similar to: udial.wav problem

I tested Vorbis encoder - beta3 version, and here are my thoughts: - In comparison to beta2, subtle high-frequency artifacts seem to be gone (though they were small in beta2). Good job there! :-) - Velvet.wav also sounds better, but transparent quality is reached at -b256+. - Horn.wav still sounds very sucky, mode -b256 gives ~100kbps (this is understandable because in this sample practically

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,

Hello everyone, ff123 compiled Monty's branch of the RC1 encoder, see his post on r3mix.net forum: http://66.96.216.160/cgi-bin/YaBB.pl?board=c&action=display&num=994299736&start=30 Anyway it only supports ~128kbps mode, so I did a quick listening test with some files that bugged vorbis beta4. grace.wav - the right channel is still a bit watery, and I think this can be seen

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

-----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

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 having a problem installing CentOS 8 from a USB drive. When the installer boots from the USB, it displays the language selection screen. After I select English and continue, the installer freezes. The USB drive flashes a couple times over the next minute or so, the stops. The mouse moves the cursor, but the installer is unresponsive to either selecting QUIT or HELP. I've

Using narrow, wideband, and ultra-wideband encoding on a short 16khz wav gave .spx's of 3,789 ... 2,935 ... and 1,875 bytes. Even after reading the manual, smaller files for the higher frequency encoding seems counter-intuitive. My mp3 at 32 kbps on the original 22khz wav is 3,866 with a quality comparable to speex wideband on the converted 16khz wav, so speex is a 24% improvement in size.

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

I couldn't catch Monty on IRC, so I'll ask here... how's RC3 coming along? Can we expect RC3 anytime soon, and what are the main improvements over RC2? -- Vorbis Xtreme | http://solair.eunet.yu/~aldov/ Ogg Vorbis - free, open source audio __________________________________________________ Do You Yahoo!? Listen to your Yahoo! Mail messages from any phone. http://phone.yahoo.com ---

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 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

Comments? ---------------------------------------- "Mark Taylor" <mt@sulaco.org> wrote: [...] > This is related to one minor objection I have to vector quantization > based codecs like Vorbis and the MPEG4 VQ codec: they do not compute > the quantization noise during the encoding process. The choice of > codebooks (use a big codebook: low quantization noise, use a

Check this clip (it's small, 373kb) http://www.geocities.com/jdxss/udialwav.zip It left oggenc, lame and MP+ encoders choking in dust. -- Vorbis Xtreme | http://solair.eunet.yu/~aldov/ Ogg Vorbis is the free, open source alternative to MP3 --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a

Hi. I think that the vorbis team should use this version of OggDrop RC3 as the official OggDrop RC3, because the menu interface is easier to use (& you can see what kind of mode you have chosen). http://www.inf.ufpr.br/~rja00/BQoggdropC1.zip See this thread for more info: http://www.hydrogenaudio.org/forums/showthread.php?s=&postid=4683 <p>I only have one problem now: what

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

In case you missed it, RC2 is on slashdot.org too: http://slashdot.org/article.pl?sid=01/08/13/1811241 BTW Gian-Carlo Pascutto wrote on /. that RC3 should appear very soon (in a week or so)? Also, it looks like Wall Street Journal article is not available on-line (at least to non-registered persons). If someone is registered or have a paper edition, please post this article somewhere so that we

Hi. Since I'm interested in quality, I did some listening tests using Vorbis beta4, mode -b256. I think that ~256kbps, ~5.5:1 compression is still a good tradeoff because I can put 6-7 albums on 1 CD-R (assuming that the average length of an album is ~50 minutes) while retaining excellent quality. My comments: 1.) most songs are transparent but there are still some glitches that appear

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..