similar to: ***SPAM*** pre-echo preventing

vorbisHi Sorry if it is too stupid to ask the following questions. I am just confused by the pychoacoustics model. I think the computation in the psy.c is to derive a curve which stands for the threshold for the mdct coefficient. Is it right ? If it is right , then what can we do with the curve ? I refered to several articles and find the aim of most pychoacoustics model was to derive the SMR

Dear All: I studied the stereo couple documents and now confused, I wonder why it can be done like that, and what is the principle behind that. I think i am confused perhaps because i have no background knowledge, can you help me out? Also, when doing the T/F transform,we used MDCT, so it means no imaginary part. And i am much puzzled about the angle in the stereo couple,what does it mean?

VorbisHi Any help is appreciate. I have two questions. First, I looked into the vorbis-?-specific and was confused by the floor1 algorithm. I think at last the aim is to derive the piecewise curve with the list X and Y,then when encodering ,why not use the selected point orderly to get the curve ? In other words, the specific use the list of [0,128,64,32,96,16,48,80] , why can not use

Hi Any help is appreciate! These days I studied ogg vorbis encoder with sample rate 44.1kbsp,and I thought the whole chart was as follows: 1. read the PCM data and store them in a buffer 2. do MDCT on those PCM data 3. do FFT on those PCM data 4. use both the FFT and MDCT coefficient to the psychoacustic modle and derive an maximum allowable noise curve, and during this curse ,do the

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

HI I am confused about the psychology acoustics model. How can we decide which window to use? I looked into the procedure,but just confused . can you tell me the basic flow or give me a flowchart? Thank you! ??????? ????????fangchao ????????fangchao_dlut@sina.com ??????????2005-09-13

Hi, I'd like to contribute to Vorbis and I think this may be of some interest for low bitrate coding. I have been experimenting with low bit-rate coding for the high-band (11 kHz to 22 kHz) and, though I haven't yet started quantizing my coefficients (a gain and an LPC filter), I expect to be able to approximate the whole 11-22 kHz band with around 1000 bits/s per channel (maybe even 500

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

aoTuV beta4 is based on libVorbis1.1.1 now!! http://www.geocities.jp/aoyoume/aotuv/

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 I installed xen-2.0.6 ,and boot up it ,when I try to run "xend start",there are some ERROR in/var/log/xend-debug.log as follows: OSError: [Errno 2] No such file or directory: ''/var/lib/xen/xend-db/domain/'' ''/var/lib/xen/xend-db/domain/'' y: ''/var/lib/xen/xend-db/domain/'' What should I do ? Anyone help me My Sytem is redhat9

Hello list, Excuse me for my poor english. I have updated on a test server dovecot 2.0.13 to dovecot 2.1.1. All works fine, but with the new version it seems that dovecot don't do hardlinks when deliver a message to multiple users. I have checked my config and the only rule I can see aboout that is maildir_copy_with_hardlinks = yes in /etc/dovecot/conf.d/10-mail.conf My mail location

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 Any help is appriatiate. I download the libvorbis-1.1.1 and find it support 2 channels only,is it the truth or i just made a mistake? i think the psychology model is model?,why not use model ?? is it because the patent question? Thank you! ______________________________________ ????2G????? http://mail.sina.com.cn/chooseMode.html ?

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

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