similar to: Vorbis decoder chip: Specs needed

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

Dear All: I studied the stereo couple documents and now confused, I wonder why it can be done like that, and what is the principle behind that. I think i am confused perhaps because i have no background knowledge, can you help me out? Also, when doing the T/F transform,we used MDCT, so it means no imaginary part. And i am much puzzled about the angle in the stereo couple,what does it mean?

Hi All, Currently I am investigating how interesting OGG is for me. The most important requirements are: - Decoding on embedded hardware, ARM7 or ARM9. - Mixing multiple audio streams/files on embedded hardware First of all how difficult is it to decode OGG on an ARM7/9 processor? I read that OGG decoding needs floating point support? Secondly I would like to know how many MIPS are needed to

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

Hello All, I extended the libNE10 optimizations for float towards mdct_backwards/opus_ifft. I am able to get about 14.26% improvement for Decode use case now on my Beaglebone Black. Please see [1] for measurements. Questions 1. Since this patch needs to go in after Encode [2] patch) should I submit this as patch series? 2. Since Jonathan Lennox posted intrinsics cleanup [3] patch, should

Here's a question I have: My understanding is that OGG is a 'containter' format, allowing for multiple streams of multiple types, of which Vorbis is only one. My question is this: Is it feasable (and, considering licencing, leagal) to create an .ogg file that contains, say Indeo or Cinepak video? (In other words, use the ogg bitstream to store the data rather than .avi).

Hello all. In the shower the other day (where most of this sort of musing gets done, eh?) I was thinking about graceful degradation of audio signals. Let me apologise in advance if these are elementary concepts or if I demonstrate a complete lack of insight -- I don't rate even a dabbler status in the area of audio codecs. Anyway: If we have a 128kbs signal coming down a *udp* channel with

Hello Vorbis folks, I'm one of the FFTW authors (www.fftw.org), and a few days ago I was playing with our codelet generator for fun and modified it to spit out hard-coded MDCTs of small sizes. The code (at jdj.mit.edu/~stevenj/mdct_128nr.c) for 256 samples (128 outputs) seems to be almost twice as fast as the Vorbis MDCT code for that size on my 2.2GHz P-IV (gcc 3.2.2 and flags "-O1

Hello, The manufacturer Cirrus has launched a chip, the CS7410, which can handle decompression of MP3, WMA and "future formats". It contains an 'audio' 16 bit DSP and an ARM core and is supposed to be a cheap solution, IE to be put into cheap $50 CD players. Here's the link to the article on it:

> From: "JD Conley" <jdc@malibuboats.com> > To: "'vorbis@xiph.org'" <vorbis@xiph.org> > Subject: RE: [vorbis] new MS codecs > Date: Thu, 14 Dec 2000 10:49:58 -0800 > X-Mailer: Internet Mail Service (5.5.2653.19) > Reply-To: vorbis@xiph.org > > Oh yeah, they have some samples on their site. Interestingly enough, they > don't

Hi vorbis-dev! I'm working with Pattara in the oggonachip project, and wondering about the implementation of mdct.c as hardware. According your recomendations about using the floating point version, I would say, we have to implement the integerized version of mdct as a core, and use the fpu only to round the input values. By doing that, you think the result would be still acceptable? How

Hi! If you were looking foe a paper about improving vorbis compression by using arithmetic coding and exploiting temporal redundancies (i.e. repeating sound) you might have a look here: http://web.interware.hu/rudas Three ogg vorbis files were compressed losslessly, with compression ratios between 2 - 8 % and compression time about 1/500 realtime (i.e. slow). Theoretical upper bound of the

Hello, I have used "MDCT_INTEGERIZED" version of mdct and the resulting bitstream is absolutely silent. Do I have to make some other changes to the source code to get it working? I mean some int->float or float->int conversion before/after call a mdct function. Thanks Rado --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage:

Dear Sebastian and all, This is a Japanese to live in Japan. I have a question. Ogg's window functon is f(x)=sin(1/2*pi*sin^2((x+0.5)/n*pi)) for 0 <= x < n But I cannot understand why this window function satisfys the equation f^2(x)+f^2(x+n/2)=2 which is for MDCT's perfect signal reconstruction. reference: Analysis/Synthesis Filter Bank Design Based on Time Domain Aliasing

Okay.. I love flac but just had a rather serious failure that really shakes my confidence. It resulted in the near loss of a master audio recording. Fortunately, I have a backup. Though there may have been other cases where I have lost original material because I have been compressing a lot of originals and deleting them after doing a 'flac -t' on them. Basically, flac failed with a

Hi there. I made a couple of noted regarding the speex API. These are not meant be seen as critisism - rather constructive suggestions. I like speex a lot, but I would like it even more if some of the following features were available. ;) Christian <p>Comments about the Speex API - from a developer's perspective ------------------------------------------------------------- ***

I've made 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 outdated,

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:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I want to encode wav file into fixed window length vorbis file. I don't know if "window" is good name for that, but i mean MDCT buffer size. This buffer is passed to mdct_backward() as "in" and it's size is determined by "init->n" variable which is also passed to this function. So, when I say "window

the Vorbis FAQ says: "mid to high quality (8kHz-48.0kHz, 16+ bit, polyphonic) audio and music at fixed and variable bitrates from 16 to 128 kbps/channel." What is the difference between Hz and bitrate? Doesn't MP3 support higher bitrates? Pointers for more reading are welcome.