similar to: Console as extension problems

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.

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! - what is the mime type for ogg? - does the current implementation (xmms/winamp plugin) support http streaming? Bye, Peter Surda (Shurdeek) <surda@bigfoot.com>, ICQ 10236103, +4369910964300 -- Press every key to continue. --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/

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, I'm new to *. I just installed my X101P card with * from the source on Mandrake 10.0 and I test it. Everything seems to work fine. When I call at my home office all the demo ivr seem to work. But I have one question regarding * using /dev/dsp0. I only have one sound card on my system and it has to be use by my personnal PVR called MythTV. I though that * did not need a sound card to work.

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_

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

I use Wine for play World of Warcraft, that support on Windows, 6 channel sounds mixing software. On Ubuntu 8.04 and Wine i get only stereo output. I've tried to do this .asoundrc for "force" use on Wine, of 6channels and I don't get any sound. Code: pcm.!default { type dmix ipc_key 1024 ipc_key_add_uid false # let multiple users share ipc_perm 0660 # IPC permissions

Hi list, I have an asterisk box running on a via C3 motherboaard/Debian Sarge. Installed version was the Debian packages one 1.0.7-bristuff. I use this box with the console dial command and it was working fine. Cards info are: cat /proc/assound/cards 0 [V8235 ]: VIA 8233 - VIA 8235 VIA 8235 at )xe400, irq 11 Now I installed the bristuff+asterisk 1.0.9 and always have in my logs

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

hello, can I get out of wine more than 16 bit output? sound_file -> player -> wine_out, which is 24/32 to linux? somethink like that, I am audiophile :( thank you

All, After installing asterisk on Linux, I run "asterisk -vvvc". But I got the following warning message: chan_oss.so] => (OSS Console Channel Driver) May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing

hi i'm a newbie in asterisk...i installed asterisk but when i tried to dial 1000 for the first time i got the following error messages and i don't hear anything... May 29 20:46:03 WARNING[262160]: chan_oss.c:413 soundcard_setinput: Unable to re-open DSP device: Device or resource busy May 29 20:46:03 WARNING[262160]: chan_oss.c:572 oss_write: Unable to set device to input mode May 29

Hey, I'm having some errors whenever I dial out and I can't dial in at all. I'm using NuFone as my provider just so you know. Jan 11 17:39:46 WARNING[1771]: chan_oss.c:413 soundcard_setinput: Unable to re-open DSP device: No such device Jan 11 17:39:46 WARNING[1771]: chan_oss.c:572 oss_write: Unable to set device to input mode Jan 11 17:39:46 WARNING[1771]: app_dial.c:359

I am trying to get roaming profiles working for my Win2K workstation and run a group login script at logon. My user account (traxx) can join and logon to the domain (DATA) but I get 2 error messages after authentication: 1 'Windows cannot create profile directory \\henry\dcarter\profile.pds. You will be loggeed on with a local profile only. Changes to the profile will not be propogated to the

Hello, does anyone have experience of running two or more sound cards in a FreeBSD machine? I'm using liveice/icecast to stream radio stations, and I'd like to get several in one box. I realise a BSD list would be a better place for this, but who else but streaming folk would ever think about multiple sound cards? After installing a second card, when I makedev snd1, it creates all the

the problem below keeps recarrying even after i have cleared this error when i run asterisk -vvv or -c the error occurs again please help ..Warning, flexible rate not heavily tested! .................WARNING[1024]: File loader.c, Line 212 (ast_load_resource): /usr/local/lib/libh323_linux_x86_r.so.1: undefined symbol: _ZN13PASN_Sequence17PreambleDecodeXERER11PXER_Stream WARNING[1024]: File

I am using alsa with asterisk. The asound.conf is below. When I start asterisk with /etc/asound.conf present I get errors on the console that: chan_alsa.c:304 alsa_card_init: snd_pcm_open failed: Invalid argument If I remove the asound.conf asterisk starts up and works. However I NEED the asound.conf for another application. What might be the issue here? THanks jerry /etc/asound.conf

Hi I had have trouble to get sound working under freebsd 8.1 amd64 arch. So i decide to dig in wine code and I create a simple "proof of concept" patch to get work my sound card.? ********* SND STAT ********FreeBSD Audio Driver (newpcm: 64bit 2009061500/amd64) Installed devices: pcm0: <HDA Realtek ALC272 PCM #0 Analog> (play/rec) default <- This is my primary snd pcm1: <HDA

I found this information elsewhere.... making it /root/.asoundrc helped. # This is .asoundrc # # this makes legacy OSS apps use alsa software mixing dmix pcm.dsp0 { type plug slave.pcm "dmix" } # mixer0 can stay unchanged, because it isn't used anyway, I guess ? ;) ctl.mixer0 { type hw card 0 } # this makes native ALSA apps default to using dmix pcm.!default {