similar to: Small files encoding problem

Hi all, Yes, it's true. A new version of rehuff, the tool that losslessly compresses Vorbis files: one that is easy to compile, and that works with newer-than-two-years-ago streams, too! On 1.0 streams, you get about 3% size reduction, and the headers get _much_ smaller (which helps for fast-start network streams). Building it should be easy (you might have to add some -I and -L for

Hi all, Yes, it's true. A new version of rehuff, the tool that losslessly compresses Vorbis files: one that is easy to compile, and that works with newer-than-two-years-ago streams, too! On 1.0 streams, you get about 3% size reduction, and the headers get _much_ smaller (which helps for fast-start network streams). Building it should be easy (you might have to add some -I and -L for

No offense, Segher, but the output quality of this thing is awful. =) I'll disregard the fact that, at least with *my* compiler, the source tarball I downloaded reduces every packet to zero bytes, which isn't terribly interesting. I decided to set the byte reduction to something constant: I started by dividing each packet's size by 2 just to see what would happen. The resulting ogg

Important (imho)! I found this site in the internet: http://www.plusv.org/ PlusV is an audio enhancement algorithm similar to SRB of Mp3Pro, but 1) better than SBR 2) Fully open-source It seems to me wise to include PlusV into new versions of Ogg Vorbis. P.S. what about some long-awaited features in Ogg Vorbis? 1) ability to turn off the frequency filter in the encoder (especially for high

Fredag, 11 oktober 2002, skrev du: >Rubbish. Been there, done that (multiple times). Look at rehuff, for >example; written without looking at libvorbis (nor at the spec). And you wrote that without any knowledge about the file format? Tor <p><p>=================================================================== EASY and FREE access to your email anywhere:

Hiya, Here is the sources to my "rehuff" program. ./rehuff in.ogg out.ogg does a lossless recoding of a vorbis stream. (It generates optimal huffman codes for the particular stream). This code is meant for developers only, until someone is kind enough to provide good build and configure support for it. I won't. And no installation help questions please. There is a little patch

I took 5 albums (Classical music) and converted them to Ogg Vorbis at "Full Bitrate" (-q10) and all 5 directories take up about 775 Megs which won't fit on a CD. So I ripped them again in WAV first (And give my friend back his CDs) but now I wanna know what quality setting should I use to fit them on 1 CD (The highest possible with total space used just under 700 Megs)

Hi there. I have a short .ogg with Vorbis stream @56kbps from a 44.1 16bits stereo sample, which weights exactly 135'781 bytes. Packed to ZIP, it goes down to 109'485 bytes, that's a 80% ratio ! Even worse with latest RAR3 archive, with which the file size goes to 104'349 (76%). <sceptical> I admit this sample is a bit repetitive, but I guess the probability of having the

Hi, When I used oggenc to encode small PCM wav files (eg <2k), the size of the .ogg file is usually larger. I assume this is because of some sort of header or dictionary that gets placed at the beginning, and some technical reason. Is there any way around this? Its not a big issue for me, since most of the files I compress are much larger, and the compression is good. It would be clever

I think that there is a very simple way to encode audio files with lossless compression around 35%. (the compression may be improved if we can lose some lsb bits around 30% for 2 and 25% for 3) Other lossless audio encoder I've seen reach at maximum a compression of 48% (*.flac, *.ape) So the questions are: 1- Is someone interested in another encoder? 2- If true n.1 may it be included in ogg

> > I found a bug with FLAC v0.6 raw encoding. It > appears that the file > > pointer in the source file is not reset after > seeking to the end for > > checking the size. I've attached a patch. > > What's the impact? Do I have bad .flac files? > If you are piping raw samples in, no. The bug only occurs when you give an input file AND use -fr. In that

Hello, I am a computer engineering student and compression hobbyst and have recently developed a new entropy coding algorithm. It can be used to achieve compression proper of arithmetic coders at very high speeds -almost like Huffman codecs-. Since it could be of interest to you, I send it as an attachment -code and technical report-. Please, drop me a line in case you have any doubt or

Hello ! A question to all 'Wheel-reinventers': I can build the huffmantrees by hand (on paper) but how to code it? Are there any good URLs out there? Or does the spec supply sufficient information? I tried figure out how oggdec (debugging) does this, but I couldn't get the clou. Thank you Dominik --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project

Hi- just wondering, what would be the best quality rating to use to fit lots of music into 10GB of HD space. The audio quality has to be high enough for it to sound good when played through PA systems (for when my college is stupid enough to forget to hire a DJ, or when the realise the DJ they always hire can't use a mixing desk properly) ranging from a functional PA system thats stuck in the

Hi! Is there any Java Vorbis player that works on MIDP 2.0 platforms (mobile phones) ? I know the pure Java players, but I'm looking for a mobile version. Regards, David --------------------------------------------------------------------- http://noepatents.eu.org/ Innovation, not litigation ! --- David Balazic mailto:david.balazic@hermes.si HERMES Softlab

Hello! I'm working on a program which purpose is to compress a single wav file using many processes. I know that my solution may not look excellent but in my opinion it's simple to implement it in a program. What I've managed to do is: -split single wav file into many parts (with appropriate headers), -create process for each part, -each process executes 'oggenc' program. It

Torsdag, 23 januar 2003, skrev Miroslav Lichvar <lichvarm@phoenix. inf.upol.cz>: >If input is 16 bit, side channel will be 17 bit (16bit - 16bit is >17bit number). And warmup samples will be (17 - wasted_bits) bit. Voila, this was the source of all my frustration, sync problems and who knows what. My decoder now works correctly for files encoded with the default settings. Are

(I've already posted this message month ago, but nobody answered, may be it was not delivered to newsgroup?) Hello, All. I wrote an encoder-decoder based on example from OGG-Vorbis SDK. This encoder can encode a large amount of small WAV-files with equal parameters into one sound archive. Then I decode this sound archive back into large amount of small WAV-files. Theese files are

I see :( That what I would call a good struct size optimisation. Please tell me there was another reason behind this being only 3 instead of 8 or 16 bits, right ? 2017-01-25 18:30 GMT+01:00 Tor-Einar Jarnbjo <tor-einar at jarnbjo.name>: > Hello Olivier, > > the limitation is in the file format itself, as the number of channels is > encoded in a 3 bit field in the streaminfo

Hi all, [For the uninitiated: a "peeler" is a program that transforms a Vorbis stream into a smaller, (somewhat) lower quality Vorbis stream, and does so quickly, by just throwing out some data.] After having prototyped several peelers that aim to peel to a certain filesize, or to a certain quality, with mixed success, I've now taken a different route: a peeler that aims for the