similar to: File differences

Dear Developer Team (Monty/Michael Smith/and others), It seems you get angry for me, but I don't understand why. I don't want to dispute with you, but maybe I forgot to say some things... Here they are: 1. My program (clone) is not a finalized, and NOT A RELEASED (public) program. This is just a test, I wrote it to myself, to the (sound quality) comparsion of MP3 and OGG. 2.

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

On Aug 22 09:38:17, u.windl at ukr.de wrote: > I wonder: What happens if you disable cURL's (default) buffering? With curl -N, I see the same behaviour. That's curl 8.1.2. What about you? I still can't see the original problem. Jan > -----Original Message----- > From: Vorbis <vorbis-bounces at xiph.org> On Behalf Of Jan Stary > Sent: Tuesday, August 22, 2023

I am trying to convert some a few WAV files to ogg format (using vorbis-tools-1.0beta4 oggenc). I tried: $ oggenc -o nash.ogg nash.wav ERROR: Wav file is unsupported subformat (must be 16 bit PCM or floating point PCM ERROR: Input file "nash.wav" is not a supported format And: splay -d - nash.wav | oggenc -r -o nash.ogg - The resulting file when played with ogg123 played so

hi i'm looking for information about the multichannel spec. of vorbis. 1/ the documentation specifies that 256 channels are available ( -c 256) ... it sounds very well but how may i specify the original files ??? (i haven't 256 physical entries on my soundcard ... arffff) so is there a way to declare something like oggenc file1.wav file2.wav file3.wav -c3 final.ogg ??? 2/ panoramics are

I'm now trying to use an m3u formatted list to pass to ezstream rather than stdin. Since all of my audio files are in .wav format, and I have oggenc, I see no reason to decode them before ogg encoding them. Is there a configuration I can use to make this happen? I tried omitting the decode parameter and ezstream complained. Here is what I have now: <ezstream>

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

Hai! I'm from Hungary, and my English is not so good. Excuse me! I'm using Linux Debian woody, with kernel 2.4.18 Now I try to encode my music files, that are in .wav format to .ogg format. I used to use the lame program to encode my music files to .mp3 format. I try it now on a Windows NT 4, and this 2 version: Lame version 3.92 MMX OggEnc v1.0 (libvorbis 1.0) I have a question, why is

Dan Nelson (dnelson@allantgroup.com) wrote: > I guess the problem might be oggenc's option parser, then. Given a > stereo 22050hz input file, I can't seem to get oggenc to encode less > than 22kbits. The lowest bitrate it will allow on the commandline (for > 22050hz/2ch input) is -b 30, but if you also add -M 1, it will generate > a file with an average bitrate of 22. $

Hi all, I've got a strange problem. I was just encoding a new CD I bought a couple of days ago (Muse - Hullabaloo Soundtrack) when I ran into a strange bug. After ripping track 8, CD 1 with cdparanoia under OpenBSD/alpha, I wanted to encode it with oggenc (latest version from the OpenBSD-current ports tree, (`oggenc -v` says 'OggEnc v1.0 (libvorbis 1.0)')). Here's the output :

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

I'm sorry if this question has been asked before; I've looked through the archives and haven't seen anything. The problem I'm seeing is that oggenc's VBR encoding doesn't seem to pay attention to any sort of bitrate limitation, either the -m or bitrate_hard_min settings. It isn't that it temporarily dips below the minimum; the average for the whole (in this case,

---------- > 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... I was very surprised when I found, that oggfiles created with RC3 are much worse than files made with RC2. I have some example sound file set up for you on the net. The difference is *obvious*. While RC2 made ogg-files, which I would say are perfect (at least I cannot hear a difference to the original), the quality of RC3 is *bad*. Let me first say, that I used libvorbis0, libogg0

Is anyone working on making oggenc use some nice library that reads lots of different audio file formats so that people can encode from files other than WAV? For example, the 'AudioFile' library (just found it via a search on freshmeat) reads wav, aiff, aiff-c, .au, and .snd. Maybe there are other better libraries (OpenAL, maybe), but this library does what I want (read AIFF). -tim ---

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

It seems, that oggenc-ing audiofile, and then flac-ing diffrences between original file and vorbis compressed file gives a little better compression than simply flac-ing. I've tested it on one file only: file.wav 55829468 bytes flac -8 file.wav file.flac 37924329 bytes (0.6793 of original) oggenc file.wav file.ogg 4784799 bytes oggdec -o ogg.wav file.ogg sox tmp.wav ogg-.wav vol

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

As the subject implies, my question is: is it possible to use the -b (or -M) flag at non-44K sample rates? I'm working with an application that is trying to optimize for very small audio filesize. I found that downsampling to 11K and then using q0 gives high compression, but won't seem to drop below 64kbps or so. It seems like the combination of downsampling, then reducing to 30kps

When I encode a monophonic wav file, I would expect the resulting ogg file to be at about half the bit rate specified on the command line, as stated in the "oggenc -h" help text: "The 6 modes are approximately 112, 128, 160, 192, 256, and 350 kbps (for stereo 44.1kHz input. Halve these numbers for mono input).". This doesn't seem to be happening, though. I took a 16-bit,