similar to: Re: [vorbis] question about codebook

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, 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 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

The following has been committed to the Vorbis I spec as of r19445: Errata 20150226: Single entry codebooks A ?single-entry codebook? is a codebook with one active codeword entry. A single-entry codebook may be either a fully populated codebook with only one declared entry, or a sparse codebook with only one entry marked used. The Vorbis I spec provides no means to specify a codeword length of

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hi kuhlen, what you said is correct. i am talking about how you are going to arrange these codewords into an array, i.e. in the function _make_decode_table. there he uses node bytes and leaf words for memory management. i got a 24 bit platform. so if i assume that max. codeword length that could be possible as 24 bits can i allocate a memory of (2 * used entries - 2), to arrange the whole tree in

Dear Sir or Maddam, we're in the progress of developing a multimedia framework for Java. We started with programming a Vorbis (ogg) codec and demultiplexer and have still some problems with this. The problem occured when trying to read the setup header of a vorbis file. After a certain time we have to decode the codebook_lengths but this does not work. We followed the instructions in the

hello all, we are developing and porting vorbis1decoder on a 24 bit platform. in the process we came across somedoubts about node bytes and leaf words. from the specification we got that we are arranging the huffman codeword tree into an array. the nodebytes are the number of bytes that are required to represent a node and leafwords are the no. of bytes required to represent the leaf i.e the

Hi, I am a bit confused on how code-words are derived from the codeword lengths. I will appreciate if someone can point me in the correct direction. I will take the example of an actual codebook that i found in a valid vorbis encoded file as shown below. [SK] +------Codebook [0] -------- [SK] Codebook Dimensions = 1 [SK] Codebook Entries = 8 [SK] Unordered [SK] 1, 6, 3, 7, 2, 5, 4, 7, [SK] NO

Hi, We are porting the Ogg Vorbis fixed point decoder to a 24 bit fixed point precision processor. We are trying to reduce the number of cycles for MIPS optimization. In this process we observed that most of the songs do not have "maximum codeword length" greater than 24 bits. We are planning to reduce most of the double word operations (in our case 32 bit data is stored as double

Hi According to vorbis I spec, there are several configurations. There are 2 formats of floor, 3 formats of residue and 3 look up types of VQ codebook, etc. I think to get all kinds of vorbis file with these configurations but what I have got from network only support some of the formats (Ex. floor1, residue 1 & 2, VQ lookup type0 & 1). I downloaded several freeware vorbis encoder

Hello, I was wondering if it is possible to change the huffman stuff? A different mapping maybe, or maybe it is not too late to change? Decoding is much more efficient if: -- the huffman tree is left-aligned (i.e., longer codes more to the left) -- and we get the bits msb, not lsb Oh, and I need some clarification about the lsp: the purpose of the lsp is to find a low-bitcount approximation of

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.

Hi, I've been using the Vorbis codec for a while and recently I got the source code and had a look at the spec (great job on the doc BTW). I don't have any former experience with audio compression, but I do know what a DCT does and I totally grok Huffman coding (so I guess I'm not that much of a newbie, am I?). Anyway I just had some questions about how VQ works on the encoding side.

Hi, all When I read the vorbis-I spec. In the section 1.1.6 Hardware profile, it mentioned that "embedded designs are allowed to deviate in limited ways from the 'full' decode specification yet still be certified compliant. These optional omissions are labelled in the spec. where relevant.". But I didn't read any relevant omissions in the whole spec and I have not seen

Hi, All help is appreciated. I am looking at the function "vorbis_staticbook_unpack" in codebook.c I guess this function just reads the codebooks in some structure (static_codebook) and does not actually map the codewords to the codevalues. Does anyone know what functions can I use in this function to actually view this mapping? I just want to display something like below: I can

I was just somewhat curious as to what lossless compression is used... is it just huffman? Tony Arcieri --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/

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

Hi radeonhd, nouveau, mesa3d developers, Firstly, thank you all very much for all the important work you do. I've been working as a part-time developer on the "Spring RTS" project (open-source game engine) which runs on linux (and other os). Some time ago I tried the engine on the open-source ATI radeonhd driver, which I understand to be partly based on mesa 3d, and all textures

hello, does anybody know whether the codewords have a special meaning? whether they have a different influence like the quantization matriy in jpeg? e.g. do lower or the first entries have a greater influence than the last or higher codewords? thanks gerald