similar to: casting/rounding ASM committed to beta3 branch

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

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

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

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 have included the oggvorbis encoding and playback stuff in the Linux-Mandrake distro back in August 2000. At that time I put the "nightly CVS" version, which worked just fine. Then on Mon Nov 27 I noticed some RPM's labelled "1.0beta3" on the website so I upgraded with these ones. Since then we have had some serious problems in encoding: approx 20% of the WAV

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

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.

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

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 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>

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

dear£º in the example£¨nomogram£©£¬I don't understand the meanings of the program which have been marked by red line.And how to compile the program(L <- .4*(sex=='male') + .045*(age-50) + (log(cholesterol - 10)-5.2)*(-2*(sex=='female') + 2*(sex=='male'))). n <- 1000 # define sample size set.seed(17) # so can reproduce the results age <- rnorm(n, 50, 10)

Hello, Iam being to compile speexenc but I have the following error: LINK : warning LNK4098: defaultlib "LIBCMTD" conflicts with use of other libs; use /NODEFAULTLIB:library libspeex.lib(lsp.obj) : error LNK2001: unresolved external symbol _spx_cos Debug/speexenc.exe : fatal error LNK1120: 1 unresolved externals Error executing link.exe. Please, help me Guillaume --- >8 ---- List

I've not so much bandwidth to add channel coding but i've few bytes to spare for error detection. With a polynomial CRC-8 i've already more than 99.6 % to detect a burst error in a frame. So let's suppose the critical fields of Speex are: Wideband bit Mode ID LSP OL pitch OL pitch gain OL Exc gain So my plans are to calculate a CRC on these fields for each frame i send, then if i

I would like to know if my vision of things is correct about frame structure. -------------------------------------------------------- Frame header -------------------------------------------------------- Subframe1 header|Subframe1 content| -------------------------------------------------------- Subframe2 header|Subframe2 content| --------------------------------------------------------