similar to: dsp | sox | oggenc

Don't know if anybody is still missing the lame oggenc features for resampling, lowpass/highpass filters etc, but I wrote a little script that uses sox to do all the stuff I need to real-time encode oggenc from the radio, or any input device. #!/bin/bash DATE=`date '+%m-%d-%Y-(%H.%M)'` DESTIN=/video/music/perftoday export DATE=$DATE'-PerformanceToday.ogg' sox -V -r 44100 -c

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

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. $

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

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. I guess the world is full of these, one more won't hurt: "The ogg-cdparanoia script is a wrapper of the programs cdparanoia (http://www.xiph.org/paranoia/), sox (http://sox.sourceforge.net/) and oggenc (http://www.vorbis.com/). The process is simple: for each track, cdparanoia is called to extract the audio to a temporary wav file. Sox is used on it to see if it is normalized.

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>

By the way, I was delighted to find out that I could join .ogg files with cat, i.e. cat {file1.ogg} file2.ogg > outputfile.ogg <p>I successfully used this capability, together with oggenc and sox, to join two .WAV files in the following manner: <p>oggenc -q9 file1.wav ... oggenc -q9 file2.wav --- cat file1.ogg file2.ogg > outfile.ogg ox outfile.ogg outfile.wav I get

Hi, I am trying to encode a 2.1gig .wav file using oggenc with the following error message: andrew@ilium:~/Desktop$ oggenc -q 6 merge.wav -o test.ogg ERROR: Cannot open input file "merge.wav": File too large I have compiled oggenc from source: andrew@ilium:~/Desktop$ oggenc -version OggEnc v1.0.2 I am hoping that this not a limitation of oggenc, rather that I have done something a

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

First of all, keep the replies on the mailing list. Are you subscribed to the mailing list? On Aug 24 09:44:56, nguyenquocthao00 at gmail.com wrote: > The file ogg is the same, I know that. So oggenc must have produced exactly the same audio. So it must be specific to the "live" playing. > And the problem is not from curl, > curl is only an example to replicate my problem.

Hi, I recently noticed that a relativly short wav file loses the last second (truncated perhaps) when I ran it through oggenc. Originally the wav was 25 seconds, the ogg is 24 seconds. This happens at various quality levels, and I saved several copys of the file on a webserver at http://array26.rockefeller.edu/www/ogg/ If anyone wants to confirm my observations. The command lines were:

---------- > 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

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

Hello everybody, I've put some functions/modifications into the Ogg Vorbis encoder source. (The new functions are in the first 500 lines of the oggenc.c, some definitions are in the oggenc.h) I hope, you can use it. Here is the full source (same source on 3 locations): http://www.geocities.com/mpxplay/oggenc.zip http://www.tar.hu/mpxplay/oggenc.zip http://www.extra.hu/galileog/oggenc.zip

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 ---

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

Hi! Excuse the delay, but I had to deal with a corrupted NTFS file system that ate many important files on an USB stick... The FLAC version of the original is almost 6MB and it can be downloaded slowly from this time-limited link: https://sbr5vjid0jgmce4q.myfritz.net:40262/nas/filelink.lua?id=0ba5a10529a6fe7b On the meaning of a logarithmic sweep: If you use foobar2000 and the

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

It is commonly said here that if I want to make AM radio-quality stuff at very low bitrates, a good way is to downsample. I downsampled a song to 11025Hz mono and encoded with -q 0, the result is about 18kbps and is at least radio quality. The downsampler I used is from Edinburgh speech tools, named ch_wave. `sox' performs terribly, so I didn't use it. However, I heard some unpleasant