similar to: oggenc only encodes in "128 kbps" mode?

---------- > Van: ORK <korsmo@solungen.hm.no> > Aan: vorbis@xiph.org > Onderwerp: [vorbis] OggEnc help file > Datum: maandag 3 september 2001 18:21 > > Somebody asked how to use OggEnc a while ago. > > OggEnc v0.8 (libvorbis rc2) > (c) 2001 Michael Smith <msmith@labyrinth.net.au) > > Usage: oggenc [options] input.wav [...] LOL! --- >8 ---- List

Hello all, I wanted to ask some experienced users some settings about oggenc. 1. How can I setup oggenc so that I can DIRECTLY rip tracks off my audio cd? As we all know, audio cd doens't really have a file format that we can directly use. Do I HAVE to convert them to wav files then convert them to ogg files? 2. The options: -b -m -M with the bitrate of the audio file. If I set something

Hi! I am very pleased with the progress that Ogg is making, expecially after I read the latest comparision tests on http://ff123.net/128test/instruct.html that put OGG on top aside with MPC. BUT the APPLAUD.WAV test case ( http://lame.sourceforge.net/download/samples/applaud.wav ) still produces **VERY** easily audible high-frequency artifacts when encoded with OGG RC3 up to q4.9 (!!!). Things

First let me say that I was absolutely astounded at the sound quality when using oggenc at q = -1 (around 50 kbps). So much so, that I'm anxious to try some of the lower bit rates that were mentioned in the announcement for 1.0: ... audio and music at fixed and variable bitrates from 16 to 128 kbps/channel. But I am having trouble finding how to use either oggdrop or oggenc to get these

A friend of mine made the following comment in a discussion I had with him that on a website we adminster we should offer a) WAV or maybe shorten files b) Ogg as a decent reference lossy encoded version He's been trying to convince me that we should offer MP3 (in lieu of WAV) and possibly Ogg. The audio files are primarily vocals I am not a physics guy but his statements don't

For those who are interested, I've run 3 EAQUAL tests on both the LAME and OGG encoders: LAME with the --alt-standard preset (with and without --nspsytune) and OGGENC (RC3) with -q 6 (my personal "sweet spot") I am somewhat unsure though about the validity of these tests, since the original and decoded files were not the same size (2 KB difference). The LAME encoded files were also

Hi there, I'm using libvorbis in my program and need to encode to target bitrate. I know libvorbis prefer to use quality scale but I can't use it. I've found something at faq http://vorbis.com/faq/#quality *For now, quality 0 is roughly equivalent to 64kbps average, 5 is roughly 160kbps, and 10 gives about 400kbps. Most people seeking very-near-CD-quality audio encode at a quality

I'm going to encode ~2000 CDs soon. All genres, but 90% of it has distorted guitars... Everything from punkrock to metal to industrial to goth to synthpop to classical to techno to whatever... I've heard that RC2 has some hearable artifacts, even in 192/256 kbps... There have been quite a few "bugreports" since RC2 with people sending samples that even I can differ from the

I converted the 128kbps tom.mp3 file into a .wav and then used oggenc -m 2 to laterally encode it to 128kbps tommy.ogg format. I noticed that it gets very close to the size of an mp3, great job guys. Then I got to thinking that maybe it could be compressed further with gzip so I tried it. I also gzipped the mp3 to compare and it looks like the mp3 gzips better but gzipping the ogg file will get

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

I would like to run oggenc on existing wavs through a perl/cgi script in apache 2.0 in OS Linux mandrake 9.1. I have a problem where the script hangs after about 20sec worth of wav, whether it would be 20sec through a 60 sec song or on the second file of a 10sec-a-file batch. The same problem occurs if i run a cron job to encode any found wav files on my hdd. Running the script in shell encodes

Hello, congratulations to the Ogg Vorbis team - RC2 sounds good. But... RC2 in 128 kbps mode seems to clip off all frequencys beyond 16 KHz. On the tracks I tested Beta 4 gave response even beyond 18 KHz. Some testings on a randomly chosen track: (other tracks gave similar results) Artist: Judas Priest Album: Jugulator Title: Bullet Train Beta4: 127 kbps, ~ 18 KHz (!) RC2: 132 kbps (!), ~ 16

Hi there ! I checked Vorbis' performace for mono files at approx. 64-80 kbps (it does a good job) and I'm wondering why the current OggEnc still uses Point-Stereo (>10kHz) for -q4 and -q5 I know, we're usually unable to percieve those phase correlations above 10 kHz, But a Dolby Prologic Decoder isn't. I think a future version of OggEnc which is able to use a user-selectable

Hi, I'm trying to record something for about 24 hours but oggenc tells me that I have 203m remaining. Quote: durand at Deuterium ~> arecord -f cd -t wav | oggenc -b 192 -o file.ogg - Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Opening with wav module: WAV file reader Encoding standard input to "file.ogg" at approximate bitrate

Hi. I'd like to ask a little about where Ogg stands today when it comes to streaming audio. The VBR has been the main argument in the incompatibility for streaming but has this changed somehow or will it? Is there something else besides quick and large scaled bandwith usage variations and the difficulty of buffering, that could cause problems? CBR could be used but some people say that it

When encoding a wav file with vorbis 1.0, I notice the following message: [eds@crash eds]$ oggenc -q 5 grind1.wav Skipping chunk of type "cue ", length 52 Opening with wav module: WAV file reader Encoding "grind1.wav" to "grind1.ogg" What's this "skipping chunk" all about? --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg

Hello folks I am encoding music to ogg vorbis format for the first time. From what i have researched, ogg vorbis is natively a vbr format. Also encoding using the -q switch is encouraged instead of specifying a bitrate through -b. Now, while encoding using -b i get this: Encoding "file.wav" to "file.ogg" at approximate bitrate 224 kbps (VBR encoding enabled) Notice

I was wondering why my CD ripping was producing short files from some tracks. A superficial examination of the code shows this piece of code in vorbis-tools/oggenc/encode.c: Line 120: /* Main encode loop - continue until end of file */ while(!eos) { [...] /* While we can get enough data from the library to analyse, one

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

Hello, I'd like to use abcde to rip my CD:s to Ogg Vorbis format - but abcde wants a "vorbiscomment" in my PATH. I assume it's a tool for inserting meta-data into Ogg Vorbis files, but where do I get it? The vorbis-tools package doesn't seem to contain it. (Platform: Linux) Thanks! -- / Peter Schuller, InfiDyne Technologies HB PGP userID: 0xE9758B7D or 'Peter