similar to: Low bitrate high-band coding...

Hi! Dear all I compressed an audiu(wave) with libvorbis1.1.1 and found that the pre-echo was longer than the one which was compressed with MP3 at the same bitrate(128kbps),I am now confused aboout the pre-echo preventing scheme. Can you tell me what method does Ogg Vorbis adopt besides the auto windowsize switch? Looking forward to your letter! Thank you! ??????? Lisa (^_^)

In case people on the list don't know it, A Vorbis tester (who's email I've lost! If you're on the list, speak up!) is maintaining a vorbis audio quality test page: http://hem.passagen.se/ingets1/vorbis.htm He's doing us the great service of hunting down samples that break Vorbis (so we don't have to :-) In any case, it turns out that the trumpets in the dr4 sample

The Italian Harpsichord files were not up as claimed to be --they were the wrong files. They were replaced Dec 14 at 14.00 PST with the correct ones including track title. My apologies. Also ogg files were missing for several recordings. They have been added. To do yet: record jacket info and more recordings. We welcome cooments and suggestions. .....Al Goldstein --- >8 ---- List archives:

Hi, having inspected one of the files found at music.ibiblio.org, I realized that the comments are quite poor or misformatted in the files. I'm telling you, because: 1) I think it's great to have some free music 2) I think the performers deserve to be recognized Generally, I'd put as much as possible into the Vorbis comments. Then you could skip creating these index.html manually:

Hello, I am author of a Morse code training software written in Perl at http://starling.ws/morse Currently I output to *.wav and, if on Unix, convert to *.ogg. And that's okay for a trainer. But I have desire to upgrade the project to a real-time communication tool...or at least a Morse code audio mail routine. And I should like to avoid *.wav. I'd like some means of writing a pure

hi; i'm a lurker & trying to make sense of the discussion... is there a glossary of terms available somewhere? mdct, wavelets, 4th dimensions & preechoes remind me of my time in the twilight zone;) thanks peter --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

Hi. Since I'm new to this mailing list, I don't know whether this was discussed in this mailing list, or if it is already fixed. Anyway, the problem is that all audio encoders out there add a small amount of silence to the start/end of the encoded file. This is usually a small amount, around 0.03 seconds. Most of the time it goes unnoticed, BUT when you have tracks that flow into one

Hi folks, some notes about goings on. After opportunity to think aloud, I have the pieces of the VQ system more or less in place. I need to finish the codebook generation core (this is not libvorbis code, but an external utility). I then need to write the encode/ decode/packing code that makes use of the codebook. This future code is already specced. Tony: I haven't worked on the xmms

Hi folks, some notes about goings on. After opportunity to think aloud, I have the pieces of the VQ system more or less in place. I need to finish the codebook generation core (this is not libvorbis code, but an external utility). I then need to write the encode/ decode/packing code that makes use of the codebook. This future code is already specced. Tony: I haven't worked on the xmms

Hi, I am the CTO of an online e-commerce site selling signed underground music. We're just about to begin with mp3 distribution by download. I have looked into Ogg Vorbis, and it is interesting for us to both avoid the mp3 patent stuff (what's the legal status of Fraunhofer looking into any possible patent infringements by Ogg?), and WMA. However, I still don't think Ogg Vorbis is

Hi, I need a psychoacoustics model for some experiments and would like to use the one included in vorbis. Is there any documentation about how to use it? Looks like the relevent code is in psy.c, but I can't go further. Also I would like to know whether I can be of any help in vorbis. I have some knowledge in speech coding (I doing my master at the University of Sherbrooke's speech

I saw from the interview on Slashdot that there is no support for channel coupling in Vorbis. I would like to help work on that. Has anything been started about that? I don't know yet whether if could work "in real life", but I have started to work on a way to do adaptative coupling between two channels (by finding the best transformation to predict one channel from the other). I

(Disclaimer: this is not an LPC vs. LSP question) After looking at the Vorbis code I was wondering why you were using LSP to quantize the spectral envelope instead of simply quantizing the cepstrum (DCT(log(envelope))) or modified cepstrum (DCT(envelope.^alpha)). To me it seems like when the information is already in the frequency domain, there's no need to go back to LPC. Also, I think a DCT

I have noticed may broken links in the vorbis documentation. For instances, none of the links in the "Vorbis analysis and synthesis" and "Vorbis coding and decoding" sections work. Where has that documentation been? Jean-Marc Valin -- Jean-Marc Valin Universite de Sherbrooke - Genie Electrique valj01@gel.usherb.ca --- >8 ---- List archives:

Ok so I finally realized that you need icecast2 for ogg streaming to work and got the latest version of ices-0.2 installed. I am now unable to ever stream a friggin mp3 never mind an ogg file. The server is up (icecast2) and I see it in the web browser, when I try to run ices-0.2 I get the following errror. I am running ices-0.2 as root and icecast2 as richard (might that be the problemo?) Both

Hi, all. I'm new to this list, and I know that my question touches on a FAQ. I'm just hoping to get more information than "it's a priority item, but it's not done yet." (Basically what the FAQ says.) I've been encoding about 20 minutes of mono audio each week to as small a file as I can, so it can be served via html. As long as the result is understandable,

Greatly concerned for size, i just ripped an audio cd @ 22050kHz and mono sound. Encoding using 'oggenc -b 1 -', the resulting Ogg file is around 30kb/s. During the encoding process i was "warned" that the encoder is tuned toward 44.1kHz and encoding files in any other may result in degraded quality. How worried should I be? Is the advantage in size lost due to crappy

Still trying to stream ogg and have not gotten past a make error in ices2. We www.freespeech.org are an independent media / news site so it would be nice to get this running and support this new format. I'd love to convert all of our 700 streaming video files from from realmedia as well but thats another story! Perhaps this is not the correct forum but it seems like there are a lot of

thanks, this works but its almost impossible to make all my programs to add up to an hour exactly so at some stage I would need to kill the a song half way through just before the news at the top of the hour. Any ideas on how to do this ??/ R <p>On Wed, 9 Oct 2002 09:00:39+1000(EST) Geoff Shang<gshang@uq.net.au> wrote: > Hi: > > Regarding refreshing the playlist. IIRC,

Actually, you don't have to check out ogg and vorbis from CVS. You can just use Vorbis 1.0, if you like. ices was updated to use the Vorbis 1.0 encoding API, and that is the source of the errors the original poster was seing. --- Stan Seibert <p>On Wed, 2002-07-31 at 15:23, D. Anthony Patrick wrote: > You need to check out ogg and vorbis from cvs and install them as well. Refer