similar to: Understanding of Vorbis coder

I've made a block diagram of the encoder because I tried to find out, how it works http://stoffke.freeshell.65535.net/ogg/block.html Although there are specifiation docs, that give very detailed information about single aspects of the encoding (or decoding) , I'm missing documenations that give a more general overview, about how the encoder works. (Vorbis Illuminated seems a bit

I write a work over Vorbis. ? ?Now I have a few questions to the function. ? ?For what is the MDCT used? ? ?For what is the FFT used? ? ?How is the expiration with the encodieren? ?In the reference to the MDCT,FFT and the hearing model . ? ?Is there a volume fragmentation in Subband as MP3? ? ?If it is possible, the answers in the detail ?Thanks ? -------------- next

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

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

Thread-split from the vorbis-mailing list ("Vorbis determined to be as good as MPC at 128 kbps!") <p>On Sun, 30 May 2004, Segher Boessenkool wrote: [Steven So] SS>> If iTunes AAC can encode castanets with much less pre-echo at SS>> ABR 128 kbps, then hopefully there will be an imaginative SS>> (and non-patented) way of doing this in Vorbis without the SS>>

I found this sentence in the Ogg format specs: http://www.xiph.org/ogg/vorbis/doc/vorbis-spec-res.html "A residue vector may represent spectral lines, spectral magnitude, spectral phase or hybrids as mixed by channel coupling." But where does the spectral phase information come from ? AFAIK MDCT doesn't provide any phase information. And in OGG-encoding, MDCT is taking place a few

Hi folks, I'm doing a merge of my current branch onto the mainline (for testing) today. I believe it to be stable. Just a little more vorbisfile testing. After the merge, I have a few more patches to apply, then onto cascading/coupling. New stuff: Floor backend 1 and residue backend 1; both are present, but the mainline modes won't use either yet. Naturally, both are enabled for

Hi Any help is appreciate! These days I studied ogg vorbis encoder with sample rate 44.1kbsp,and I thought the whole chart was as follows: 1. read the PCM data and store them in a buffer 2. do MDCT on those PCM data 3. do FFT on those PCM data 4. use both the FFT and MDCT coefficient to the psychoacustic modle and derive an maximum allowable noise curve, and during this curse ,do the

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

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

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 Guys, I want to compile libvorbis, but I get this error message if I made a make. I use Sun Solaris 9 on a UltraSPARC Server and gcc 3.3 can anyone help me, please? <p>Best Thanks Daniel Here the message output from compiling. ----------------------------------------- /usr/ccs/bin/ld -G -z defs -h libvorbis.so.0 -o .libs/libvorbis.so.0.3.0 mdct.lo smallft.lo block.lo envelope.lo

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

Is anyone working on an Ogg Vorbis decoder that uses only integer arithmetic? Would it be hard to do? It will be required at some point (for example, many portable MP3 players are based on an ARM processor without the floating-point coprocessor), but I don't claim that now is the time to start work on such a thing. I'm just curious. Edmund --- >8 ---- List archives:

Is anybody interested in cleanly separating encode and decode in vorbis? I'm trying to take a stab at it myself from the head of cvs as of this morning, but it looks like a lot of files contain both encode-only and decode-only functions. An obvious one is mdct.c, which contains mdct_forward and mdct_backward (which isn't that big of a deal, since both are pretty well-contained, but it

libvorbis does not currently have any simd/vectorization. Following patches add generic framework for simd/vectorization and on top, add ARM-NEON simd vectorization using intrinsics. I was able to get over 34% performance improvement on my Beaglebone Black which is single Cortex-A8 based CPU. You can find more information on metrics and procedure I used to measure at

ltconfig placeholder shoud be removed. Makefile.am : some stuff copied from branch_beta3 ogg lib must be added only where necesary. vq subdir Makefile.am'ized, (but installs nothing) made distclean target slightly more clean -- check it by hand first, please --- also you'll want to remove vq/Makefile if the ...am passes inspection ------------------------------------------------- diff

Hi, Keep up the good work! The file sizes of Theora's output is amazing! I have just joined the list, and would like to contribute to the

(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

Well, here you are. 24k; sorry if I'm not supposed to put this size things in your mailbox, didn't know where else to put it. And you all are subscribed to vorbis-dev, after all. I'm not that good at breaking patches apart, so it's one big patch. Sorry. Overview: configure.in make profiling easier & more useful decoder-example.c (#if 0'ed) dither output;