similar to: One codebook for all audiofiles?

Hello. Earlier I asked if it would be possible to use one common large codebook for all audiofiles. Can you estimate how the following system would work? 1. The audio would be divided to 16 samples blocks. 2. For each block, the first sample would be stored without any encoding. 3. Linear interpolations between the stored samples are computed. 4. The residues (length of 16 samples) between the

I've tried to understand the codebook matter (but haven't yet had time to read the source). I want to understand how ibg is the flexibility resulting from packing the codebook in the header of every bitstream. Can we, for example, change the volume of the whole sound by scaling the vectors in the codebook? (My guess is that this wouldn't be enough, as we have to scale the floor of

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, 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

Hi guys, This is my first post, I?m a brazilian programmer and I think speex is a great project. :) I'm working on a free software project and I was planning to use speex on it. I have used the java speex version in another project and that worked well, and this time I was planning to use a JNI version to get the latest features. ok, lets get to the point. :) I would need to mix some audio

heh...:) maybe I?ll just answer "the number is 42". ;) well... some guys did mixing using *G.722 *encoded frames... maybe we could ask them: 8P http://ieeexplore.ieee.org/Xplore/login.jsp?url=/iel5/4054516/4054517/04054870.pdf?isnumber=4054517&prod=CNF&arnumber=4054870&arSt=2045&ared=2048&arAuthor=G.+Agnello%3B+R.+M.+Dansereau and a little off-topic, but fun for

Anexed is my function for reading codebooks and the Vq Table. My problem is: I read a codeBook number of 41, so i called this function above 41 times. The firtst time, the sync patterns (BCV) is alligned correctly, so it reads correctly. the sencond time it is executed, the patter is not there anymore and there are only 8 BCV in my file. function fReadBits1 is the same as the ogg's read bit

I'm copying the flac-dev list to see if anyone has any feedback also... --- Juhana Sadeharju <kouhia@nic.funet.fi> wrote: > Hello again. I had time to check the paper out. I have filled the > steps given in the paper with formulae, and then written a piece of > C code. It is not complete code, but could be a reasonable start. > Maybe there is one typo in the paper -- I have

I've been trying to get some data from the National Survey for Family Growth into R - however, the data is in a .dat file and the data I need doesn't have any spaces or commas separating fields - rather you have to look into the codebook and what number of digits along the line the data you need is. The data I want are the following, where 1,12,int means that the data I'm interested

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, can someone help me understand and experiment with Speex in a better way than my knowledge. I want to change the LPC filter order of the Speex algorithm. I think I need to change the LPC order( or lpcSize) in modes.c file. but to make it work I need to modify the codebook also, can someone tell me how to go about changing the codebook entries to support the changed order of prediction.

Hi folks, I've been dickering around again, wasting the time away.... and today Brian Zisk (chief Vorbis Evangelist among other things) said, "Hey! We need to demo to the world on the 4th of April!" Hooo boy. Time to suck it up and bang it out. The Vorbis libs are not a problem. I'm in process of checking in my new psychoacoustics (they're real pretty!) based on

Is there a currently existing method in an R package for creating a codebook from a dataframe? Preferably, I would like to be able to export to a text file all relevant information. Thanks, Harold [[alternative HTML version deleted]]

Hi, I am trying to implement ogg vorbis real time decoder on SHARC ADSP 21364 processor. I am writing my own version of the decoder using the documentation vorbis_I_spec.pdf from xiph.org. However I am facing certain hurdles in floor curve decoding during audio packet decode. In the spec it says that [cval] = read from packet using codebook number [so and so] in *scalar context*.(page no 43).

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

I have been using libvorbis for the past few weeks and have been asked to summarise what I have discovered about the codec. There is an early draft of the document at http://www.geocities.com/gatewaystation/vorbis/vorbis.htm - Please forgive the dodgy formatting (it was formerly a MS word document that got converted with their 'save as html' feature). I still have some additions to

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

Hi, all I am studying the ogg-vorbis algorithm but I have less knowledge about the codebook structure of vorbis. After tracing the release source code, I found it doesn't use huffman tree to decode the huffman code instead of using a special mapping and pre-look. The memory requirement seems similar with huffman tree decoding. Could anyone tell me how the vorbis huffman decoding works

Hello, There is a coodebook for the pitch gains stored in pgain_table.h. How can we get such a table? I have read in the CELT's paper that the codebook is trained, but I do not understand what that realy means. Can anyone give me an idea about that? Thank you very much. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Monty, Sorry for the wrong subject title last time. A little more clarification needed. When reading bit-by-bit, there is no MSb/LSb, just 'first bit' and 'last bit'. The Huffman tree is traversed from 'first bit' to last bit'. This is a conceptual layer above the bitpacker; all you need to know is what order the bits come out, which happens to be LSb-first.