similar to: using vcut on split ogg files

Hello I have experienced some problems with vcut (media-sound/vorbis-tools-1.0-r1 package from Gentoo/Linux). I have a long live record I have encoded in OGG (maybe it would have been easier to cut the WAV file and encode the parts, but still). The file was encoded with oggenc from the same package as vcut. I have therefore tried to use vcut to split the long OGG file in several smaller files.

Hi, I have a question regarding vcut. In the man page, it says: DESCRIPTION vcut reads an Ogg Vorbis audio file and splits it at the given cutpoint, which is a sample number. <p>What I want to do, is to cut an Ogg Vorbis file at a given time. Say, I have an Ogg Vorbis file, which is 1 hour long, and I want to cut it into two half hour parts. With a fixed bitrate

Dear vorbis-dev I tried to split an ogg file which was downloaded as a part of a stream with mplayer -dumpstream -dumpfile file.ogg http://..../stream.ogg. The metadata or headers or what it is is then of course in error. (I hate this problem.) Both vcut (I'd post version if vcut supported a --version or -V option) and oggsplt (= mp3splt (same about version)) (according to this page, they

I'm running ices-kh to stream from jack at 64kbps, and also using oggenc (with ecasound via jack) to record the audio to disk at the same time. This is also running at 64kbps. ices is using virtually no cpu (0.0%), but oggenc is using 15-16%. I can't see why there should be such a difference - both are recording the same audio stream in real time at the same bitrate. oggenc is getting its

I'm running ices-kh to stream from jack at 64kbps, and also using oggenc (with ecasound via jack) to record the audio to disk at the same time. This is also running at 64kbps. ices is using virtually no cpu (0.0%), but oggenc is using 15-16%. I can't see why there should be such a difference - both are recording the same audio stream in real time at the same bitrate. oggenc is getting its

How does one use the vcut from Vorbis-tools package? I tried to enter samples, miliseconds, seconds, h:mm:ss as cut point, but nothing worked (I even tried to enter 1 as cutpoint, but I always got "Cutpoint not within stream." message)... I'm doing this on Windows, if it matters... -- Jernej Simoncic, jernej.simoncic@guest.arnes.si http://www2.arnes.si/~sopjsimo/ ICQ: 26266467

Some detailed feedback that looks useful. ----- Forwarded message from Michael Semon <wayfarer42@postmaster.co.uk> ----- Delivery-Date: Tue Jul 16 05:29:37 2002 X-Mailer: MIME-tools 5.41 (Entity 5.404) From: "Michael Semon" <wayfarer42@postmaster.co.uk> Subject: CVS: ogg123 rocks! vcut no so much so... To: feedback@vorbis.com Hi! Being impatient for RC4, I downloaded

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this is 6.0-STABLE as for Mar 17. KDB: debugger backends: ddb KDB: current backend: ddb Copyright (c) 1992-2006 The FreeBSD Project. Copyright (c) 1979, 1980, 1983, 1986, 1988, 1989, 1991, 1992, 1993, 1994 The Regents of the University of California. All rights reserved. FreeBSD 6.1-PRERELEASE #2: Fri Mar 17 11:05:32 UTC 2006 vlad@host:/usr/src/sys/amd64/compile/DEF_WEB Timecounter

Here's an updated version of my last vcut patch, which adds support for chained streams. The patch is against vorbis-tools 1.3.0b1 and fixes all issues that were identified by TODO and XXX comments in the preliminary patch. It also fixes a false positive for the "bad GP range" error and avoids using stderr for messages that aren't errors or warnings. I believe this version is

Hey everyone, A while back I worked on a project to measure call quality. I've finally gotten around to releasing it and I'm calling it recqual (Real Call Quality). There isn't much to it and it should be considered alpha quality. I'm hoping some of the bright minds on the list can help me out with it. I'll include the intro text from the README in the tarball: ----

On Mon, 18 Oct 2004 16:15:44 +0100, EvilOverlord wrote: >> The setup I'm working with is an ecasound session relaying audio between >> its input and output ports, with its output connected to all the ices >> clients for the different streams we're running. Then I'm >> switching the ecasound input between different stream sources (line in, >> recorded

I've been having a problem where ices-kh (the jack'ified version) disconnects from its jack input source unexpectedly. This happens mainly while other jack clients are being started/stopped, or connected/disconnected, but also at other times (e.g. switching between different X sessions). I'm planning to do a bit more work on tuning up the jack setup to see if I can get rid of the

Hi every body , ( Please forgive my dumb poor english ) I got many tracks in ogg/vorbis format . I would like to make a "preview" of them . I mean : I have a track : foo.ogg ( wich is 3:00 minutes long ) i want to have : preview_foo.ogg ( wich is foo.ogg but from 1:00 -> 1:15 ) I read man oggdec , man ogg123 , man oggenc but i did not see what suits me . I use zsh shell , on a

On Sun, 31 Oct 2004 22:32:17 +0000, andy wrote: > On Mon, 18 Oct 2004 16:06:00 +0100, Andy Baxter wrote: > >> I've been having a problem where ices-kh (the jack'ified version) >> disconnects from its jack input source unexpectedly. This happens mainly >> while other jack clients are being started/stopped, or >> connected/disconnected, but also at other times

Hi, I've got alot of long 1-2hr files which I'd like to split up into 5-10 minute chunks. ?I think this should be possible using a small shell script and vcut, but my scripting abilities are lacking. ?Does anyone know of the existence of a script which would do this, or know how I'd go about making one? So say I had a 60 min file (60mins.ogg) I'd like to issue a command

On Aug 23, 2004, at 9:35 PM, Ralph Giles wrote: > On Mon, Aug 23, 2004 at 11:49:27AM -0700, Eric Smith wrote: > >> I've tracked the problem down to abuse of the C type size_t in >> aplay.c. So I'm pretty sure it's not a problem in oggenc. > > Hmm. Wouldn't have expected that from alsa, though if it's saving > as 'wav' you have to sort of

Dear vorbis-dev lurkers, I bring you the first beta release of vorbis-tools 1.3.0. It has quite a few new features and the final version will have even more, so I am sure it is to everyone's benefit that it gets as much tested as possible so we can do a quick release. Here's the changelog so far: * Fixed an error in configure.ac; --with-speex/flac work again (#1319) * Corrected

Please excuse me, I haven't yet dug into the code too far. What I'd like to do: I'm using ecasound to record for 24 hour periods, which pipes the output directly to flac. I would like to be able to decode sections of this (using until/skip) during recording/encoding. I understand currently that flac can't do this, unless I pipe into dd to skip after decoding (which would be an

Sounds like you need to fix the problem at the source first: balancing the levels going *into* your mixing board so that they're not all over the place coming out. You'll never fix that with any hardware or software device. I use a program called ecasound to do software dynamics processing, but you'd have to hack liveice pretty extensively to use it in that situation. Software