request 39 and then a segmentation fault. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Did not send this to speex-dev mailing list by error... -----Urspr?ngliche Nachricht----- Von: "Frank Lorenz" <Frank_wtal at web.de> Gesendet: 14.01.10 14:03:22 An: Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> Betreff: Re: [Speex-dev] Fwd: Re: Fixed Point on wideband-mode: Single Frame loss on 2000 Hz sine causes "freak off" Hi Jean-Marc, The codec mostly

I finally got RC3 built on my home machine (had to sneakernet in a tarball of libcurl to get it to build :-( ) and started testing it out on the range of quality levels, side by side with lame using it's range of vbr quailty modes. Overall, it sounds pretty good, but I haven't really tested a good range yet, as I ended up starting on udial.wav... vorbis handles it ok up to quality 3,

Greetings fellow listers, I have an application where I have approximately 300 files that I playback individually or in blocks to simulate "text-to-speech" in a "less mechanical" voice than normal Allison files provide. These files are presently in GSM format and sound pretty good when I play them on my computer speakers or on my in-house

After playing with the vorbis code for a while and doing tons of hacks and analysis on it, I've found it to perform very poorly with impulse signals. The MDCT seems to cause lots of spreading, and it seems to result in much worse impulse performance then mp3. What is the current plan on handling this? Will a smart quantizer be able to avoid it? I've been looking at various ways of