similar to: just an idea about quality evaluation

I was recently talking to a friend of mine who produces what's widely referred to as "techno", music that's heavy on electronically generated loops and effects, samples, and has plenty of bass. He doesn't like ogg for a few reasons - he sent me a rather long email, and I've stripped out the nonsense from it (he has some technically irrelevant reasons for not liking

hello, people! could you advice me some kind of computer audio recording programme to write music 'on-the-fly' from line-in sound card exit to ogg vorbis file directly. for example, there is an 'absolute mp3 recorder', but i would like to record audio in ogg vorbis. -- kind regards, maxim abalenkov http://maniac.times.lv copy writer mail to: vault13@inbox.lv

On Thu, 2003-01-02 at 10:48, David Wheeler wrote: > > > Unfortunately, an Ogg file can (and will) contain different codecs in > > the same file. Imagine a file that contains two chained groups of > > multiplexed bitstreams. The first group contains vorbis, vp3, speex, and > > subtitle streams. The second group contains FLAC, MNG, and MIDI streams. > > [The fact

On Thu, 2003-01-02 at 10:48, David Wheeler wrote: > > > Unfortunately, an Ogg file can (and will) contain different codecs in > > the same file. Imagine a file that contains two chained groups of > > multiplexed bitstreams. The first group contains vorbis, vp3, speex, and > > subtitle streams. The second group contains FLAC, MNG, and MIDI streams. > > [The fact

[ I'm not in the list because I didn't find a digested version; please move the lists to sourceforge.net, and we would have the digested version. I read the replies from the archive. ] Hello. Would it be possible to allow Vorbis use the same codebook for multiple files? I could keep a 650 MB codebook on CD-R and use that for all my audiofiles. If that is possible, how much the

I quote from "Principles of Digital Audio" by Ken C. Pohlmann: "Because perceptual coders tailor the coded signal to the ear's acuity, they similarly tailor the required response of the playback system itself. Live music does not pass through amplifiers and loudspeakers, it goes directly to the ear. But recorded music must pass through the playback signal chain. Much of the

Hi All, What is the status of the oggpcm project? I'm investigation solutions to the following problem: losslessly encode double-precision mutli-channel timeseries data in a format that is compatible with free (libre) internet streaming technologies and that permits diverse metadata to be encoded with the stream. flac isn't suitable because it only supports integer data, lossy

Hello, I wanted to find out more about square polar mapping, cause that whole stuff is a bit tricky - to me - and not so easy to understand. Searching the web did't get me any results, so I wonder what the background of square polar mapping is. Is it derived from some mathematical publications or just the result of a brilliant intuition of a Vorbis programmer ? ;) Thanks. Stoffke

Hi all! I would like to use Ogg-Vorbis to encode a non audio waveform. My waveform is in .wav format, on 16 bit mono, with frequency range from 100Hz to 100MHz. It's about 100MB lenght. I need to compact it with lossy for net transfer. Is there something like this, already done, that can help me ?? How can I measure the distortion that Vorbis introduce? I'm sorry for my bad english.

Hello everyone! Please bear with me, I am a complete audio-newbie, and I apologise if I am asking a question that I should know. As most of you all know, DVD's have 5.1 audio, and I want to preserve that in my voribs files. So, to ask the question : Does it support Surround sound? Thank you very much!

Hello, we are using speex 1.0.3 under windows. We code samples with values from -32.000 to 32.000 (signed short), but when we decode these samples the values range from -45.000 to 45.000. What is the problem? <p> Thanks --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message

Cool, thanks for all the great comments. I think we agree now on that the "find mp3 before encoding" feature would not be a good idea to implement in the flac core. As Brian pointed out, it might be a better idea to create a program that automatically checks if a flac might have been an mp3 source. My first suggestion was to use FFT, because I know that 128kbps mp3 have a low-pass

?Hi to everybody. I have data in a buffer. when i encode it and decode it i does not obtain the same data. i think encode(decode(data)) = decode(encode(data)) = data. I also try it with trivial example in opus boundle downloded on the official website. My probleme is that how can i write two function myencode and mydecode such as myencode(mydecode(data)) = mydecode(myencode(data)) = data? Thank

Unintended Consequences of MP3 Compression to Hearing: http://www.informatik.fh-hamburg.de/~windle_c/Logologie/MP3-Gefahr/MP3-risk.html Based in some real facts or just someone makeing stuff up? Chris --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-request@xiph.org'

We all know that currently Vorbis is the only codec out there that doesn't add any silence to the decoded file, so when you decode your .ogg file the resulting .wav will be exactly the same length as the original one. This is very useful when encoding gapless songs (like live concerts...). But since Vorbis is a lossy codec, even though there is no silence added, sometimes you can still hear a

I sent a little mail some time ago asking if there was going to be an ogg pic-format, and you replied that PNG, MNG and JNG is good enough (sorry for this late answer btw).. But, consider this: The ogg video-format (tarkin ? where do you get these names from anyway ? :) ) needs a way to compress its frames. Are you going to use MNG for that ? :) .. If you had an ogg pic format, that format could

It's my understanding that the CELT layer of Opus has a maximum input sample rate of 48k, and frequencies above 20k are effectively not encoded. I've been trying to get up to speed on the specification, and studying its operation, but as far as I can infer, there is a fixed set of 21 bands distributed logarithmically to encode DC to 20k. If I were inclined to encode at say, 96k, and pass

For those who haven't yet seen this: EarGuy's Digital Ear (physiological model of the ear based on work by Frank Baumgarte) has just finished rating the sound quality of Vorbis beta 4 at 128 kbs using 30 random 10-second selections of music: http://pub41.ezboard.com/fr3mixfrm4.showMessage?topicID=33.topic Two samples on which the Ear says Vorbis performed uncharacteristically poorly

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 So you're the encoder. Tou get two vectors in (for some band), L and R. One thing you could do with this is compute M = L+R and S = L-R. (Yes, I know, this is not how the encoder actually works. Bear with me.) Then let m = normalize(M) and s = normalize(S). You transmit, m, s, |L|, and |R|. The decoder needs to find unknown positive constants

Hello, I am interested in recording my LP's from my stereo to my computer and then burning to cd. If I understand right theres no way to save as oog and then burn ? Thanks NewbieMark