similar to: State of the code 19991022

OK, into the nitty-gritty, albeit a high-level version. If it sounds like I'm glossing important details, you're right. This discusses only the basic DSP; precise coding, framing, sync, etc, will be in another mail. Vorbis is a hybrid transform domain general purpose audio encoder, like MPEG in some respects (it is rooted in much of the same basic theory). For the most part, the

I've made block diagram of the encoder because I tried to find out, how it works http://stoffke.freeshell.65535.net/ogg/block.html Although there are specifiation docs, that give very detailed information about single aspects of the encoding (or decoding) , I'm missing documenations that give a more general overview, about how the encoder works. (Vorbis Illuminated seems a bit outdated,

I've made a block diagram of the encoder because I tried to find out, how it works http://stoffke.freeshell.65535.net/ogg/block.html Although there are specifiation docs, that give very detailed information about single aspects of the encoding (or decoding) , I'm missing documenations that give a more general overview, about how the encoder works. (Vorbis Illuminated seems a bit

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

> I have one doubt again, that is Vorbis use DCT/MDCT based algorithm > and also use psychoacoustic model so this is lossy codec. Speex is also a lossy codec. > And I dont think it ca regenerate a better matching waveform than > speex. At bit-rates above 32 kbps, Vorbis tends to produce better results than Speex, even for speech. The only advantages of Speex over Vorbis at these

Hi, I chose speex initially because i had some work in VQ on speex i.e. modifying split VQ to GMM based parametric VQ and I thought If I train the GMM based VQ codebooks with audio signal and then do audio coding with speex, I probably get a better(smaller) residual signal even with speex. But I couldnt get that. I was trying to get a lossless bitstream by MUXing the speex-bitstream and the

Thread-split from the vorbis-mailing list ("Vorbis determined to be as good as MPC at 128 kbps!") <p>On Sun, 30 May 2004, Segher Boessenkool wrote: [Steven So] SS>> If iTunes AAC can encode castanets with much less pre-echo at SS>> ABR 128 kbps, then hopefully there will be an imaginative SS>> (and non-patented) way of doing this in Vorbis without the SS>>

Hi everyone, I've just made the first public release of some new *experimental* codec work I've been doing (part of the vague Ghost project) with help from Monty and Timothy. This is mainly intended with developers with DSP knowledge, not for doing anything useful with it (but it does encode and decode already). Also, the main idea is *not* to replace either Speex or Vorbis, but to code

Hi everyone, I've just made the first public release of some new *experimental* codec work I've been doing (part of the vague Ghost project) with help from Monty and Timothy. This is mainly intended with developers with DSP knowledge, not for doing anything useful with it (but it does encode and decode already). Also, the main idea is *not* to replace either Speex or Vorbis, but to code

Hi everyone, I've just made the first public release of some new *experimental* codec work I've been doing (part of the vague Ghost project) with help from Monty and Timothy. This is mainly intended with developers with DSP knowledge, not for doing anything useful with it (but it does encode and decode already). Also, the main idea is *not* to replace either Speex or Vorbis, but to code

Hi everyone, I've just made the first public release of some new *experimental* codec work I've been doing (part of the vague Ghost project) with help from Monty and Timothy. This is mainly intended with developers with DSP knowledge, not for doing anything useful with it (but it does encode and decode already). Also, the main idea is *not* to replace either Speex or Vorbis, but to code

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> <HEAD><META http-equiv=Content-Type content="text/html; charset=iso-8859-1"></HEAD> <BODY bgColor=#ffffff><FONT face="verdana,arial" size="2"> Hi there,</P> one step in the encoding process (if I got it right), would becomputing a "floor

Hello! I´m just interested if there exists a matlab implementation of the vorbis encoder / decoder or at least of some parts of it (psychoacoustic model, MDCT, VQ), would need it for my diploma thesis. Thanks Stefan Bayer <p>--- >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

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

I'm working on getting the Darwin/Mac OS X port of Vorbis up and running again and have run into problems building libvorbisfile. I have installed the latest libtool (1.3.5) and modified it to handle Darwin/Mac OS X better (to create shared libraries, dylibs, that can be linked against other dylibs instead of creating bundles which cannot be used as input to the linker). But, when it tries

--- libcelt/Makefile.am | 6 +- libcelt/bands.c | 26 +++++++++- libcelt/bands.h | 2 +- libcelt/celt.c | 23 +++----- libcelt/pgain_table.h | 133 ------------------------------------------------- libcelt/quant_pitch.c | 117 ------------------------------------------- libcelt/quant_pitch.h | 44 ---------------- 7 files changed, 37 insertions(+), 314

The VQ codebook generation utilities are more or less ready to go. Minor things are left to do, but those will have to be handled as I to them :-) The next thing to do is complete the interface to get compression modes and codebooks to the library for encoding (minor stuff), and then arrange how exactly the MDCT residue is filtered through the codebooks. VQ codebook util instructions:

Hello folks, The codebook infrastructure for Vorbis is now well underway. This is it! The last piece needed for *real* bitstreams. To that end, I have a decent VQ codebook generator running. I was originally using some farily typical merge/split algorithms and then decided that the right way to do this was to model a VQ codebook as an m-dimensional set of bubbles (like a foam). I know there

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:

Hello Martin, > how a number of samples with no defined size is translated into 8, 16, or 32 bit values in my buffer Although I do not know in detail about the actual implementation of libvorbis, most of the decoders should internally produce single-precision (32bit) floating point values, or 32bit integers as the result of PCM decoding. They are downcast to PCM samples with 8bit, 16bit, or