similar to: Vector quantization - is it a bad thing?

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

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

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>>

---- "Aleksandar Dovnikovic" <aldov@EUnet.yu> wrote: > I think that AAC wasn't developed to be used as 'another' > MP3, it's usage is focused towards companies that sell audio > over the internet, for portable players, satellite communications... A-ha! That explains a lot. There used to be a project called FAAC but I presume it's gone. After seeing

Hi I am a student and working on Speech Coding and Speex. I just want to know where to get the document describing subvector quantization of innovation. Is there any document which can help me know how C code works or any paper exactly explainin the idea? regards Faisal __________________________________ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger.

Hi, I'm studying at ETH Zurich and want to use a pyramid vector quantization for my semester thesis. Unfortunately, I have some troubles with generating the codebooks for high dimensions. For vectors with dimension bigger than 12 it takes such a long computation time to generate. I thought maybe you could help me out with a efficient way to do this? Is there some code available? Thanks for

After reading http://mp3.radified.com/mp3.htm I sent Rad an explanation of some things as I understand them. He liked it and posted it on his site (still unlinked, use the URL below). Can somebody with better understanding of psycho-acoustic terms and the vorbis model check it and comment on it? In particular I didn't know how vorbis handles quantization noise. If you reply with

---- "Bacchus 13" <bacchus13@zdnetonebox.com> wrote: > There used to be a project called FAAC but I presume it's gone. Simply, WRONG INFORMATION! It's not gone but the source only publisized. The same tactics as Lame's. Sorry, the FAAC project team. http://sourceforge.net/projects/faac ___________________________________________________________________ To get

Hi, Does anyone have an idea about the possibility to apply psychoacoustic models as the ones in mp3 or AAC to a CELP coder? Thanks! /Pontus <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 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the

Dear Stefan, In the paper "Improved Forward-Adaptive Prediction for MPEG-4 Audio Lossless Coding", a non-linear compander is applied to the parcor coefficients prior to quantization. This compander is designed in order to minimize quantization error, especially for magnitudes close to unity. If you determine the typical distribution of magnitudes of the LPC coefficients, you could

(This is almost out of topic but anyway...) It is surprising how little research effort have been put into psy-acou models for CELP. The basic problem lies in that it is not easy to alter the LP model without distroying the minimum-phase property (ie. the stability of the predictor). That leaves us with psy-acou modelling of the noise-part only. However, my own research is in constrained

Hi all, I'd like to quantize a variable to map it into a limited set of integers for use with a colormap. "image" and filled.contour" do this mapping inside somewhere, but I'd like to choose the colors for plotting a set of polygons. Is there a pre-existing function that does something like this well? i.e., is capable of using 'breaks'?

Hi I have created a small website trying to document my Ogg/Vorbis support for MPEG4IP. You can find it here. http://www.maersk-moller.net/mpeg4ip/index.html Kind regards Peter Maersk-Moller --- >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 'vorbis-request@xiph.org' containing

Hello everybody, my name is Francesco, I am an italian student graduating in physics. I'm studing the Theora video codec and making some comparative tests. Unfortunately i'm not able to find some informations that could be useful. If someone could make clear some concepts, (or tell me where i can find these informations), i'll be very happy. ( I've yet read the Theora Format

Monkey's Audio lossless compressor (currently win32 only, free but not open-source except decoder) author is thinking to implement a kind of audiophile-quality lossy compression which would filter "noise bits" that are hard to encode lossless but which are (or should be) inaudible and thus improve lossless compression (avg. 300-450kbps). I think that implementing something like this

Hi, I've tested Theora on an HD-ready material, with quality 7, single pass. Resulting bitrate was 1.8Mb/s. Most parts of a movie encoded this way look fine, usually better than a DVD, and in daylight scenes, have no noticeable artifacts at all. But it seems that red surfaces are too much compressed; blocking artifacts are clearly visible on them. Also, white people have slightly smeared

Hello, I have improved quality for -l1->-l4 settings.I have properly set up the wavelet coeff threshold of the dead-zone according to the different wavelet subbands, I have then saved another 1Ko per .nhw files without noticeable difference and so I have raised again the quantization.The reconstructed images have now a better precision and still a good neatness. Update at:

Hello, Really hope that I am not disturbing you, but for those interested, I have improved quantization in NHW, and so now it is visually better, with a better balance between neatness and precision. As it is a small modification and correction, I did not release a new version but I have updated the binaries. More at: http://nhwcodec.blogspot.com/ I know I am publishing a lot of incremental

Well, in the world of audio compression it's never boring... :-) Full press release from Dolby: http://www.aac-audio.com/press/aac.pr.0104.AACencoder.html My comments: Looks like Dolby will start licensing AAC encoder that targets consumers and the products they use like CD rippers, jukebox players, etc. Also, it looks like that this encoder will be Low Complexity AAC ("The AAC Consumer

Hello! I'm an Italian student in computer engineering (University "Federico II" of Naples) and I have to write my graduation thesis in the field of signal processing. My teacher (Prof. Giovanni Poggi http://www.diet.unina.it/giovanni.poggi/) has published a research about multiple description vector quantizers by means of the self organizing maps, applying his idea to the images and