similar to: vorbis vs. mp3 vs. realaudio

Can someone who really knows the Ogg command-line encoder, help recommend the best setting for 33.6k modem stereo music streaming? (56k doesn't count cuz many people's 56k modems don't work at a full 56k, and I want them to be able to surf CD Baby at the same time as listening. 2 minutes / 120 seconds of audio should be about 400k.) I'm at my wit's end: tried everything I

In testing, I have found that both MP3 and RealAudio 3(!) are much better sounding at bitrates from 8 to 40-something kbps than Ogg Vorbis. This definatly needs drastic improvents and I know that 1.1 will be focusing on improvements in those areas but that seems pretty far off(a few months minimum) which leaves low-bitrate streams sounding pretty bad. --- >8 ---- List archives:

dear vorbis developers, it would be very useful, if oggenc would support constant low-bandwith bitrates for testing. something from 20kbps to 44kbps, so anyone could test streaming over modem and isdn. audio quality doesn't matter. mörk --- >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

Many of you have read and my "introduction to compressed audio with Vorbis" and given me excellent feedback. But with 1.0 about to hit the streets, I figured I'd freshen it up a bit. [ http://grahammitchell.net/writings/vorbis_intro.html ] In particular, it contains two paragraphs discussing the relative quality of Vorbis vs mp3 at a certain target bitrate. These figures seem

did another test of many. NeroAAC q=1 @400kbps and Vorbis q=10 @412kbps shared 2nd place. OPUS @330 kbps - 3rd place. LAME MP3 q=0 @320 kbps - 1st place. ---JPEG file attached--- Please disable speech synthezation in OPUS for 96 kbps and up. I don't want my music sound like from a phone speaker! Or what is the problem? Modern codec at high bitrates should produce nearly bit-exact sound, not

Guys (and girls), I\'ve been testing Ogg Vorbis quality at home, and I thought I\'d share the results with you. For those who want the conclusion, Vorbis 1.0 RC3 sounds better than LAME 3.70 vbr/cbr (which I\'m told now that it\'s broken for VBR), at less bitrate, however, it\'s not perfect (and probably never will, since it\'s a lossy audio codec) and could use some

In my efforts to convince people to hear how good Vorbis is when compared to MP3, I'm gonna be encoding tracks from my CD collection to both formats, then writing them to audio CD, so people don't have to install extra software to hear the difference. Has anyone got some suggested specs for the options for MP3 encoding when using CDex or Chun-Yu's MP3 writer for winamp (preferably in

This mail depends upon the fact that I don't have a couple of good earphones ;-) I read in the site that q=6 is a very high quality, but does it contain perceivable differencies from the original? (for 95% of people, of course). I also found q=6 to produce files slightly bigger (1/10 bigger) than those produced with lame VBR q=2 (about 192 bps on average). I always thought LAME VBR q=2

I tried darkice 0.9 with icecast 1.3.12 and lame 3.91. I specified a 64k stream and similar options to what works for us with liveice. Streaming worked successfully, and I applaud darkice for being fairly simple to build and configure. But the stream sounded like mud. It was worse than our 16k mono RealAudio stream. I re-examined my config, I rebuilt everything with gcc 2.96, then with

[ 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

One thing I must note before I begin is that, I probably just am not that well skilled at listening, but I can hear no difference between vorbis, mp3, and wav at the bitrates I encoded them at. lame 3.83 (beta) was used to encode the mp3s and mpg123 was used to decode. The current (Jun 8 00) encoder/decoder examples were used for vorbis. mp3s were encoded at 128 kbps, and vorbis is unknown

After reading http://mp3.radified.com/mp3.htm I sent Rad an explanation of some things as I understand them. He liked it and posted it on his site (still unlinked, use the URL below). Can somebody with better understanding of psycho-acoustic terms and the vorbis model check it and comment on it? In particular I didn't know how vorbis handles quantization noise. If you reply with

Hello, I've been postponing some of my encoding for when joint stereo gets implemented. The reason I've been doing this, is that I am under the impression this is the largest step in the quality/bitrate ratio that's left. Now I'm wondering if I am correct in thinking this. lame's documentation seems to imply it doesn't make much of a difference at higher than 128kbps (I

Hi there, I have a question on ogglame, especially on its compression quality. I've download the binary package of ogglame from www.vorbis.com. It works fine. I've also tried LAME 3.85 beta that supports the encoding and decoding of Vorbis streams. I believe that ogglame is a beta version of LAME. I built LAME using libvorbis in the CVS nightly snapshot fetched on July 19th. I fixed

Hi, (this is my first post here) A previous thread, starting Date: Tue 19 Nov 2002 - 06:09:56 EST "[vorbis] need speech and music in one" http://www.xiph.org/archives/vorbis/200211/0142.html expressed needs similar to mine, to encode a lengthy speech at low bitrate. I did some tests initially in September then concluded in December, and I was surprised to find Vorbis to be the best

On Fri, 6 Apr 2001, John Griffiths wrote: > ok so 24kbps for 56k modems... > > can i go any lower and get the 28 k modems? (still a lot of them about) or will 24 be good enough fo that? As others have said, 16kbps should do the trick. Keep in mind though that the quality of the sound will also depend on the sampling rate. MP3 will handle some higher sampling rates higher than some of

Hello everyone, I am very tempted to try and use icecast as the weapon of choice to realize a live audio stream for the radio station I work for. Now, the last time I tried the FreeBSD port of Icecast, it immediately consumed something like 95% CPU time, even without any client connected. I think someone else reported that problem, but I somehow lost track of the issue - has this been

I'd like to use the Bitrate and Reencode features of ices and understand I must recompile with lamelib. However, it's Very Unclear where to begin here. This is noted in the ices-0.2.3 INSTALL file: I installed lame as a debian package so lame.h is in /usr/include/lame/lame.h and libmp3lame.a is in /usr/lib. configure finds everything automatically. <p>I assume we

On Wed, 6 Feb 2002, Brendan Cully wrote: > Strange. The different bitrates shouldn't be a problem, as long as the > music beneath is 44.1kHz stereo (which I would expect at 128+ kbps). I > reencode from 128 to 192 regularly, and ices never hiccups at all. What > version of LAME did you link against? 3.89 beta 1. And, yes, everything's at 44100. Like I said, it's no big

Shawn sent me a copy of the non-Direct Sound version of this output plugin, and I'm hosting it on my web site's rare software page: http://members.lycos.co.uk/bhafool1/rarities/ I've found it adds far more smoothness and clarity to various low sample-rate audio files, including an 80-minute speech I mentioned previously, due to my soundcard's approach to D-to-A conversion