similar to: beta3 problems

Hi, I encoded some songs into 350 kbit/s with beta2 and beta3 too, and I don't know why, but it seems the b2 sounds better (makes a better quality ogg file)... The b2 contains more the b3 contains less high sounds. And with the b3, the size of file is smaller with 4-5%... Do you know this problem, or do you think so, that the sound quality is good (or better in the b3) ?? Please be carefull

I thought this might be interesting to you, it's an extract from the latest streaming media newsletter, I was intrested to note that MS are claiming cd quality audio at 48kbps whichi is obviously very low, I didn't think much of it at first because nobody uses MS codecs for audio anyway! However (a bit I missed off this quote by the look of it) I then read mention of portable audio players

I tested Vorbis encoder - beta3 version, and here are my thoughts: - In comparison to beta2, subtle high-frequency artifacts seem to be gone (though they were small in beta2). Good job there! :-) - Velvet.wav also sounds better, but transparent quality is reached at -b256+. - Horn.wav still sounds very sucky, mode -b256 gives ~100kbps (this is understandable because in this sample practically

Hi there, I encoded the hard to encode piece ":wumpscut: - mother" with the new prebeta4 encoder (windows). As I crashed my harddrive I've to reinstall linsuxx again and double check the results, maybe there's a difference between the platforms (as far as I remember there was such a thing before). This song is pretty good for fast listening-tests as there are plenty of

Hi there, I encoded the hard to encode piece ":wumpscut: - mother" with the new prebeta4 encoder (windows). As I crashed my harddrive I've to reinstall linsuxx again and double check the results, maybe there's a difference between the platforms (as far as I remember there was such a thing before). This song is pretty good for fast listening-tests as there are plenty of

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

Greetings: Another test. I've re-encoded a batch of mono 22050 Hz WAVs, average bitrate seems to hang around 30-40 kbps. Icecast.xml is set for three connects, and the URL is here: http://64.108.112.145:8000/shouter.ogg IMO, the sound quality is much better now. Again, the stream should be up for about 45 minutes or so. Please let me know how it works for you... Best regards,

Greetings: Another test. I've re-encoded a batch of mono 22050 Hz WAVs, average bitrate seems to hang around 30-40 kbps. Icecast.xml is set for three connects, and the URL is here: http://64.108.112.145:8000/shouter.ogg IMO, the sound quality is much better now. Again, the stream should be up for about 45 minutes or so. Please let me know how it works for you... Best regards,

Here are some minor things I noticed when updating the vorbis module for GoldWave: Bitwise.c, line 175 and 207 Warning: Negative unsigned value ret=-1UL; Info.c, line 385 Warning: Unreachable code break; vorbisfile.c, line 1407 Warning: Call to function with no prototype int host_endian = host_is_big_endian(); fix: add 'void' to line 1339: static int host_is_big_endian( void )

Some thoughts for the new year: 1) MDCT is good for image coding 2) image coding and audio coding are two very different things 3) combine 1 and 2 4) if a psycho model is good, after leaving out what it tells you you can without hurting quality, applying the same model should yield the same results as you got before 5) from 4: decode -> encode -> decode should result in (almost) the

Ubuntu 8.10 Desktop, Wine 1.1.13 (upgraded from 1.0.1 to try to solve my problem). I have used used Guitar Pro very recently without a problem. After a PSU failure, I had to reinstall the OS and downloaded and installed software, including Wine. Guitar Pro appeared to install properly and its splash screen was normal. Immediately the full program ran, a fatal error stopped it. I have a screen

Hello all, Finally I succeeded in uploading the test case I promised. It's at http://home.wanadoo.nl/segher/test1.wav.bz2 (It is a wav, the headers are a bit inconsistent, but encoder_example will be ok with it, as it just skips them). I did some thinking, and a possible solution is decreasing the ATH_Bark_dB[] for the lower frequencies. As the comments say, it's not really an ATH, but

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

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

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

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

what are the main fields where optimisation will take place to improve the CPU use when decoding Ogg Vorbis files? -- Venlig hilsen/Kind regards Thomas Kirk ARKENA thomas@arkena.com http://www.arkena.com "I was drunk last night, crawled home across the lawn. By accident I put the car key in the door lock. The house started up. So I figured what the hell, and drove it around the block a

Form the vorbis.com homepage: "Ogg Vorbis does not compromise quality for freedom. Its stunning quality is coupled with aggressive features like fast bitrate scaling, surround channels, and fast sample-granularity seek and decode." This makes me wonder... -- What is "fast bitrate scaling"? -- I've never seen anything in the source supporting surround channels. -- I

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