similar to: A new introduction attempt.

Hello vorbis developer! I'm not very experienced in informatics, cause i'm only an audio engineer and sorry for my broken english but i have an idea... After the filterbank the maximum aplitude in each band decides about the nessesary bits, then the psychoacoustic model decides how much of the signal can be truncated without hearable noise. What do you think about quantising the

I intend to finish my Vorbis document on Monday so I will leave it over the weekend to see if there are any more modifications suggested. Although the document will be internal I will put a copy of the final version at the same address - I know firsthand how annoying it is to find loads of dead links when trying to find background info on Vorbis (I know google has probably already got a link

We've implemented a vorbis decoder based on Tremor and as part of the documentation we're also writing about psycho acoustic models and encoding. We're quite up to date with the decoding process and psycho acoustics in general but unfortunately not on the psycho acoustic encoding used in Vorbis. We have a few questions that would we would be very thankful to have answer to: Which

Hi everyone, first of all, I searched through the archives for any posts resembling this, I didn't read all posts about the patents so if this has been suggested before I apologise. I read in an article on C|Net (I think, it was linked from Slashdot anyway) that Thompson are threatening to sue you if Ogg Vorbis becomes a success. Which is evil, and I'm also mad at them because they never

I have developed a flexible arithmetic coder implementation and I am now looking to incorporate it into a compression system for some proper testing. It is my intention to incorporate the coder into the Vorbis codec to attempt to improve on the current Huffman tree system. It would be useful if someone could point me at the main c and h files that deal with the entropy coding as I am currently

Hi guys, Here's two patches I'm planning to apply to simplify the pitch quantisation. Can you test them and let me know how it works? Thanks, Jean-Marc -------------- next part -------------- A non-text attachment was scrubbed... Name: 0001-Moved-the-application-of-the-pitch-gain-to-un-quant.patch Type: text/x-patch Size: 8182 bytes Desc: not available Url :

http://www.xiph.org/ogg/vorbis/doc/draft-moffitt-vorbis-rtp-00.txt This is the Internet-Draft I'll be submitting tomorrow and hopefully presenting at the March IETF meeting. If you see anything major, let me know right away, I'll be submitting this in the morning. jack. --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To

Hi Guys, I want to compile libvorbis, but I get this error message if I made a make. I use Sun Solaris 9 on a UltraSPARC Server and gcc 3.3 can anyone help me, please? <p>Best Thanks Daniel Here the message output from compiling. ----------------------------------------- /usr/ccs/bin/ld -G -z defs -h libvorbis.so.0 -o .libs/libvorbis.so.0.3.0 mdct.lo smallft.lo block.lo envelope.lo

As part of the documentation of a Vorbis decoding project, a quick explanation of the encoding procedure is required. I quite up to date with the decoding process but not with the encoding part. Is there any documentation explaining the different steps during encoding? What I'm looking for is a list of encoding steps, very briefly explained and more details on a few fundamental steps, like

Ha ha! I'm don't know what change I made that cleared up the errors, but I can figure that out later. Of course, I immediately tested my new oggenc. Using the same input file and default options, I get these results: [MSVC] File length: 4m 21.0s Elapsed time: 0m 54.0s Rate: 4.8333 Average bitrate: 129.6 kb/s File size: 4,229,428 [BCB]

I've been reading up on wavelet transforms the past week, and I plan to start on some video compression stuff next week, if it's any good (small chance :)) for Tarkin. So far I think I know what's happening, however there's one small thing I don't quite understand yet. If I understand correctly, you can do a 2d wavelet transform (I'm assuming a Haar transform here for

Hi, I've been reading through the Theora format spec and API reference trying to find an answer to this but haven't come across anything certain. I'm wondering whether it would be theoretically possible to change the encoded bit rate (or target video qiality) during a live stream without a standard theora decoder having any problems. I had assumed that this must be possible, given

Me again, this time not about patents but about wavelets. I've been doing some work on an image compression method which uses Haar wavelets plus VQ and entropy coding, and Segher Boessenkool told me in a private message that you are also doing a video codec. So I searched the archives and read a bunch of posts, but I never did find out any web address. And it's not on xiph.org either. So

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

[ I'm not in the list because I didn't find a digested version; please move the lists to sourceforge.net, and we would have the digested version. I read the replies from the archive. ] Hello. Would it be possible to allow Vorbis use the same codebook for multiple files? I could keep a 650 MB codebook on CD-R and use that for all my audiofiles. If that is possible, how much the

Hi everybody: Here is the newest round of status updates from Xiph.org. The HTML version is at http://www.vorbis.com/ot/20030930.html. Enjoy! -Carsten <p>Ogg Traffic for Tuesday, September 30, 2003 Author: Carsten "Purple" Haese Contact: [1]carsten@xiph.org Date: September 30, 2003 Table of Contents 1 Status Updates 1.1 Monty 1.2

Hi, I have one doubt again, that is Vorbis use DCT/MDCT based algorithm and also use psychoacoustic model so this is lossy codec. And I dont think it ca regenerate a better matching waveform than speex. Then there comes FLAC which is the perfect answer to my question, I suppose. But my concern is this that FLAC use simple prediction algorithm and doesnt use any CELP based algo which could have

This idea is so simple that I'm sure it must have been thought of before, and discarded, since AFAIK it's not used anywhere. I did a quick web search but that didn't turn up much, so I figured I'd put it up for discussion here anyway. How about using large-scale repetition in audio compression? I'm thinking of redundancy in repeated pieces of a song, ie a chorus.

Hi I have gone through the document available in the net regarding the Vorbis encoder /Decoder. Based on that i have prepared a understanding document on the encoder/decoder block. I would like to know whether my understanding of the coder is OK. If there are any other additional block /information pl. provide me with the same. Thanks and regards S.Padmashri <HR NOSHADE> <UL>

Hi, I have looked through the documentation available from the downloads and the web and haven't yet come across a block diagram which shows exactly how the process of going from a .wav file to a .ogg file works. I suspect it is the same as the mp3 process (with a different psychoacoustic model). I found this document: