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

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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 everyone, In the 1.1.x branch, I've kept adding more stuff to libspeex: preprocessor, AEC, etc. I'm now considering moving all those to a separate library (libvoip, libspeech, whatever). Anyone on this list has good reasons I should consider for either splitting or not splitting libspeex? Cheers, Jean-Marc

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, 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, 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

Hi, I was just trying to use speex for sampling frequency >48KHz. In the original Speex-1.0.5 its restricted only upto 48KHz. I tired to modify it by changing the boundary conditions( the error conditions, i.e. if sampling freq >48KHz, it gives error) in /src/speexenc.c and then it atleast doesnt give the error, there is flow in decoding or encoding(i think). I suspect there are other

hi, I have a small and quick question. When speex divide the UWB speech signal into High and low bands, JMV said to me in HA that : "For ultra-wideband (0-16 kHz, 32 kHz sampling), I first split the band into wideband (0-8 kHz) and "very high band" (8-16 kHz). Then, the wideband itself is split into low (0-4 kHz) and high (4-8 kHz) band. So there's a total of 3 bands

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

Dear R users, codebook (v. 0.5.8) was just released on CRAN. https://CRAN.R-project.org/package=codebook It uses the metadata/attributes that some other packages add (item labels, value labels, labelled missings) to automatically generate a codebook that is both human- and machine-readable. It combines metadata about the variables with data summaries. Many people will import this kind of

Dear R users, codebook (v. 0.5.8) was just released on CRAN. https://CRAN.R-project.org/package=codebook It uses the metadata/attributes that some other packages add (item labels, value labels, labelled missings) to automatically generate a codebook that is both human- and machine-readable. It combines metadata about the variables with data summaries. Many people will import this kind of

> ok, lets get to the point. :) > I would need to mix some audio streams realtime. After googling a little > I had only two answers: > 1 - decode,mix,encode > 2 - lower quality, send streams at the same time and mix at the client. > > I didnt like either solutions. > > I?m not an audio expert, so I?m just saying something silly. I readed > that speex uses a fixed

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]]

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

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

Hi, All. I am a new one to vorbis codec, I want to write a vorbis decoder, so I read the vorbis specification, but when I read the paragraph about codebook stream format. I am totally confused, can any one give me some detailed information about the codebook. Thanks in advance He Yang

Hi, All. I am a new one to vorbis codec, I want to write a vorbis decoder, so I read the vorbis specification, but when I read the paragraph about codebook stream format. I am totally confused, can any one give me some detailed information about the codebook. Thanks in advance He Yang

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:

Bj?rn, I would be very interested in looking at your data and the R file. Actually, I think a lot of people would. You might want to post it on a website... CB -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of Bj?rn Thalheim Sent: Monday, September 11, 2006 4:43 AM To: speex-dev@xiph.org Subject: [Speex-dev] Distances in codebook

Áô©~ªº ¦Ñ¤H wrote: > Dear Segher > > Thanks for your information. Besides, the mechanism of the huffman > decoding seems can not work in a 24-bit processor if the code length > exceeds 24, though I have checked several vorbis file from network and > they are all below 24 bit True; but you can special case the >24 bit codebook to use a more "traditional" decoding