similar to: Speex Command line, Changing the LPC order and modifying the codebook

>There's plenty of areas for improvements that don't require incompatible changes like this one. can u please tell me what do I do to make it more exact waveform coder for music rather than speech. I understand that its meant for speech, but I was just using it for music... I am interested in getting the residual as small as possible using speex. Can you please tell me the areas to

Okay, Not exactly the answer I was looking for. This sounds like a big change. I don't mind re-writing the LSP quantizer but re-training code books and breaking compatibility is not what I want to do. I am working on an optimization for an ARM cortex-A8. It is desirable to process things in 4 element blocks. Is there a simpler approach you could recommend? Vasant Shridhar

I would like to make lpcSize a multiple of 4. In the current library for narrow band mode lpcSize is currently set to 10. I would like to increase this 12 for optimization reasons. Is it a simple matter of just changing lpcSize in the structures in mode.c or are there other implications to doing this? I did try and change the value but I seem to be getting some runtime errors indicating a

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

Hi, Can someone please tell me how should I go about changing the frame size which is hardcoded to 160 for NB and WB and 320 for UWB. For NB speech(8KHz) the framesize of 160 is 20ms frame but for WB and UWB its 10ms. What are the parameters being affected by simply changing the framesize and sub-frame size in "modes.c" How to change the buffer size and how its affected. can we have a

Hello, Jean-Marc Valin schrieb: >>I am planning on creating a steganographic extension for speex/CELP. >>Since speex/CELP is lossy, there should be quite many bits that one >>can use for hiding data. > > Quite the opposite. Speex uses very few bits, so every bit you corrupt > is likely to be heard. You would need to use "fractions of bits" at a > time.

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

Hello kind developers, I am planning on creating a steganographic extension for speex/CELP. Since speex/CELP is lossy, there should be quite many bits that one can use for hiding data. I'm familiar with the principles of CELP (two filter loops, one for pitch, the other for formants, let the raw speech frame run through those filter loops, take residue signal, map it on a codebook entry,

Hi Jean-Marc, Hi Jim, I have also seen some problems with the 1.1.8 release on the C55x. So far I have boiled down the issues to the following: 1) We need our own "fixed_xx.h" header file. I don't know why, and haven't had time to investigate, but there is a definite improvement when I use the attached fixed_c55x.h file which has turned all the maths into inline functions.

hi, basically I am anyway using GMM based parametric VQ for LSF quantization so that is not a problem. Can you plz. tell me how to retrain the pitch codebook and innovation codebook. what are the tools to train the pitch and innovation codebook? Regarding the pitch prediction for high-band, I understand your reason of not using it for high-band. There is one paper, "A 14 kb/s wideband

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

ltconfig placeholder shoud be removed. Makefile.am : some stuff copied from branch_beta3 ogg lib must be added only where necesary. vq subdir Makefile.am'ized, (but installs nothing) made distclean target slightly more clean -- check it by hand first, please --- also you'll want to remove vq/Makefile if the ...am passes inspection ------------------------------------------------- diff

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

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 Guys, I want to compile libvorbis, but I get this error message if I made a make. I use Sun Solaris 9 on a UltraSPARC Server and gcc 3.3 can anyone help me, please? <p>Best Thanks Daniel Here the message output from compiling. ----------------------------------------- /usr/ccs/bin/ld -G -z defs -h libvorbis.so.0 -o .libs/libvorbis.so.0.3.0 mdct.lo smallft.lo block.lo envelope.lo

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 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

Hi I have gone through the document available in the net regarding the Vorbis encoder /Decoder. Based on that i have prepared a understanding document on the encoder/decoder block. I would like to know whether my understanding of the coder is OK. If there are any other additional block /information pl. provide me with the same. Thanks and regards S.Padmashri <HR NOSHADE> <UL>

[ I'm not in the list because I didn't find a digested version; please move the lists to sourceforge.net, and we would have the digested version. I read the replies from the archive. ] Hello. Would it be possible to allow Vorbis use the same codebook for multiple files? I could keep a 650 MB codebook on CD-R and use that for all my audiofiles. If that is possible, how much the