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VorbisHi Any help is appreciate. I have two questions. First, I looked into the vorbis-?-specific and was confused by the floor1 algorithm. I think at last the aim is to derive the piecewise curve with the list X and Y,then when encodering ,why not use the selected point orderly to get the curve ? In other words, the specific use the list of [0,128,64,32,96,16,48,80] , why can not use

Hi, Can any one explain me basic concept of Floor Generation in Vorbis encoder? Ravi __________________________________________________________ Yahoo! India Matrimony: Find your partner online. Go to http://yahoo.shaadi.com

Dear All, Floor 1 decoder uses FLOOR_fromdB_LOOKUP table. The purpose of this table is not clear to me. It is also not explicitly explained in the vorbis specification. As i have understood is that it is used to dequantise the "Y" values which are decoded from the bitstream during curve synthesis. Is this correct ? If so why does the table have only 256 elements ? This would imply

vorbisHi Sorry if it is too stupid to ask the following questions. I am just confused by the pychoacoustics model. I think the computation in the psy.c is to derive a curve which stands for the threshold for the mdct coefficient. Is it right ? If it is right , then what can we do with the curve ? I refered to several articles and find the aim of most pychoacoustics model was to derive the SMR

Hi Any help is appreciate! These days I studied ogg vorbis encoder with sample rate 44.1kbsp,and I thought the whole chart was as follows: 1. read the PCM data and store them in a buffer 2. do MDCT on those PCM data 3. do FFT on those PCM data 4. use both the FFT and MDCT coefficient to the psychoacustic modle and derive an maximum allowable noise curve, and during this curse ,do the

Dear All: I studied the stereo couple documents and now confused, I wonder why it can be done like that, and what is the principle behind that. I think i am confused perhaps because i have no background knowledge, can you help me out? Also, when doing the T/F transform,we used MDCT, so it means no imaginary part. And i am much puzzled about the angle in the stereo couple,what does it mean?

Hi! Dear all I compressed an audiu(wave) with libvorbis1.1.1 and found that the pre-echo was longer than the one which was compressed with MP3 at the same bitrate(128kbps),I am now confused aboout the pre-echo preventing scheme. Can you tell me what method does Ogg Vorbis adopt besides the auto windowsize switch? Looking forward to your letter! Thank you! ??????? Lisa (^_^)

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

Hi all, I would like to split an audio stream into more encoded flows (for example 1, 2 and 3). This division isn't time based (Maybe frequency based). The substreams that have these features: 1) The flows must be re-assemblable and must generate the original stream (1+2+3 = original stream) 2) I can listen only the flow number 1 (even if the audio quality is worst than the original, of

Dear All, As i have understood it , the Huffman codebook uses a huffman code present in the bitstream ( lets us say it is decoding a floor 1 Y value using the specifed codebook number ) to walk through the huffman code book tree till it hits a leaf node which is nothing but the entry number. If we consider our floor 1 Y value example , this entry number is directly being used as the Y value.

I can't seem to find the right set of commands to enable me to do perform a regression with cluster-adjusted standard-errors. There seems to be nothing in the archives about this -- so this thread could help generate some useful content. I've searched everywhere. Can anyone point me to the right set of commands? An example would be most helpful as well. Bo [[alternative HTML version

Hi I'm new with ogg and also with using third party libraries. I'm making a C++ PocketPC program and I need a simple wav->ogg encoder function. But as I have tried to use ogg vorbis libraries I don't know what or how to use them. I think that best way would be to use libogg 1.1.3 and libvorbis 1.1.2. Do I even need both of them, since I only need to have capability of

Hi, Does anybody know how codebooks are generated in OggVorbis encoder? We are porting oggorbis encoder on embedded platform for which VQ codebook memory is hugeeee to imagine. How can we reduce that? Can we do VQ with less codebooks and if yes how? If any help available? Parul Embedded Engineer Einfochips Ltd

Hi all, I am in the thinking phase of Vorbis 'file' to 'buffer' conversion. ie, accessing the stream data from the static buffer, rather than accessing the file from the decoder( the way it is implemented in the vorbis decoder). I am putting my understanding and the respective query below, As the general convention , there are two methods by which i can do this. 1) Assume what

It obviously would be nice to have such a mode available, for e.g. DVD audio compression. Apparently, the list doesn''t tell me too much about it. My questions are: 1. What is the current status of the 5.1 channel coupling in Vorbis? 2. If I''ll be interested in participation in its development, what is the recommended reading? -------------- next part -------------- An HTML