similar to: HTTP streaming / mime type

I'm trying to set up the console as an extension (so I can set up overhead paging), but I can't seem to get it to work. When I call my paging extension, I get an error that it can't open the device: -- Executing Ringing("Zap/9-1", "") in new stack -- Executing Dial("Zap/9-1", "Console/dsp0|18|A(new/whistle)") in new stack << Call

I was looking into adding Ogg support to gnome-vfs (virtual file system library for GNOME). What I currently do for mime-magic checking is check for the "OggS" part at the beginning. Is this correct? Secondly - what is the "canonical" mime-type? Right now I am using 'audio/x-ogg' - is this correct? Regards, Ali --- >8 ---- List archives:

Hi everyone, Over the last two weeks or so, I've been thinking about how to add surround sound to Ogg -- and more than that, to do it in the best way possible. With this in mind, I started considering using Ambisonic surround sound. The advantages of this format are considerable: a) It was developed in the early to mid '70s, so the patents should be expired by now.

Hi people! I think it would be cool if ogg123 had a mode in which you could conrol it with stdin. pausing, quitting/stopping and seeking would be enough imho. mpg123 already has something like this (I think -e). I'd do it myself but I'm busy. Shouldn't take much time. Bye, Peter Surda (Shurdeek) <surda@bigfoot.com>, ICQ 10236103, +4369910964300 -- The dark ages

Dear developers, I patched ogg123 so that it can use esd (only local socket currently). In order to compile you need to copy esd.h from the esound-devel to the vorbis-tools directory. Usage: ogg123 -e -d something file.ogg the "something" is not important I don't intend to maintain this patch, but perhaps somebody could make it more nicer and incorporate it in the

Hi! I want to pipe stdout/stderr from vorbize for wrapper scripts. But as it doesn't send any "\n"s it doesn't work correctly. I solved it by adding if (param.verbose == 1) { fprintf (stderr, "\n"); } into appropriate position in vorbize.c. Could somebody please implement this nicely? Something like "-e" option in cdparanoia. Thnx. Bye, Peter Surda

Hello ppl! I asked some time ago how about adding the option to control ogg123 interactively from command line, similar to mpg123. So I got fed up waiting and added it myself today. The code initially comes from mpg123, I already adapted it to work with aviplay and vlc so it's proven and shouldn't introduce many problems. The only problems I see now: - in order to compile ogg123 now

Hi! The version of ogg123 (or perhaps libao I think) says that the wav output can't go to STDOUT. Is this feature supported already in the current CVS? Bye, Peter Surda (Shurdeek) <surda@bigfoot.com>, ICQ 10236103, +436505122023 -- The three Rs of Microsoft support: Retry, Reboot, Reinstall. <HR NOSHADE> <UL> <LI>application/pgp-signature attachment: stored

Hello! I was wondering if someone took a look at my patch I posted on tuesday. I just checked and it applies to current cvs too. Anyone has comments about functionality, idea or whether it should/should not be commited? Sorry if this looks like I'm impatient but I just want to be assured someone responsible will eventually take a look and then give me a yes or a no so I can stop worrying and

Hi! I've been wondering, is there a tool (preferably under linux) that allows me simple audio editing of ogg+vorbis file? Say I have a song that takes 2:00 and the last 20 seconds is silence and I want to get rid of it without reencoding. Or could this be done simply in 10 lines in C? Bye, Peter Surda (Shurdeek) <shurdeek@panorama.sth.ac.at>, ICQ 10236103, +436505122023 --

Hi! Sorry for being somewhat offtopic but I'm hoping someone can help me. I need to download some music I can use in a one-time non-commercial multimedia presentation (technically speaking I'll be demo-ing ogg123 :-)). I don't care what it is, only it musn't suck and be somewhat suitable for general audience. RMS's speeches simply don't cut :-) So far all music I found on

Date: Mon, 10 Jul 2000 14:51:12 +0100 (BST) From: DG Malham <dgm2@york.ac.uk> To: vorbis-dev@xiph.org Cc: DG Malham <dgm2@york.ac.uk>, Rob Fletcher <rpf1@york.ac.uk> Subject: Re: [vorbis-dev] True surround sound for Ogg -- a proposal (fwd) In-Reply-To: <Pine.SGI.3.95L.1000710092216.9043693B-100000@turpin.york.ac.uk> Message-ID:

Hi all, As I read into the forum, NATting (SNAT) is expensive, using iptables to translate IP sucks the performance of the system. I just want to know if IProute2 can handle NATting and if it handels NATting can it performed faster than iptables? thanks, Wennie

This came out of the OggPCM discussion, but I think it needs to be addressed on a wider scale. Let's start here, 5 years ago.. http://lists.xiph.org/pipermail/vorbis-dev/2000-July/009513.html (I included this email, below) I emailed David (author of that email) and asked him to join this list. I'm thinking, as I look at the problem, that surround sound needs to be defined _outside_

Ok, the story is this, I''m running an Unreal Tournament 2004 server. It''s running on a system that has two external NICs that are connected to different ISPs. I would like players to be able to connect to the server from either link. My default gateway goes through NIC1. For traffic through NIC2 to work I simply added an extra routing table and rule. The extra routing table has

Hello, I'm considering using Asterisk with some type of Cisco phone, and currently considering either the 7940 or 7960 because of its more-complete functionality (compared to the 7905). I'm currently wondering: Do all the expected functions (transfer, conference, voice mail, message waiting indicator, etc.) work normally with Asterisk over SIP? What caveats are known about using

Hello, I''m looking how (if) can I solve the following problem using HTB and iproute2: I need to assing the same bandwidth limit to every client, but the problem is that clients will be random - i.e. I know niether number of clients no IP or MAC addresses. If anybody knows FreeBSD''s ipfw2 - I''m looking for something like "ipfw pipe 150 config mask dst-ip 0xffffffff

The latest CVS update of libvorbis fails to compile on SuSE Linux 6.4. The error is below. Also, what happened to the xmms plugin directory in cvs? It complains that it is missing when I do a 'cvs update'. gcc -DPACKAGE=\"libvorbis\" -DVERSION=\"1.0beta3\" -DHAVE_ALLOCA_H=1 -DHAVE_ ALLOCA=1 -I. -I. -I../include -O20 -ffast-math -mno-ieee-fp -D_REENTRANT -fs

what is the limit on the arp cache entires ?! Does someone have very big LANs with linux-routers ? How big ? (i mean flat L2 network, not segmented) I want to know how much can my net can scale ? thanx _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

https://bugzilla.mindrot.org/show_bug.cgi?id=2006 Bug #: 2006 Summary: AIX 5.2 /32 bit - a windows Putty session will not connect to AIX box Classification: Unclassified Product: Portable OpenSSH Version: 6.0p1 Platform: All OS/Version: All Status: NEW Severity: normal