similar to: Vorbis

After playing with the vorbis code for a while and doing tons of hacks and analysis on it, I've found it to perform very poorly with impulse signals. The MDCT seems to cause lots of spreading, and it seems to result in much worse impulse performance then mp3. What is the current plan on handling this? Will a smart quantizer be able to avoid it? I've been looking at various ways of

Hi, I need a psychoacoustics model for some experiments and would like to use the one included in vorbis. Is there any documentation about how to use it? Looks like the relevent code is in psy.c, but I can't go further. Also I would like to know whether I can be of any help in vorbis. I have some knowledge in speech coding (I doing my master at the University of Sherbrooke's speech

Hi, I have two time series, y and x. Diff(y) and Diff(x) both show no autocorrelation. But durbin.watson(lm(Diff(y)~lag(Diff(x),k=-4)) gives a DW value of zero. How come the residule is autocorrelated while Diff(y) and Diff(x) are not? Does anyone know if in my case a DW of zero indicates serial correlation, or is it telling me that the DW statistics is not the appropriate statistics to use here?

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>

Hi folks, My wife and I are taking a long weekend to celibrate our anniversary. I won't be around again until Wednesday to answer email or do anything else on Vorbis. Yeah, I know, I usually answer my email once a month anyway (so likely no one would notice me being gone), but just so folks know if anything really juicy comes up :-) I'll be merging my latest branch with mainline

I upgraded my versions of llvm, clang and compiler-rt to the top-of-tree versions from last night (r180162, April 24). I recompiled debug versions of llvm, clang and my code. I then regenerated my test case and the results were the same - I can list lines of dwarf1.lsp in lldb but I can't set break-points or do anything else (what else should I be able to do?). The updated file that

One other thing that may or may not illuminate the situation. When I run under gdb (on OS X 10.8.3 this is an ancient version of gdb 6.3.5 - but it works with clang compiled C++ code) I get the following error when I try to list a line in dwarf1.lsp: Dwarf Error: Cannot handle DW_FORM_<unknown> in DWARF reader [in module /Users/meister/Development/cando/src/tests/core/dwarf1.bundle] (gdb)

(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

It seems to me that in this version of the function using float p=.7071067812f; float q=.7071067812f; will just leads after applying all the products and squaring to p and q having a common factor 0.5 which can be extracted from the formulaes. So I propose to replace: float p=.7071067812f; float q=.7071067812f; float w=vorbis_coslook(wdel*k); float *ftmp=lsp; int

Hi there, I would like to ask something about how to avoid a possible error message within a for loop. I am running a simulation and in some repetitions there may be an error that will cause a crash and stop the whole procedure, what I want is to simply move on to the next iteration automatically and discard the "bad" repetitions from my results. I used the "try" function to

Hello I'm working on a fix-point decoder for AOS/Bluebottle - the proprietary OS of ETH Zurich. I've some problems with Floor0-curve-computation: I don't know what the little omega [cos(w)] stands for, line 3 of the official Vorbis documentation. Does anybody knows what this value means? I've tried to get the answer from the jOrbis and Tremor, but both seem to do - among

Hi, I'd like to contribute to Vorbis and I think this may be of some interest for low bitrate coding. I have been experimenting with low bit-rate coding for the high-band (11 kHz to 22 kHz) and, though I haven't yet started quantizing my coefficients (a gain and an LPC filter), I expect to be able to approximate the whole 11-22 kHz band with around 1000 bits/s per channel (maybe even 500

Hi all, I have a very large music collection that I keep on a portable hard drive (to plug in to car USB, carry with when I'm at my office, etc). All but a few dozen files are part of an album, and not a single audio file. This translates to an insane amount of files stored on my hard drive. I am very strict about how I label and file my media (I loathe unlabeled audio tracks), but

So, it turns out (and another implementation actually explicitly mentions it) that LSP->LPC computation using the FIR algorithm is very sensitive to noise (iterative algorithm) and really really requires doubles [we're not kidding]. This was complicating things for folks pursuing fixed point implementations, and also was a potential source for bugs if FP optimizations got out of hand. This

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

Jean-Marc, >> > It's odd that it "almost" works with the fixed_generic.h. The easiest >> > thing would be to gradually replace routines and see which one causes >> > problem. It's most likely (though I'm not 100% sure) that somewhere in >> > the code, I have a 16-bit value that gets sent to a function/macro that >> > expects a

All, Am running asterisk on an Opteron 165 with 4GB RAM and 1x80GB and 1x320GB SATA for a call center application (running VICIDIAL). Asterisk CLI (accessed by screen logging asterisk on startup and entering the allocated screen) gives me 'Warning LSP Low' and the voice quality goes down when this message pops up! That is, to start, we use: `/usr/bin/screen -L -d -m -S asterisk

Jean-Marc, Last week I tried the SVN code (build 11700) on the TI C55x DSP, and found that operation was broken again. I traced this to build 11522, committed on 5 June. The problem is in lsp.c, function lsp_to_lpc(). The line (lsp.c line 461 in build 11700): xin = 1<<(QIMP-1); /* 0.5 in QIMP format */ evaluates to zero. The following change corrects the problem: xin =

Hi, I've downloaded the Speex-1.1.5 code and am trying to compile the Speexdec code using Microsoft VC++ 6.0 IDE. When I try to open the speexdec.dsw workspace, I get a dialog asking to locate the ogg_static file. I simply cancel that message, and then the project is opened in the VC++ 6.0 IDE. Now, when I simply try to compile the program, I receive this 5 error messages. error1.

Jack, I got the asm we were working last night committed ont he beta 3 branch. It's in use in both vorbisfile:ov_read() and the lsp lookup (when using float lookups) I'd like to have a patch of the other things you did in the past day or so, even if you don't want to commit yet. Monty --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: