similar to: Please help me understand what values of FFMPEG's "compression_level" preset generate subset FLAC stream and what not-subset?

> check the resulting file for subset blocksizes, sample rates, bits per sample, LPC filter order and rice partition order. That's all parameters that I need to check? FLAC tools have a lots of warnings about non-subset files during encoding, but unfortunately don't have easy way to check/test existing FLAC stream for subset compliance. "flac -a" generates the big text file

Hey. The problems with Levinson -type algorithms and integrization are well known. Older codecs have usually solved the problem by using Schur decomposition for inversion of the autocorrelation matrix instead of Levinson. Another solution sometimes used is to use double precision - in this case you probably could code some pseudo-64bit implementation (using only 32bit commands), but that

Hi Jean-Marc, I did what you proposed. I changed the levinson durbin algorithm to: { int i, j; spx_word16_t r; spx_word16_t error = ac[0]; for (i = 0; i < p; i++) lpc[i] = 0; if (ac[0] == 0) { //for (i = 0; i < p; i++) // lpc[i] = 0; return 0; } for (i = 0; i < p; i++) { /* Sum up this iteration's reflection coefficient

hi, am trying to understand speex's algo. have a few questions. 1) autocorrelation: in the function, _spx_autocorr (for floating point version), there is a line ac[0] += 10; correct me if i am wrong, i suppose the addition of 10 is used to condition the autocorrelation matrix. wonder how the value of 10 is arrived at? 2) levinson durbin (LD) algo in the function _spx_lpc, i referred

Hi there. Just a little status update how that integerization is coming along. I am trying to limit myself to 32 bit arithmetics. That means not using any __int64 or long long datatypes at any point. I have now replaced all steps up to including the estimation of the LPC filter coefficients with integer code. That is about a quarter of the total work completed, I would say. One problem that i

Hi, I'm researching lossless compression for a highschool mathematics research essay and am fairly confused about how the linear prediction coefficients are solved for within flac. As far as I understand, Levinson Durbin Recursion is used to solve for these coefficients, however, what I don't understand is what the toeplitz matrix is composed of. I found sources using samples from within

Hi, I wanted to understand exactly how acf and pacf works, so I tried to calculate ac and pac manually. For ac, I used the standard acf formula: acf(k) = sum(X(t)-Xbar)(X(t-k)-Xbar))/sum(X(t)-Xbar)^2. But for pac, I could not figure out how to calculate it by hand. I understand that in both R and EVIEWS, it is done using the Durbin-Levinson algorithm by the computer. However, I don't

(Disclaimer: this is not an LPC vs. LSP question) After looking at the Vorbis code I was wondering why you were using LSP to quantize the spectral envelope instead of simply quantizing the cepstrum (DCT(log(envelope))) or modified cepstrum (DCT(envelope.^alpha)). To me it seems like when the information is already in the frequency domain, there's no need to go back to LPC. Also, I think a DCT

Hi, I am currently encoding 32 Khz/Qual-10/UWB encode mode, with MANUAL_ALLOC enabled ( similar to the c5x/c6x configuration). In file sb_celp.c, I noticed the scratch memory grow during recursive (UWB -> WB) calls to 'sb_encode'. 1. 'stack' was not tracked (with 'tmp_stack' as done at other palaces) after/before - auto-correlation/Levinson-Durbin scope (~line

I scanned the website with linkchecker and found quite a lot of dead links. This commit fixes or removes them. --- developers.html | 2 +- documentation_tasks.html | 2 +- download.html | 12 ++++++------ faq.html | 2 +- features.html | 2 +- feeds/feed.xml | 8 ++++++++ format.html | 8 ++++----

Hi Jean-Marc I'm looking at the 1.0.5 source, and I'm not seeing an _spx_lpc(). There's an _spx_autocorr(), which is in lpc.c and is called near the start of the encoder function in nb_celp.c. The encoder seems to call the autocorr() function, then calls wld() to do something called Levinson-Durbin. Am I right in thinking that after the call to wld(), the st->lpc[] array

O.k., some more info: I just tested bandwidth widening to fix this. But I need to go to gamma values below 0.9 to become stable -- clearly too much widening, I think. I looked inside the Levinson-Durbin algorithm next. The lines #ifdef FIXED_POINT r = DIV32_16(rr+PSHR32(error,1),ADD16(error,8)); #else r = rr/(error+.003*ac[0]); #endif look interesting. While for floating point,

I scanned the website with linkchecker and found quite a lot of dead links. This commit fixes or removes them. --- developers.html | 2 +- documentation_tasks.html | 2 +- download.html | 14 +++++++------- faq.html | 2 +- features.html | 2 +- feeds/feed.xml | 8 ++++++++ format.html | 8 ++++----

Hello. I'm trying to make k-step-ahead forecasts using ARFIMA(0, d, 0) models by taking the first T+k-1 coefficients in the binomial expansion of (1-B)^d, regarding (1-B)^d x(T+k) as an AR(T+k-1) on x(T+k), where x(T) is the series value at time T and k = 1, 2, 3, . That is, I forecast the series k values forward using the first T+k-1 coefficients in the binomial expansion of (1-B)^d as

Bonjour, D?apr?s votre question ? Jean_Marc : ? Je m'int?resse ? la source 1.0.5, et je ne vois pas une _spx_lpc (). Il ya une _spx_autocorr (), qui est en lpc.c et est appel? ? proximit? du d?part de la Codeur fonction dans nb_celp.c. L'encodeur semble appeler la autocorr () la fonction, puis appelle wld () pour faire quelque chose qui s'appelle Levinson-Durbin.

>From: Josh Coalson <j_coalson@yahoo.com> > >I'm copying the flac-dev list to see if anyone has any >feedback also... I'm supposed to be there myself since yesterday but have not got the first digest yet. >First, the results they show are for compression of data >that has already been lossily quantized to fewer bits per >sample, e.g. u-Law and A-Law are

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

Since two people have asked me to open the source code on my work-in-progress Integer Speex project, I have put it up onto my site www.openacm.org as a download in the project tree. the suggested directory structure is as follows, i.e. the speex source folder should be on the same directory level as the sphinx filder. peex-1.0.1 (ideally the folder from within the SpeexACM source archive)

Hi, in "save.image", it would be nice if there was a "compression_level" argument that is passed along to "save". Or is there a reason for disabling the "compression_level" option for saving workspaces, but enabling it for manually saving individual objects? Thanks, Andreas

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