similar to: format spec unclear on how to quantize the LPC prediction

Hi, I am kind of new to DTrace , I have written a script to time function calls in our application library , Wanted to know how to interpret the output from quantize of the elapsed time in each function call , here is a sample Here is the entry and return function for the library that is being traced pid$1:libswduar::entry { duarEntry[probefunc] = timestamp;

1) What are you doing "mathematically" in a procedure x86/x86enquant: oc_enc_quantize_sse2()? This - the assembler code, and I do not understand mathematically - that's going on there. --- A: 120 121 28 73 -20 -99 -98 -100 123 122 112 108 73 -32 -102 -98 123 123 117 121 100

(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

Hello, I joined because this list is the preferred method of communication according to web site, and I would like to tell you about something I've been working with that perhaps you would like to include among the links on the FLAC web site? Since 2014, I have a system that runs as a service on a computer connected to my stereo to play FLAC files, exposing a web interface for playback

Hi all, I'd like to quantize a variable to map it into a limited set of integers for use with a colormap. "image" and filled.contour" do this mapping inside somewhere, but I'd like to choose the colors for plotting a set of polygons. Is there a pre-existing function that does something like this well? i.e., is capable of using 'breaks'?

Dear Stefan, In the paper "Improved Forward-Adaptive Prediction for MPEG-4 Audio Lossless Coding", a non-linear compander is applied to the parcor coefficients prior to quantization. This compander is designed in order to minimize quantization error, especially for magnitudes close to unity. If you determine the typical distribution of magnitudes of the LPC coefficients, you could

Refs http://sourceforge.net/p/flac/bugs/424/ --- doc/html/format.html | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/doc/html/format.html b/doc/html/format.html index 8598941..2ce82c2 100644 --- a/doc/html/format.html +++ b/doc/html/format.html @@ -1578,7 +1578,7 @@ &lt;5&gt; </td> <td> - Quantized linear predictor coefficient shift needed

Tristan Matthews wrote: > Refs http://sourceforge.net/p/flac/bugs/424/ > - Quantized linear predictor coefficient shift needed in bits (NOTE: this number is signed two's-complement). > + Quantized linear predictor coefficient shift needed in bits (NOTE: this number is signed two's-complement and must be non-negative). Maybe it's better to explain why? At least

Well I'm writing my own source client to automatize song selection, decoding/encoding and maintaning a few tasks on my website (listeners requests, current playing song and playlist, etc...) I've understood that basically icecast protocol is http, then pure ogg or mp3 data. I've found a lot of usefull library (JOgg, JFlac, etc). Thanks for the help, I'm digging into

Hi Jesper, Thanks for reaching out. The links page on the FLAC website needs some pruning. I'm afraid there are a lot of dead links there. I can include your effort here as a library wrapper though: https://github.com/ietf-wg-cellar/flac-specification/wiki/Implementations Anyway, nice to see that libFLAC can be used for Java. As far as I am aware, libFLAC was only available for Java through

Hi, there ! I'm pretty sure that the following text (and attached picture) explains the reason for the gain of high frequencies. when quantizing a vector scalar by scalar by simply rounding each scalar to the nearest level, the quantization-error-vector and the original signal-vector can be assumed to be orthogonal (average case) This is a problem when we want to preserve the energy level

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.

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 the FLAC format description: <5> Quantized linear predictor coefficient shift needed in bits (NOTE: this number is signed two's-complement). What happens when the shift is negative? Problem is that libFLAC decoder (and possibly all other decoders) uses always '>>' operator and this operator shifts only to right. (int)x >> -1 is the same as (int)x

When compiling lpc.c for x86-64 architecture MSVS issues warnings: warning C4028: formal parameter _N_ different from declaration because of flac_restrict modifier. This patch suppresses these warnings. -------------- next part -------------- A non-text attachment was scrubbed... Name: lpc-msvs-warnings.patch Type: application/octet-stream Size: 1025 bytes Desc: not available Url :

Op 03-12-14 om 15:49 schreef Olivier Tristan: > [...] > If you want to check this on a single piano note, I would be > happy to know if this improves the monophonic use case as well. This sample is indeed a case where the retuning of FLAC 1.3.1 shows a severe regression. It seems the patch does work quite well in this case, but it does not fully fix the regression. FLAC 1.3.0 -5:

On Dec 3, 2014, at 8:30 AM, Martijn van Beurden <mvanb1 at gmail.com> wrote: > Op 03-12-14 om 16:48 schreef Olivier Tristan: >> This patch changes the settings associated with compression >> levels 6, 7 and 8. With this patch, -e is no longer used, but >> instead apodization functions are added. This should improve >> compression with at least 95% of all material.

Op 03-12-14 om 18:14 schreef Brian Willoughby: > I would like to point out that 'quiet' material is common when making original recordings. This is probably a misunderstanding: I said that the material that suffers is usually quiet, not that all quiet material suffers. > I do a great deal of live recording, and the levels are always quiet in order to guard against clipping. These

On 12/08/14 12:13 PM, Erik de Castro Lopo wrote: > Martijn van Beurden wrote: > >> >Sorry, most of the testmaterial isn't copylefted, and I don't >> >think it is possible to get a nice copylefted test library. > I don't think we need it to be copy left. We would keep it as > a separate git repo of test files, and we could probably just > use snippets

I want to use the LPC to estimate the vocal tract of various speakers making short utterances. Can anyone point me in the direction of a script that extracts the correct features Stephen -- Internal Virus Database is out-of-date. Checked by AVG Anti-Virus. [[alternative HTML version deleted]]